From 5409fb4e327a84972483047ecf4fb41f279453e2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 7 Apr 2009 18:45:21 +0100 Subject: ASoC: Add WM8988 CODEC driver The WM8988 is a low power, high quality stereo CODEC designed for portable digital audio applications. The device integrates complete interfaces to 2 stereo headphone or line out ports. External component requirements are drastically reduced as no separate headphone amplifiers are required. Advanced on-chip digital signal processing performs graphic equaliser, 3-D sound enhancement and automatic level control for the microphone or line input. The WM8988 can operate as a master or a slave, with various master clock frequencies including 12 or 24MHz for USB devices, or standard 256fs rates like 12.288MHz and 24.576MHz. Different audio sample rates such as 96kHz, 48kHz, 44.1kHz are generated directly from the master clock without the need for an external PLL. Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8988.c | 1097 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8988.h | 60 +++ 4 files changed, 1163 insertions(+) create mode 100644 sound/soc/codecs/wm8988.c create mode 100644 sound/soc/codecs/wm8988.h (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b6c7f7a01cb0..ab364854675b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -36,6 +36,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8900 if I2C select SND_SOC_WM8903 if I2C select SND_SOC_WM8971 if I2C + select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8990 if I2C select SND_SOC_WM9705 if SND_SOC_AC97_BUS select SND_SOC_WM9712 if SND_SOC_AC97_BUS @@ -141,6 +142,9 @@ config SND_SOC_WM8903 config SND_SOC_WM8971 tristate +config SND_SOC_WM8988 + tristate + config SND_SOC_WM8990 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 030d2454725f..a72548dc1885 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -24,6 +24,7 @@ snd-soc-wm8753-objs := wm8753.o snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o snd-soc-wm8971-objs := wm8971.o +snd-soc-wm8988-objs := wm8988.o snd-soc-wm8990-objs := wm8990.o snd-soc-wm9705-objs := wm9705.o snd-soc-wm9712-objs := wm9712.o @@ -55,6 +56,7 @@ obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o +obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.o obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c new file mode 100644 index 000000000000..c05f71803aa8 --- /dev/null +++ b/sound/soc/codecs/wm8988.c @@ -0,0 +1,1097 @@ +/* + * wm8988.c -- WM8988 ALSA SoC audio driver + * + * Copyright 2009 Wolfson Microelectronics plc + * Copyright 2005 Openedhand Ltd. + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm8988.h" + +/* + * wm8988 register cache + * We can't read the WM8988 register space when we + * are using 2 wire for device control, so we cache them instead. + */ +static const u16 wm8988_reg[] = { + 0x0097, 0x0097, 0x0079, 0x0079, /* 0 */ + 0x0000, 0x0008, 0x0000, 0x000a, /* 4 */ + 0x0000, 0x0000, 0x00ff, 0x00ff, /* 8 */ + 0x000f, 0x000f, 0x0000, 0x0000, /* 12 */ + 0x0000, 0x007b, 0x0000, 0x0032, /* 16 */ + 0x0000, 0x00c3, 0x00c3, 0x00c0, /* 20 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 24 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 28 */ + 0x0000, 0x0000, 0x0050, 0x0050, /* 32 */ + 0x0050, 0x0050, 0x0050, 0x0050, /* 36 */ + 0x0079, 0x0079, 0x0079, /* 40 */ +}; + +/* codec private data */ +struct wm8988_priv { + unsigned int sysclk; + struct snd_soc_codec codec; + struct snd_pcm_hw_constraint_list *sysclk_constraints; + u16 reg_cache[WM8988_NUM_REG]; +}; + + +/* + * read wm8988 register cache + */ +static inline unsigned int wm8988_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg > WM8988_NUM_REG) + return -1; + return cache[reg]; +} + +/* + * write wm8988 register cache + */ +static inline void wm8988_write_reg_cache(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg > WM8988_NUM_REG) + return; + cache[reg] = value; +} + +static int wm8988_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D9 WM8753 register offset + * D8...D0 register data + */ + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + wm8988_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +#define wm8988_reset(c) wm8988_write(c, WM8988_RESET, 0) + +/* + * WM8988 Controls + */ + +static const char *bass_boost_txt[] = {"Linear Control", "Adaptive Boost"}; +static const struct soc_enum bass_boost = + SOC_ENUM_SINGLE(WM8988_BASS, 7, 2, bass_boost_txt); + +static const char *bass_filter_txt[] = { "130Hz @ 48kHz", "200Hz @ 48kHz" }; +static const struct soc_enum bass_filter = + SOC_ENUM_SINGLE(WM8988_BASS, 6, 2, bass_filter_txt); + +static const char *treble_txt[] = {"8kHz", "4kHz"}; +static const struct soc_enum treble = + SOC_ENUM_SINGLE(WM8988_TREBLE, 6, 2, treble_txt); + +static const char *stereo_3d_lc_txt[] = {"200Hz", "500Hz"}; +static const struct soc_enum stereo_3d_lc = + SOC_ENUM_SINGLE(WM8988_3D, 5, 2, stereo_3d_lc_txt); + +static const char *stereo_3d_uc_txt[] = {"2.2kHz", "1.5kHz"}; +static const struct soc_enum stereo_3d_uc = + SOC_ENUM_SINGLE(WM8988_3D, 6, 2, stereo_3d_uc_txt); + +static const char *stereo_3d_func_txt[] = {"Capture", "Playback"}; +static const struct soc_enum stereo_3d_func = + SOC_ENUM_SINGLE(WM8988_3D, 7, 2, stereo_3d_func_txt); + +static const char *alc_func_txt[] = {"Off", "Right", "Left", "Stereo"}; +static const struct soc_enum alc_func = + SOC_ENUM_SINGLE(WM8988_ALC1, 7, 4, alc_func_txt); + +static const char *ng_type_txt[] = {"Constant PGA Gain", + "Mute ADC Output"}; +static const struct soc_enum ng_type = + SOC_ENUM_SINGLE(WM8988_NGATE, 1, 2, ng_type_txt); + +static const char *deemph_txt[] = {"None", "32Khz", "44.1Khz", "48Khz"}; +static const struct soc_enum deemph = + SOC_ENUM_SINGLE(WM8988_ADCDAC, 1, 4, deemph_txt); + +static const char *adcpol_txt[] = {"Normal", "L Invert", "R Invert", + "L + R Invert"}; +static const struct soc_enum adcpol = + SOC_ENUM_SINGLE(WM8988_ADCDAC, 5, 4, adcpol_txt); + +static const DECLARE_TLV_DB_SCALE(pga_tlv, -1725, 75, 0); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1); +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); +static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0); + +static const struct snd_kcontrol_new wm8988_snd_controls[] = { + +SOC_ENUM("Bass Boost", bass_boost), +SOC_ENUM("Bass Filter", bass_filter), +SOC_SINGLE("Bass Volume", WM8988_BASS, 0, 15, 1), + +SOC_SINGLE("Treble Volume", WM8988_TREBLE, 0, 15, 0), +SOC_ENUM("Treble Cut-off", treble), + +SOC_SINGLE("3D Switch", WM8988_3D, 0, 1, 0), +SOC_SINGLE("3D Volume", WM8988_3D, 1, 15, 0), +SOC_ENUM("3D Lower Cut-off", stereo_3d_lc), +SOC_ENUM("3D Upper Cut-off", stereo_3d_uc), +SOC_ENUM("3D Mode", stereo_3d_func), + +SOC_SINGLE("ALC Capture Target Volume", WM8988_ALC1, 0, 7, 0), +SOC_SINGLE("ALC Capture Max Volume", WM8988_ALC1, 4, 7, 0), +SOC_ENUM("ALC Capture Function", alc_func), +SOC_SINGLE("ALC Capture ZC Switch", WM8988_ALC2, 7, 1, 0), +SOC_SINGLE("ALC Capture Hold Time", WM8988_ALC2, 0, 15, 0), +SOC_SINGLE("ALC Capture Decay Time", WM8988_ALC3, 4, 15, 0), +SOC_SINGLE("ALC Capture Attack Time", WM8988_ALC3, 0, 15, 0), +SOC_SINGLE("ALC Capture NG Threshold", WM8988_NGATE, 3, 31, 0), +SOC_ENUM("ALC Capture NG Type", ng_type), +SOC_SINGLE("ALC Capture NG Switch", WM8988_NGATE, 0, 1, 0), + +SOC_SINGLE("ZC Timeout Switch", WM8988_ADCTL1, 0, 1, 0), + +SOC_DOUBLE_R_TLV("Capture Digital Volume", WM8988_LADC, WM8988_RADC, + 0, 255, 0, adc_tlv), +SOC_DOUBLE_R_TLV("Capture Volume", WM8988_LINVOL, WM8988_RINVOL, + 0, 63, 0, pga_tlv), +SOC_DOUBLE_R("Capture ZC Switch", WM8988_LINVOL, WM8988_RINVOL, 6, 1, 0), +SOC_DOUBLE_R("Capture Switch", WM8988_LINVOL, WM8988_RINVOL, 7, 1, 1), + +SOC_ENUM("Playback De-emphasis", deemph), + +SOC_ENUM("Capture Polarity", adcpol), +SOC_SINGLE("Playback 6dB Attenuate", WM8988_ADCDAC, 7, 1, 0), +SOC_SINGLE("Capture 6dB Attenuate", WM8988_ADCDAC, 8, 1, 0), + +SOC_DOUBLE_R_TLV("PCM Volume", WM8988_LDAC, WM8988_RDAC, 0, 255, 0, dac_tlv), + +SOC_SINGLE_TLV("Left Mixer Left Bypass Volume", WM8988_LOUTM1, 4, 7, 1, + bypass_tlv), +SOC_SINGLE_TLV("Left Mixer Right Bypass Volume", WM8988_LOUTM2, 4, 7, 1, + bypass_tlv), +SOC_SINGLE_TLV("Right Mixer Left Bypass Volume", WM8988_ROUTM1, 4, 7, 1, + bypass_tlv), +SOC_SINGLE_TLV("Right Mixer Right Bypass Volume", WM8988_ROUTM2, 4, 7, 1, + bypass_tlv), + +SOC_DOUBLE_R("Output 1 Playback ZC Switch", WM8988_LOUT1V, + WM8988_ROUT1V, 7, 1, 0), +SOC_DOUBLE_R_TLV("Output 1 Playback Volume", WM8988_LOUT1V, WM8988_ROUT1V, + 0, 127, 0, out_tlv), + +SOC_DOUBLE_R("Output 2 Playback ZC Switch", WM8988_LOUT2V, + WM8988_ROUT2V, 7, 1, 0), +SOC_DOUBLE_R_TLV("Output 2 Playback Volume", WM8988_LOUT2V, WM8988_ROUT2V, + 0, 127, 0, out_tlv), + +}; + +/* + * DAPM Controls + */ + +static int wm8988_lrc_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + u16 adctl2 = wm8988_read_reg_cache(codec, WM8988_ADCTL2); + + /* Use the DAC to gate LRC if active, otherwise use ADC */ + if (wm8988_read_reg_cache(codec, WM8988_PWR2) & 0x180) + adctl2 &= ~0x4; + else + adctl2 |= 0x4; + + return wm8988_write(codec, WM8988_ADCTL2, adctl2); +} + +static const char *wm8988_line_texts[] = { + "Line 1", "Line 2", "PGA", "Differential"}; + +static const unsigned int wm8988_line_values[] = { + 0, 1, 3, 4}; + +static const struct soc_enum wm8988_lline_enum = + SOC_VALUE_ENUM_SINGLE(WM8988_LOUTM1, 0, 7, + ARRAY_SIZE(wm8988_line_texts), + wm8988_line_texts, + wm8988_line_values); +static const struct snd_kcontrol_new wm8988_left_line_controls = + SOC_DAPM_VALUE_ENUM("Route", wm8988_lline_enum); + +static const struct soc_enum wm8988_rline_enum = + SOC_VALUE_ENUM_SINGLE(WM8988_ROUTM1, 0, 7, + ARRAY_SIZE(wm8988_line_texts), + wm8988_line_texts, + wm8988_line_values); +static const struct snd_kcontrol_new wm8988_right_line_controls = + SOC_DAPM_VALUE_ENUM("Route", wm8988_lline_enum); + +/* Left Mixer */ +static const struct snd_kcontrol_new wm8988_left_mixer_controls[] = { + SOC_DAPM_SINGLE("Playback Switch", WM8988_LOUTM1, 8, 1, 0), + SOC_DAPM_SINGLE("Left Bypass Switch", WM8988_LOUTM1, 7, 1, 0), + SOC_DAPM_SINGLE("Right Playback Switch", WM8988_LOUTM2, 8, 1, 0), + SOC_DAPM_SINGLE("Right Bypass Switch", WM8988_LOUTM2, 7, 1, 0), +}; + +/* Right Mixer */ +static const struct snd_kcontrol_new wm8988_right_mixer_controls[] = { + SOC_DAPM_SINGLE("Left Playback Switch", WM8988_ROUTM1, 8, 1, 0), + SOC_DAPM_SINGLE("Left Bypass Switch", WM8988_ROUTM1, 7, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", WM8988_ROUTM2, 8, 1, 0), + SOC_DAPM_SINGLE("Right Bypass Switch", WM8988_ROUTM2, 7, 1, 0), +}; + +static const char *wm8988_pga_sel[] = {"Line 1", "Line 2", "Differential"}; +static const unsigned int wm8988_pga_val[] = { 0, 1, 3 }; + +/* Left PGA Mux */ +static const struct soc_enum wm8988_lpga_enum = + SOC_VALUE_ENUM_SINGLE(WM8988_LADCIN, 6, 3, + ARRAY_SIZE(wm8988_pga_sel), + wm8988_pga_sel, + wm8988_pga_val); +static const struct snd_kcontrol_new wm8988_left_pga_controls = + SOC_DAPM_VALUE_ENUM("Route", wm8988_lpga_enum); + +/* Right PGA Mux */ +static const struct soc_enum wm8988_rpga_enum = + SOC_VALUE_ENUM_SINGLE(WM8988_RADCIN, 6, 3, + ARRAY_SIZE(wm8988_pga_sel), + wm8988_pga_sel, + wm8988_pga_val); +static const struct snd_kcontrol_new wm8988_right_pga_controls = + SOC_DAPM_VALUE_ENUM("Route", wm8988_rpga_enum); + +/* Differential Mux */ +static const char *wm8988_diff_sel[] = {"Line 1", "Line 2"}; +static const struct soc_enum diffmux = + SOC_ENUM_SINGLE(WM8988_ADCIN, 8, 2, wm8988_diff_sel); +static const struct snd_kcontrol_new wm8988_diffmux_controls = + SOC_DAPM_ENUM("Route", diffmux); + +/* Mono ADC Mux */ +static const char *wm8988_mono_mux[] = {"Stereo", "Mono (Left)", + "Mono (Right)", "Digital Mono"}; +static const struct soc_enum monomux = + SOC_ENUM_SINGLE(WM8988_ADCIN, 6, 4, wm8988_mono_mux); +static const struct snd_kcontrol_new wm8988_monomux_controls = + SOC_DAPM_ENUM("Route", monomux); + +static const struct snd_soc_dapm_widget wm8988_dapm_widgets[] = { + SND_SOC_DAPM_MICBIAS("Mic Bias", WM8988_PWR1, 1, 0), + + SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0, + &wm8988_diffmux_controls), + SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0, + &wm8988_monomux_controls), + SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0, + &wm8988_monomux_controls), + + SND_SOC_DAPM_MUX("Left PGA Mux", WM8988_PWR1, 5, 0, + &wm8988_left_pga_controls), + SND_SOC_DAPM_MUX("Right PGA Mux", WM8988_PWR1, 4, 0, + &wm8988_right_pga_controls), + + SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0, + &wm8988_left_line_controls), + SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0, + &wm8988_right_line_controls), + + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8988_PWR1, 2, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8988_PWR1, 3, 0), + + SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8988_PWR2, 7, 0), + SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8988_PWR2, 8, 0), + + SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0, + &wm8988_left_mixer_controls[0], + ARRAY_SIZE(wm8988_left_mixer_controls)), + SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0, + &wm8988_right_mixer_controls[0], + ARRAY_SIZE(wm8988_right_mixer_controls)), + + SND_SOC_DAPM_PGA("Right Out 2", WM8988_PWR2, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("Left Out 2", WM8988_PWR2, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Out 1", WM8988_PWR2, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("Left Out 1", WM8988_PWR2, 6, 0, NULL, 0), + + SND_SOC_DAPM_POST("LRC control", wm8988_lrc_control), + + SND_SOC_DAPM_OUTPUT("LOUT1"), + SND_SOC_DAPM_OUTPUT("ROUT1"), + SND_SOC_DAPM_OUTPUT("LOUT2"), + SND_SOC_DAPM_OUTPUT("ROUT2"), + SND_SOC_DAPM_OUTPUT("VREF"), + + SND_SOC_DAPM_INPUT("LINPUT1"), + SND_SOC_DAPM_INPUT("LINPUT2"), + SND_SOC_DAPM_INPUT("RINPUT1"), + SND_SOC_DAPM_INPUT("RINPUT2"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + + { "Left Line Mux", "Line 1", "LINPUT1" }, + { "Left Line Mux", "Line 2", "LINPUT2" }, + { "Left Line Mux", "PGA", "Left PGA Mux" }, + { "Left Line Mux", "Differential", "Differential Mux" }, + + { "Right Line Mux", "Line 1", "RINPUT1" }, + { "Right Line Mux", "Line 2", "RINPUT2" }, + { "Right Line Mux", "PGA", "Right PGA Mux" }, + { "Right Line Mux", "Differential", "Differential Mux" }, + + { "Left PGA Mux", "Line 1", "LINPUT1" }, + { "Left PGA Mux", "Line 2", "LINPUT2" }, + { "Left PGA Mux", "Differential", "Differential Mux" }, + + { "Right PGA Mux", "Line 1", "RINPUT1" }, + { "Right PGA Mux", "Line 2", "RINPUT2" }, + { "Right PGA Mux", "Differential", "Differential Mux" }, + + { "Differential Mux", "Line 1", "LINPUT1" }, + { "Differential Mux", "Line 1", "RINPUT1" }, + { "Differential Mux", "Line 2", "LINPUT2" }, + { "Differential Mux", "Line 2", "RINPUT2" }, + + { "Left ADC Mux", "Stereo", "Left PGA Mux" }, + { "Left ADC Mux", "Mono (Left)", "Left PGA Mux" }, + { "Left ADC Mux", "Digital Mono", "Left PGA Mux" }, + + { "Right ADC Mux", "Stereo", "Right PGA Mux" }, + { "Right ADC Mux", "Mono (Right)", "Right PGA Mux" }, + { "Right ADC Mux", "Digital Mono", "Right PGA Mux" }, + + { "Left ADC", NULL, "Left ADC Mux" }, + { "Right ADC", NULL, "Right ADC Mux" }, + + { "Left Line Mux", "Line 1", "LINPUT1" }, + { "Left Line Mux", "Line 2", "LINPUT2" }, + { "Left Line Mux", "PGA", "Left PGA Mux" }, + { "Left Line Mux", "Differential", "Differential Mux" }, + + { "Right Line Mux", "Line 1", "RINPUT1" }, + { "Right Line Mux", "Line 2", "RINPUT2" }, + { "Right Line Mux", "PGA", "Right PGA Mux" }, + { "Right Line Mux", "Differential", "Differential Mux" }, + + { "Left Mixer", "Playback Switch", "Left DAC" }, + { "Left Mixer", "Left Bypass Switch", "Left Line Mux" }, + { "Left Mixer", "Right Playback Switch", "Right DAC" }, + { "Left Mixer", "Right Bypass Switch", "Right Line Mux" }, + + { "Right Mixer", "Left Playback Switch", "Left DAC" }, + { "Right Mixer", "Left Bypass Switch", "Left Line Mux" }, + { "Right Mixer", "Playback Switch", "Right DAC" }, + { "Right Mixer", "Right Bypass Switch", "Right Line Mux" }, + + { "Left Out 1", NULL, "Left Mixer" }, + { "LOUT1", NULL, "Left Out 1" }, + { "Right Out 1", NULL, "Right Mixer" }, + { "ROUT1", NULL, "Right Out 1" }, + + { "Left Out 2", NULL, "Left Mixer" }, + { "LOUT2", NULL, "Left Out 2" }, + { "Right Out 2", NULL, "Right Mixer" }, + { "ROUT2", NULL, "Right Out 2" }, +}; + +struct _coeff_div { + u32 mclk; + u32 rate; + u16 fs; + u8 sr:5; + u8 usb:1; +}; + +/* codec hifi mclk clock divider coefficients */ +static const struct _coeff_div coeff_div[] = { + /* 8k */ + {12288000, 8000, 1536, 0x6, 0x0}, + {11289600, 8000, 1408, 0x16, 0x0}, + {18432000, 8000, 2304, 0x7, 0x0}, + {16934400, 8000, 2112, 0x17, 0x0}, + {12000000, 8000, 1500, 0x6, 0x1}, + + /* 11.025k */ + {11289600, 11025, 1024, 0x18, 0x0}, + {16934400, 11025, 1536, 0x19, 0x0}, + {12000000, 11025, 1088, 0x19, 0x1}, + + /* 16k */ + {12288000, 16000, 768, 0xa, 0x0}, + {18432000, 16000, 1152, 0xb, 0x0}, + {12000000, 16000, 750, 0xa, 0x1}, + + /* 22.05k */ + {11289600, 22050, 512, 0x1a, 0x0}, + {16934400, 22050, 768, 0x1b, 0x0}, + {12000000, 22050, 544, 0x1b, 0x1}, + + /* 32k */ + {12288000, 32000, 384, 0xc, 0x0}, + {18432000, 32000, 576, 0xd, 0x0}, + {12000000, 32000, 375, 0xa, 0x1}, + + /* 44.1k */ + {11289600, 44100, 256, 0x10, 0x0}, + {16934400, 44100, 384, 0x11, 0x0}, + {12000000, 44100, 272, 0x11, 0x1}, + + /* 48k */ + {12288000, 48000, 256, 0x0, 0x0}, + {18432000, 48000, 384, 0x1, 0x0}, + {12000000, 48000, 250, 0x0, 0x1}, + + /* 88.2k */ + {11289600, 88200, 128, 0x1e, 0x0}, + {16934400, 88200, 192, 0x1f, 0x0}, + {12000000, 88200, 136, 0x1f, 0x1}, + + /* 96k */ + {12288000, 96000, 128, 0xe, 0x0}, + {18432000, 96000, 192, 0xf, 0x0}, + {12000000, 96000, 125, 0xe, 0x1}, +}; + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + + return -EINVAL; +} + +/* The set of rates we can generate from the above for each SYSCLK */ + +static unsigned int rates_12288[] = { + 8000, 12000, 16000, 24000, 24000, 32000, 48000, 96000, +}; + +static struct snd_pcm_hw_constraint_list constraints_12288 = { + .count = ARRAY_SIZE(rates_12288), + .list = rates_12288, +}; + +static unsigned int rates_112896[] = { + 8000, 11025, 22050, 44100, +}; + +static struct snd_pcm_hw_constraint_list constraints_112896 = { + .count = ARRAY_SIZE(rates_112896), + .list = rates_112896, +}; + +static unsigned int rates_12[] = { + 8000, 11025, 12000, 16000, 22050, 2400, 32000, 41100, 48000, + 48000, 88235, 96000, +}; + +static struct snd_pcm_hw_constraint_list constraints_12 = { + .count = ARRAY_SIZE(rates_12), + .list = rates_12, +}; + +/* + * Note that this should be called from init rather than from hw_params. + */ +static int wm8988_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 wm8988_priv *wm8988 = codec->private_data; + + switch (freq) { + case 11289600: + case 18432000: + case 22579200: + case 36864000: + wm8988->sysclk_constraints = &constraints_112896; + wm8988->sysclk = freq; + return 0; + + case 12288000: + case 16934400: + case 24576000: + case 33868800: + wm8988->sysclk_constraints = &constraints_12288; + wm8988->sysclk = freq; + return 0; + + case 12000000: + case 24000000: + wm8988->sysclk_constraints = &constraints_12; + wm8988->sysclk = freq; + return 0; + } + return -EINVAL; +} + +static int wm8988_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 = 0x0040; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x0002; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0001; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x0003; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= 0x0013; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x0090; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x0080; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x0010; + break; + default: + return -EINVAL; + } + + wm8988_write(codec, WM8988_IFACE, iface); + return 0; +} + +static int wm8988_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8988_priv *wm8988 = codec->private_data; + + /* The set of sample rates that can be supported depends on the + * MCLK supplied to the CODEC - enforce this. + */ + if (!wm8988->sysclk) { + dev_err(codec->dev, + "No MCLK configured, call set_sysclk() on init\n"); + return -EINVAL; + } + + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + wm8988->sysclk_constraints); + + return 0; +} + +static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct wm8988_priv *wm8988 = codec->private_data; + u16 iface = wm8988_read_reg_cache(codec, WM8988_IFACE) & 0x1f3; + u16 srate = wm8988_read_reg_cache(codec, WM8988_SRATE) & 0x180; + int coeff; + + coeff = get_coeff(wm8988->sysclk, params_rate(params)); + if (coeff < 0) { + coeff = get_coeff(wm8988->sysclk / 2, params_rate(params)); + srate |= 0x40; + } + if (coeff < 0) { + dev_err(codec->dev, + "Unable to configure sample rate %dHz with %dHz MCLK\n", + params_rate(params), wm8988->sysclk); + return coeff; + } + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x0004; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x0008; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= 0x000c; + break; + } + + /* set iface & srate */ + wm8988_write(codec, WM8988_IFACE, iface); + if (coeff >= 0) + wm8988_write(codec, WM8988_SRATE, srate | + (coeff_div[coeff].sr << 1) | coeff_div[coeff].usb); + + return 0; +} + +static int wm8988_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = wm8988_read_reg_cache(codec, WM8988_ADCDAC) & 0xfff7; + + if (mute) + wm8988_write(codec, WM8988_ADCDAC, mute_reg | 0x8); + else + wm8988_write(codec, WM8988_ADCDAC, mute_reg); + return 0; +} + +static int wm8988_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 pwr_reg = wm8988_read_reg_cache(codec, WM8988_PWR1) & ~0x1c1; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* VREF, VMID=2x50k, digital enabled */ + wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x00c0); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* VREF, VMID=2x5k */ + wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x1c1); + + /* Charge caps */ + msleep(100); + } + + /* VREF, VMID=2*500k, digital stopped */ + wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x0141); + break; + + case SND_SOC_BIAS_OFF: + wm8988_write(codec, WM8988_PWR1, 0x0000); + break; + } + codec->bias_level = level; + return 0; +} + +#define WM8988_RATES SNDRV_PCM_RATE_8000_96000 + +#define WM8988_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops wm8988_ops = { + .startup = wm8988_pcm_startup, + .hw_params = wm8988_pcm_hw_params, + .set_fmt = wm8988_set_dai_fmt, + .set_sysclk = wm8988_set_dai_sysclk, + .digital_mute = wm8988_mute, +}; + +struct snd_soc_dai wm8988_dai = { + .name = "WM8988", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8988_RATES, + .formats = WM8988_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8988_RATES, + .formats = WM8988_FORMATS, + }, + .ops = &wm8988_ops, + .symmetric_rates = 1, +}; +EXPORT_SYMBOL_GPL(wm8988_dai); + +static int wm8988_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8988_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8988_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < WM8988_NUM_REG; i++) { + if (i == WM8988_RESET) + continue; + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + + wm8988_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static struct snd_soc_codec *wm8988_codec; + +static int wm8988_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8988_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8988_codec; + codec = wm8988_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, wm8988_snd_controls, + ARRAY_SIZE(wm8988_snd_controls)); + snd_soc_dapm_new_controls(codec, wm8988_dapm_widgets, + ARRAY_SIZE(wm8988_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_new_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +static int wm8988_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8988 = { + .probe = wm8988_probe, + .remove = wm8988_remove, + .suspend = wm8988_suspend, + .resume = wm8988_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8988); + +static int wm8988_register(struct wm8988_priv *wm8988) +{ + struct snd_soc_codec *codec = &wm8988->codec; + int ret; + u16 reg; + + if (wm8988_codec) { + dev_err(codec->dev, "Another WM8988 is registered\n"); + ret = -EINVAL; + goto err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8988; + codec->name = "WM8988"; + codec->owner = THIS_MODULE; + codec->read = wm8988_read_reg_cache; + codec->write = wm8988_write; + codec->dai = &wm8988_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(wm8988->reg_cache); + codec->reg_cache = &wm8988->reg_cache; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8988_set_bias_level; + + memcpy(codec->reg_cache, wm8988_reg, + sizeof(wm8988_reg)); + + ret = wm8988_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + return ret; + } + + /* set the update bits (we always update left then right) */ + reg = wm8988_read_reg_cache(codec, WM8988_RADC); + wm8988_write(codec, WM8988_RADC, reg | 0x100); + reg = wm8988_read_reg_cache(codec, WM8988_RDAC); + wm8988_write(codec, WM8988_RDAC, reg | 0x0100); + reg = wm8988_read_reg_cache(codec, WM8988_ROUT1V); + wm8988_write(codec, WM8988_ROUT1V, reg | 0x0100); + reg = wm8988_read_reg_cache(codec, WM8988_ROUT2V); + wm8988_write(codec, WM8988_ROUT2V, reg | 0x0100); + reg = wm8988_read_reg_cache(codec, WM8988_RINVOL); + wm8988_write(codec, WM8988_RINVOL, reg | 0x0100); + + wm8988_set_bias_level(&wm8988->codec, SND_SOC_BIAS_STANDBY); + + wm8988_dai.dev = codec->dev; + + wm8988_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + ret = snd_soc_register_dai(&wm8988_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + return ret; + } + + return 0; + +err: + kfree(wm8988); + return ret; +} + +static void wm8988_unregister(struct wm8988_priv *wm8988) +{ + wm8988_set_bias_level(&wm8988->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&wm8988_dai); + snd_soc_unregister_codec(&wm8988->codec); + kfree(wm8988); + wm8988_codec = NULL; +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static int wm8988_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8988_priv *wm8988; + struct snd_soc_codec *codec; + + wm8988 = kzalloc(sizeof(struct wm8988_priv), GFP_KERNEL); + if (wm8988 == NULL) + return -ENOMEM; + + codec = &wm8988->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8988); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + + return wm8988_register(wm8988); +} + +static int wm8988_i2c_remove(struct i2c_client *client) +{ + struct wm8988_priv *wm8988 = i2c_get_clientdata(client); + wm8988_unregister(wm8988); + return 0; +} + +static const struct i2c_device_id wm8988_i2c_id[] = { + { "wm8988", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8988_i2c_id); + +static struct i2c_driver wm8988_i2c_driver = { + .driver = { + .name = "WM8988", + .owner = THIS_MODULE, + }, + .probe = wm8988_i2c_probe, + .remove = wm8988_i2c_remove, + .id_table = wm8988_i2c_id, +}; +#endif + +#if defined(CONFIG_SPI_MASTER) +static int wm8988_spi_write(struct spi_device *spi, const char *data, int len) +{ + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; + + if (len <= 0) + return 0; + + msg[0] = data[0]; + msg[1] = data[1]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} + +static int __devinit wm8988_spi_probe(struct spi_device *spi) +{ + struct wm8988_priv *wm8988; + struct snd_soc_codec *codec; + + wm8988 = kzalloc(sizeof(struct wm8988_priv), GFP_KERNEL); + if (wm8988 == NULL) + return -ENOMEM; + + codec = &wm8988->codec; + codec->hw_write = (hw_write_t)wm8988_spi_write; + codec->control_data = spi; + codec->dev = &spi->dev; + + spi->dev.driver_data = wm8988; + + return wm8988_register(wm8988); +} + +static int __devexit wm8988_spi_remove(struct spi_device *spi) +{ + struct wm8988_priv *wm8988 = spi->dev.driver_data; + + wm8988_unregister(wm8988); + + return 0; +} + +static struct spi_driver wm8988_spi_driver = { + .driver = { + .name = "wm8988", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8988_spi_probe, + .remove = __devexit_p(wm8988_spi_remove), +}; +#endif + +static int __init wm8988_modinit(void) +{ + int ret; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&wm8988_i2c_driver); + if (ret != 0) + pr_err("WM8988: Unable to register I2C driver: %d\n", ret); +#endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&wm8988_spi_driver); + if (ret != 0) + pr_err("WM8988: Unable to register SPI driver: %d\n", ret); +#endif + return ret; +} +module_init(wm8988_modinit); + +static void __exit wm8988_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8988_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8988_spi_driver); +#endif +} +module_exit(wm8988_exit); + + +MODULE_DESCRIPTION("ASoC WM8988 driver"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8988.h b/sound/soc/codecs/wm8988.h new file mode 100644 index 000000000000..4552d37fdd41 --- /dev/null +++ b/sound/soc/codecs/wm8988.h @@ -0,0 +1,60 @@ +/* + * Copyright 2005 Openedhand Ltd. + * + * Author: Richard Purdie + * + * Based on WM8753.h + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef _WM8988_H +#define _WM8988_H + +/* WM8988 register space */ + +#define WM8988_LINVOL 0x00 +#define WM8988_RINVOL 0x01 +#define WM8988_LOUT1V 0x02 +#define WM8988_ROUT1V 0x03 +#define WM8988_ADCDAC 0x05 +#define WM8988_IFACE 0x07 +#define WM8988_SRATE 0x08 +#define WM8988_LDAC 0x0a +#define WM8988_RDAC 0x0b +#define WM8988_BASS 0x0c +#define WM8988_TREBLE 0x0d +#define WM8988_RESET 0x0f +#define WM8988_3D 0x10 +#define WM8988_ALC1 0x11 +#define WM8988_ALC2 0x12 +#define WM8988_ALC3 0x13 +#define WM8988_NGATE 0x14 +#define WM8988_LADC 0x15 +#define WM8988_RADC 0x16 +#define WM8988_ADCTL1 0x17 +#define WM8988_ADCTL2 0x18 +#define WM8988_PWR1 0x19 +#define WM8988_PWR2 0x1a +#define WM8988_ADCTL3 0x1b +#define WM8988_ADCIN 0x1f +#define WM8988_LADCIN 0x20 +#define WM8988_RADCIN 0x21 +#define WM8988_LOUTM1 0x22 +#define WM8988_LOUTM2 0x23 +#define WM8988_ROUTM1 0x24 +#define WM8988_ROUTM2 0x25 +#define WM8988_LOUT2V 0x28 +#define WM8988_ROUT2V 0x29 +#define WM8988_LPPB 0x43 +#define WM8988_NUM_REG 0x44 + +#define WM8988_SYSCLK 0 + +extern struct snd_soc_dai wm8988_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8988; + +#endif -- cgit v1.2.3 From 894bf92fdec9909fefcfe907786c6c6944a22052 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 9 Apr 2009 12:34:40 +0300 Subject: ASoC: tlv320aic23: add DSP_A format support Add DSP_A interface format support by setting the LRP bit in DSP mode. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index c3f4afb5d017..21f69df9994c 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -523,6 +523,8 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, case SND_SOC_DAIFMT_I2S: iface_reg |= TLV320AIC23_FOR_I2S; break; + case SND_SOC_DAIFMT_DSP_A: + iface_reg |= TLV320AIC23_LRP_ON; case SND_SOC_DAIFMT_DSP_B: iface_reg |= TLV320AIC23_FOR_DSP; break; -- cgit v1.2.3 From f4976116a98f108bf385f217332aadb3ca98fe66 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 13 Apr 2009 10:53:02 +0100 Subject: ASoC: WM9713 requires symmetric rates on the voice DAI Signed-off-by: Mark Brown --- sound/soc/codecs/wm9713.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 523bad077fa0..aa94cc68f0c0 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1069,6 +1069,7 @@ struct snd_soc_dai wm9713_dai[] = { .rates = WM9713_PCM_RATES, .formats = WM9713_PCM_FORMATS,}, .ops = &wm9713_dai_ops_voice, + .symmetric_rates = 1, }, }; EXPORT_SYMBOL_GPL(wm9713_dai); -- cgit v1.2.3 From 6bbcb459cd50807511491ddf96bca1ef92006bf8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 13 Apr 2009 11:29:10 +0100 Subject: ASoC: Move the WM9713 voice DAC powerdown to a DAPM event This ensures that we sync with the DAPM powerdown sequencing properly and don't need to bounce the power on the voice DAC so often. Signed-off-by: Mark Brown --- sound/soc/codecs/wm9713.c | 39 ++++++++++++++++++++++----------------- 1 file changed, 22 insertions(+), 17 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index aa94cc68f0c0..a6feb7842314 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -189,6 +189,26 @@ SOC_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0), SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1), }; +static int wm9713_voice_shutdown(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + u16 status, rate; + + BUG_ON(event != SND_SOC_DAPM_PRE_PMD); + + /* Gracefully shut down the voice interface. */ + status = ac97_read(codec, AC97_EXTENDED_MID) | 0x1000; + rate = ac97_read(codec, AC97_HANDSET_RATE) & 0xF0FF; + ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0200); + schedule_timeout_interruptible(msecs_to_jiffies(1)); + ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0F00); + ac97_write(codec, AC97_EXTENDED_MID, status); + + return 0; +} + + /* We have to create a fake left and right HP mixers because * the codec only has a single control that is shared by both channels. * This makes it impossible to determine the audio path using the current @@ -400,7 +420,8 @@ SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("HP Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("Line Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("Capture Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), -SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1), +SND_SOC_DAPM_DAC_E("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1, + wm9713_voice_shutdown, SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", AC97_EXTENDED_MID, 11, 1), SND_SOC_DAPM_PGA("Left ADC", AC97_EXTENDED_MID, 5, 1, NULL, 0), SND_SOC_DAPM_PGA("Right ADC", AC97_EXTENDED_MID, 4, 1, NULL, 0), @@ -936,21 +957,6 @@ static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static void wm9713_voiceshutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - u16 status, rate; - - /* Gracefully shut down the voice interface. */ - status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000; - rate = ac97_read(codec, AC97_HANDSET_RATE) & 0xF0FF; - ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0200); - schedule_timeout_interruptible(msecs_to_jiffies(1)); - ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0F00); - ac97_write(codec, AC97_EXTENDED_MID, status); -} - static int ac97_hifi_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -1019,7 +1025,6 @@ static struct snd_soc_dai_ops wm9713_dai_ops_aux = { static struct snd_soc_dai_ops wm9713_dai_ops_voice = { .hw_params = wm9713_pcm_hw_params, - .shutdown = wm9713_voiceshutdown, .set_clkdiv = wm9713_set_dai_clkdiv, .set_pll = wm9713_set_dai_pll, .set_fmt = wm9713_set_dai_fmt, -- cgit v1.2.3 From f2644a2c00a06236a9c5e85488b0680825bad39c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 7 Apr 2009 19:20:14 +0100 Subject: ASoC: Add WM8960 CODEC driver The WM8960 is a low power, high quality stereo codec designed for portable digital audio applications. Stereo class D speaker drivers provide 1W per channel into 8W loads. Guaranteed low leakage, excellent PSRR and pop/click suppression mechanisms enable direct battery connection for the speaker supply. The device also integrates a complete microphone interface and a stereo headphone driver. External component requirements are drastically reduced as no separate microphone, speaker or headphone amplifiers are required. Advanced on-chip digital signal processing performs automatic level control for the microphone or line input. Stereo 24-bit sigma-delta ADCs and DACs are used with low power over-sampling digital interpolation and decimation filters and a flexible digital audio interface. The master clock can be input directly or generated internally by an onboard PLL, supporting most commonly-used clocking schemes. This driver was originally written by Liam Girdwood, with substantial subsequent additions and updates for feature completeness and changes in the ASoC framework from me. Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8960.c | 969 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8960.h | 127 ++++++ 4 files changed, 1102 insertions(+) create mode 100644 sound/soc/codecs/wm8960.c create mode 100644 sound/soc/codecs/wm8960.h (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ab364854675b..121d63f13dbb 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -35,6 +35,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8900 if I2C select SND_SOC_WM8903 if I2C + select SND_SOC_WM8960 if I2C select SND_SOC_WM8971 if I2C select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8990 if I2C @@ -139,6 +140,9 @@ config SND_SOC_WM8900 config SND_SOC_WM8903 tristate +config SND_SOC_WM8960 + tristate + config SND_SOC_WM8971 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a72548dc1885..d8e15a47711e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -23,6 +23,7 @@ snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o +snd-soc-wm8960-objs := wm8960.o snd-soc-wm8971-objs := wm8971.o snd-soc-wm8988-objs := wm8988.o snd-soc-wm8990-objs := wm8990.o @@ -56,6 +57,7 @@ obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o +obj-$(CONFIG_SND_SOC_WM8960) += snd-soc-wm8960.o obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.o diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c new file mode 100644 index 000000000000..e224d8add170 --- /dev/null +++ b/sound/soc/codecs/wm8960.c @@ -0,0 +1,969 @@ +/* + * wm8960.c -- WM8960 ALSA SoC Audio driver + * + * Author: Liam Girdwood + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm8960.h" + +#define AUDIO_NAME "wm8960" + +struct snd_soc_codec_device soc_codec_dev_wm8960; + +/* R25 - Power 1 */ +#define WM8960_VREF 0x40 + +/* R28 - Anti-pop 1 */ +#define WM8960_POBCTRL 0x80 +#define WM8960_BUFDCOPEN 0x10 +#define WM8960_BUFIOEN 0x08 +#define WM8960_SOFT_ST 0x04 +#define WM8960_HPSTBY 0x01 + +/* R29 - Anti-pop 2 */ +#define WM8960_DISOP 0x40 + +/* + * wm8960 register cache + * We can't read the WM8960 register space when we are + * using 2 wire for device control, so we cache them instead. + */ +static const u16 wm8960_reg[WM8960_CACHEREGNUM] = { + 0x0097, 0x0097, 0x0000, 0x0000, + 0x0000, 0x0008, 0x0000, 0x000a, + 0x01c0, 0x0000, 0x00ff, 0x00ff, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x007b, 0x0100, 0x0032, + 0x0000, 0x00c3, 0x00c3, 0x01c0, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0100, 0x0100, 0x0050, 0x0050, + 0x0050, 0x0050, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0040, 0x0000, + 0x0000, 0x0050, 0x0050, 0x0000, + 0x0002, 0x0037, 0x004d, 0x0080, + 0x0008, 0x0031, 0x0026, 0x00e9, +}; + +struct wm8960_priv { + u16 reg_cache[WM8960_CACHEREGNUM]; + struct snd_soc_codec codec; +}; + +/* + * read wm8960 register cache + */ +static inline unsigned int wm8960_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg == WM8960_RESET) + return 0; + if (reg >= WM8960_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write wm8960 register cache + */ +static inline void wm8960_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= WM8960_CACHEREGNUM) + return; + cache[reg] = value; +} + +static inline unsigned int wm8960_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + return wm8960_read_reg_cache(codec, reg); +} + +/* + * write to the WM8960 register space + */ +static int wm8960_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D9 WM8960 register offset + * D8...D0 register data + */ + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + wm8960_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +#define wm8960_reset(c) wm8960_write(c, WM8960_RESET, 0) + +/* enumerated controls */ +static const char *wm8960_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; +static const char *wm8960_polarity[] = {"No Inversion", "Left Inverted", + "Right Inverted", "Stereo Inversion"}; +static const char *wm8960_3d_upper_cutoff[] = {"High", "Low"}; +static const char *wm8960_3d_lower_cutoff[] = {"Low", "High"}; +static const char *wm8960_alcfunc[] = {"Off", "Right", "Left", "Stereo"}; +static const char *wm8960_alcmode[] = {"ALC", "Limiter"}; + +static const struct soc_enum wm8960_enum[] = { + SOC_ENUM_SINGLE(WM8960_DACCTL1, 1, 4, wm8960_deemph), + SOC_ENUM_SINGLE(WM8960_DACCTL1, 5, 4, wm8960_polarity), + SOC_ENUM_SINGLE(WM8960_DACCTL2, 5, 4, wm8960_polarity), + SOC_ENUM_SINGLE(WM8960_3D, 6, 2, wm8960_3d_upper_cutoff), + SOC_ENUM_SINGLE(WM8960_3D, 5, 2, wm8960_3d_lower_cutoff), + SOC_ENUM_SINGLE(WM8960_ALC1, 7, 4, wm8960_alcfunc), + SOC_ENUM_SINGLE(WM8960_ALC3, 8, 2, wm8960_alcmode), +}; + +static const DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 50, 0); +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 50, 1); +static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0); +static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); + +static const struct snd_kcontrol_new wm8960_snd_controls[] = { +SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL, + 0, 63, 0, adc_tlv), +SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL, + 6, 1, 0), +SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL, + 7, 1, 0), + +SOC_DOUBLE_R_TLV("Playback Volume", WM8960_LDAC, WM8960_RDAC, + 0, 255, 0, dac_tlv), + +SOC_DOUBLE_R_TLV("Headphone Playback Volume", WM8960_LOUT1, WM8960_ROUT1, + 0, 127, 0, out_tlv), +SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8960_LOUT1, WM8960_ROUT1, + 7, 1, 0), + +SOC_DOUBLE_R_TLV("Speaker Playback Volume", WM8960_LOUT2, WM8960_ROUT2, + 0, 127, 0, out_tlv), +SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8960_LOUT2, WM8960_ROUT2, + 7, 1, 0), +SOC_SINGLE("Speaker DC Volume", WM8960_CLASSD3, 3, 5, 0), +SOC_SINGLE("Speaker AC Volume", WM8960_CLASSD3, 0, 5, 0), + +SOC_SINGLE("PCM Playback -6dB Switch", WM8960_DACCTL1, 7, 1, 0), +SOC_ENUM("ADC Polarity", wm8960_enum[1]), +SOC_ENUM("Playback De-emphasis", wm8960_enum[0]), +SOC_SINGLE("ADC High Pass Filter Switch", WM8960_DACCTL1, 0, 1, 0), + +SOC_ENUM("DAC Polarity", wm8960_enum[2]), + +SOC_ENUM("3D Filter Upper Cut-Off", wm8960_enum[3]), +SOC_ENUM("3D Filter Lower Cut-Off", wm8960_enum[4]), +SOC_SINGLE("3D Volume", WM8960_3D, 1, 15, 0), +SOC_SINGLE("3D Switch", WM8960_3D, 0, 1, 0), + +SOC_ENUM("ALC Function", wm8960_enum[5]), +SOC_SINGLE("ALC Max Gain", WM8960_ALC1, 4, 7, 0), +SOC_SINGLE("ALC Target", WM8960_ALC1, 0, 15, 1), +SOC_SINGLE("ALC Min Gain", WM8960_ALC2, 4, 7, 0), +SOC_SINGLE("ALC Hold Time", WM8960_ALC2, 0, 15, 0), +SOC_ENUM("ALC Mode", wm8960_enum[6]), +SOC_SINGLE("ALC Decay", WM8960_ALC3, 4, 15, 0), +SOC_SINGLE("ALC Attack", WM8960_ALC3, 0, 15, 0), + +SOC_SINGLE("Noise Gate Threshold", WM8960_NOISEG, 3, 31, 0), +SOC_SINGLE("Noise Gate Switch", WM8960_NOISEG, 0, 1, 0), + +SOC_DOUBLE_R("ADC PCM Capture Volume", WM8960_LINPATH, WM8960_RINPATH, + 0, 127, 0), + +SOC_SINGLE_TLV("Left Output Mixer Boost Bypass Volume", + WM8960_BYPASS1, 4, 7, 1, bypass_tlv), +SOC_SINGLE_TLV("Left Output Mixer LINPUT3 Volume", + WM8960_LOUTMIX, 4, 7, 1, bypass_tlv), +SOC_SINGLE_TLV("Right Output Mixer Boost Bypass Volume", + WM8960_BYPASS2, 4, 7, 1, bypass_tlv), +SOC_SINGLE_TLV("Right Output Mixer RINPUT3 Volume", + WM8960_ROUTMIX, 4, 7, 1, bypass_tlv), +}; + +static const struct snd_kcontrol_new wm8960_lin_boost[] = { +SOC_DAPM_SINGLE("LINPUT2 Switch", WM8960_LINPATH, 6, 1, 0), +SOC_DAPM_SINGLE("LINPUT3 Switch", WM8960_LINPATH, 7, 1, 0), +SOC_DAPM_SINGLE("LINPUT1 Switch", WM8960_LINPATH, 8, 1, 0), +}; + +static const struct snd_kcontrol_new wm8960_lin[] = { +SOC_DAPM_SINGLE("Boost Switch", WM8960_LINPATH, 3, 1, 0), +}; + +static const struct snd_kcontrol_new wm8960_rin_boost[] = { +SOC_DAPM_SINGLE("RINPUT2 Switch", WM8960_RINPATH, 6, 1, 0), +SOC_DAPM_SINGLE("RINPUT3 Switch", WM8960_RINPATH, 7, 1, 0), +SOC_DAPM_SINGLE("RINPUT1 Switch", WM8960_RINPATH, 8, 1, 0), +}; + +static const struct snd_kcontrol_new wm8960_rin[] = { +SOC_DAPM_SINGLE("Boost Switch", WM8960_RINPATH, 3, 1, 0), +}; + +static const struct snd_kcontrol_new wm8960_loutput_mixer[] = { +SOC_DAPM_SINGLE("PCM Playback Switch", WM8960_LOUTMIX, 8, 1, 0), +SOC_DAPM_SINGLE("LINPUT3 Switch", WM8960_LOUTMIX, 7, 1, 0), +SOC_DAPM_SINGLE("Boost Bypass Switch", WM8960_BYPASS1, 7, 1, 0), +}; + +static const struct snd_kcontrol_new wm8960_routput_mixer[] = { +SOC_DAPM_SINGLE("PCM Playback Switch", WM8960_ROUTMIX, 8, 1, 0), +SOC_DAPM_SINGLE("RINPUT3 Switch", WM8960_ROUTMIX, 7, 1, 0), +SOC_DAPM_SINGLE("Boost Bypass Switch", WM8960_BYPASS2, 7, 1, 0), +}; + +static const struct snd_kcontrol_new wm8960_mono_out[] = { +SOC_DAPM_SINGLE("Left Switch", WM8960_MONOMIX1, 7, 1, 0), +SOC_DAPM_SINGLE("Right Switch", WM8960_MONOMIX2, 7, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8960_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("LINPUT1"), +SND_SOC_DAPM_INPUT("RINPUT1"), +SND_SOC_DAPM_INPUT("LINPUT2"), +SND_SOC_DAPM_INPUT("RINPUT2"), +SND_SOC_DAPM_INPUT("LINPUT3"), +SND_SOC_DAPM_INPUT("RINPUT3"), + +SND_SOC_DAPM_MICBIAS("MICB", WM8960_POWER1, 1, 0), + +SND_SOC_DAPM_MIXER("Left Boost Mixer", WM8960_POWER1, 5, 0, + wm8960_lin_boost, ARRAY_SIZE(wm8960_lin_boost)), +SND_SOC_DAPM_MIXER("Right Boost Mixer", WM8960_POWER1, 4, 0, + wm8960_rin_boost, ARRAY_SIZE(wm8960_rin_boost)), + +SND_SOC_DAPM_MIXER("Left Input Mixer", WM8960_POWER3, 5, 0, + wm8960_lin, ARRAY_SIZE(wm8960_lin)), +SND_SOC_DAPM_MIXER("Right Input Mixer", WM8960_POWER3, 4, 0, + wm8960_rin, ARRAY_SIZE(wm8960_rin)), + +SND_SOC_DAPM_ADC("Left ADC", "Capture", WM8960_POWER2, 3, 0), +SND_SOC_DAPM_ADC("Right ADC", "Capture", WM8960_POWER2, 2, 0), + +SND_SOC_DAPM_DAC("Left DAC", "Playback", WM8960_POWER2, 8, 0), +SND_SOC_DAPM_DAC("Right DAC", "Playback", WM8960_POWER2, 7, 0), + +SND_SOC_DAPM_MIXER("Left Output Mixer", WM8960_POWER3, 3, 0, + &wm8960_loutput_mixer[0], + ARRAY_SIZE(wm8960_loutput_mixer)), +SND_SOC_DAPM_MIXER("Right Output Mixer", WM8960_POWER3, 2, 0, + &wm8960_routput_mixer[0], + ARRAY_SIZE(wm8960_routput_mixer)), + +SND_SOC_DAPM_MIXER("Mono Output Mixer", WM8960_POWER2, 1, 0, + &wm8960_mono_out[0], + ARRAY_SIZE(wm8960_mono_out)), + +SND_SOC_DAPM_PGA("LOUT1 PGA", WM8960_POWER2, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("ROUT1 PGA", WM8960_POWER2, 5, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Left Speaker PGA", WM8960_POWER2, 4, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right Speaker PGA", WM8960_POWER2, 3, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Right Speaker Output", WM8960_CLASSD1, 7, 0, NULL, 0), +SND_SOC_DAPM_PGA("Left Speaker Output", WM8960_CLASSD1, 6, 0, NULL, 0), + +SND_SOC_DAPM_OUTPUT("SPK_LP"), +SND_SOC_DAPM_OUTPUT("SPK_LN"), +SND_SOC_DAPM_OUTPUT("HP_L"), +SND_SOC_DAPM_OUTPUT("HP_R"), +SND_SOC_DAPM_OUTPUT("SPK_RP"), +SND_SOC_DAPM_OUTPUT("SPK_RN"), +SND_SOC_DAPM_OUTPUT("OUT3"), +}; + +static const struct snd_soc_dapm_route audio_paths[] = { + { "Left Boost Mixer", "LINPUT1 Switch", "LINPUT1" }, + { "Left Boost Mixer", "LINPUT2 Switch", "LINPUT2" }, + { "Left Boost Mixer", "LINPUT3 Switch", "LINPUT3" }, + + { "Left Input Mixer", "Boost Switch", "Left Boost Mixer", }, + { "Left Input Mixer", NULL, "LINPUT1", }, /* Really Boost Switch */ + { "Left Input Mixer", NULL, "LINPUT2" }, + { "Left Input Mixer", NULL, "LINPUT3" }, + + { "Right Boost Mixer", "RINPUT1 Switch", "RINPUT1" }, + { "Right Boost Mixer", "RINPUT2 Switch", "RINPUT2" }, + { "Right Boost Mixer", "RINPUT3 Switch", "RINPUT3" }, + + { "Right Input Mixer", "Boost Switch", "Right Boost Mixer", }, + { "Right Input Mixer", NULL, "RINPUT1", }, /* Really Boost Switch */ + { "Right Input Mixer", NULL, "RINPUT2" }, + { "Right Input Mixer", NULL, "LINPUT3" }, + + { "Left ADC", NULL, "Left Input Mixer" }, + { "Right ADC", NULL, "Right Input Mixer" }, + + { "Left Output Mixer", "LINPUT3 Switch", "LINPUT3" }, + { "Left Output Mixer", "Boost Bypass Switch", "Left Boost Mixer"} , + { "Left Output Mixer", "PCM Playback Switch", "Left DAC" }, + + { "Right Output Mixer", "RINPUT3 Switch", "RINPUT3" }, + { "Right Output Mixer", "Boost Bypass Switch", "Right Boost Mixer" } , + { "Right Output Mixer", "PCM Playback Switch", "Right DAC" }, + + { "Mono Output Mixer", "Left Switch", "Left Output Mixer" }, + { "Mono Output Mixer", "Right Switch", "Right Output Mixer" }, + + { "LOUT1 PGA", NULL, "Left Output Mixer" }, + { "ROUT1 PGA", NULL, "Right Output Mixer" }, + + { "HP_L", NULL, "LOUT1 PGA" }, + { "HP_R", NULL, "ROUT1 PGA" }, + + { "Left Speaker PGA", NULL, "Left Output Mixer" }, + { "Right Speaker PGA", NULL, "Right Output Mixer" }, + + { "Left Speaker Output", NULL, "Left Speaker PGA" }, + { "Right Speaker Output", NULL, "Right Speaker PGA" }, + + { "SPK_LN", NULL, "Left Speaker Output" }, + { "SPK_LP", NULL, "Left Speaker Output" }, + { "SPK_RN", NULL, "Right Speaker Output" }, + { "SPK_RP", NULL, "Right Speaker Output" }, + + { "OUT3", NULL, "Mono Output Mixer", } +}; + +static int wm8960_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8960_dapm_widgets, + ARRAY_SIZE(wm8960_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int wm8960_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 |= 0x0040; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x0002; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0001; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x0003; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= 0x0013; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x0090; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x0080; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x0010; + break; + default: + return -EINVAL; + } + + /* set iface */ + wm8960_write(codec, WM8960_IFACE1, iface); + return 0; +} + +static int wm8960_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + u16 iface = wm8960_read(codec, WM8960_IFACE1) & 0xfff3; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x0004; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x0008; + break; + } + + /* set iface */ + wm8960_write(codec, WM8960_IFACE1, iface); + return 0; +} + +static int wm8960_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = wm8960_read(codec, WM8960_DACCTL1) & 0xfff7; + + if (mute) + wm8960_write(codec, WM8960_DACCTL1, mute_reg | 0x8); + else + wm8960_write(codec, WM8960_DACCTL1, mute_reg); + return 0; +} + +static int wm8960_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm8960_data *pdata = codec->dev->platform_data; + u16 reg; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* Set VMID to 2x50k */ + reg = wm8960_read(codec, WM8960_POWER1); + reg &= ~0x180; + reg |= 0x80; + wm8960_write(codec, WM8960_POWER1, reg); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Enable anti-pop features */ + wm8960_write(codec, WM8960_APOP1, + WM8960_POBCTRL | WM8960_SOFT_ST | + WM8960_BUFDCOPEN | WM8960_BUFIOEN); + + /* Discharge HP output */ + reg = WM8960_DISOP; + if (pdata) + reg |= pdata->dres << 4; + wm8960_write(codec, WM8960_APOP2, reg); + + msleep(400); + + wm8960_write(codec, WM8960_APOP2, 0); + + /* Enable & ramp VMID at 2x50k */ + reg = wm8960_read(codec, WM8960_POWER1); + reg |= 0x80; + wm8960_write(codec, WM8960_POWER1, reg); + msleep(100); + + /* Enable VREF */ + wm8960_write(codec, WM8960_POWER1, reg | WM8960_VREF); + + /* Disable anti-pop features */ + wm8960_write(codec, WM8960_APOP1, WM8960_BUFIOEN); + } + + /* Set VMID to 2x250k */ + reg = wm8960_read(codec, WM8960_POWER1); + reg &= ~0x180; + reg |= 0x100; + wm8960_write(codec, WM8960_POWER1, reg); + break; + + case SND_SOC_BIAS_OFF: + /* Enable anti-pop features */ + wm8960_write(codec, WM8960_APOP1, + WM8960_POBCTRL | WM8960_SOFT_ST | + WM8960_BUFDCOPEN | WM8960_BUFIOEN); + + /* Disable VMID and VREF, let them discharge */ + wm8960_write(codec, WM8960_POWER1, 0); + msleep(600); + + wm8960_write(codec, WM8960_APOP1, 0); + break; + } + + codec->bias_level = level; + + return 0; +} + +/* PLL divisors */ +struct _pll_div { + u32 pre_div:1; + u32 n:4; + u32 k:24; +}; + +/* The size in bits of the pll divide multiplied by 10 + * to allow rounding later */ +#define FIXED_PLL_SIZE ((1 << 24) * 10) + +static int pll_factors(unsigned int source, unsigned int target, + struct _pll_div *pll_div) +{ + unsigned long long Kpart; + unsigned int K, Ndiv, Nmod; + + pr_debug("WM8960 PLL: setting %dHz->%dHz\n", source, target); + + /* Scale up target to PLL operating frequency */ + target *= 4; + + Ndiv = target / source; + if (Ndiv < 6) { + source >>= 1; + pll_div->pre_div = 1; + Ndiv = target / source; + } else + pll_div->pre_div = 0; + + if ((Ndiv < 6) || (Ndiv > 12)) { + pr_err("WM8960 PLL: Unsupported N=%d\n", Ndiv); + return -EINVAL; + } + + pll_div->n = Ndiv; + Nmod = target % source; + Kpart = FIXED_PLL_SIZE * (long long)Nmod; + + do_div(Kpart, source); + + K = Kpart & 0xFFFFFFFF; + + /* Check if we need to round */ + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + K /= 10; + + pll_div->k = K; + + pr_debug("WM8960 PLL: N=%x K=%x pre_div=%d\n", + pll_div->n, pll_div->k, pll_div->pre_div); + + return 0; +} + +static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + static struct _pll_div pll_div; + int ret; + + if (freq_in && freq_out) { + ret = pll_factors(freq_in, freq_out, &pll_div); + if (ret != 0) + return ret; + } + + /* Disable the PLL: even if we are changing the frequency the + * PLL needs to be disabled while we do so. */ + wm8960_write(codec, WM8960_CLOCK1, + wm8960_read(codec, WM8960_CLOCK1) & ~1); + wm8960_write(codec, WM8960_POWER2, + wm8960_read(codec, WM8960_POWER2) & ~1); + + if (!freq_in || !freq_out) + return 0; + + reg = wm8960_read(codec, WM8960_PLL1) & ~0x3f; + reg |= pll_div.pre_div << 4; + reg |= pll_div.n; + + if (pll_div.k) { + reg |= 0x20; + + wm8960_write(codec, WM8960_PLL2, (pll_div.k >> 18) & 0x3f); + wm8960_write(codec, WM8960_PLL3, (pll_div.k >> 9) & 0x1ff); + wm8960_write(codec, WM8960_PLL4, pll_div.k & 0x1ff); + } + wm8960_write(codec, WM8960_PLL1, reg); + + /* Turn it on */ + wm8960_write(codec, WM8960_POWER2, + wm8960_read(codec, WM8960_POWER2) | 1); + msleep(250); + wm8960_write(codec, WM8960_CLOCK1, + wm8960_read(codec, WM8960_CLOCK1) | 1); + + return 0; +} + +static int wm8960_set_dai_clkdiv(struct snd_soc_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + switch (div_id) { + case WM8960_SYSCLKSEL: + reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1fe; + wm8960_write(codec, WM8960_CLOCK1, reg | div); + break; + case WM8960_SYSCLKDIV: + reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1f9; + wm8960_write(codec, WM8960_CLOCK1, reg | div); + break; + case WM8960_DACDIV: + reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1c7; + wm8960_write(codec, WM8960_CLOCK1, reg | div); + break; + case WM8960_OPCLKDIV: + reg = wm8960_read(codec, WM8960_PLL1) & 0x03f; + wm8960_write(codec, WM8960_PLL1, reg | div); + break; + case WM8960_DCLKDIV: + reg = wm8960_read(codec, WM8960_CLOCK2) & 0x03f; + wm8960_write(codec, WM8960_CLOCK2, reg | div); + break; + case WM8960_TOCLKSEL: + reg = wm8960_read(codec, WM8960_ADDCTL1) & 0x1fd; + wm8960_write(codec, WM8960_ADDCTL1, reg | div); + break; + default: + return -EINVAL; + } + + return 0; +} + +#define WM8960_RATES SNDRV_PCM_RATE_8000_48000 + +#define WM8960_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops wm8960_dai_ops = { + .hw_params = wm8960_hw_params, + .digital_mute = wm8960_mute, + .set_fmt = wm8960_set_dai_fmt, + .set_clkdiv = wm8960_set_dai_clkdiv, + .set_pll = wm8960_set_dai_pll, +}; + +struct snd_soc_dai wm8960_dai = { + .name = "WM8960", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8960_RATES, + .formats = WM8960_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8960_RATES, + .formats = WM8960_FORMATS,}, + .ops = &wm8960_dai_ops, + .symmetric_rates = 1, +}; +EXPORT_SYMBOL_GPL(wm8960_dai); + +static int wm8960_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8960_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8960_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8960_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + + wm8960_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8960_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +static struct snd_soc_codec *wm8960_codec; + +static int wm8960_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8960_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8960_codec; + codec = wm8960_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, wm8960_snd_controls, + ARRAY_SIZE(wm8960_snd_controls)); + wm8960_add_widgets(codec); + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +/* power down chip */ +static int wm8960_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8960 = { + .probe = wm8960_probe, + .remove = wm8960_remove, + .suspend = wm8960_suspend, + .resume = wm8960_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8960); + +static int wm8960_register(struct wm8960_priv *wm8960) +{ + struct wm8960_data *pdata = wm8960->codec.dev->platform_data; + struct snd_soc_codec *codec = &wm8960->codec; + int ret; + u16 reg; + + if (wm8960_codec) { + dev_err(codec->dev, "Another WM8960 is registered\n"); + return -EINVAL; + } + + if (!pdata) { + dev_warn(codec->dev, "No platform data supplied\n"); + } else { + if (pdata->dres > WM8960_DRES_MAX) { + dev_err(codec->dev, "Invalid DRES: %d\n", pdata->dres); + pdata->dres = 0; + } + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8960; + codec->name = "WM8960"; + codec->owner = THIS_MODULE; + codec->read = wm8960_read_reg_cache; + codec->write = wm8960_write; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8960_set_bias_level; + codec->dai = &wm8960_dai; + codec->num_dai = 1; + codec->reg_cache_size = WM8960_CACHEREGNUM; + codec->reg_cache = &wm8960->reg_cache; + + memcpy(codec->reg_cache, wm8960_reg, sizeof(wm8960_reg)); + + ret = wm8960_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + return ret; + } + + wm8960_dai.dev = codec->dev; + + wm8960_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Latch the update bits */ + reg = wm8960_read(codec, WM8960_LINVOL); + wm8960_write(codec, WM8960_LINVOL, reg | 0x100); + reg = wm8960_read(codec, WM8960_RINVOL); + wm8960_write(codec, WM8960_RINVOL, reg | 0x100); + reg = wm8960_read(codec, WM8960_LADC); + wm8960_write(codec, WM8960_LADC, reg | 0x100); + reg = wm8960_read(codec, WM8960_RADC); + wm8960_write(codec, WM8960_RADC, reg | 0x100); + reg = wm8960_read(codec, WM8960_LDAC); + wm8960_write(codec, WM8960_LDAC, reg | 0x100); + reg = wm8960_read(codec, WM8960_RDAC); + wm8960_write(codec, WM8960_RDAC, reg | 0x100); + reg = wm8960_read(codec, WM8960_LOUT1); + wm8960_write(codec, WM8960_LOUT1, reg | 0x100); + reg = wm8960_read(codec, WM8960_ROUT1); + wm8960_write(codec, WM8960_ROUT1, reg | 0x100); + reg = wm8960_read(codec, WM8960_LOUT2); + wm8960_write(codec, WM8960_LOUT2, reg | 0x100); + reg = wm8960_read(codec, WM8960_ROUT2); + wm8960_write(codec, WM8960_ROUT2, reg | 0x100); + + wm8960_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + ret = snd_soc_register_dai(&wm8960_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + return ret; + } + + return 0; +} + +static void wm8960_unregister(struct wm8960_priv *wm8960) +{ + wm8960_set_bias_level(&wm8960->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&wm8960_dai); + snd_soc_unregister_codec(&wm8960->codec); + kfree(wm8960); + wm8960_codec = NULL; +} + +static __devinit int wm8960_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8960_priv *wm8960; + struct snd_soc_codec *codec; + + wm8960 = kzalloc(sizeof(struct wm8960_priv), GFP_KERNEL); + if (wm8960 == NULL) + return -ENOMEM; + + codec = &wm8960->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8960); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + + return wm8960_register(wm8960); +} + +static __devexit int wm8960_i2c_remove(struct i2c_client *client) +{ + struct wm8960_priv *wm8960 = i2c_get_clientdata(client); + wm8960_unregister(wm8960); + return 0; +} + +static const struct i2c_device_id wm8960_i2c_id[] = { + { "wm8960", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8960_i2c_id); + +static struct i2c_driver wm8960_i2c_driver = { + .driver = { + .name = "WM8960 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = wm8960_i2c_probe, + .remove = __devexit_p(wm8960_i2c_remove), + .id_table = wm8960_i2c_id, +}; + +static int __init wm8960_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&wm8960_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8960 I2C driver: %d\n", + ret); + } + + return ret; +} +module_init(wm8960_modinit); + +static void __exit wm8960_exit(void) +{ + i2c_del_driver(&wm8960_i2c_driver); +} +module_exit(wm8960_exit); + + +MODULE_DESCRIPTION("ASoC WM8960 driver"); +MODULE_AUTHOR("Liam Girdwood"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8960.h b/sound/soc/codecs/wm8960.h new file mode 100644 index 000000000000..c9af56c9d9d4 --- /dev/null +++ b/sound/soc/codecs/wm8960.h @@ -0,0 +1,127 @@ +/* + * wm8960.h -- WM8960 Soc Audio driver + * + * 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 _WM8960_H +#define _WM8960_H + +/* WM8960 register space */ + + +#define WM8960_CACHEREGNUM 56 + +#define WM8960_LINVOL 0x0 +#define WM8960_RINVOL 0x1 +#define WM8960_LOUT1 0x2 +#define WM8960_ROUT1 0x3 +#define WM8960_CLOCK1 0x4 +#define WM8960_DACCTL1 0x5 +#define WM8960_DACCTL2 0x6 +#define WM8960_IFACE1 0x7 +#define WM8960_CLOCK2 0x8 +#define WM8960_IFACE2 0x9 +#define WM8960_LDAC 0xa +#define WM8960_RDAC 0xb + +#define WM8960_RESET 0xf +#define WM8960_3D 0x10 +#define WM8960_ALC1 0x11 +#define WM8960_ALC2 0x12 +#define WM8960_ALC3 0x13 +#define WM8960_NOISEG 0x14 +#define WM8960_LADC 0x15 +#define WM8960_RADC 0x16 +#define WM8960_ADDCTL1 0x17 +#define WM8960_ADDCTL2 0x18 +#define WM8960_POWER1 0x19 +#define WM8960_POWER2 0x1a +#define WM8960_ADDCTL3 0x1b +#define WM8960_APOP1 0x1c +#define WM8960_APOP2 0x1d + +#define WM8960_LINPATH 0x20 +#define WM8960_RINPATH 0x21 +#define WM8960_LOUTMIX 0x22 + +#define WM8960_ROUTMIX 0x25 +#define WM8960_MONOMIX1 0x26 +#define WM8960_MONOMIX2 0x27 +#define WM8960_LOUT2 0x28 +#define WM8960_ROUT2 0x29 +#define WM8960_MONO 0x2a +#define WM8960_INBMIX1 0x2b +#define WM8960_INBMIX2 0x2c +#define WM8960_BYPASS1 0x2d +#define WM8960_BYPASS2 0x2e +#define WM8960_POWER3 0x2f +#define WM8960_ADDCTL4 0x30 +#define WM8960_CLASSD1 0x31 + +#define WM8960_CLASSD3 0x33 +#define WM8960_PLL1 0x34 +#define WM8960_PLL2 0x35 +#define WM8960_PLL3 0x36 +#define WM8960_PLL4 0x37 + + +/* + * WM8960 Clock dividers + */ +#define WM8960_SYSCLKDIV 0 +#define WM8960_DACDIV 1 +#define WM8960_OPCLKDIV 2 +#define WM8960_DCLKDIV 3 +#define WM8960_TOCLKSEL 4 +#define WM8960_SYSCLKSEL 5 + +#define WM8960_SYSCLK_DIV_1 (0 << 1) +#define WM8960_SYSCLK_DIV_2 (2 << 1) + +#define WM8960_SYSCLK_MCLK (0 << 0) +#define WM8960_SYSCLK_PLL (1 << 0) + +#define WM8960_DAC_DIV_1 (0 << 3) +#define WM8960_DAC_DIV_1_5 (1 << 3) +#define WM8960_DAC_DIV_2 (2 << 3) +#define WM8960_DAC_DIV_3 (3 << 3) +#define WM8960_DAC_DIV_4 (4 << 3) +#define WM8960_DAC_DIV_5_5 (5 << 3) +#define WM8960_DAC_DIV_6 (6 << 3) + +#define WM8960_DCLK_DIV_1_5 (0 << 6) +#define WM8960_DCLK_DIV_2 (1 << 6) +#define WM8960_DCLK_DIV_3 (2 << 6) +#define WM8960_DCLK_DIV_4 (3 << 6) +#define WM8960_DCLK_DIV_6 (4 << 6) +#define WM8960_DCLK_DIV_8 (5 << 6) +#define WM8960_DCLK_DIV_12 (6 << 6) +#define WM8960_DCLK_DIV_16 (7 << 6) + +#define WM8960_TOCLK_F19 (0 << 1) +#define WM8960_TOCLK_F21 (1 << 1) + +#define WM8960_OPCLK_DIV_1 (0 << 0) +#define WM8960_OPCLK_DIV_2 (1 << 0) +#define WM8960_OPCLK_DIV_3 (2 << 0) +#define WM8960_OPCLK_DIV_4 (3 << 0) +#define WM8960_OPCLK_DIV_5_5 (4 << 0) +#define WM8960_OPCLK_DIV_6 (5 << 0) + +extern struct snd_soc_dai wm8960_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8960; + +#define WM8960_DRES_400R 0 +#define WM8960_DRES_200R 1 +#define WM8960_DRES_600R 2 +#define WM8960_DRES_150R 3 +#define WM8960_DRES_MAX 3 + +struct wm8960_data { + int dres; +}; + +#endif -- cgit v1.2.3 From 0d960e8891459f5af85e5781bce3f1da5f7db0fb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Apr 2009 10:08:39 +0100 Subject: ASoC: Request shared rates for WM8903 It has a shared LRCLK. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 8cf571f1a803..c5391841d41f 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1523,6 +1523,7 @@ struct snd_soc_dai wm8903_dai = { .formats = WM8903_FORMATS, }, .ops = &wm8903_dai_ops, + .symmetric_rates = 1, }; EXPORT_SYMBOL_GPL(wm8903_dai); -- cgit v1.2.3 From 6b87a91f5417226c7fe62100b0e7217e7096b789 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 17 Apr 2009 15:55:08 +0300 Subject: ASoC: TWL4030: Fix for the constraint handling The original implementation of the constraints were good against sane applications. If the opening sequence is: stream1_open, stream1_hw_params, stream2_open, stream2_hw_params -> the constraints are set correctly for stream2. But if the sequence is: stream1_open, stream2_open, stream2_hw_params, stream1_hw_params -> than stream2 would receive constraint rate = 0, sample_bits = 0, since the stream1 has not yet called hw_params... The command to trigger this event: gst-launch-0.10 alsasrc device=hw:0 ! alsasink device=hw:0 sync=false This patch does some 'black magic' in order to always set the correct constraints and sets it only when it is needed for the other stream. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 85 +++++++++++++++++++++++++++++++++++----------- 1 file changed, 66 insertions(+), 19 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 921b205de28a..a1b76d7fd130 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -125,6 +125,11 @@ struct twl4030_priv { struct snd_pcm_substream *master_substream; struct snd_pcm_substream *slave_substream; + + unsigned int configured; + unsigned int rate; + unsigned int sample_bits; + unsigned int channels; }; /* @@ -1220,6 +1225,36 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec, return 0; } +static void twl4030_constraints(struct twl4030_priv *twl4030, + struct snd_pcm_substream *mst_substream) +{ + struct snd_pcm_substream *slv_substream; + + /* Pick the stream, which need to be constrained */ + if (mst_substream == twl4030->master_substream) + slv_substream = twl4030->slave_substream; + else if (mst_substream == twl4030->slave_substream) + slv_substream = twl4030->master_substream; + else /* This should not happen.. */ + return; + + /* Set the constraints according to the already configured stream */ + snd_pcm_hw_constraint_minmax(slv_substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + twl4030->rate, + twl4030->rate); + + snd_pcm_hw_constraint_minmax(slv_substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + twl4030->sample_bits, + twl4030->sample_bits); + + snd_pcm_hw_constraint_minmax(slv_substream->runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, + twl4030->channels, + twl4030->channels); +} + static int twl4030_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -1228,26 +1263,16 @@ static int twl4030_startup(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = socdev->card->codec; struct twl4030_priv *twl4030 = codec->private_data; - /* If we already have a playback or capture going then constrain - * this substream to match it. - */ if (twl4030->master_substream) { - struct snd_pcm_runtime *master_runtime; - master_runtime = twl4030->master_substream->runtime; - - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - master_runtime->rate, - master_runtime->rate); - - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - master_runtime->sample_bits, - master_runtime->sample_bits); - twl4030->slave_substream = substream; - } else + /* The DAI has one configuration for playback and capture, so + * if the DAI has been already configured then constrain this + * substream to match it. */ + if (twl4030->configured) + twl4030_constraints(twl4030, twl4030->master_substream); + } else { twl4030->master_substream = substream; + } return 0; } @@ -1264,6 +1289,13 @@ static void twl4030_shutdown(struct snd_pcm_substream *substream, twl4030->master_substream = twl4030->slave_substream; twl4030->slave_substream = NULL; + + /* If all streams are closed, or the remaining stream has not yet + * been configured than set the DAI as not configured. */ + if (!twl4030->master_substream) + twl4030->configured = 0; + else if (!twl4030->master_substream->runtime->channels) + twl4030->configured = 0; } static int twl4030_hw_params(struct snd_pcm_substream *substream, @@ -1276,8 +1308,8 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, struct twl4030_priv *twl4030 = codec->private_data; u8 mode, old_mode, format, old_format; - if (substream == twl4030->slave_substream) - /* Ignoring hw_params for slave substream */ + if (twl4030->configured) + /* Ignoring hw_params for already configured DAI */ return 0; /* bit rate */ @@ -1357,6 +1389,21 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, /* set CODECPDZ afterwards */ twl4030_codec_enable(codec, 1); } + + /* Store the important parameters for the DAI configuration and set + * the DAI as configured */ + twl4030->configured = 1; + twl4030->rate = params_rate(params); + twl4030->sample_bits = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min; + twl4030->channels = params_channels(params); + + /* If both playback and capture streams are open, and one of them + * is setting the hw parameters right now (since we are here), set + * constraints to the other stream to match the current one. */ + if (twl4030->slave_substream) + twl4030_constraints(twl4030, substream); + return 0; } -- cgit v1.2.3 From 7154b3e80203ee91f9ba7d0a43d3daa05c49d9e9 Mon Sep 17 00:00:00 2001 From: Joonyoung Shim Date: Mon, 20 Apr 2009 19:21:35 +0900 Subject: ASoC: TWL4030: Add support Voice DAI Add Voice DAI to support the PCM voice interface of the twl4030 codec. The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX. The PCM voice interface has two modes - PCM mode1 : This uses the normal FS polarity and the rising edge of the clock signal. - PCM mode2 : This uses the FS polarity inverted and the falling edge of the clock signal. If the system master clock is not 26MHz or the twl4030 codec mode is not option2, the voice PCM interface is not available. Signed-off-by: Joonyoung Shim Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 173 ++++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/twl4030.h | 18 ++++- sound/soc/omap/omap2evm.c | 2 +- sound/soc/omap/omap3beagle.c | 2 +- sound/soc/omap/omap3pandora.c | 4 +- sound/soc/omap/overo.c | 2 +- sound/soc/omap/sdp3430.c | 2 +- 7 files changed, 191 insertions(+), 12 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index a1b76d7fd130..cc2968cf6409 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1484,6 +1484,144 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +static int twl4030_voice_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + u8 infreq; + u8 mode; + + /* If the system master clock is not 26MHz, the voice PCM interface is + * not avilable. + */ + infreq = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL) + & TWL4030_APLL_INFREQ; + + if (infreq != TWL4030_APLL_INFREQ_26000KHZ) { + printk(KERN_ERR "TWL4030 voice startup: " + "MCLK is not 26MHz, call set_sysclk() on init\n"); + return -EINVAL; + } + + /* If the codec mode is not option2, the voice PCM interface is not + * avilable. + */ + mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) + & TWL4030_OPT_MODE; + + if (mode != TWL4030_OPTION_2) { + printk(KERN_ERR "TWL4030 voice startup: " + "the codec mode is not option2\n"); + return -EINVAL; + } + + return 0; +} + +static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + u8 old_mode, mode; + + /* bit rate */ + old_mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) + & ~(TWL4030_CODECPDZ); + mode = old_mode; + + switch (params_rate(params)) { + case 8000: + mode &= ~(TWL4030_SEL_16K); + break; + case 16000: + mode |= TWL4030_SEL_16K; + break; + default: + printk(KERN_ERR "TWL4030 voice hw params: unknown rate %d\n", + params_rate(params)); + return -EINVAL; + } + + if (mode != old_mode) { + /* change rate and set CODECPDZ */ + twl4030_codec_enable(codec, 0); + twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); + twl4030_codec_enable(codec, 1); + } + + return 0; +} + +static int twl4030_voice_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; + u8 infreq; + + switch (freq) { + case 26000000: + infreq = TWL4030_APLL_INFREQ_26000KHZ; + break; + default: + printk(KERN_ERR "TWL4030 voice set sysclk: unknown rate %d\n", + freq); + return -EINVAL; + } + + infreq |= TWL4030_APLL_EN; + twl4030_write(codec, TWL4030_REG_APLL_CTL, infreq); + + return 0; +} + +static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 old_format, format; + + /* get format */ + old_format = twl4030_read_reg_cache(codec, TWL4030_REG_VOICE_IF); + format = old_format; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFM: + format &= ~(TWL4030_VIF_SLAVE_EN); + break; + case SND_SOC_DAIFMT_CBS_CFS: + format |= TWL4030_VIF_SLAVE_EN; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + format &= ~(TWL4030_VIF_FORMAT); + break; + case SND_SOC_DAIFMT_NB_IF: + format |= TWL4030_VIF_FORMAT; + break; + default: + return -EINVAL; + } + + if (format != old_format) { + /* change format and set CODECPDZ */ + twl4030_codec_enable(codec, 0); + twl4030_write(codec, TWL4030_REG_VOICE_IF, format); + twl4030_codec_enable(codec, 1); + } + + return 0; +} + #define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) @@ -1495,7 +1633,15 @@ static struct snd_soc_dai_ops twl4030_dai_ops = { .set_fmt = twl4030_set_dai_fmt, }; -struct snd_soc_dai twl4030_dai = { +static struct snd_soc_dai_ops twl4030_dai_voice_ops = { + .startup = twl4030_voice_startup, + .hw_params = twl4030_voice_hw_params, + .set_sysclk = twl4030_voice_set_dai_sysclk, + .set_fmt = twl4030_voice_set_dai_fmt, +}; + +struct snd_soc_dai twl4030_dai[] = { +{ .name = "twl4030", .playback = { .stream_name = "Playback", @@ -1510,6 +1656,23 @@ struct snd_soc_dai twl4030_dai = { .rates = TWL4030_RATES, .formats = TWL4030_FORMATS,}, .ops = &twl4030_dai_ops, +}, +{ + .name = "twl4030 Voice", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &twl4030_dai_voice_ops, +}, }; EXPORT_SYMBOL_GPL(twl4030_dai); @@ -1550,8 +1713,8 @@ static int twl4030_init(struct snd_soc_device *socdev) codec->read = twl4030_read_reg_cache; codec->write = twl4030_write; codec->set_bias_level = twl4030_set_bias_level; - codec->dai = &twl4030_dai; - codec->num_dai = 1; + codec->dai = twl4030_dai; + codec->num_dai = ARRAY_SIZE(twl4030_dai), codec->reg_cache_size = sizeof(twl4030_reg); codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg), GFP_KERNEL); @@ -1645,13 +1808,13 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030); static int __init twl4030_modinit(void) { - return snd_soc_register_dai(&twl4030_dai); + return snd_soc_register_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); } module_init(twl4030_modinit); static void __exit twl4030_exit(void) { - snd_soc_unregister_dai(&twl4030_dai); + snd_soc_unregister_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); } module_exit(twl4030_exit); diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index cb63765db1df..981ec609495b 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -113,6 +113,8 @@ #define TWL4030_SEL_16K 0x04 #define TWL4030_CODECPDZ 0x02 #define TWL4030_OPT_MODE 0x01 +#define TWL4030_OPTION_1 (1 << 0) +#define TWL4030_OPTION_2 (0 << 0) /* TWL4030_REG_MICBIAS_CTL (0x04) Fields */ @@ -171,6 +173,17 @@ #define TWL4030_CLK256FS_EN 0x02 #define TWL4030_AIF_EN 0x01 +/* VOICE_IF (0x0F) Fields */ + +#define TWL4030_VIF_SLAVE_EN 0x80 +#define TWL4030_VIF_DIN_EN 0x40 +#define TWL4030_VIF_DOUT_EN 0x20 +#define TWL4030_VIF_SWAP 0x10 +#define TWL4030_VIF_FORMAT 0x08 +#define TWL4030_VIF_TRI_EN 0x04 +#define TWL4030_VIF_SUB_EN 0x02 +#define TWL4030_VIF_EN 0x01 + /* EAR_CTL (0x21) */ #define TWL4030_EAR_GAIN 0x30 @@ -236,7 +249,10 @@ #define TWL4030_SMOOTH_ANAVOL_EN 0x02 #define TWL4030_DIGMIC_LR_SWAP_EN 0x01 -extern struct snd_soc_dai twl4030_dai; +#define TWL4030_DAI_HIFI 0 +#define TWL4030_DAI_VOICE 1 + +extern struct snd_soc_dai twl4030_dai[2]; extern struct snd_soc_codec_device soc_codec_dev_twl4030; #endif /* End of __TWL4030_AUDIO_H__ */ diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c index 0c2322dcf02a..027e1a40f8a1 100644 --- a/sound/soc/omap/omap2evm.c +++ b/sound/soc/omap/omap2evm.c @@ -86,7 +86,7 @@ static struct snd_soc_dai_link omap2evm_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap2evm_ops, }; diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c index fd24a4acd2f5..6aa428e07d86 100644 --- a/sound/soc/omap/omap3beagle.c +++ b/sound/soc/omap/omap3beagle.c @@ -83,7 +83,7 @@ static struct snd_soc_dai_link omap3beagle_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap3beagle_ops, }; diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index fe282d4ef422..ad219aaf7cb8 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -228,14 +228,14 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { .name = "PCM1773", .stream_name = "HiFi Out", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap3pandora_out_ops, .init = omap3pandora_out_init, }, { .name = "TWL4030", .stream_name = "Line/Mic In", .cpu_dai = &omap_mcbsp_dai[1], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap3pandora_in_ops, .init = omap3pandora_in_init, } diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c index a72dc4e159e5..ec4f8fd8b3a2 100644 --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -83,7 +83,7 @@ static struct snd_soc_dai_link overo_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &overo_ops, }; diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 10f1c867f11d..1c7974101a0b 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -197,7 +197,7 @@ static struct snd_soc_dai_link sdp3430_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .init = sdp3430_twl4030_init, .ops = &sdp3430_ops, }; -- cgit v1.2.3 From 42768a12822c3a0a6d7db69445281db975938294 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 22 Apr 2009 18:39:39 +0100 Subject: ASoC: Use DAPM supply widget for WM8903 charge pump Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 53 ++++++++++++++++++----------------------------- 1 file changed, 20 insertions(+), 33 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index c5391841d41f..a3a489da008f 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -217,7 +217,6 @@ struct wm8903_priv { int sysclk; /* Reference counts */ - int charge_pump_users; int class_w_users; int playback_active; int capture_active; @@ -373,6 +372,15 @@ static void wm8903_reset(struct snd_soc_codec *codec) #define WM8903_OUTPUT_INT 0x2 #define WM8903_OUTPUT_IN 0x1 +static int wm8903_cp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + WARN_ON(event != SND_SOC_DAPM_POST_PMU); + mdelay(4); + + return 0; +} + /* * Event for headphone and line out amplifier power changes. Special * power up/down sequences are required in order to maximise pop/click @@ -382,12 +390,9 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - struct wm8903_priv *wm8903 = codec->private_data; - struct i2c_client *i2c = codec->control_data; u16 val; u16 reg; int shift; - u16 cp_reg = wm8903_read(codec, WM8903_CHARGE_PUMP_0); switch (w->reg) { case WM8903_POWER_MANAGEMENT_2: @@ -419,18 +424,6 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w, /* Short the output */ val &= ~(WM8903_OUTPUT_SHORT << shift); wm8903_write(codec, reg, val); - - wm8903->charge_pump_users++; - - dev_dbg(&i2c->dev, "Charge pump use count now %d\n", - wm8903->charge_pump_users); - - if (wm8903->charge_pump_users == 1) { - dev_dbg(&i2c->dev, "Enabling charge pump\n"); - wm8903_write(codec, WM8903_CHARGE_PUMP_0, - cp_reg | WM8903_CP_ENA); - mdelay(4); - } } if (event & SND_SOC_DAPM_POST_PMU) { @@ -464,19 +457,6 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w, wm8903_write(codec, reg, val); } - if (event & SND_SOC_DAPM_POST_PMD) { - wm8903->charge_pump_users--; - - dev_dbg(&i2c->dev, "Charge pump use count now %d\n", - wm8903->charge_pump_users); - - if (wm8903->charge_pump_users == 0) { - dev_dbg(&i2c->dev, "Disabling charge pump\n"); - wm8903_write(codec, WM8903_CHARGE_PUMP_0, - cp_reg & ~WM8903_CP_ENA); - } - } - return 0; } @@ -844,26 +824,28 @@ SND_SOC_DAPM_MIXER("Right Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 0, 0, SND_SOC_DAPM_PGA_E("Left Headphone Output PGA", WM8903_POWER_MANAGEMENT_2, 1, 0, NULL, 0, wm8903_output_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA_E("Right Headphone Output PGA", WM8903_POWER_MANAGEMENT_2, 0, 0, NULL, 0, wm8903_output_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA_E("Left Line Output PGA", WM8903_POWER_MANAGEMENT_3, 1, 0, NULL, 0, wm8903_output_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA_E("Right Line Output PGA", WM8903_POWER_MANAGEMENT_3, 0, 0, NULL, 0, wm8903_output_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA("Left Speaker PGA", WM8903_POWER_MANAGEMENT_5, 1, 0, NULL, 0), SND_SOC_DAPM_PGA("Right Speaker PGA", WM8903_POWER_MANAGEMENT_5, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("Charge Pump", WM8903_CHARGE_PUMP_0, 0, 0, + wm8903_cp_event, SND_SOC_DAPM_POST_PMU), }; static const struct snd_soc_dapm_route intercon[] = { @@ -951,6 +933,11 @@ static const struct snd_soc_dapm_route intercon[] = { { "ROP", NULL, "Right Speaker PGA" }, { "RON", NULL, "Right Speaker PGA" }, + + { "Left Headphone Output PGA", NULL, "Charge Pump" }, + { "Right Headphone Output PGA", NULL, "Charge Pump" }, + { "Left Line Output PGA", NULL, "Charge Pump" }, + { "Right Line Output PGA", NULL, "Charge Pump" }, }; static int wm8903_add_widgets(struct snd_soc_codec *codec) -- cgit v1.2.3 From c2aef4ffd24dab5c8e94c66e4042ad39d38bcf39 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 22 Apr 2009 20:04:44 +0100 Subject: ASoC: Support CLK_DSP in WM8903 CLK_DSP provides a master clock for the DAC and ADC related functionality on the device. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index a3a489da008f..27c8b94c0551 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -846,6 +846,7 @@ SND_SOC_DAPM_PGA("Right Speaker PGA", WM8903_POWER_MANAGEMENT_5, 0, 0, SND_SOC_DAPM_SUPPLY("Charge Pump", WM8903_CHARGE_PUMP_0, 0, 0, wm8903_cp_event, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8903_CLOCK_RATES_2, 1, 0, NULL, 0), }; static const struct snd_soc_dapm_route intercon[] = { @@ -891,7 +892,12 @@ static const struct snd_soc_dapm_route intercon[] = { { "Right Input PGA", NULL, "Right Input Mode Mux" }, { "ADCL", NULL, "Left Input PGA" }, + { "ADCL", NULL, "CLK_DSP" }, { "ADCR", NULL, "Right Input PGA" }, + { "ADCR", NULL, "CLK_DSP" }, + + { "DACL", NULL, "CLK_DSP" }, + { "DACR", NULL, "CLK_DSP" }, { "Left Output Mixer", "Left Bypass Switch", "Left Input PGA" }, { "Left Output Mixer", "Right Bypass Switch", "Right Input PGA" }, -- cgit v1.2.3 From 4dbfe8097157fde1f8054f48f991ea45833852cd Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 22 Apr 2009 20:32:40 +0100 Subject: ASoC: Optimise configuration of WM8903 DC servo Modify the default startup sequence in the chip to set the DC servo dither level for optimal performance. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 27c8b94c0551..de0a58507202 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -978,6 +978,11 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec, wm8903_write(codec, WM8903_CLOCK_RATES_2, WM8903_CLK_SYS_ENA); + /* Change DC servo dither level in startup sequence */ + wm8903_write(codec, WM8903_WRITE_SEQUENCER_0, 0x11); + wm8903_write(codec, WM8903_WRITE_SEQUENCER_1, 0x1257); + wm8903_write(codec, WM8903_WRITE_SEQUENCER_2, 0x2); + wm8903_run_sequence(codec, 0); wm8903_sync_reg_cache(codec, codec->reg_cache); -- cgit v1.2.3 From d7d5c5476a12333a33b7a14ebb10eccc729c01cb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 22 Apr 2009 21:03:50 +0100 Subject: ASoC: Actively manage the DC servo for WM8903 Save a little extra power by enabling the DC servo offset correction for the output channels only when the relevant channels are enabled. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index de0a58507202..0bab5c6bd64a 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -392,14 +392,18 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w, struct snd_soc_codec *codec = w->codec; u16 val; u16 reg; + u16 dcs_reg; + u16 dcs_bit; int shift; switch (w->reg) { case WM8903_POWER_MANAGEMENT_2: reg = WM8903_ANALOGUE_HP_0; + dcs_bit = 0 + w->shift; break; case WM8903_POWER_MANAGEMENT_3: reg = WM8903_ANALOGUE_LINEOUT_0; + dcs_bit = 2 + w->shift; break; default: BUG(); @@ -439,6 +443,11 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w, val |= (WM8903_OUTPUT_OUT << shift); wm8903_write(codec, reg, val); + /* Enable the DC servo */ + dcs_reg = wm8903_read(codec, WM8903_DC_SERVO_0); + dcs_reg |= dcs_bit; + wm8903_write(codec, WM8903_DC_SERVO_0, dcs_reg); + /* Remove the short */ val |= (WM8903_OUTPUT_SHORT << shift); wm8903_write(codec, reg, val); @@ -451,6 +460,11 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w, val &= ~(WM8903_OUTPUT_SHORT << shift); wm8903_write(codec, reg, val); + /* Disable the DC servo */ + dcs_reg = wm8903_read(codec, WM8903_DC_SERVO_0); + dcs_reg &= ~dcs_bit; + wm8903_write(codec, WM8903_DC_SERVO_0, dcs_reg); + /* Then disable the intermediate and output stages */ val &= ~((WM8903_OUTPUT_OUT | WM8903_OUTPUT_INT | WM8903_OUTPUT_IN) << shift); -- cgit v1.2.3 From 727fb909e541ebd09d5b552afef02a147978c151 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 22 Apr 2009 21:06:14 +0100 Subject: ASoC: Remove redundant rate constraint for WM8903 This is now handled by symmetric_rates. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 0bab5c6bd64a..bec418af97c9 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1289,14 +1289,8 @@ static int wm8903_startup(struct snd_pcm_substream *substream, if (wm8903->master_substream) { master_runtime = wm8903->master_substream->runtime; - dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n", - master_runtime->sample_bits, - master_runtime->rate); - - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - master_runtime->rate, - master_runtime->rate); + dev_dbg(&i2c->dev, "Constraining to %d bits\n", + master_runtime->sample_bits); snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -- cgit v1.2.3 From 291ce18ceb84aca79368369885eec2d329ae16c5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 22 Apr 2009 21:36:14 +0100 Subject: ASoC: Implement WM8903 digital sidetone support Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index bec418af97c9..d8a9222fbf74 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -533,6 +533,7 @@ static int wm8903_class_w_put(struct snd_kcontrol *kcontrol, /* ALSA can only do steps of .01dB */ static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1); +static const DECLARE_TLV_DB_SCALE(digital_sidetone_tlv, -3600, 300, 0); static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0); static const DECLARE_TLV_DB_SCALE(drc_tlv_thresh, 0, 75, 0); @@ -651,6 +652,16 @@ static const struct soc_enum rinput_inv_enum = SOC_ENUM_SINGLE(WM8903_ANALOGUE_RIGHT_INPUT_1, 4, 3, rinput_mux_text); +static const char *sidetone_text[] = { + "None", "Left", "Right" +}; + +static const struct soc_enum lsidetone_enum = + SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 2, 3, sidetone_text); + +static const struct soc_enum rsidetone_enum = + SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 0, 3, sidetone_text); + static const struct snd_kcontrol_new wm8903_snd_controls[] = { /* Input PGAs - No TLV since the scale depends on PGA mode */ @@ -694,6 +705,9 @@ SOC_DOUBLE_R_TLV("Digital Capture Volume", WM8903_ADC_DIGITAL_VOLUME_LEFT, SOC_ENUM("ADC Companding Mode", adc_companding), SOC_SINGLE("ADC Companding Switch", WM8903_AUDIO_INTERFACE_0, 3, 1, 0), +SOC_DOUBLE_TLV("Digital Sidetone Volume", WM8903_DAC_DIGITAL_0, 4, 8, + 12, 0, digital_sidetone_tlv), + /* DAC */ SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8903_DAC_DIGITAL_VOLUME_LEFT, WM8903_DAC_DIGITAL_VOLUME_RIGHT, 1, 120, 0, digital_tlv), @@ -756,6 +770,12 @@ static const struct snd_kcontrol_new rinput_mux = static const struct snd_kcontrol_new rinput_inv_mux = SOC_DAPM_ENUM("Right Inverting Input Mux", rinput_inv_enum); +static const struct snd_kcontrol_new lsidetone_mux = + SOC_DAPM_ENUM("DACL Sidetone Mux", lsidetone_enum); + +static const struct snd_kcontrol_new rsidetone_mux = + SOC_DAPM_ENUM("DACR Sidetone Mux", rsidetone_enum); + static const struct snd_kcontrol_new left_output_mixer[] = { SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_LEFT_MIX_0, 3, 1, 0), SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_LEFT_MIX_0, 2, 1, 0), @@ -822,6 +842,9 @@ SND_SOC_DAPM_PGA("Right Input PGA", WM8903_POWER_MANAGEMENT_0, 0, 0, NULL, 0), SND_SOC_DAPM_ADC("ADCL", "Left HiFi Capture", WM8903_POWER_MANAGEMENT_6, 1, 0), SND_SOC_DAPM_ADC("ADCR", "Right HiFi Capture", WM8903_POWER_MANAGEMENT_6, 0, 0), +SND_SOC_DAPM_MUX("DACL Sidetone", SND_SOC_NOPM, 0, 0, &lsidetone_mux), +SND_SOC_DAPM_MUX("DACR Sidetone", SND_SOC_NOPM, 0, 0, &rsidetone_mux), + SND_SOC_DAPM_DAC("DACL", "Left Playback", WM8903_POWER_MANAGEMENT_6, 3, 0), SND_SOC_DAPM_DAC("DACR", "Right Playback", WM8903_POWER_MANAGEMENT_6, 2, 0), @@ -910,7 +933,14 @@ static const struct snd_soc_dapm_route intercon[] = { { "ADCR", NULL, "Right Input PGA" }, { "ADCR", NULL, "CLK_DSP" }, + { "DACL Sidetone", "Left", "ADCL" }, + { "DACL Sidetone", "Right", "ADCR" }, + { "DACR Sidetone", "Left", "ADCL" }, + { "DACR Sidetone", "Right", "ADCR" }, + + { "DACL", NULL, "DACL Sidetone" }, { "DACL", NULL, "CLK_DSP" }, + { "DACR", NULL, "DACR Sidetone" }, { "DACR", NULL, "CLK_DSP" }, { "Left Output Mixer", "Left Bypass Switch", "Left Input PGA" }, -- cgit v1.2.3 From 1a787e7ad242312af0afb2156596d42ee5e0c6bc Mon Sep 17 00:00:00 2001 From: Joonyoung Shim Date: Wed, 22 Apr 2009 13:13:34 +0900 Subject: ASoC: TWL4030: Add VDL path support Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 256 ++++++++++++++++++++++----------------------- 1 file changed, 126 insertions(+), 130 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index cc2968cf6409..fdf88dfbcff9 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -321,104 +321,60 @@ static void twl4030_power_down(struct snd_soc_codec *codec) } /* Earpiece */ -static const char *twl4030_earpiece_texts[] = - {"Off", "DACL1", "DACL2", "DACR1"}; - -static const unsigned int twl4030_earpiece_values[] = - {0x0, 0x1, 0x2, 0x4}; - -static const struct soc_enum twl4030_earpiece_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_EAR_CTL, 1, 0x7, - ARRAY_SIZE(twl4030_earpiece_texts), - twl4030_earpiece_texts, - twl4030_earpiece_values); - -static const struct snd_kcontrol_new twl4030_dapm_earpiece_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_earpiece_enum); +static const struct snd_kcontrol_new twl4030_dapm_earpiece_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_EAR_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_EAR_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_EAR_CTL, 2, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_EAR_CTL, 3, 1, 0), +}; /* PreDrive Left */ -static const char *twl4030_predrivel_texts[] = - {"Off", "DACL1", "DACL2", "DACR2"}; - -static const unsigned int twl4030_predrivel_values[] = - {0x0, 0x1, 0x2, 0x4}; - -static const struct soc_enum twl4030_predrivel_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDL_CTL, 1, 0x7, - ARRAY_SIZE(twl4030_predrivel_texts), - twl4030_predrivel_texts, - twl4030_predrivel_values); - -static const struct snd_kcontrol_new twl4030_dapm_predrivel_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_predrivel_enum); +static const struct snd_kcontrol_new twl4030_dapm_predrivel_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PREDL_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_PREDL_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PREDL_CTL, 2, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PREDL_CTL, 3, 1, 0), +}; /* PreDrive Right */ -static const char *twl4030_predriver_texts[] = - {"Off", "DACR1", "DACR2", "DACL2"}; - -static const unsigned int twl4030_predriver_values[] = - {0x0, 0x1, 0x2, 0x4}; - -static const struct soc_enum twl4030_predriver_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDR_CTL, 1, 0x7, - ARRAY_SIZE(twl4030_predriver_texts), - twl4030_predriver_texts, - twl4030_predriver_values); - -static const struct snd_kcontrol_new twl4030_dapm_predriver_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_predriver_enum); +static const struct snd_kcontrol_new twl4030_dapm_predriver_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PREDR_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_PREDR_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PREDR_CTL, 2, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PREDR_CTL, 3, 1, 0), +}; /* Headset Left */ -static const char *twl4030_hsol_texts[] = - {"Off", "DACL1", "DACL2"}; - -static const struct soc_enum twl4030_hsol_enum = - SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 1, - ARRAY_SIZE(twl4030_hsol_texts), - twl4030_hsol_texts); - -static const struct snd_kcontrol_new twl4030_dapm_hsol_control = -SOC_DAPM_ENUM("Route", twl4030_hsol_enum); +static const struct snd_kcontrol_new twl4030_dapm_hsol_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_HS_SEL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_HS_SEL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_HS_SEL, 2, 1, 0), +}; /* Headset Right */ -static const char *twl4030_hsor_texts[] = - {"Off", "DACR1", "DACR2"}; - -static const struct soc_enum twl4030_hsor_enum = - SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 4, - ARRAY_SIZE(twl4030_hsor_texts), - twl4030_hsor_texts); - -static const struct snd_kcontrol_new twl4030_dapm_hsor_control = -SOC_DAPM_ENUM("Route", twl4030_hsor_enum); +static const struct snd_kcontrol_new twl4030_dapm_hsor_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_HS_SEL, 3, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_HS_SEL, 4, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_HS_SEL, 5, 1, 0), +}; /* Carkit Left */ -static const char *twl4030_carkitl_texts[] = - {"Off", "DACL1", "DACL2"}; - -static const struct soc_enum twl4030_carkitl_enum = - SOC_ENUM_SINGLE(TWL4030_REG_PRECKL_CTL, 1, - ARRAY_SIZE(twl4030_carkitl_texts), - twl4030_carkitl_texts); - -static const struct snd_kcontrol_new twl4030_dapm_carkitl_control = -SOC_DAPM_ENUM("Route", twl4030_carkitl_enum); +static const struct snd_kcontrol_new twl4030_dapm_carkitl_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PRECKL_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_PRECKL_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PRECKL_CTL, 2, 1, 0), +}; /* Carkit Right */ -static const char *twl4030_carkitr_texts[] = - {"Off", "DACR1", "DACR2"}; - -static const struct soc_enum twl4030_carkitr_enum = - SOC_ENUM_SINGLE(TWL4030_REG_PRECKR_CTL, 1, - ARRAY_SIZE(twl4030_carkitr_texts), - twl4030_carkitr_texts); - -static const struct snd_kcontrol_new twl4030_dapm_carkitr_control = -SOC_DAPM_ENUM("Route", twl4030_carkitr_enum); +static const struct snd_kcontrol_new twl4030_dapm_carkitr_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PRECKR_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_PRECKR_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PRECKR_CTL, 2, 1, 0), +}; /* Handsfree Left */ static const char *twl4030_handsfreel_texts[] = - {"Voice", "DACL1", "DACL2", "DACR2"}; + {"Voice", "AudioL1", "AudioL2", "AudioR2"}; static const struct soc_enum twl4030_handsfreel_enum = SOC_ENUM_SINGLE(TWL4030_REG_HFL_CTL, 0, @@ -430,7 +386,7 @@ SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum); /* Handsfree Right */ static const char *twl4030_handsfreer_texts[] = - {"Voice", "DACR1", "DACR2", "DACL2"}; + {"Voice", "AudioR1", "AudioR2", "AudioL2"}; static const struct soc_enum twl4030_handsfreer_enum = SOC_ENUM_SINGLE(TWL4030_REG_HFR_CTL, 0, @@ -828,6 +784,12 @@ static DECLARE_TLV_DB_SCALE(digital_fine_tlv, -6300, 100, 1); */ static DECLARE_TLV_DB_SCALE(digital_coarse_tlv, 0, 600, 0); +/* + * Voice Downlink GAIN volume control: + * from -37 to 12 dB in 1 dB steps (mute instead of -37 dB) + */ +static DECLARE_TLV_DB_SCALE(digital_voice_downlink_tlv, -3700, 100, 1); + /* * Analog playback gain * -24 dB to 12 dB in 2 dB steps @@ -892,6 +854,16 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL, 1, 1, 0), + /* Common voice downlink gain controls */ + SOC_SINGLE_TLV("DAC Voice Digital Downlink Volume", + TWL4030_REG_VRXPGA, 0, 0x31, 0, digital_voice_downlink_tlv), + + SOC_SINGLE_TLV("DAC Voice Analog Downlink Volume", + TWL4030_REG_VDL_APGA_CTL, 3, 0x12, 1, analog_tlv), + + SOC_SINGLE("DAC Voice Analog Downlink Switch", + TWL4030_REG_VDL_APGA_CTL, 1, 1, 0), + /* Separate output gain controls */ SOC_DOUBLE_R_TLV_TWL4030("PreDriv Playback Volume", TWL4030_REG_PREDL_CTL, TWL4030_REG_PREDR_CTL, @@ -956,6 +928,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Voice", "Voice Playback", + TWL4030_REG_AVDAC_CTL, 4, 0), /* Analog PGAs */ SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL, @@ -966,6 +940,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { 0, 0, NULL, 0), SND_SOC_DAPM_PGA("ARXL2_APGA", TWL4030_REG_ARXL2_APGA_CTL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("VDL_APGA", TWL4030_REG_VDL_APGA_CTL, + 0, 0, NULL, 0), /* Analog bypasses */ SND_SOC_DAPM_SWITCH_E("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0, @@ -998,26 +974,35 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", TWL4030_REG_AVDAC_CTL, 3, 0, NULL, 0), - /* Output MUX controls */ + /* Output MIXER controls */ /* Earpiece */ - SND_SOC_DAPM_VALUE_MUX("Earpiece Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_earpiece_control), + SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_earpiece_controls[0], + ARRAY_SIZE(twl4030_dapm_earpiece_controls)), /* PreDrivL/R */ - SND_SOC_DAPM_VALUE_MUX("PredriveL Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_predrivel_control), - SND_SOC_DAPM_VALUE_MUX("PredriveR Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_predriver_control), + SND_SOC_DAPM_MIXER("PredriveL Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_predrivel_controls[0], + ARRAY_SIZE(twl4030_dapm_predrivel_controls)), + SND_SOC_DAPM_MIXER("PredriveR Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_predriver_controls[0], + ARRAY_SIZE(twl4030_dapm_predriver_controls)), /* HeadsetL/R */ - SND_SOC_DAPM_MUX_E("HeadsetL Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_hsol_control, headsetl_event, - SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_MUX("HeadsetR Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_hsor_control), + SND_SOC_DAPM_MIXER_E("HeadsetL Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_hsol_controls[0], + ARRAY_SIZE(twl4030_dapm_hsol_controls), headsetl_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER("HeadsetR Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_hsor_controls[0], + ARRAY_SIZE(twl4030_dapm_hsor_controls)), /* CarkitL/R */ - SND_SOC_DAPM_MUX("CarkitL Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_carkitl_control), - SND_SOC_DAPM_MUX("CarkitR Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_carkitr_control), + SND_SOC_DAPM_MIXER("CarkitL Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_carkitl_controls[0], + ARRAY_SIZE(twl4030_dapm_carkitl_controls)), + SND_SOC_DAPM_MIXER("CarkitR Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_carkitr_controls[0], + ARRAY_SIZE(twl4030_dapm_carkitr_controls)), + + /* Output MUX controls */ /* HandsfreeL/R */ SND_SOC_DAPM_MUX_E("HandsfreeL Mux", TWL4030_REG_HFL_CTL, 5, 0, &twl4030_dapm_handsfreel_control, handsfree_event, @@ -1082,50 +1067,61 @@ static const struct snd_soc_dapm_route intercon[] = { {"ARXL2_APGA", NULL, "Analog L2 Playback Mixer"}, {"ARXR2_APGA", NULL, "Analog R2 Playback Mixer"}, + {"VDL_APGA", NULL, "DAC Voice"}, + /* Internal playback routings */ /* Earpiece */ - {"Earpiece Mux", "DACL1", "ARXL1_APGA"}, - {"Earpiece Mux", "DACL2", "ARXL2_APGA"}, - {"Earpiece Mux", "DACR1", "ARXR1_APGA"}, + {"Earpiece Mixer", "Voice", "VDL_APGA"}, + {"Earpiece Mixer", "AudioL1", "ARXL1_APGA"}, + {"Earpiece Mixer", "AudioL2", "ARXL2_APGA"}, + {"Earpiece Mixer", "AudioR1", "ARXR1_APGA"}, /* PreDrivL */ - {"PredriveL Mux", "DACL1", "ARXL1_APGA"}, - {"PredriveL Mux", "DACL2", "ARXL2_APGA"}, - {"PredriveL Mux", "DACR2", "ARXR2_APGA"}, + {"PredriveL Mixer", "Voice", "VDL_APGA"}, + {"PredriveL Mixer", "AudioL1", "ARXL1_APGA"}, + {"PredriveL Mixer", "AudioL2", "ARXL2_APGA"}, + {"PredriveL Mixer", "AudioR2", "ARXR2_APGA"}, /* PreDrivR */ - {"PredriveR Mux", "DACR1", "ARXR1_APGA"}, - {"PredriveR Mux", "DACR2", "ARXR2_APGA"}, - {"PredriveR Mux", "DACL2", "ARXL2_APGA"}, + {"PredriveR Mixer", "Voice", "VDL_APGA"}, + {"PredriveR Mixer", "AudioR1", "ARXR1_APGA"}, + {"PredriveR Mixer", "AudioR2", "ARXR2_APGA"}, + {"PredriveR Mixer", "AudioL2", "ARXL2_APGA"}, /* HeadsetL */ - {"HeadsetL Mux", "DACL1", "ARXL1_APGA"}, - {"HeadsetL Mux", "DACL2", "ARXL2_APGA"}, + {"HeadsetL Mixer", "Voice", "VDL_APGA"}, + {"HeadsetL Mixer", "AudioL1", "ARXL1_APGA"}, + {"HeadsetL Mixer", "AudioL2", "ARXL2_APGA"}, /* HeadsetR */ - {"HeadsetR Mux", "DACR1", "ARXR1_APGA"}, - {"HeadsetR Mux", "DACR2", "ARXR2_APGA"}, + {"HeadsetR Mixer", "Voice", "VDL_APGA"}, + {"HeadsetR Mixer", "AudioR1", "ARXR1_APGA"}, + {"HeadsetR Mixer", "AudioR2", "ARXR2_APGA"}, /* CarkitL */ - {"CarkitL Mux", "DACL1", "ARXL1_APGA"}, - {"CarkitL Mux", "DACL2", "ARXL2_APGA"}, + {"CarkitL Mixer", "Voice", "VDL_APGA"}, + {"CarkitL Mixer", "AudioL1", "ARXL1_APGA"}, + {"CarkitL Mixer", "AudioL2", "ARXL2_APGA"}, /* CarkitR */ - {"CarkitR Mux", "DACR1", "ARXR1_APGA"}, - {"CarkitR Mux", "DACR2", "ARXR2_APGA"}, + {"CarkitR Mixer", "Voice", "VDL_APGA"}, + {"CarkitR Mixer", "AudioR1", "ARXR1_APGA"}, + {"CarkitR Mixer", "AudioR2", "ARXR2_APGA"}, /* HandsfreeL */ - {"HandsfreeL Mux", "DACL1", "ARXL1_APGA"}, - {"HandsfreeL Mux", "DACL2", "ARXL2_APGA"}, - {"HandsfreeL Mux", "DACR2", "ARXR2_APGA"}, + {"HandsfreeL Mux", "Voice", "VDL_APGA"}, + {"HandsfreeL Mux", "AudioL1", "ARXL1_APGA"}, + {"HandsfreeL Mux", "AudioL2", "ARXL2_APGA"}, + {"HandsfreeL Mux", "AudioR2", "ARXR2_APGA"}, /* HandsfreeR */ - {"HandsfreeR Mux", "DACR1", "ARXR1_APGA"}, - {"HandsfreeR Mux", "DACR2", "ARXR2_APGA"}, - {"HandsfreeR Mux", "DACL2", "ARXL2_APGA"}, + {"HandsfreeR Mux", "Voice", "VDL_APGA"}, + {"HandsfreeR Mux", "AudioR1", "ARXR1_APGA"}, + {"HandsfreeR Mux", "AudioR2", "ARXR2_APGA"}, + {"HandsfreeR Mux", "AudioL2", "ARXL2_APGA"}, /* outputs */ {"OUTL", NULL, "ARXL2_APGA"}, {"OUTR", NULL, "ARXR2_APGA"}, - {"EARPIECE", NULL, "Earpiece Mux"}, - {"PREDRIVEL", NULL, "PredriveL Mux"}, - {"PREDRIVER", NULL, "PredriveR Mux"}, - {"HSOL", NULL, "HeadsetL Mux"}, - {"HSOR", NULL, "HeadsetR Mux"}, - {"CARKITL", NULL, "CarkitL Mux"}, - {"CARKITR", NULL, "CarkitR Mux"}, + {"EARPIECE", NULL, "Earpiece Mixer"}, + {"PREDRIVEL", NULL, "PredriveL Mixer"}, + {"PREDRIVER", NULL, "PredriveR Mixer"}, + {"HSOL", NULL, "HeadsetL Mixer"}, + {"HSOR", NULL, "HeadsetR Mixer"}, + {"CARKITL", NULL, "CarkitL Mixer"}, + {"CARKITR", NULL, "CarkitR Mixer"}, {"HFL", NULL, "HandsfreeL Mux"}, {"HFR", NULL, "HandsfreeR Mux"}, -- cgit v1.2.3 From 8a1f936acdfd53cb0a981f3f80483863dcd84fa9 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 23 Apr 2009 14:36:49 +0300 Subject: ASoC: TWL4030: Add 4 channel TDM support Support for 4 channel TDM (SND_SOC_DAIFMT_DSP_A) for twl4030 codec. The channel allocations are: Playback: TDM i2s TWL RX Channel 1 Left SDRL2 Channel 3 Right SDRR2 Channel 2 -- SDRL1 Channel 4 -- SDRR1 Capture: TDM i2s TWL TX Channel 1 Left TXL1 Channel 3 Right TXR1 Channel 2 -- TXL2 Channel 4 -- TXR2 Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 52 ++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/twl4030.h | 11 ++++++++++ 2 files changed, 61 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index fdf88dfbcff9..e23c20c42f19 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1251,6 +1251,28 @@ static void twl4030_constraints(struct twl4030_priv *twl4030, twl4030->channels); } +/* In case of 4 channel mode, the RX1 L/R for playback and the TX2 L/R for + * capture has to be enabled/disabled. */ +static void twl4030_tdm_enable(struct snd_soc_codec *codec, int direction, + int enable) +{ + u8 reg, mask; + + reg = twl4030_read_reg_cache(codec, TWL4030_REG_OPTION); + + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + mask = TWL4030_ARXL1_VRX_EN | TWL4030_ARXR1_EN; + else + mask = TWL4030_ATXL2_VTXL_EN | TWL4030_ATXR2_VTXR_EN; + + if (enable) + reg |= mask; + else + reg &= ~mask; + + twl4030_write(codec, TWL4030_REG_OPTION, reg); +} + static int twl4030_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -1267,6 +1289,15 @@ static int twl4030_startup(struct snd_pcm_substream *substream, if (twl4030->configured) twl4030_constraints(twl4030, twl4030->master_substream); } else { + if (!(twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) & + TWL4030_OPTION_1)) { + /* In option2 4 channel is not supported, set the + * constraint for the first stream for channels, the + * second stream will 'inherit' this cosntraint */ + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, + 2, 2); + } twl4030->master_substream = substream; } @@ -1292,6 +1323,10 @@ static void twl4030_shutdown(struct snd_pcm_substream *substream, twl4030->configured = 0; else if (!twl4030->master_substream->runtime->channels) twl4030->configured = 0; + + /* If the closing substream had 4 channel, do the necessary cleanup */ + if (substream->runtime->channels == 4) + twl4030_tdm_enable(codec, substream->stream, 0); } static int twl4030_hw_params(struct snd_pcm_substream *substream, @@ -1304,6 +1339,16 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, struct twl4030_priv *twl4030 = codec->private_data; u8 mode, old_mode, format, old_format; + /* If the substream has 4 channel, do the necessary setup */ + if (params_channels(params) == 4) { + /* Safety check: are we in the correct operating mode? */ + if ((twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) & + TWL4030_OPTION_1)) + twl4030_tdm_enable(codec, substream->stream, 1); + else + return -EINVAL; + } + if (twl4030->configured) /* Ignoring hw_params for already configured DAI */ return 0; @@ -1461,6 +1506,9 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, case SND_SOC_DAIFMT_I2S: format |= TWL4030_AIF_FORMAT_CODEC; break; + case SND_SOC_DAIFMT_DSP_A: + format |= TWL4030_AIF_FORMAT_TDM; + break; default: return -EINVAL; } @@ -1642,13 +1690,13 @@ struct snd_soc_dai twl4030_dai[] = { .playback = { .stream_name = "Playback", .channels_min = 2, - .channels_max = 2, + .channels_max = 4, .rates = TWL4030_RATES | SNDRV_PCM_RATE_96000, .formats = TWL4030_FORMATS,}, .capture = { .stream_name = "Capture", .channels_min = 2, - .channels_max = 2, + .channels_max = 4, .rates = TWL4030_RATES, .formats = TWL4030_FORMATS,}, .ops = &twl4030_dai_ops, diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 981ec609495b..3441115136f6 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -116,6 +116,17 @@ #define TWL4030_OPTION_1 (1 << 0) #define TWL4030_OPTION_2 (0 << 0) +/* TWL4030_OPTION (0x02) Fields */ + +#define TWL4030_ATXL1_EN (1 << 0) +#define TWL4030_ATXR1_EN (1 << 1) +#define TWL4030_ATXL2_VTXL_EN (1 << 2) +#define TWL4030_ATXR2_VTXR_EN (1 << 3) +#define TWL4030_ARXL1_VRX_EN (1 << 4) +#define TWL4030_ARXR1_EN (1 << 5) +#define TWL4030_ARXL2_EN (1 << 6) +#define TWL4030_ARXR2_EN (1 << 7) + /* TWL4030_REG_MICBIAS_CTL (0x04) Fields */ #define TWL4030_MICBIAS2_CTL 0x40 -- cgit v1.2.3 From 008db442efa542357314593c71ab9966be909855 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 27 Apr 2009 19:17:08 +0100 Subject: ASoC: Include WM8350 register definitions in CODEC header It's expected behaviour for the CODEC header to provide them but the WM8350 doesn't due to having all the registers together under drivers/mfd. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8350.h | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8350.h b/sound/soc/codecs/wm8350.h index d11bd9288cf9..d088eb4b88bb 100644 --- a/sound/soc/codecs/wm8350.h +++ b/sound/soc/codecs/wm8350.h @@ -13,6 +13,7 @@ #define _WM8350_H #include +#include extern struct snd_soc_dai wm8350_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8350; -- cgit v1.2.3 From 0b5e92c5e020ee7437fa5304a8451d6dd08d1a26 Mon Sep 17 00:00:00 2001 From: Jonathan Cameron Date: Mon, 27 Apr 2009 13:49:44 +0000 Subject: ASoC WM8940 Driver Signed-off-by: Jonathan Cameron Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8940.c | 955 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8940.h | 104 +++++ 4 files changed, 1065 insertions(+) create mode 100644 sound/soc/codecs/wm8940.c create mode 100644 sound/soc/codecs/wm8940.h (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 121d63f13dbb..1c19ad54a9f9 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -35,6 +35,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8900 if I2C select SND_SOC_WM8903 if I2C + select SND_SOC_WM8940 if I2C select SND_SOC_WM8960 if I2C select SND_SOC_WM8971 if I2C select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI @@ -140,6 +141,9 @@ config SND_SOC_WM8900 config SND_SOC_WM8903 tristate +config SND_SOC_WM8940 + tristate + config SND_SOC_WM8960 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 811696861d31..3d31b6bea834 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -23,6 +23,7 @@ snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o +snd-soc-wm8940-objs := wm8940.o snd-soc-wm8960-objs := wm8960.o snd-soc-wm8971-objs := wm8971.o snd-soc-wm8988-objs := wm8988.o @@ -57,6 +58,7 @@ obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o +obj-$(CONFIG_SND_SOC_WM8940) += snd-soc-wm8940.o obj-$(CONFIG_SND_SOC_WM8960) += snd-soc-wm8960.o obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c new file mode 100644 index 000000000000..26987dcd8d55 --- /dev/null +++ b/sound/soc/codecs/wm8940.c @@ -0,0 +1,955 @@ +/* + * wm8940.c -- WM8940 ALSA Soc Audio driver + * + * Author: Jonathan Cameron + * + * Based on wm8510.c + * Copyright 2006 Wolfson Microelectronics PLC. + * Author: Liam Girdwood + * + * 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. + * + * Not currently handled: + * Notch filter control + * AUXMode (inverting vs mixer) + * No means to obtain current gain if alc enabled. + * No use made of gpio + * Fast VMID discharge for power down + * Soft Start + * DLR and ALR Swaps not enabled + * Digital Sidetone not supported + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm8940.h" + +struct wm8940_priv { + unsigned int sysclk; + u16 reg_cache[WM8940_CACHEREGNUM]; + struct snd_soc_codec codec; +}; + +static u16 wm8940_reg_defaults[] = { + 0x8940, /* Soft Reset */ + 0x0000, /* Power 1 */ + 0x0000, /* Power 2 */ + 0x0000, /* Power 3 */ + 0x0010, /* Interface Control */ + 0x0000, /* Companding Control */ + 0x0140, /* Clock Control */ + 0x0000, /* Additional Controls */ + 0x0000, /* GPIO Control */ + 0x0002, /* Auto Increment Control */ + 0x0000, /* DAC Control */ + 0x00FF, /* DAC Volume */ + 0, + 0, + 0x0100, /* ADC Control */ + 0x00FF, /* ADC Volume */ + 0x0000, /* Notch Filter 1 Control 1 */ + 0x0000, /* Notch Filter 1 Control 2 */ + 0x0000, /* Notch Filter 2 Control 1 */ + 0x0000, /* Notch Filter 2 Control 2 */ + 0x0000, /* Notch Filter 3 Control 1 */ + 0x0000, /* Notch Filter 3 Control 2 */ + 0x0000, /* Notch Filter 4 Control 1 */ + 0x0000, /* Notch Filter 4 Control 2 */ + 0x0032, /* DAC Limit Control 1 */ + 0x0000, /* DAC Limit Control 2 */ + 0, + 0, + 0, + 0, + 0, + 0, + 0x0038, /* ALC Control 1 */ + 0x000B, /* ALC Control 2 */ + 0x0032, /* ALC Control 3 */ + 0x0000, /* Noise Gate */ + 0x0041, /* PLLN */ + 0x000C, /* PLLK1 */ + 0x0093, /* PLLK2 */ + 0x00E9, /* PLLK3 */ + 0, + 0, + 0x0030, /* ALC Control 4 */ + 0, + 0x0002, /* Input Control */ + 0x0050, /* PGA Gain */ + 0, + 0x0002, /* ADC Boost Control */ + 0, + 0x0002, /* Output Control */ + 0x0000, /* Speaker Mixer Control */ + 0, + 0, + 0, + 0x0079, /* Speaker Volume */ + 0, + 0x0000, /* Mono Mixer Control */ +}; + +static inline unsigned int wm8940_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + + if (reg >= ARRAY_SIZE(wm8940_reg_defaults)) + return -1; + + return cache[reg]; +} + +static inline int wm8940_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + + if (reg >= ARRAY_SIZE(wm8940_reg_defaults)) + return -1; + + cache[reg] = value; + + return 0; +} + +static int wm8940_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + int ret; + u8 data[3] = { reg, + (value & 0xff00) >> 8, + (value & 0x00ff) + }; + + wm8940_write_reg_cache(codec, reg, value); + + ret = codec->hw_write(codec->control_data, data, 3); + + if (ret < 0) + return ret; + else if (ret != 3) + return -EIO; + return 0; +} + +static const char *wm8940_companding[] = { "Off", "NC", "u-law", "A-law" }; +static const struct soc_enum wm8940_adc_companding_enum += SOC_ENUM_SINGLE(WM8940_COMPANDINGCTL, 1, 4, wm8940_companding); +static const struct soc_enum wm8940_dac_companding_enum += SOC_ENUM_SINGLE(WM8940_COMPANDINGCTL, 3, 4, wm8940_companding); + +static const char *wm8940_alc_mode_text[] = {"ALC", "Limiter"}; +static const struct soc_enum wm8940_alc_mode_enum += SOC_ENUM_SINGLE(WM8940_ALC3, 8, 2, wm8940_alc_mode_text); + +static const char *wm8940_mic_bias_level_text[] = {"0.9", "0.65"}; +static const struct soc_enum wm8940_mic_bias_level_enum += SOC_ENUM_SINGLE(WM8940_INPUTCTL, 8, 2, wm8940_mic_bias_level_text); + +static const char *wm8940_filter_mode_text[] = {"Audio", "Application"}; +static const struct soc_enum wm8940_filter_mode_enum += SOC_ENUM_SINGLE(WM8940_ADC, 7, 2, wm8940_filter_mode_text); + +DECLARE_TLV_DB_SCALE(wm8940_spk_vol_tlv, -5700, 100, 1); +DECLARE_TLV_DB_SCALE(wm8940_att_tlv, -1000, 1000, 0); +DECLARE_TLV_DB_SCALE(wm8940_pga_vol_tlv, -1200, 75, 0); +DECLARE_TLV_DB_SCALE(wm8940_alc_min_tlv, -1200, 600, 0); +DECLARE_TLV_DB_SCALE(wm8940_alc_max_tlv, 675, 600, 0); +DECLARE_TLV_DB_SCALE(wm8940_alc_tar_tlv, -2250, 50, 0); +DECLARE_TLV_DB_SCALE(wm8940_lim_boost_tlv, 0, 100, 0); +DECLARE_TLV_DB_SCALE(wm8940_lim_thresh_tlv, -600, 100, 0); +DECLARE_TLV_DB_SCALE(wm8940_adc_tlv, -12750, 50, 1); +DECLARE_TLV_DB_SCALE(wm8940_capture_boost_vol_tlv, 0, 2000, 0); + +static const struct snd_kcontrol_new wm8940_snd_controls[] = { + SOC_SINGLE("Digital Loopback Switch", WM8940_COMPANDINGCTL, + 6, 1, 0), + SOC_ENUM("DAC Companding", wm8940_dac_companding_enum), + SOC_ENUM("ADC Companding", wm8940_adc_companding_enum), + + SOC_ENUM("ALC Mode", wm8940_alc_mode_enum), + SOC_SINGLE("ALC Switch", WM8940_ALC1, 8, 1, 0), + SOC_SINGLE_TLV("ALC Capture Max Gain", WM8940_ALC1, + 3, 7, 1, wm8940_alc_max_tlv), + SOC_SINGLE_TLV("ALC Capture Min Gain", WM8940_ALC1, + 0, 7, 0, wm8940_alc_min_tlv), + SOC_SINGLE_TLV("ALC Capture Target", WM8940_ALC2, + 0, 14, 0, wm8940_alc_tar_tlv), + SOC_SINGLE("ALC Capture Hold", WM8940_ALC2, 4, 10, 0), + SOC_SINGLE("ALC Capture Decay", WM8940_ALC3, 4, 10, 0), + SOC_SINGLE("ALC Capture Attach", WM8940_ALC3, 0, 10, 0), + SOC_SINGLE("ALC ZC Switch", WM8940_ALC4, 1, 1, 0), + SOC_SINGLE("ALC Capture Noise Gate Switch", WM8940_NOISEGATE, + 3, 1, 0), + SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8940_NOISEGATE, + 0, 7, 0), + + SOC_SINGLE("DAC Playback Limiter Switch", WM8940_DACLIM1, 8, 1, 0), + SOC_SINGLE("DAC Playback Limiter Attack", WM8940_DACLIM1, 0, 9, 0), + SOC_SINGLE("DAC Playback Limiter Decay", WM8940_DACLIM1, 4, 11, 0), + SOC_SINGLE_TLV("DAC Playback Limiter Threshold", WM8940_DACLIM2, + 4, 9, 1, wm8940_lim_thresh_tlv), + SOC_SINGLE_TLV("DAC Playback Limiter Boost", WM8940_DACLIM2, + 0, 12, 0, wm8940_lim_boost_tlv), + + SOC_SINGLE("Capture PGA ZC Switch", WM8940_PGAGAIN, 7, 1, 0), + SOC_SINGLE_TLV("Capture PGA Volume", WM8940_PGAGAIN, + 0, 63, 0, wm8940_pga_vol_tlv), + SOC_SINGLE_TLV("Digital Playback Volume", WM8940_DACVOL, + 0, 255, 0, wm8940_adc_tlv), + SOC_SINGLE_TLV("Digital Capture Volume", WM8940_ADCVOL, + 0, 255, 0, wm8940_adc_tlv), + SOC_ENUM("Mic Bias Level", wm8940_mic_bias_level_enum), + SOC_SINGLE_TLV("Capture Boost Volue", WM8940_ADCBOOST, + 8, 1, 0, wm8940_capture_boost_vol_tlv), + SOC_SINGLE_TLV("Speaker Playback Volume", WM8940_SPKVOL, + 0, 63, 0, wm8940_spk_vol_tlv), + SOC_SINGLE("Speaker Playback Switch", WM8940_SPKVOL, 6, 1, 1), + + SOC_SINGLE_TLV("Speaker Mixer Line Bypass Volume", WM8940_SPKVOL, + 8, 1, 1, wm8940_att_tlv), + SOC_SINGLE("Speaker Playback ZC Switch", WM8940_SPKVOL, 7, 1, 0), + + SOC_SINGLE("Mono Out Switch", WM8940_MONOMIX, 6, 1, 1), + SOC_SINGLE_TLV("Mono Mixer Line Bypass Volume", WM8940_MONOMIX, + 7, 1, 1, wm8940_att_tlv), + + SOC_SINGLE("High Pass Filter Switch", WM8940_ADC, 8, 1, 0), + SOC_ENUM("High Pass Filter Mode", wm8940_filter_mode_enum), + SOC_SINGLE("High Pass Filter Cut Off", WM8940_ADC, 4, 7, 0), + SOC_SINGLE("ADC Inversion Switch", WM8940_ADC, 0, 1, 0), + SOC_SINGLE("DAC Inversion Switch", WM8940_DAC, 0, 1, 0), + SOC_SINGLE("DAC Auto Mute Switch", WM8940_DAC, 2, 1, 0), + SOC_SINGLE("ZC Timeout Clock Switch", WM8940_ADDCNTRL, 0, 1, 0), +}; + +static const struct snd_kcontrol_new wm8940_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", WM8940_SPKMIX, 1, 1, 0), + SOC_DAPM_SINGLE("Aux Playback Switch", WM8940_SPKMIX, 5, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", WM8940_SPKMIX, 0, 1, 0), +}; + +static const struct snd_kcontrol_new wm8940_mono_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", WM8940_MONOMIX, 1, 1, 0), + SOC_DAPM_SINGLE("Aux Playback Switch", WM8940_MONOMIX, 2, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", WM8940_MONOMIX, 0, 1, 0), +}; + +DECLARE_TLV_DB_SCALE(wm8940_boost_vol_tlv, -1500, 300, 1); +static const struct snd_kcontrol_new wm8940_input_boost_controls[] = { + SOC_DAPM_SINGLE("Mic PGA Switch", WM8940_PGAGAIN, 6, 1, 1), + SOC_DAPM_SINGLE_TLV("Aux Volume", WM8940_ADCBOOST, + 0, 7, 0, wm8940_boost_vol_tlv), + SOC_DAPM_SINGLE_TLV("Mic Volume", WM8940_ADCBOOST, + 4, 7, 0, wm8940_boost_vol_tlv), +}; + +static const struct snd_kcontrol_new wm8940_micpga_controls[] = { + SOC_DAPM_SINGLE("AUX Switch", WM8940_INPUTCTL, 2, 1, 0), + SOC_DAPM_SINGLE("MICP Switch", WM8940_INPUTCTL, 0, 1, 0), + SOC_DAPM_SINGLE("MICN Switch", WM8940_INPUTCTL, 1, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8940_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Speaker Mixer", WM8940_POWER3, 2, 0, + &wm8940_speaker_mixer_controls[0], + ARRAY_SIZE(wm8940_speaker_mixer_controls)), + SND_SOC_DAPM_MIXER("Mono Mixer", WM8940_POWER3, 3, 0, + &wm8940_mono_mixer_controls[0], + ARRAY_SIZE(wm8940_mono_mixer_controls)), + SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8940_POWER3, 0, 0), + + SND_SOC_DAPM_PGA("SpkN Out", WM8940_POWER3, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("SpkP Out", WM8940_POWER3, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mono Out", WM8940_POWER3, 7, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("MONOOUT"), + SND_SOC_DAPM_OUTPUT("SPKOUTP"), + SND_SOC_DAPM_OUTPUT("SPKOUTN"), + + SND_SOC_DAPM_PGA("Aux Input", WM8940_POWER1, 6, 0, NULL, 0), + SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8940_POWER2, 0, 0), + SND_SOC_DAPM_MIXER("Mic PGA", WM8940_POWER2, 2, 0, + &wm8940_micpga_controls[0], + ARRAY_SIZE(wm8940_micpga_controls)), + SND_SOC_DAPM_MIXER("Boost Mixer", WM8940_POWER2, 4, 0, + &wm8940_input_boost_controls[0], + ARRAY_SIZE(wm8940_input_boost_controls)), + SND_SOC_DAPM_MICBIAS("Mic Bias", WM8940_POWER1, 4, 0), + + SND_SOC_DAPM_INPUT("MICN"), + SND_SOC_DAPM_INPUT("MICP"), + SND_SOC_DAPM_INPUT("AUX"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Mono output mixer */ + {"Mono Mixer", "PCM Playback Switch", "DAC"}, + {"Mono Mixer", "Aux Playback Switch", "Aux Input"}, + {"Mono Mixer", "Line Bypass Switch", "Boost Mixer"}, + + /* Speaker output mixer */ + {"Speaker Mixer", "PCM Playback Switch", "DAC"}, + {"Speaker Mixer", "Aux Playback Switch", "Aux Input"}, + {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"}, + + /* Outputs */ + {"Mono Out", NULL, "Mono Mixer"}, + {"MONOOUT", NULL, "Mono Out"}, + {"SpkN Out", NULL, "Speaker Mixer"}, + {"SpkP Out", NULL, "Speaker Mixer"}, + {"SPKOUTN", NULL, "SpkN Out"}, + {"SPKOUTP", NULL, "SpkP Out"}, + + /* Microphone PGA */ + {"Mic PGA", "MICN Switch", "MICN"}, + {"Mic PGA", "MICP Switch", "MICP"}, + {"Mic PGA", "AUX Switch", "AUX"}, + + /* Boost Mixer */ + {"Boost Mixer", "Mic PGA Switch", "Mic PGA"}, + {"Boost Mixer", "Mic Volume", "MICP"}, + {"Boost Mixer", "Aux Volume", "Aux Input"}, + + {"ADC", NULL, "Boost Mixer"}, +}; + +static int wm8940_add_widgets(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_dapm_new_controls(codec, 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)); + if (ret) + goto error_ret; + ret = snd_soc_dapm_new_widgets(codec); + +error_ret: + return ret; +} + +#define wm8940_reset(c) wm8940_write(c, WM8940_SOFTRESET, 0); + +static int wm8940_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = wm8940_read_reg_cache(codec, WM8940_IFACE) & 0xFE67; + u16 clk = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0x1fe; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + clk |= 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + wm8940_write(codec, WM8940_CLOCK, clk); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= (2 << 3); + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= (1 << 3); + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= (3 << 3); + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= (3 << 3) | (1 << 7); + break; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= (1 << 7); + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= (1 << 8); + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= (1 << 8) | (1 << 7); + break; + } + + wm8940_write(codec, WM8940_IFACE, iface); + + return 0; +} + +static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + u16 iface = wm8940_read_reg_cache(codec, WM8940_IFACE) & 0xFD9F; + u16 addcntrl = wm8940_read_reg_cache(codec, WM8940_ADDCNTRL) & 0xFFF1; + u16 companding = wm8940_read_reg_cache(codec, + WM8940_COMPANDINGCTL) & 0xFFDF; + int ret; + + /* LoutR control */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE + && params_channels(params) == 2) + iface |= (1 << 9); + + switch (params_rate(params)) { + case SNDRV_PCM_RATE_8000: + addcntrl |= (0x5 << 1); + break; + case SNDRV_PCM_RATE_11025: + addcntrl |= (0x4 << 1); + break; + case SNDRV_PCM_RATE_16000: + addcntrl |= (0x3 << 1); + break; + case SNDRV_PCM_RATE_22050: + addcntrl |= (0x2 << 1); + break; + case SNDRV_PCM_RATE_32000: + addcntrl |= (0x1 << 1); + break; + case SNDRV_PCM_RATE_44100: + case SNDRV_PCM_RATE_48000: + break; + } + ret = wm8940_write(codec, WM8940_ADDCNTRL, addcntrl); + if (ret) + goto error_ret; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + companding = companding | (1 << 5); + break; + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= (1 << 5); + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= (2 << 5); + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= (3 << 5); + break; + } + ret = wm8940_write(codec, WM8940_COMPANDINGCTL, companding); + if (ret) + goto error_ret; + ret = wm8940_write(codec, WM8940_IFACE, iface); + +error_ret: + return ret; +} + +static int wm8940_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = wm8940_read_reg_cache(codec, WM8940_DAC) & 0xffbf; + + if (mute) + mute_reg |= 0x40; + + return wm8940_write(codec, WM8940_DAC, mute_reg); +} + +static int wm8940_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 val; + u16 pwr_reg = wm8940_read_reg_cache(codec, WM8940_POWER1) & 0x1F0; + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + /* ensure bufioen and biasen */ + pwr_reg |= (1 << 2) | (1 << 3); + /* Enable thermal shutdown */ + val = wm8940_read_reg_cache(codec, WM8940_OUTPUTCTL); + ret = wm8940_write(codec, WM8940_OUTPUTCTL, val | 0x2); + if (ret) + break; + /* set vmid to 75k */ + ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x1); + break; + case SND_SOC_BIAS_PREPARE: + /* ensure bufioen and biasen */ + pwr_reg |= (1 << 2) | (1 << 3); + ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x1); + break; + case SND_SOC_BIAS_STANDBY: + /* ensure bufioen and biasen */ + pwr_reg |= (1 << 2) | (1 << 3); + /* set vmid to 300k for standby */ + ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x2); + break; + case SND_SOC_BIAS_OFF: + ret = wm8940_write(codec, WM8940_POWER1, pwr_reg); + break; + } + + return ret; +} + +struct pll_ { + unsigned int pre_scale:2; + unsigned int n:4; + unsigned int k; +}; + +static struct pll_ pll_div; + +/* The size in bits of the pll divide multiplied by 10 + * to allow rounding later */ +#define FIXED_PLL_SIZE ((1 << 24) * 10) +static void pll_factors(unsigned int target, unsigned int source) +{ + unsigned long long Kpart; + unsigned int K, Ndiv, Nmod; + /* The left shift ist to avoid accuracy loss when right shifting */ + Ndiv = target / source; + + if (Ndiv > 12) { + source <<= 1; + /* Multiply by 2 */ + pll_div.pre_scale = 0; + Ndiv = target / source; + } else if (Ndiv < 3) { + source >>= 2; + /* Divide by 4 */ + pll_div.pre_scale = 3; + Ndiv = target / source; + } else if (Ndiv < 6) { + source >>= 1; + /* divide by 2 */ + pll_div.pre_scale = 2; + Ndiv = target / source; + } else + pll_div.pre_scale = 1; + + if ((Ndiv < 6) || (Ndiv > 12)) + printk(KERN_WARNING + "WM8940 N value %d outwith recommended range!d\n", + Ndiv); + + pll_div.n = Ndiv; + Nmod = target % source; + Kpart = FIXED_PLL_SIZE * (long long)Nmod; + + do_div(Kpart, source); + + K = Kpart & 0xFFFFFFFF; + + /* Check if we need to round */ + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + K /= 10; + + pll_div.k = K; +} + +/* Untested at the moment */ +static int wm8940_set_dai_pll(struct snd_soc_dai *codec_dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + /* Turn off PLL */ + reg = wm8940_read_reg_cache(codec, WM8940_POWER1); + wm8940_write(codec, WM8940_POWER1, reg & 0x1df); + + if (freq_in == 0 || freq_out == 0) { + /* Clock CODEC directly from MCLK */ + reg = wm8940_read_reg_cache(codec, WM8940_CLOCK); + wm8940_write(codec, WM8940_CLOCK, reg & 0x0ff); + /* Pll power down */ + wm8940_write(codec, WM8940_PLLN, (1 << 7)); + return 0; + } + + /* Pll is followed by a frequency divide by 4 */ + pll_factors(freq_out*4, freq_in); + if (pll_div.k) + wm8940_write(codec, WM8940_PLLN, + (pll_div.pre_scale << 4) | pll_div.n | (1 << 6)); + else /* No factional component */ + wm8940_write(codec, WM8940_PLLN, + (pll_div.pre_scale << 4) | pll_div.n); + wm8940_write(codec, WM8940_PLLK1, pll_div.k >> 18); + wm8940_write(codec, WM8940_PLLK2, (pll_div.k >> 9) & 0x1ff); + wm8940_write(codec, WM8940_PLLK3, pll_div.k & 0x1ff); + /* Enable the PLL */ + reg = wm8940_read_reg_cache(codec, WM8940_POWER1); + wm8940_write(codec, WM8940_POWER1, reg | 0x020); + + /* Run CODEC from PLL instead of MCLK */ + reg = wm8940_read_reg_cache(codec, WM8940_CLOCK); + wm8940_write(codec, WM8940_CLOCK, reg | 0x100); + + return 0; +} + +static int wm8940_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 wm8940_priv *wm8940 = codec->private_data; + + switch (freq) { + case 11289600: + case 12000000: + case 12288000: + case 16934400: + case 18432000: + wm8940->sysclk = freq; + return 0; + } + return -EINVAL; +} + +static int wm8940_set_dai_clkdiv(struct snd_soc_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + int ret = 0; + + switch (div_id) { + case WM8940_BCLKDIV: + reg = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0xFFEF3; + ret = wm8940_write(codec, WM8940_CLOCK, reg | (div << 2)); + break; + case WM8940_MCLKDIV: + reg = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0xFF1F; + ret = wm8940_write(codec, WM8940_CLOCK, reg | (div << 5)); + break; + case WM8940_OPCLKDIV: + reg = wm8940_read_reg_cache(codec, WM8940_ADDCNTRL) & 0xFFCF; + ret = wm8940_write(codec, WM8940_ADDCNTRL, reg | (div << 4)); + break; + } + return ret; +} + +#define WM8940_RATES SNDRV_PCM_RATE_8000_48000 + +#define WM8940_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + 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 wm8940_dai_ops = { + .hw_params = wm8940_i2s_hw_params, + .set_sysclk = wm8940_set_dai_sysclk, + .digital_mute = wm8940_mute, + .set_fmt = wm8940_set_dai_fmt, + .set_clkdiv = wm8940_set_dai_clkdiv, + .set_pll = wm8940_set_dai_pll, +}; + +struct snd_soc_dai wm8940_dai = { + .name = "WM8940", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8940_RATES, + .formats = WM8940_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8940_RATES, + .formats = WM8940_FORMATS, + }, + .ops = &wm8940_dai_ops, + .symmetric_rates = 1, +}; +EXPORT_SYMBOL_GPL(wm8940_dai); + +static int wm8940_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + return wm8940_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static int wm8940_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + int i; + int ret; + u8 data[3]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware + * Could use auto incremented writes to speed this up + */ + for (i = 0; i < ARRAY_SIZE(wm8940_reg_defaults); i++) { + data[0] = i; + data[1] = (cache[i] & 0xFF00) >> 8; + data[2] = cache[i] & 0x00FF; + ret = codec->hw_write(codec->control_data, data, 3); + if (ret < 0) + goto error_ret; + else if (ret != 3) { + ret = -EIO; + goto error_ret; + } + } + ret = wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (ret) + goto error_ret; + ret = wm8940_set_bias_level(codec, codec->suspend_bias_level); + +error_ret: + return ret; +} + +static struct snd_soc_codec *wm8940_codec; + +static int wm8940_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + + int ret = 0; + + if (wm8940_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8940_codec; + codec = wm8940_codec; + + mutex_init(&codec->mutex); + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + ret = snd_soc_add_controls(codec, wm8940_snd_controls, + ARRAY_SIZE(wm8940_snd_controls)); + if (ret) + goto error_free_pcms; + ret = wm8940_add_widgets(codec); + if (ret) + goto error_free_pcms; + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto error_free_pcms; + } + + return ret; + +error_free_pcms: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +static int wm8940_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8940 = { + .probe = wm8940_probe, + .remove = wm8940_remove, + .suspend = wm8940_suspend, + .resume = wm8940_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8940); + +static int wm8940_register(struct wm8940_priv *wm8940) +{ + struct wm8940_setup_data *pdata = wm8940->codec.dev->platform_data; + struct snd_soc_codec *codec = &wm8940->codec; + int ret; + u16 reg; + if (wm8940_codec) { + dev_err(codec->dev, "Another WM8940 is registered\n"); + return -EINVAL; + } + + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8940; + codec->name = "WM8940"; + codec->owner = THIS_MODULE; + codec->read = wm8940_read_reg_cache; + codec->write = wm8940_write; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8940_set_bias_level; + codec->dai = &wm8940_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(wm8940_reg_defaults); + codec->reg_cache = &wm8940->reg_cache; + + memcpy(codec->reg_cache, wm8940_reg_defaults, + sizeof(wm8940_reg_defaults)); + + ret = wm8940_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + return ret; + } + + wm8940_dai.dev = codec->dev; + + wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + ret = wm8940_write(codec, WM8940_POWER1, 0x180); + if (ret < 0) + return ret; + + if (!pdata) + dev_warn(codec->dev, "No platform data supplied\n"); + else { + reg = wm8940_read_reg_cache(codec, WM8940_OUTPUTCTL); + ret = wm8940_write(codec, WM8940_OUTPUTCTL, reg | pdata->vroi); + if (ret < 0) + return ret; + } + + + wm8940_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + ret = snd_soc_register_dai(&wm8940_dai); + if (ret) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + return ret; + } + + return 0; +} + +static void wm8940_unregister(struct wm8940_priv *wm8940) +{ + wm8940_set_bias_level(&wm8940->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&wm8940_dai); + snd_soc_unregister_codec(&wm8940->codec); + kfree(wm8940); + wm8940_codec = NULL; +} + +static int wm8940_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8940_priv *wm8940; + struct snd_soc_codec *codec; + + wm8940 = kzalloc(sizeof *wm8940, GFP_KERNEL); + if (wm8940 == NULL) + return -ENOMEM; + + codec = &wm8940->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + i2c_set_clientdata(i2c, wm8940); + codec->control_data = i2c; + codec->dev = &i2c->dev; + + return wm8940_register(wm8940); +} + +static int wm8940_i2c_remove(struct i2c_client *client) +{ + struct wm8940_priv *wm8940 = i2c_get_clientdata(client); + + wm8940_unregister(wm8940); + + return 0; +} + +static const struct i2c_device_id wm8940_i2c_id[] = { + { "wm8940", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8940_i2c_id); + +static struct i2c_driver wm8940_i2c_driver = { + .driver = { + .name = "WM8940 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = wm8940_i2c_probe, + .remove = __devexit_p(wm8940_i2c_remove), + .id_table = wm8940_i2c_id, +}; + +static int __init wm8940_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&wm8940_i2c_driver); + if (ret) + printk(KERN_ERR "Failed to register WM8940 I2C driver: %d\n", + ret); + return ret; +} +module_init(wm8940_modinit); + +static void __exit wm8940_exit(void) +{ + i2c_del_driver(&wm8940_i2c_driver); +} +module_exit(wm8940_exit); + +MODULE_DESCRIPTION("ASoC WM8940 driver"); +MODULE_AUTHOR("Jonathan Cameron"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8940.h b/sound/soc/codecs/wm8940.h new file mode 100644 index 000000000000..8410eed3ef84 --- /dev/null +++ b/sound/soc/codecs/wm8940.h @@ -0,0 +1,104 @@ +/* + * wm8940.h -- WM8940 Soc Audio driver + * + * 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 _WM8940_H +#define _WM8940_H + +struct wm8940_setup_data { + /* Vref to analogue output resistance */ +#define WM8940_VROI_1K 0 +#define WM8940_VROI_30K 1 + unsigned int vroi:1; +}; +extern struct snd_soc_dai wm8940_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8940; + +/* WM8940 register space */ +#define WM8940_SOFTRESET 0x00 +#define WM8940_POWER1 0x01 +#define WM8940_POWER2 0x02 +#define WM8940_POWER3 0x03 +#define WM8940_IFACE 0x04 +#define WM8940_COMPANDINGCTL 0x05 +#define WM8940_CLOCK 0x06 +#define WM8940_ADDCNTRL 0x07 +#define WM8940_GPIO 0x08 +#define WM8940_CTLINT 0x09 +#define WM8940_DAC 0x0A +#define WM8940_DACVOL 0x0B + +#define WM8940_ADC 0x0E +#define WM8940_ADCVOL 0x0F +#define WM8940_NOTCH1 0x10 +#define WM8940_NOTCH2 0x11 +#define WM8940_NOTCH3 0x12 +#define WM8940_NOTCH4 0x13 +#define WM8940_NOTCH5 0x14 +#define WM8940_NOTCH6 0x15 +#define WM8940_NOTCH7 0x16 +#define WM8940_NOTCH8 0x17 +#define WM8940_DACLIM1 0x18 +#define WM8940_DACLIM2 0x19 + +#define WM8940_ALC1 0x20 +#define WM8940_ALC2 0x21 +#define WM8940_ALC3 0x22 +#define WM8940_NOISEGATE 0x23 +#define WM8940_PLLN 0x24 +#define WM8940_PLLK1 0x25 +#define WM8940_PLLK2 0x26 +#define WM8940_PLLK3 0x27 + +#define WM8940_ALC4 0x2A + +#define WM8940_INPUTCTL 0x2C +#define WM8940_PGAGAIN 0x2D + +#define WM8940_ADCBOOST 0x2F + +#define WM8940_OUTPUTCTL 0x31 +#define WM8940_SPKMIX 0x32 + +#define WM8940_SPKVOL 0x36 + +#define WM8940_MONOMIX 0x38 + +#define WM8940_CACHEREGNUM 0x57 + + +/* Clock divider Id's */ +#define WM8940_BCLKDIV 0 +#define WM8940_MCLKDIV 1 +#define WM8940_OPCLKDIV 2 + +/* MCLK clock dividers */ +#define WM8940_MCLKDIV_1 0 +#define WM8940_MCLKDIV_1_5 1 +#define WM8940_MCLKDIV_2 2 +#define WM8940_MCLKDIV_3 3 +#define WM8940_MCLKDIV_4 4 +#define WM8940_MCLKDIV_6 5 +#define WM8940_MCLKDIV_8 6 +#define WM8940_MCLKDIV_12 7 + +/* BCLK clock dividers */ +#define WM8940_BCLKDIV_1 0 +#define WM8940_BCLKDIV_2 1 +#define WM8940_BCLKDIV_4 2 +#define WM8940_BCLKDIV_8 3 +#define WM8940_BCLKDIV_16 4 +#define WM8940_BCLKDIV_32 5 + +/* PLL Out Dividers */ +#define WM8940_OPCLKDIV_1 0 +#define WM8940_OPCLKDIV_2 1 +#define WM8940_OPCLKDIV_3 2 +#define WM8940_OPCLKDIV_4 3 + +#endif /* _WM8940_H */ + -- cgit v1.2.3 From 9c935386512a3faa1be1c3d81cba38b7259a43f5 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 24 Apr 2009 15:00:25 +0200 Subject: ASoC: cs4270: fix Master Capture Switch polarity The control modifies the MUTE register, hence the polarity must be inverted. Signed-off-by: Daniel Mack Acked-By: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 7fa09a387622..3c34fe67c3d7 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -486,7 +486,7 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = { SOC_SINGLE("Zero Cross Switch", CS4270_TRANS, 5, 1, 0), SOC_SINGLE("Popguard Switch", CS4270_MODE, 0, 1, 1), SOC_SINGLE("Auto-Mute Switch", CS4270_MUTE, 5, 1, 0), - SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 0) + SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 1) }; /* -- cgit v1.2.3 From 1a4ba05ec8369d62c10155a8931e81267bfbd31c Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 24 Apr 2009 16:37:45 +0200 Subject: ASoC: cs4270: add Master Playback Switch This adds a new control named 'Master Playback Switch' for cs4270 codecs. It is implemented using the new SOC_DOUBLE_EXT macro to catch the put function and store the information about manually set mute controls from userspace. When a manual mute is set, we don't want the soc core to un-mute the outputs. Renamed cs4270_mute() to cs4270_dai_mute() to avoid confusion. Signed-off-by: Daniel Mack Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 44 +++++++++++++++++++++++++++++++++++++++----- 1 file changed, 39 insertions(+), 5 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 3c34fe67c3d7..ece6ed6a844f 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -109,6 +109,7 @@ struct cs4270_private { unsigned int mclk; /* Input frequency of the MCLK pin */ unsigned int mode; /* The mode (I2S or left-justified) */ unsigned int slave_mode; + unsigned int manual_mute; }; /** @@ -453,7 +454,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, } /** - * cs4270_mute - enable/disable the CS4270 external mute + * cs4270_dai_mute - enable/disable the CS4270 external mute * @dai: the SOC DAI * @mute: 0 = disable mute, 1 = enable mute * @@ -462,21 +463,52 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, * board does not have the MUTEA or MUTEB pins connected to such circuitry, * then this function will do nothing. */ -static int cs4270_mute(struct snd_soc_dai *dai, int mute) +static int cs4270_dai_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; + struct cs4270_private *cs4270 = codec->private_data; int reg6; reg6 = snd_soc_read(codec, CS4270_MUTE); if (mute) reg6 |= CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B; - else + else { reg6 &= ~(CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B); + reg6 |= cs4270->manual_mute; + } return snd_soc_write(codec, CS4270_MUTE, reg6); } +/** + * cs4270_soc_put_mute - put callback for the 'Master Playback switch' + * alsa control. + * @kcontrol: mixer control + * @ucontrol: control element information + * + * This function basically passes the arguments on to the generic + * snd_soc_put_volsw() function and saves the mute information in + * our private data structure. This is because we want to prevent + * cs4270_dai_mute() neglecting the user's decision to manually + * mute the codec's output. + * + * Returns 0 for success. + */ +static int cs4270_soc_put_mute(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs4270_private *cs4270 = codec->private_data; + int left = !ucontrol->value.integer.value[0]; + int right = !ucontrol->value.integer.value[1]; + + cs4270->manual_mute = (left ? CS4270_MUTE_DAC_A : 0) | + (right ? CS4270_MUTE_DAC_B : 0); + + return snd_soc_put_volsw(kcontrol, ucontrol); +} + /* A list of non-DAPM controls that the CS4270 supports */ static const struct snd_kcontrol_new cs4270_snd_controls[] = { SOC_DOUBLE_R("Master Playback Volume", @@ -486,7 +518,9 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = { SOC_SINGLE("Zero Cross Switch", CS4270_TRANS, 5, 1, 0), SOC_SINGLE("Popguard Switch", CS4270_MODE, 0, 1, 1), SOC_SINGLE("Auto-Mute Switch", CS4270_MUTE, 5, 1, 0), - SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 1) + SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 1), + SOC_DOUBLE_EXT("Master Playback Switch", CS4270_MUTE, 0, 1, 1, 1, + snd_soc_get_volsw, cs4270_soc_put_mute), }; /* @@ -506,7 +540,7 @@ static struct snd_soc_dai_ops cs4270_dai_ops = { .hw_params = cs4270_hw_params, .set_sysclk = cs4270_set_dai_sysclk, .set_fmt = cs4270_set_dai_fmt, - .digital_mute = cs4270_mute, + .digital_mute = cs4270_dai_mute, }; struct snd_soc_dai cs4270_dai = { -- cgit v1.2.3 From 6be01cfb854818298753bfce65543dbc81d51d5a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 27 Apr 2009 20:57:42 +0100 Subject: ASoC: Staticise TLV values in WM8940 Signed-off-by: Mark Brown --- sound/soc/codecs/wm8940.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 26987dcd8d55..a66dacc7cc83 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -168,16 +168,16 @@ static const char *wm8940_filter_mode_text[] = {"Audio", "Application"}; static const struct soc_enum wm8940_filter_mode_enum = SOC_ENUM_SINGLE(WM8940_ADC, 7, 2, wm8940_filter_mode_text); -DECLARE_TLV_DB_SCALE(wm8940_spk_vol_tlv, -5700, 100, 1); -DECLARE_TLV_DB_SCALE(wm8940_att_tlv, -1000, 1000, 0); -DECLARE_TLV_DB_SCALE(wm8940_pga_vol_tlv, -1200, 75, 0); -DECLARE_TLV_DB_SCALE(wm8940_alc_min_tlv, -1200, 600, 0); -DECLARE_TLV_DB_SCALE(wm8940_alc_max_tlv, 675, 600, 0); -DECLARE_TLV_DB_SCALE(wm8940_alc_tar_tlv, -2250, 50, 0); -DECLARE_TLV_DB_SCALE(wm8940_lim_boost_tlv, 0, 100, 0); -DECLARE_TLV_DB_SCALE(wm8940_lim_thresh_tlv, -600, 100, 0); -DECLARE_TLV_DB_SCALE(wm8940_adc_tlv, -12750, 50, 1); -DECLARE_TLV_DB_SCALE(wm8940_capture_boost_vol_tlv, 0, 2000, 0); +static DECLARE_TLV_DB_SCALE(wm8940_spk_vol_tlv, -5700, 100, 1); +static DECLARE_TLV_DB_SCALE(wm8940_att_tlv, -1000, 1000, 0); +static DECLARE_TLV_DB_SCALE(wm8940_pga_vol_tlv, -1200, 75, 0); +static DECLARE_TLV_DB_SCALE(wm8940_alc_min_tlv, -1200, 600, 0); +static DECLARE_TLV_DB_SCALE(wm8940_alc_max_tlv, 675, 600, 0); +static DECLARE_TLV_DB_SCALE(wm8940_alc_tar_tlv, -2250, 50, 0); +static DECLARE_TLV_DB_SCALE(wm8940_lim_boost_tlv, 0, 100, 0); +static DECLARE_TLV_DB_SCALE(wm8940_lim_thresh_tlv, -600, 100, 0); +static DECLARE_TLV_DB_SCALE(wm8940_adc_tlv, -12750, 50, 1); +static DECLARE_TLV_DB_SCALE(wm8940_capture_boost_vol_tlv, 0, 2000, 0); static const struct snd_kcontrol_new wm8940_snd_controls[] = { SOC_SINGLE("Digital Loopback Switch", WM8940_COMPANDINGCTL, @@ -253,7 +253,7 @@ static const struct snd_kcontrol_new wm8940_mono_mixer_controls[] = { SOC_DAPM_SINGLE("PCM Playback Switch", WM8940_MONOMIX, 0, 1, 0), }; -DECLARE_TLV_DB_SCALE(wm8940_boost_vol_tlv, -1500, 300, 1); +static DECLARE_TLV_DB_SCALE(wm8940_boost_vol_tlv, -1500, 300, 1); static const struct snd_kcontrol_new wm8940_input_boost_controls[] = { SOC_DAPM_SINGLE("Mic PGA Switch", WM8940_PGAGAIN, 6, 1, 1), SOC_DAPM_SINGLE_TLV("Aux Volume", WM8940_ADCBOOST, -- cgit v1.2.3 From 33f503c96c976fd585dedb76514ca6cb286e60d9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 2 May 2009 12:24:55 +0100 Subject: ASoC: Use a shared define for AC97 CODEC data formats The AC97 wire format is completely fixed so CODECs don't have any choice about the formats they accept but controllers accept a variety of data formats and render them down onto the bus. Have a shared define so all the CODEC drivers will interoperate with any of our controller drivers. Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 3 +++ sound/soc/codecs/ac97.c | 4 ++-- sound/soc/codecs/ad1980.c | 4 ++-- sound/soc/codecs/wm9705.c | 4 ++-- sound/soc/codecs/wm9712.c | 6 +++--- sound/soc/codecs/wm9713.c | 6 +++--- 6 files changed, 15 insertions(+), 12 deletions(-) (limited to 'sound/soc/codecs') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 22b729fbbf84..ea07b4bd5161 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -96,6 +96,9 @@ struct snd_pcm_substream; #define SND_SOC_CLOCK_IN 0 #define SND_SOC_CLOCK_OUT 1 +#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + struct snd_soc_dai_ops; struct snd_soc_dai; struct snd_ac97_bus_ops; diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index b0d4af145b87..932299bb5d1e 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -53,13 +53,13 @@ struct snd_soc_dai ac97_dai = { .channels_min = 1, .channels_max = 2, .rates = STD_AC97_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .formats = SND_SOC_STD_AC97_FMTS,}, .capture = { .stream_name = "AC97 Capture", .channels_min = 1, .channels_max = 2, .rates = STD_AC97_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .formats = SND_SOC_STD_AC97_FMTS,}, .ops = &ac97_dai_ops, }; EXPORT_SYMBOL_GPL(ac97_dai); diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index ddb3b08ac23c..d7440a982d22 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -137,13 +137,13 @@ struct snd_soc_dai ad1980_dai = { .channels_min = 2, .channels_max = 6, .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, }, + .formats = SND_SOC_STD_AC97_FMTS, }, .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, }, + .formats = SND_SOC_STD_AC97_FMTS, }, }; EXPORT_SYMBOL_GPL(ad1980_dai); diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index c2d1a7a18fa3..fa88b463e71f 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -282,14 +282,14 @@ struct snd_soc_dai wm9705_dai[] = { .channels_min = 1, .channels_max = 2, .rates = WM9705_AC97_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .formats = SND_SOC_STD_AC97_FMTS, }, .capture = { .stream_name = "HiFi Capture", .channels_min = 1, .channels_max = 2, .rates = WM9705_AC97_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .formats = SND_SOC_STD_AC97_FMTS, }, .ops = &wm9705_dai_ops, }, diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 765cf1e7369e..550c903f23bf 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -534,13 +534,13 @@ struct snd_soc_dai wm9712_dai[] = { .channels_min = 1, .channels_max = 2, .rates = WM9712_AC97_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .formats = SND_SOC_STD_AC97_FMTS,}, .capture = { .stream_name = "HiFi Capture", .channels_min = 1, .channels_max = 2, .rates = WM9712_AC97_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .formats = SND_SOC_STD_AC97_FMTS,}, .ops = &wm9712_dai_ops_hifi, }, { @@ -550,7 +550,7 @@ struct snd_soc_dai wm9712_dai[] = { .channels_min = 1, .channels_max = 1, .rates = WM9712_AC97_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .formats = SND_SOC_STD_AC97_FMTS,}, .ops = &wm9712_dai_ops_aux, } }; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index a6feb7842314..d1744e96f303 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1040,13 +1040,13 @@ struct snd_soc_dai wm9713_dai[] = { .channels_min = 1, .channels_max = 2, .rates = WM9713_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .formats = SND_SOC_STD_AC97_FMTS,}, .capture = { .stream_name = "HiFi Capture", .channels_min = 1, .channels_max = 2, .rates = WM9713_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .formats = SND_SOC_STD_AC97_FMTS,}, .ops = &wm9713_dai_ops_hifi, }, { @@ -1056,7 +1056,7 @@ struct snd_soc_dai wm9713_dai[] = { .channels_min = 1, .channels_max = 1, .rates = WM9713_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .formats = SND_SOC_STD_AC97_FMTS,}, .ops = &wm9713_dai_ops_aux, }, { -- cgit v1.2.3 From fcd274a345875b05c348ba19bc6b3dd48ecbb7d0 Mon Sep 17 00:00:00 2001 From: "Lopez Cruz, Misael" Date: Thu, 30 Apr 2009 21:47:22 -0500 Subject: ASoC: TWL4030: Add VDL analog bypass This patch adds voice downlink analog bypass switch. It follows the same approach as in other analog bypass switches. DAC switch is moved from 'DAC Voice' to 'Analog Voice Playback Mixer', that will also allow voice DAC to be powered in digital voice loopback (sidetone). Signed-off-by: Misael Lopez Cruz Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 37 +++++++++++++++++++++++++++++++------ 1 file changed, 31 insertions(+), 6 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index efa1a80b806c..efb371f6f01c 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -468,6 +468,10 @@ static const struct snd_kcontrol_new twl4030_dapm_abypassr2_control = static const struct snd_kcontrol_new twl4030_dapm_abypassl2_control = SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXL2_APGA_CTL, 2, 1, 0); +/* Analog bypass for Voice */ +static const struct snd_kcontrol_new twl4030_dapm_abypassv_control = + SOC_DAPM_SINGLE("Switch", TWL4030_REG_VDL_APGA_CTL, 2, 1, 0); + /* Digital bypass gain, 0 mutes the bypass */ static const unsigned int twl4030_dapm_dbypass_tlv[] = { TLV_DB_RANGE_HEAD(2), @@ -585,7 +589,7 @@ static int bypass_event(struct snd_soc_dapm_widget *w, struct soc_mixer_control *m = (struct soc_mixer_control *)w->kcontrols->private_value; struct twl4030_priv *twl4030 = w->codec->private_data; - unsigned char reg; + unsigned char reg, misc; reg = twl4030_read_reg_cache(w->codec, m->reg); @@ -597,14 +601,28 @@ static int bypass_event(struct snd_soc_dapm_widget *w, else twl4030->bypass_state &= ~(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL)); + } else if (m->reg == TWL4030_REG_VDL_APGA_CTL) { + /* Analog voice bypass */ + if (reg & (1 << m->shift)) + twl4030->bypass_state |= (1 << 4); + else + twl4030->bypass_state &= ~(1 << 4); } else { /* Digital bypass */ if (reg & (0x7 << m->shift)) - twl4030->bypass_state |= (1 << (m->shift ? 5 : 4)); + twl4030->bypass_state |= (1 << (m->shift ? 6 : 5)); else - twl4030->bypass_state &= ~(1 << (m->shift ? 5 : 4)); + twl4030->bypass_state &= ~(1 << (m->shift ? 6 : 5)); } + /* Enable master analog loopback mode if any analog switch is enabled*/ + misc = twl4030_read_reg_cache(w->codec, TWL4030_REG_MISC_SET_1); + if (twl4030->bypass_state & 0x1F) + misc |= TWL4030_FMLOOP_EN; + else + misc &= ~TWL4030_FMLOOP_EN; + twl4030_write(w->codec, TWL4030_REG_MISC_SET_1, misc); + if (w->codec->bias_level == SND_SOC_BIAS_STANDBY) { if (twl4030->bypass_state) twl4030_codec_mute(w->codec, 0); @@ -935,7 +953,7 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC Voice", "Voice Playback", - TWL4030_REG_AVDAC_CTL, 4, 0), + SND_SOC_NOPM, 0, 0), /* Analog PGAs */ SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL, @@ -962,6 +980,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_SWITCH_E("Left2 Analog Loopback", SND_SOC_NOPM, 0, 0, &twl4030_dapm_abypassl2_control, bypass_event, SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_SWITCH_E("Voice Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassv_control, + bypass_event, SND_SOC_DAPM_POST_REG), /* Digital bypasses */ SND_SOC_DAPM_SWITCH_E("Left Digital Loopback", SND_SOC_NOPM, 0, 0, @@ -979,6 +1000,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { 2, 0, NULL, 0), SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", TWL4030_REG_AVDAC_CTL, 3, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Analog Voice Playback Mixer", TWL4030_REG_AVDAC_CTL, + 4, 0, NULL, 0), /* Output MIXER controls */ /* Earpiece */ @@ -1067,13 +1090,13 @@ static const struct snd_soc_dapm_route intercon[] = { {"Analog R1 Playback Mixer", NULL, "DAC Right1"}, {"Analog L2 Playback Mixer", NULL, "DAC Left2"}, {"Analog R2 Playback Mixer", NULL, "DAC Right2"}, + {"Analog Voice Playback Mixer", NULL, "DAC Voice"}, {"ARXL1_APGA", NULL, "Analog L1 Playback Mixer"}, {"ARXR1_APGA", NULL, "Analog R1 Playback Mixer"}, {"ARXL2_APGA", NULL, "Analog L2 Playback Mixer"}, {"ARXR2_APGA", NULL, "Analog R2 Playback Mixer"}, - - {"VDL_APGA", NULL, "DAC Voice"}, + {"VDL_APGA", NULL, "Analog Voice Playback Mixer"}, /* Internal playback routings */ /* Earpiece */ @@ -1169,11 +1192,13 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left1 Analog Loopback", "Switch", "Analog Left Capture Route"}, {"Right2 Analog Loopback", "Switch", "Analog Right Capture Route"}, {"Left2 Analog Loopback", "Switch", "Analog Left Capture Route"}, + {"Voice Analog Loopback", "Switch", "Analog Left Capture Route"}, {"Analog R1 Playback Mixer", NULL, "Right1 Analog Loopback"}, {"Analog L1 Playback Mixer", NULL, "Left1 Analog Loopback"}, {"Analog R2 Playback Mixer", NULL, "Right2 Analog Loopback"}, {"Analog L2 Playback Mixer", NULL, "Left2 Analog Loopback"}, + {"Analog Voice Playback Mixer", NULL, "Voice Analog Loopback"}, /* Digital bypass routes */ {"Right Digital Loopback", "Volume", "TX1 Capture Route"}, -- cgit v1.2.3 From ee8f6894f358b6a04d8190fd78990749de98a498 Mon Sep 17 00:00:00 2001 From: "Lopez Cruz, Misael" Date: Thu, 30 Apr 2009 21:48:08 -0500 Subject: ASoC: TWL4030: Add voice digital loopback: sidetone This patch add voice digital loopback (sidetone) to the twl4030 driver. It mixes voice uplink attenuated (by sidetone gain) with voice downlink when the codec is working in option2 (voice/audio mode). Signed-off-by: Misael Lopez Cruz Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 27 +++++++++++++++++++++++++-- 1 file changed, 25 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index efb371f6f01c..23bae74ebf0a 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -491,6 +491,18 @@ static const struct snd_kcontrol_new twl4030_dapm_dbypassr_control = TWL4030_REG_ATX2ARXPGA, 0, 7, 0, twl4030_dapm_dbypass_tlv); +/* + * Voice Sidetone GAIN volume control: + * from -51 to -10 dB in 1 dB steps (mute instead of -51 dB) + */ +static DECLARE_TLV_DB_SCALE(twl4030_dapm_dbypassv_tlv, -5100, 100, 1); + +/* Digital bypass voice: sidetone (VUL -> VDL)*/ +static const struct snd_kcontrol_new twl4030_dapm_dbypassv_control = + SOC_DAPM_SINGLE_TLV("Volume", + TWL4030_REG_VSTPGA, 0, 0x29, 0, + twl4030_dapm_dbypassv_tlv); + static int micpath_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -607,12 +619,18 @@ static int bypass_event(struct snd_soc_dapm_widget *w, twl4030->bypass_state |= (1 << 4); else twl4030->bypass_state &= ~(1 << 4); + } else if (m->reg == TWL4030_REG_VSTPGA) { + /* Voice digital bypass */ + if (reg) + twl4030->bypass_state |= (1 << 5); + else + twl4030->bypass_state &= ~(1 << 5); } else { /* Digital bypass */ if (reg & (0x7 << m->shift)) - twl4030->bypass_state |= (1 << (m->shift ? 6 : 5)); + twl4030->bypass_state |= (1 << (m->shift ? 7 : 6)); else - twl4030->bypass_state &= ~(1 << (m->shift ? 6 : 5)); + twl4030->bypass_state &= ~(1 << (m->shift ? 7 : 6)); } /* Enable master analog loopback mode if any analog switch is enabled*/ @@ -991,6 +1009,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_SWITCH_E("Right Digital Loopback", SND_SOC_NOPM, 0, 0, &twl4030_dapm_dbypassr_control, bypass_event, SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_SWITCH_E("Voice Digital Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_dbypassv_control, bypass_event, + SND_SOC_DAPM_POST_REG), SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", TWL4030_REG_AVDAC_CTL, 0, 0, NULL, 0), @@ -1203,9 +1224,11 @@ static const struct snd_soc_dapm_route intercon[] = { /* Digital bypass routes */ {"Right Digital Loopback", "Volume", "TX1 Capture Route"}, {"Left Digital Loopback", "Volume", "TX1 Capture Route"}, + {"Voice Digital Loopback", "Volume", "TX2 Capture Route"}, {"Analog R2 Playback Mixer", NULL, "Right Digital Loopback"}, {"Analog L2 Playback Mixer", NULL, "Left Digital Loopback"}, + {"Analog Voice Playback Mixer", NULL, "Voice Digital Loopback"}, }; -- cgit v1.2.3 From 376f7839b72ec526173cafb5d8eadfc61e2effdf Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 5 May 2009 08:55:47 +0300 Subject: ASoC: TWL4030: Add VIBRA output This patch adds support for the VIBRA output on TWL4030 codec. The VIBRA output can be driven with audio data or with local vibrator driver. Add the needed DAPM elements and routes for the VIBRA output and controls for the VIBRA driver configuration. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 61 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 61 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 23bae74ebf0a..1a00e4b20390 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -396,6 +396,31 @@ static const struct soc_enum twl4030_handsfreer_enum = static const struct snd_kcontrol_new twl4030_dapm_handsfreer_control = SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum); +/* Vibra */ +/* Vibra audio path selection */ +static const char *twl4030_vibra_texts[] = + {"AudioL1", "AudioR1", "AudioL2", "AudioR2"}; + +static const struct soc_enum twl4030_vibra_enum = + SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 2, + ARRAY_SIZE(twl4030_vibra_texts), + twl4030_vibra_texts); + +static const struct snd_kcontrol_new twl4030_dapm_vibra_control = +SOC_DAPM_ENUM("Route", twl4030_vibra_enum); + +/* Vibra path selection: local vibrator (PWM) or audio driven */ +static const char *twl4030_vibrapath_texts[] = + {"Local vibrator", "Audio"}; + +static const struct soc_enum twl4030_vibrapath_enum = + SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 4, + ARRAY_SIZE(twl4030_vibrapath_texts), + twl4030_vibrapath_texts); + +static const struct snd_kcontrol_new twl4030_dapm_vibrapath_control = +SOC_DAPM_ENUM("Route", twl4030_vibrapath_enum); + /* Left analog microphone selection */ static const char *twl4030_analoglmic_texts[] = {"Off", "Main mic", "Headset mic", "AUXL", "Carkit mic"}; @@ -867,6 +892,26 @@ static const struct soc_enum twl4030_rampdelay_enum = ARRAY_SIZE(twl4030_rampdelay_texts), twl4030_rampdelay_texts); +/* Vibra H-bridge direction mode */ +static const char *twl4030_vibradirmode_texts[] = { + "Vibra H-bridge direction", "Audio data MSB", +}; + +static const struct soc_enum twl4030_vibradirmode_enum = + SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 5, + ARRAY_SIZE(twl4030_vibradirmode_texts), + twl4030_vibradirmode_texts); + +/* Vibra H-bridge direction */ +static const char *twl4030_vibradir_texts[] = { + "Positive polarity", "Negative polarity", +}; + +static const struct soc_enum twl4030_vibradir_enum = + SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 1, + ARRAY_SIZE(twl4030_vibradir_texts), + twl4030_vibradir_texts); + static const struct snd_kcontrol_new twl4030_snd_controls[] = { /* Common playback gain controls */ SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume", @@ -933,6 +978,9 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { 0, 3, 5, 0, input_gain_tlv), SOC_ENUM("HS ramp delay", twl4030_rampdelay_enum), + + SOC_ENUM("Vibra H-bridge mode", twl4030_vibradirmode_enum), + SOC_ENUM("Vibra H-bridge direction", twl4030_vibradir_enum), }; static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { @@ -960,6 +1008,7 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("CARKITR"), SND_SOC_DAPM_OUTPUT("HFL"), SND_SOC_DAPM_OUTPUT("HFR"), + SND_SOC_DAPM_OUTPUT("VIBRA"), /* DACs */ SND_SOC_DAPM_DAC("DAC Right1", "Right Front Playback", @@ -1060,6 +1109,11 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_MUX_E("HandsfreeR Mux", TWL4030_REG_HFR_CTL, 5, 0, &twl4030_dapm_handsfreer_control, handsfree_event, SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + /* Vibra */ + SND_SOC_DAPM_MUX("Vibra Mux", TWL4030_REG_VIBRA_CTL, 0, 0, + &twl4030_dapm_vibra_control), + SND_SOC_DAPM_MUX("Vibra Route", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_vibrapath_control), /* Introducing four virtual ADC, since TWL4030 have four channel for capture */ @@ -1161,6 +1215,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"HandsfreeR Mux", "AudioR1", "ARXR1_APGA"}, {"HandsfreeR Mux", "AudioR2", "ARXR2_APGA"}, {"HandsfreeR Mux", "AudioL2", "ARXL2_APGA"}, + /* Vibra */ + {"Vibra Mux", "AudioL1", "DAC Left1"}, + {"Vibra Mux", "AudioR1", "DAC Right1"}, + {"Vibra Mux", "AudioL2", "DAC Left2"}, + {"Vibra Mux", "AudioR2", "DAC Right2"}, /* outputs */ {"OUTL", NULL, "ARXL2_APGA"}, @@ -1174,6 +1233,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"CARKITR", NULL, "CarkitR Mixer"}, {"HFL", NULL, "HandsfreeL Mux"}, {"HFR", NULL, "HandsfreeR Mux"}, + {"Vibra Route", "Audio", "Vibra Mux"}, + {"VIBRA", NULL, "Vibra Route"}, /* Capture path */ {"Analog Left Capture Route", "Main mic", "MAINMIC"}, -- cgit v1.2.3 From 80ab8817bf9b740df1f0778c41875e93151409bf Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 5 May 2009 11:25:00 +0200 Subject: ASoC: cs4270: introduce CS4270_I2C_INCR Replace the magic 0x80 value with a suitable macro definition. Signed-off-by: Daniel Mack Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index ece6ed6a844f..153124b2e3b1 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -56,6 +56,7 @@ #define CS4270_FIRSTREG 0x01 #define CS4270_LASTREG 0x08 #define CS4270_NUMREGS (CS4270_LASTREG - CS4270_FIRSTREG + 1) +#define CS4270_I2C_INCR 0x80 /* Bit masks for the CS4270 registers */ #define CS4270_CHIPID_ID 0xF0 @@ -296,7 +297,7 @@ static int cs4270_fill_cache(struct snd_soc_codec *codec) s32 length; length = i2c_smbus_read_i2c_block_data(i2c_client, - CS4270_FIRSTREG | 0x80, CS4270_NUMREGS, cache); + CS4270_FIRSTREG | CS4270_I2C_INCR, CS4270_NUMREGS, cache); if (length != CS4270_NUMREGS) { dev_err(codec->dev, "i2c read failure, addr=0x%x\n", -- cgit v1.2.3 From 5e7c03442574ed0376c0621bfb0c477d79c12c71 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 6 May 2009 01:26:01 +0200 Subject: ASoC: cs4270: add power management support Signed-off-by: Daniel Mack Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 58 ++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 57 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 153124b2e3b1..a32b8226c8a4 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -18,7 +18,7 @@ * - The machine driver's 'startup' function must call * cs4270_set_dai_sysclk() with the value of MCLK. * - Only I2S and left-justified modes are supported - * - Power management is not supported + * - Power management is supported */ #include @@ -27,6 +27,7 @@ #include #include #include +#include #include "cs4270.h" @@ -65,6 +66,8 @@ #define CS4270_PWRCTL_PDN_ADC 0x20 #define CS4270_PWRCTL_PDN_DAC 0x02 #define CS4270_PWRCTL_PDN 0x01 +#define CS4270_PWRCTL_PDN_ALL \ + (CS4270_PWRCTL_PDN_ADC | CS4270_PWRCTL_PDN_DAC | CS4270_PWRCTL_PDN) #define CS4270_MODE_SPEED_MASK 0x30 #define CS4270_MODE_1X 0x00 #define CS4270_MODE_2X 0x10 @@ -788,6 +791,57 @@ static struct i2c_device_id cs4270_id[] = { }; MODULE_DEVICE_TABLE(i2c, cs4270_id); +#ifdef CONFIG_PM + +/* This suspend/resume implementation can handle both - a simple standby + * where the codec remains powered, and a full suspend, where the voltage + * domain the codec is connected to is teared down and/or any other hardware + * reset condition is asserted. + * + * The codec's own power saving features are enabled in the suspend callback, + * and all registers are written back to the hardware when resuming. + */ + +static int cs4270_i2c_suspend(struct i2c_client *client, pm_message_t mesg) +{ + struct cs4270_private *cs4270 = i2c_get_clientdata(client); + struct snd_soc_codec *codec = &cs4270->codec; + int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL; + + return snd_soc_write(codec, CS4270_PWRCTL, reg); +} + +static int cs4270_i2c_resume(struct i2c_client *client) +{ + struct cs4270_private *cs4270 = i2c_get_clientdata(client); + struct snd_soc_codec *codec = &cs4270->codec; + int reg; + + /* In case the device was put to hard reset during sleep, we need to + * wait 500ns here before any I2C communication. */ + ndelay(500); + + /* first restore the entire register cache ... */ + for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) { + u8 val = snd_soc_read(codec, reg); + + if (i2c_smbus_write_byte_data(client, reg, val)) { + dev_err(codec->dev, "i2c write failed\n"); + return -EIO; + } + } + + /* ... then disable the power-down bits */ + reg = snd_soc_read(codec, CS4270_PWRCTL); + reg &= ~CS4270_PWRCTL_PDN_ALL; + + return snd_soc_write(codec, CS4270_PWRCTL, reg); +} +#else +#define cs4270_i2c_suspend NULL +#define cs4270_i2c_resume NULL +#endif /* CONFIG_PM */ + /* * cs4270_i2c_driver - I2C device identification * @@ -802,6 +856,8 @@ static struct i2c_driver cs4270_i2c_driver = { .id_table = cs4270_id, .probe = cs4270_i2c_probe, .remove = cs4270_i2c_remove, + .suspend = cs4270_i2c_suspend, + .resume = cs4270_i2c_resume, }; /* -- cgit v1.2.3 From c198d811812417961582d4e25360372ca1eccdae Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 7 May 2009 14:32:00 +0300 Subject: ASoC: TWL4030: Fix typo in twl4030_codec_mute function Copy-paste error: TWL4030_PRECKL_GAIN >> TWL4030_PRECKR_GAIN It has not caused problems, since TWL4030_PRECKL_GAIN == TWL4030_PRECKR_GAIN == 0x30 Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 1a00e4b20390..fd392c65f475 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -237,7 +237,7 @@ static void twl4030_codec_mute(struct snd_soc_codec *codec, int mute) TWL4030_REG_PRECKL_CTL); reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PRECKR_CTL); twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, - reg_val & (~TWL4030_PRECKL_GAIN), + reg_val & (~TWL4030_PRECKR_GAIN), TWL4030_REG_PRECKR_CTL); /* Disable PLL */ -- cgit v1.2.3 From 31cb31f76e030ae05ed45f409ce5eb68ef2987f6 Mon Sep 17 00:00:00 2001 From: Roel Kluin Date: Mon, 11 May 2009 21:57:08 +0200 Subject: ASoC: remove driver_data direct access of struct device Signed-off-by: Mark Brown --- sound/soc/codecs/wm8400.c | 4 ++-- sound/soc/codecs/wm8731.c | 4 ++-- sound/soc/codecs/wm8753.c | 4 ++-- 3 files changed, 6 insertions(+), 6 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 510efa604008..e4547de8eec2 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1473,8 +1473,8 @@ static int wm8400_codec_probe(struct platform_device *dev) codec = &priv->codec; codec->private_data = priv; - codec->control_data = dev->dev.driver_data; - priv->wm8400 = dev->dev.driver_data; + codec->control_data = dev_get_drvdata(&dev->dev); + priv->wm8400 = dev_get_drvdata(&dev->dev); ret = regulator_bulk_get(priv->wm8400->dev, ARRAY_SIZE(power), &power[0]); diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index e043e3f60008..7a205876ef4f 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -666,14 +666,14 @@ static int __devinit wm8731_spi_probe(struct spi_device *spi) codec->hw_write = (hw_write_t)wm8731_spi_write; codec->dev = &spi->dev; - spi->dev.driver_data = wm8731; + dev_set_drvdata(&spi->dev, wm8731); return wm8731_register(wm8731); } static int __devexit wm8731_spi_remove(struct spi_device *spi) { - struct wm8731_priv *wm8731 = spi->dev.driver_data; + struct wm8731_priv *wm8731 = dev_get_drvdata(&spi->dev); wm8731_unregister(wm8731); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index a6e8f3f7f052..d121e58cae2b 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1822,14 +1822,14 @@ static int __devinit wm8753_spi_probe(struct spi_device *spi) codec->hw_write = (hw_write_t)wm8753_spi_write; codec->dev = &spi->dev; - spi->dev.driver_data = wm8753; + dev_set_drvdata(&spi->dev, wm8753); return wm8753_register(wm8753); } static int __devexit wm8753_spi_remove(struct spi_device *spi) { - struct wm8753_priv *wm8753 = spi->dev.driver_data; + struct wm8753_priv *wm8753 = dev_get_drvdata(&spi->dev); wm8753_unregister(wm8753); return 0; } -- cgit v1.2.3 From 97b8096dc92ae62b1d40e6bec7e7b257d2b30161 Mon Sep 17 00:00:00 2001 From: Joonyoung Shim Date: Mon, 11 May 2009 20:36:08 +0900 Subject: ASoC: TWL4030: change DAPM for analog microphone selection The inputs of the twl4030 codec can be mixed, so we will use the mixer DAPM for the analog microphone registers(0x05, 0x06), but if we enable more than one input at the same time, the input impedance of the input amplifier will be reduced. Signed-off-by: Joonyoung Shim Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 52 +++++++++++++++++----------------------------- 1 file changed, 19 insertions(+), 33 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index fd392c65f475..eaf91ab465b4 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -422,36 +422,18 @@ static const struct snd_kcontrol_new twl4030_dapm_vibrapath_control = SOC_DAPM_ENUM("Route", twl4030_vibrapath_enum); /* Left analog microphone selection */ -static const char *twl4030_analoglmic_texts[] = - {"Off", "Main mic", "Headset mic", "AUXL", "Carkit mic"}; - -static const unsigned int twl4030_analoglmic_values[] = - {0x0, 0x1, 0x2, 0x4, 0x8}; - -static const struct soc_enum twl4030_analoglmic_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICL, 0, 0xf, - ARRAY_SIZE(twl4030_analoglmic_texts), - twl4030_analoglmic_texts, - twl4030_analoglmic_values); - -static const struct snd_kcontrol_new twl4030_dapm_analoglmic_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_analoglmic_enum); +static const struct snd_kcontrol_new twl4030_dapm_analoglmic_controls[] = { + SOC_DAPM_SINGLE("Main mic", TWL4030_REG_ANAMICL, 0, 1, 0), + SOC_DAPM_SINGLE("Headset mic", TWL4030_REG_ANAMICL, 1, 1, 0), + SOC_DAPM_SINGLE("AUXL", TWL4030_REG_ANAMICL, 2, 1, 0), + SOC_DAPM_SINGLE("Carkit mic", TWL4030_REG_ANAMICL, 3, 1, 0), +}; /* Right analog microphone selection */ -static const char *twl4030_analogrmic_texts[] = - {"Off", "Sub mic", "AUXR"}; - -static const unsigned int twl4030_analogrmic_values[] = - {0x0, 0x1, 0x4}; - -static const struct soc_enum twl4030_analogrmic_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICR, 0, 0x5, - ARRAY_SIZE(twl4030_analogrmic_texts), - twl4030_analogrmic_texts, - twl4030_analogrmic_values); - -static const struct snd_kcontrol_new twl4030_dapm_analogrmic_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_analogrmic_enum); +static const struct snd_kcontrol_new twl4030_dapm_analogrmic_controls[] = { + SOC_DAPM_SINGLE("Sub mic", TWL4030_REG_ANAMICR, 0, 1, 0), + SOC_DAPM_SINGLE("AUXR", TWL4030_REG_ANAMICR, 1, 1, 0), +}; /* TX1 L/R Analog/Digital microphone selection */ static const char *twl4030_micpathtx1_texts[] = @@ -1138,11 +1120,15 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD| SND_SOC_DAPM_POST_REG), - /* Analog input muxes with switch for the capture amplifiers */ - SND_SOC_DAPM_VALUE_MUX("Analog Left Capture Route", - TWL4030_REG_ANAMICL, 4, 0, &twl4030_dapm_analoglmic_control), - SND_SOC_DAPM_VALUE_MUX("Analog Right Capture Route", - TWL4030_REG_ANAMICR, 4, 0, &twl4030_dapm_analogrmic_control), + /* Analog input mixers for the capture amplifiers */ + SND_SOC_DAPM_MIXER("Analog Left Capture Route", + TWL4030_REG_ANAMICL, 4, 0, + &twl4030_dapm_analoglmic_controls[0], + ARRAY_SIZE(twl4030_dapm_analoglmic_controls)), + SND_SOC_DAPM_MIXER("Analog Right Capture Route", + TWL4030_REG_ANAMICR, 4, 0, + &twl4030_dapm_analogrmic_controls[0], + ARRAY_SIZE(twl4030_dapm_analogrmic_controls)), SND_SOC_DAPM_PGA("ADC Physical Left", TWL4030_REG_AVADC_CTL, 3, 0, NULL, 0), -- cgit v1.2.3 From 63c26baa2aa624b023892d66ed696525fc787560 Mon Sep 17 00:00:00 2001 From: Marek Vasut Date: Thu, 14 May 2009 20:52:46 +0100 Subject: ASoC: Support AC97 link off by default on WM9712 The WM9712 can be configured by resistor strapping GPIO4 to behave like the WM9713 and default to leaving the AC97 link disabled after cold reset until a warm reset occurs. In this configuration we need to issue a warm reset after cold to bring the link up so do so. The warm reset will be harmless on systems that don't need it. [Changelog rewritten to document the reasoning. -- broonie] Signed-off-by: Marek Vasut Signed-off-by: Mark Brown --- sound/soc/codecs/wm9712.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 550c903f23bf..1fd4e88f50cf 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -585,6 +585,8 @@ static int wm9712_reset(struct snd_soc_codec *codec, int try_warm) } soc_ac97_ops.reset(codec->ac97); + if (soc_ac97_ops.warm_reset) + soc_ac97_ops.warm_reset(codec->ac97); if (ac97_read(codec, 0) != wm9712_reg[0]) goto err; return 0; -- cgit v1.2.3 From 2baaec28068d07db3d4ae6ba885fa07255b2ad79 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 15 May 2009 12:18:47 +0200 Subject: ASoC: Add missing __devexit in wm8940.c Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm8940.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index a66dacc7cc83..b8e17d6bc1f7 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -907,7 +907,7 @@ static int wm8940_i2c_probe(struct i2c_client *i2c, return wm8940_register(wm8940); } -static int wm8940_i2c_remove(struct i2c_client *client) +static int __devexit wm8940_i2c_remove(struct i2c_client *client) { struct wm8940_priv *wm8940 = i2c_get_clientdata(client); -- cgit v1.2.3 From b7a755a8a145a7e34e735bda9c452317de7a538a Mon Sep 17 00:00:00 2001 From: Misael Lopez Cruz Date: Sun, 17 May 2009 20:02:31 -0500 Subject: ASoC: TWL4030: Enable/disable voice digital filters Enable TWL4030 VTXL/VTXR and VRX digital filters for uplink and downlink paths, respectively. This patch also corrects voice 8/16kHz mode selection bit (SEL_16K) of CODEC_MODE register. Signed-off-by: Misael Lopez Cruz Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 37 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl4030.h | 2 +- 2 files changed, 38 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index eaf91ab465b4..e4d683daa450 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1629,6 +1629,28 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +/* In case of voice mode, the RX1 L(VRX) for downlink and the TX2 L/R + * (VTXL, VTXR) for uplink has to be enabled/disabled. */ +static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction, + int enable) +{ + u8 reg, mask; + + reg = twl4030_read_reg_cache(codec, TWL4030_REG_OPTION); + + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + mask = TWL4030_ARXL1_VRX_EN; + else + mask = TWL4030_ATXL2_VTXL_EN | TWL4030_ATXR2_VTXR_EN; + + if (enable) + reg |= mask; + else + reg &= ~mask; + + twl4030_write(codec, TWL4030_REG_OPTION, reg); +} + static int twl4030_voice_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -1665,6 +1687,17 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, return 0; } +static void twl4030_voice_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + + /* Enable voice digital filters */ + twl4030_voice_enable(codec, substream->stream, 0); +} + static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -1673,6 +1706,9 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = socdev->card->codec; u8 old_mode, mode; + /* Enable voice digital filters */ + twl4030_voice_enable(codec, substream->stream, 1); + /* bit rate */ old_mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) & ~(TWL4030_CODECPDZ); @@ -1780,6 +1816,7 @@ static struct snd_soc_dai_ops twl4030_dai_ops = { static struct snd_soc_dai_ops twl4030_dai_voice_ops = { .startup = twl4030_voice_startup, + .shutdown = twl4030_voice_shutdown, .hw_params = twl4030_voice_hw_params, .set_sysclk = twl4030_voice_set_dai_sysclk, .set_fmt = twl4030_voice_set_dai_fmt, diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 3441115136f6..9668bdf430fb 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -110,7 +110,7 @@ #define TWL4030_APLL_RATE_44100 0x90 #define TWL4030_APLL_RATE_48000 0xA0 #define TWL4030_APLL_RATE_96000 0xE0 -#define TWL4030_SEL_16K 0x04 +#define TWL4030_SEL_16K 0x08 #define TWL4030_CODECPDZ 0x02 #define TWL4030_OPT_MODE 0x01 #define TWL4030_OPTION_1 (1 << 0) -- cgit v1.2.3 From 181da78cd048ce866b05a2e0208ea09d2f80e721 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 19 May 2009 10:51:03 +0300 Subject: ASoC: TWL4030: Fix Analog capture path for AUXR AUXR is selected by bit 2 and not by bit 1 in the ANAMICR register. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index e4d683daa450..abf691493f43 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -432,7 +432,7 @@ static const struct snd_kcontrol_new twl4030_dapm_analoglmic_controls[] = { /* Right analog microphone selection */ static const struct snd_kcontrol_new twl4030_dapm_analogrmic_controls[] = { SOC_DAPM_SINGLE("Sub mic", TWL4030_REG_ANAMICR, 0, 1, 0), - SOC_DAPM_SINGLE("AUXR", TWL4030_REG_ANAMICR, 1, 1, 0), + SOC_DAPM_SINGLE("AUXR", TWL4030_REG_ANAMICR, 2, 1, 0), }; /* TX1 L/R Analog/Digital microphone selection */ -- cgit v1.2.3 From b74bd40fa4ae018898c8a6429c2a7daf61516524 Mon Sep 17 00:00:00 2001 From: "Lopez Cruz, Misael" Date: Mon, 18 May 2009 11:52:55 -0500 Subject: ASoC: TWL4030: Add control for selecting codec operation mode Add a control for selecting the codec operation mode. TWL4030 codec has two modes: - Option 1. Audio only (4 audio DACs) - Option 2. Voice/Audio (2 audio DACs and voice ADC/DAC) Control is restricted when a stream is ongoing, since codec's operation mode cannot be changed on-the-fly. Signed-off-by: Misael Lopez Cruz Acked-by: Peter Ujflausi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 47 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 47 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index abf691493f43..731534c19b7f 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -814,6 +814,48 @@ static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol, return err; } +/* Codec operation modes */ +static const char *twl4030_op_modes_texts[] = { + "Option 2 (voice/audio)", "Option 1 (audio)" +}; + +static const struct soc_enum twl4030_op_modes_enum = + SOC_ENUM_SINGLE(TWL4030_REG_CODEC_MODE, 0, + ARRAY_SIZE(twl4030_op_modes_texts), + twl4030_op_modes_texts); + +int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct twl4030_priv *twl4030 = codec->private_data; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned short val; + unsigned short mask, bitmask; + + if (twl4030->configured) { + printk(KERN_ERR "twl4030 operation mode cannot be " + "changed on-the-fly\n"); + return -EBUSY; + } + + for (bitmask = 1; bitmask < e->max; bitmask <<= 1) + ; + if (ucontrol->value.enumerated.item[0] > e->max - 1) + return -EINVAL; + + val = ucontrol->value.enumerated.item[0] << e->shift_l; + mask = (bitmask - 1) << e->shift_l; + if (e->shift_l != e->shift_r) { + if (ucontrol->value.enumerated.item[1] > e->max - 1) + return -EINVAL; + val |= ucontrol->value.enumerated.item[1] << e->shift_r; + mask |= (bitmask - 1) << e->shift_r; + } + + return snd_soc_update_bits(codec, e->reg, mask, val); +} + /* * FGAIN volume control: * from -62 to 0 dB in 1 dB steps (mute instead of -63 dB) @@ -895,6 +937,11 @@ static const struct soc_enum twl4030_vibradir_enum = twl4030_vibradir_texts); static const struct snd_kcontrol_new twl4030_snd_controls[] = { + /* Codec operation mode control */ + SOC_ENUM_EXT("Codec Operation Mode", twl4030_op_modes_enum, + snd_soc_get_enum_double, + snd_soc_put_twl4030_opmode_enum_double), + /* Common playback gain controls */ SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume", TWL4030_REG_ARXL1PGA, TWL4030_REG_ARXR1PGA, -- cgit v1.2.3 From 4005d39a5f5549f1f6afe88abceed78b2ab225b6 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 18 May 2009 16:02:04 +0300 Subject: ASoC: TWL4030: Change DAPM routings and controls for DACs and PGAs Restructuring the twl4030 codec's DAPM routing to be able to handle the power sequences correctly. The twl4030 codec internal implementation have this order: DAC -> Analog PGA -> Mixer/Mux While the ASoC framework expects the following order: DAC -> Mixer -> Analog PGA This patch moves the Analog PGA handling from SND_SOC_DAPM_PGA to _MIXER and adds two levels of mixer to handle the digital and analog loopback functionality. Now the analog loopback does not powers on any of the DACs. Signed-off-by: Peter Ujfalusi Tested-by: Anuj Aggarwal Tested-by: Jarkko Nikula Tested-by: Misael Lopez Cruz Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 141 +++++++++++++++++++++++---------------------- 1 file changed, 71 insertions(+), 70 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 731534c19b7f..99fe44f70507 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1051,18 +1051,6 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC Voice", "Voice Playback", SND_SOC_NOPM, 0, 0), - /* Analog PGAs */ - SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL, - 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("ARXL1_APGA", TWL4030_REG_ARXL1_APGA_CTL, - 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("ARXR2_APGA", TWL4030_REG_ARXR2_APGA_CTL, - 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("ARXL2_APGA", TWL4030_REG_ARXL2_APGA_CTL, - 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("VDL_APGA", TWL4030_REG_VDL_APGA_CTL, - 0, 0, NULL, 0), - /* Analog bypasses */ SND_SOC_DAPM_SWITCH_E("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0, &twl4030_dapm_abypassr1_control, bypass_event, @@ -1091,16 +1079,29 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { &twl4030_dapm_dbypassv_control, bypass_event, SND_SOC_DAPM_POST_REG), - SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", TWL4030_REG_AVDAC_CTL, - 0, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Analog L1 Playback Mixer", TWL4030_REG_AVDAC_CTL, - 1, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Analog R2 Playback Mixer", TWL4030_REG_AVDAC_CTL, - 2, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", TWL4030_REG_AVDAC_CTL, - 3, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Analog Voice Playback Mixer", TWL4030_REG_AVDAC_CTL, - 4, 0, NULL, 0), + /* Digital mixers, power control for the physical DACs */ + SND_SOC_DAPM_MIXER("Digital R1 Playback Mixer", + TWL4030_REG_AVDAC_CTL, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Digital L1 Playback Mixer", + TWL4030_REG_AVDAC_CTL, 1, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Digital R2 Playback Mixer", + TWL4030_REG_AVDAC_CTL, 2, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Digital L2 Playback Mixer", + TWL4030_REG_AVDAC_CTL, 3, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Digital Voice Playback Mixer", + TWL4030_REG_AVDAC_CTL, 4, 0, NULL, 0), + + /* Analog mixers, power control for the physical PGAs */ + SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", + TWL4030_REG_ARXR1_APGA_CTL, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Analog L1 Playback Mixer", + TWL4030_REG_ARXL1_APGA_CTL, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Analog R2 Playback Mixer", + TWL4030_REG_ARXR2_APGA_CTL, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", + TWL4030_REG_ARXL2_APGA_CTL, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Analog Voice Playback Mixer", + TWL4030_REG_VDL_APGA_CTL, 0, 0, NULL, 0), /* Output MIXER controls */ /* Earpiece */ @@ -1194,60 +1195,60 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { }; static const struct snd_soc_dapm_route intercon[] = { - {"Analog L1 Playback Mixer", NULL, "DAC Left1"}, - {"Analog R1 Playback Mixer", NULL, "DAC Right1"}, - {"Analog L2 Playback Mixer", NULL, "DAC Left2"}, - {"Analog R2 Playback Mixer", NULL, "DAC Right2"}, - {"Analog Voice Playback Mixer", NULL, "DAC Voice"}, - - {"ARXL1_APGA", NULL, "Analog L1 Playback Mixer"}, - {"ARXR1_APGA", NULL, "Analog R1 Playback Mixer"}, - {"ARXL2_APGA", NULL, "Analog L2 Playback Mixer"}, - {"ARXR2_APGA", NULL, "Analog R2 Playback Mixer"}, - {"VDL_APGA", NULL, "Analog Voice Playback Mixer"}, + {"Digital L1 Playback Mixer", NULL, "DAC Left1"}, + {"Digital R1 Playback Mixer", NULL, "DAC Right1"}, + {"Digital L2 Playback Mixer", NULL, "DAC Left2"}, + {"Digital R2 Playback Mixer", NULL, "DAC Right2"}, + {"Digital Voice Playback Mixer", NULL, "DAC Voice"}, + + {"Analog L1 Playback Mixer", NULL, "Digital L1 Playback Mixer"}, + {"Analog R1 Playback Mixer", NULL, "Digital R1 Playback Mixer"}, + {"Analog L2 Playback Mixer", NULL, "Digital L2 Playback Mixer"}, + {"Analog R2 Playback Mixer", NULL, "Digital R2 Playback Mixer"}, + {"Analog Voice Playback Mixer", NULL, "Digital Voice Playback Mixer"}, /* Internal playback routings */ /* Earpiece */ - {"Earpiece Mixer", "Voice", "VDL_APGA"}, - {"Earpiece Mixer", "AudioL1", "ARXL1_APGA"}, - {"Earpiece Mixer", "AudioL2", "ARXL2_APGA"}, - {"Earpiece Mixer", "AudioR1", "ARXR1_APGA"}, + {"Earpiece Mixer", "Voice", "Analog Voice Playback Mixer"}, + {"Earpiece Mixer", "AudioL1", "Analog L1 Playback Mixer"}, + {"Earpiece Mixer", "AudioL2", "Analog L2 Playback Mixer"}, + {"Earpiece Mixer", "AudioR1", "Analog R1 Playback Mixer"}, /* PreDrivL */ - {"PredriveL Mixer", "Voice", "VDL_APGA"}, - {"PredriveL Mixer", "AudioL1", "ARXL1_APGA"}, - {"PredriveL Mixer", "AudioL2", "ARXL2_APGA"}, - {"PredriveL Mixer", "AudioR2", "ARXR2_APGA"}, + {"PredriveL Mixer", "Voice", "Analog Voice Playback Mixer"}, + {"PredriveL Mixer", "AudioL1", "Analog L1 Playback Mixer"}, + {"PredriveL Mixer", "AudioL2", "Analog L2 Playback Mixer"}, + {"PredriveL Mixer", "AudioR2", "Analog R2 Playback Mixer"}, /* PreDrivR */ - {"PredriveR Mixer", "Voice", "VDL_APGA"}, - {"PredriveR Mixer", "AudioR1", "ARXR1_APGA"}, - {"PredriveR Mixer", "AudioR2", "ARXR2_APGA"}, - {"PredriveR Mixer", "AudioL2", "ARXL2_APGA"}, + {"PredriveR Mixer", "Voice", "Analog Voice Playback Mixer"}, + {"PredriveR Mixer", "AudioR1", "Analog R1 Playback Mixer"}, + {"PredriveR Mixer", "AudioR2", "Analog R2 Playback Mixer"}, + {"PredriveR Mixer", "AudioL2", "Analog L2 Playback Mixer"}, /* HeadsetL */ - {"HeadsetL Mixer", "Voice", "VDL_APGA"}, - {"HeadsetL Mixer", "AudioL1", "ARXL1_APGA"}, - {"HeadsetL Mixer", "AudioL2", "ARXL2_APGA"}, + {"HeadsetL Mixer", "Voice", "Analog Voice Playback Mixer"}, + {"HeadsetL Mixer", "AudioL1", "Analog L1 Playback Mixer"}, + {"HeadsetL Mixer", "AudioL2", "Analog L2 Playback Mixer"}, /* HeadsetR */ - {"HeadsetR Mixer", "Voice", "VDL_APGA"}, - {"HeadsetR Mixer", "AudioR1", "ARXR1_APGA"}, - {"HeadsetR Mixer", "AudioR2", "ARXR2_APGA"}, + {"HeadsetR Mixer", "Voice", "Analog Voice Playback Mixer"}, + {"HeadsetR Mixer", "AudioR1", "Analog R1 Playback Mixer"}, + {"HeadsetR Mixer", "AudioR2", "Analog R2 Playback Mixer"}, /* CarkitL */ - {"CarkitL Mixer", "Voice", "VDL_APGA"}, - {"CarkitL Mixer", "AudioL1", "ARXL1_APGA"}, - {"CarkitL Mixer", "AudioL2", "ARXL2_APGA"}, + {"CarkitL Mixer", "Voice", "Analog Voice Playback Mixer"}, + {"CarkitL Mixer", "AudioL1", "Analog L1 Playback Mixer"}, + {"CarkitL Mixer", "AudioL2", "Analog L2 Playback Mixer"}, /* CarkitR */ - {"CarkitR Mixer", "Voice", "VDL_APGA"}, - {"CarkitR Mixer", "AudioR1", "ARXR1_APGA"}, - {"CarkitR Mixer", "AudioR2", "ARXR2_APGA"}, + {"CarkitR Mixer", "Voice", "Analog Voice Playback Mixer"}, + {"CarkitR Mixer", "AudioR1", "Analog R1 Playback Mixer"}, + {"CarkitR Mixer", "AudioR2", "Analog R2 Playback Mixer"}, /* HandsfreeL */ - {"HandsfreeL Mux", "Voice", "VDL_APGA"}, - {"HandsfreeL Mux", "AudioL1", "ARXL1_APGA"}, - {"HandsfreeL Mux", "AudioL2", "ARXL2_APGA"}, - {"HandsfreeL Mux", "AudioR2", "ARXR2_APGA"}, + {"HandsfreeL Mux", "Voice", "Analog Voice Playback Mixer"}, + {"HandsfreeL Mux", "AudioL1", "Analog L1 Playback Mixer"}, + {"HandsfreeL Mux", "AudioL2", "Analog L2 Playback Mixer"}, + {"HandsfreeL Mux", "AudioR2", "Analog R2 Playback Mixer"}, /* HandsfreeR */ - {"HandsfreeR Mux", "Voice", "VDL_APGA"}, - {"HandsfreeR Mux", "AudioR1", "ARXR1_APGA"}, - {"HandsfreeR Mux", "AudioR2", "ARXR2_APGA"}, - {"HandsfreeR Mux", "AudioL2", "ARXL2_APGA"}, + {"HandsfreeR Mux", "Voice", "Analog Voice Playback Mixer"}, + {"HandsfreeR Mux", "AudioR1", "Analog R1 Playback Mixer"}, + {"HandsfreeR Mux", "AudioR2", "Analog R2 Playback Mixer"}, + {"HandsfreeR Mux", "AudioL2", "Analog L2 Playback Mixer"}, /* Vibra */ {"Vibra Mux", "AudioL1", "DAC Left1"}, {"Vibra Mux", "AudioR1", "DAC Right1"}, @@ -1255,8 +1256,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"Vibra Mux", "AudioR2", "DAC Right2"}, /* outputs */ - {"OUTL", NULL, "ARXL2_APGA"}, - {"OUTR", NULL, "ARXR2_APGA"}, + {"OUTL", NULL, "Analog L2 Playback Mixer"}, + {"OUTR", NULL, "Analog R2 Playback Mixer"}, {"EARPIECE", NULL, "Earpiece Mixer"}, {"PREDRIVEL", NULL, "PredriveL Mixer"}, {"PREDRIVER", NULL, "PredriveR Mixer"}, @@ -1320,9 +1321,9 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left Digital Loopback", "Volume", "TX1 Capture Route"}, {"Voice Digital Loopback", "Volume", "TX2 Capture Route"}, - {"Analog R2 Playback Mixer", NULL, "Right Digital Loopback"}, - {"Analog L2 Playback Mixer", NULL, "Left Digital Loopback"}, - {"Analog Voice Playback Mixer", NULL, "Voice Digital Loopback"}, + {"Digital R2 Playback Mixer", NULL, "Right Digital Loopback"}, + {"Digital L2 Playback Mixer", NULL, "Left Digital Loopback"}, + {"Digital Voice Playback Mixer", NULL, "Voice Digital Loopback"}, }; -- cgit v1.2.3 From 6943c92e87c4aa2a6d7a1f4dbd79cf4a0b5fd67b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 18 May 2009 16:02:05 +0300 Subject: ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles the headset ramp up and down sequences needed for the pop noise removal. With this patch the order of the internal components in the twl4030 codec is turned on and off in a correct order. Signed-off-by: Peter Ujfalusi Tested-by: Anuj Aggarwal Tested-by: Jarkko Nikula Tested-by: Misael Lopez Cruz Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 116 +++++++++++++++++++++++++++++++++++---------- 1 file changed, 91 insertions(+), 25 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 99fe44f70507..f554672f67c1 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -130,6 +130,12 @@ struct twl4030_priv { unsigned int rate; unsigned int sample_bits; unsigned int channels; + + unsigned int sysclk; + + /* Headset output state handling */ + unsigned int hsl_enabled; + unsigned int hsr_enabled; }; /* @@ -564,39 +570,85 @@ static int handsfree_event(struct snd_soc_dapm_widget *w, return 0; } -static int headsetl_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static void headset_ramp(struct snd_soc_codec *codec, int ramp) { unsigned char hs_gain, hs_pop; + struct twl4030_priv *twl4030 = codec->private_data; + /* Base values for ramp delay calculation: 2^19 - 2^26 */ + unsigned int ramp_base[] = {524288, 1048576, 2097152, 4194304, + 8388608, 16777216, 33554432, 67108864}; - /* Save the current volume */ - hs_gain = twl4030_read_reg_cache(w->codec, TWL4030_REG_HS_GAIN_SET); - hs_pop = twl4030_read_reg_cache(w->codec, TWL4030_REG_HS_POPN_SET); + hs_gain = twl4030_read_reg_cache(codec, TWL4030_REG_HS_GAIN_SET); + hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); - switch (event) { - case SND_SOC_DAPM_POST_PMU: - /* Do the anti-pop/bias ramp enable according to the TRM */ + if (ramp) { + /* Headset ramp-up according to the TRM */ hs_pop |= TWL4030_VMID_EN; - twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop); - /* Is this needed? Can we just use whatever gain here? */ - twl4030_write(w->codec, TWL4030_REG_HS_GAIN_SET, - (hs_gain & (~0x0f)) | 0x0a); + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); + twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hs_gain); hs_pop |= TWL4030_RAMP_EN; - twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop); - - /* Restore the original volume */ - twl4030_write(w->codec, TWL4030_REG_HS_GAIN_SET, hs_gain); - break; - case SND_SOC_DAPM_POST_PMD: - /* Do the anti-pop/bias ramp disable according to the TRM */ + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); + } else { + /* Headset ramp-down _not_ according to + * the TRM, but in a way that it is working */ hs_pop &= ~TWL4030_RAMP_EN; - twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop); + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); + /* Wait ramp delay time + 1, so the VMID can settle */ + mdelay((ramp_base[(hs_pop & TWL4030_RAMP_DELAY) >> 2] / + twl4030->sysclk) + 1); /* Bypass the reg_cache to mute the headset */ twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, hs_gain & (~0x0f), TWL4030_REG_HS_GAIN_SET); + hs_pop &= ~TWL4030_VMID_EN; - twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop); + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); + } +} + +static int headsetlpga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct twl4030_priv *twl4030 = w->codec->private_data; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* Do the ramp-up only once */ + if (!twl4030->hsr_enabled) + headset_ramp(w->codec, 1); + + twl4030->hsl_enabled = 1; + break; + case SND_SOC_DAPM_POST_PMD: + /* Do the ramp-down only if both headsetL/R is disabled */ + if (!twl4030->hsr_enabled) + headset_ramp(w->codec, 0); + + twl4030->hsl_enabled = 0; + break; + } + return 0; +} + +static int headsetrpga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct twl4030_priv *twl4030 = w->codec->private_data; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* Do the ramp-up only once */ + if (!twl4030->hsl_enabled) + headset_ramp(w->codec, 1); + + twl4030->hsr_enabled = 1; + break; + case SND_SOC_DAPM_POST_PMD: + /* Do the ramp-down only if both headsetL/R is disabled */ + if (!twl4030->hsl_enabled) + headset_ramp(w->codec, 0); + + twl4030->hsr_enabled = 0; break; } return 0; @@ -1116,13 +1168,18 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { &twl4030_dapm_predriver_controls[0], ARRAY_SIZE(twl4030_dapm_predriver_controls)), /* HeadsetL/R */ - SND_SOC_DAPM_MIXER_E("HeadsetL Mixer", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_MIXER("HeadsetL Mixer", SND_SOC_NOPM, 0, 0, &twl4030_dapm_hsol_controls[0], - ARRAY_SIZE(twl4030_dapm_hsol_controls), headsetl_event, + ARRAY_SIZE(twl4030_dapm_hsol_controls)), + SND_SOC_DAPM_PGA_E("HeadsetL PGA", SND_SOC_NOPM, + 0, 0, NULL, 0, headsetlpga_event, SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MIXER("HeadsetR Mixer", SND_SOC_NOPM, 0, 0, &twl4030_dapm_hsor_controls[0], ARRAY_SIZE(twl4030_dapm_hsor_controls)), + SND_SOC_DAPM_PGA_E("HeadsetR PGA", SND_SOC_NOPM, + 0, 0, NULL, 0, headsetrpga_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), /* CarkitL/R */ SND_SOC_DAPM_MIXER("CarkitL Mixer", SND_SOC_NOPM, 0, 0, &twl4030_dapm_carkitl_controls[0], @@ -1227,10 +1284,12 @@ static const struct snd_soc_dapm_route intercon[] = { {"HeadsetL Mixer", "Voice", "Analog Voice Playback Mixer"}, {"HeadsetL Mixer", "AudioL1", "Analog L1 Playback Mixer"}, {"HeadsetL Mixer", "AudioL2", "Analog L2 Playback Mixer"}, + {"HeadsetL PGA", NULL, "HeadsetL Mixer"}, /* HeadsetR */ {"HeadsetR Mixer", "Voice", "Analog Voice Playback Mixer"}, {"HeadsetR Mixer", "AudioR1", "Analog R1 Playback Mixer"}, {"HeadsetR Mixer", "AudioR2", "Analog R2 Playback Mixer"}, + {"HeadsetR PGA", NULL, "HeadsetR Mixer"}, /* CarkitL */ {"CarkitL Mixer", "Voice", "Analog Voice Playback Mixer"}, {"CarkitL Mixer", "AudioL1", "Analog L1 Playback Mixer"}, @@ -1261,8 +1320,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"EARPIECE", NULL, "Earpiece Mixer"}, {"PREDRIVEL", NULL, "PredriveL Mixer"}, {"PREDRIVER", NULL, "PredriveR Mixer"}, - {"HSOL", NULL, "HeadsetL Mixer"}, - {"HSOR", NULL, "HeadsetR Mixer"}, + {"HSOL", NULL, "HeadsetL PGA"}, + {"HSOR", NULL, "HeadsetR PGA"}, {"CARKITL", NULL, "CarkitL Mixer"}, {"CARKITR", NULL, "CarkitR Mixer"}, {"HFL", NULL, "HandsfreeL Mux"}, @@ -1601,17 +1660,21 @@ static int twl4030_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 twl4030_priv *twl4030 = codec->private_data; u8 infreq; switch (freq) { case 19200000: infreq = TWL4030_APLL_INFREQ_19200KHZ; + twl4030->sysclk = 19200; break; case 26000000: infreq = TWL4030_APLL_INFREQ_26000KHZ; + twl4030->sysclk = 26000; break; case 38400000: infreq = TWL4030_APLL_INFREQ_38400KHZ; + twl4030->sysclk = 38400; break; default: printk(KERN_ERR "TWL4030 set sysclk: unknown rate %d\n", @@ -2000,6 +2063,9 @@ static int twl4030_probe(struct platform_device *pdev) kfree(codec); return -ENOMEM; } + /* Set default sysclk (used by the headsetl/rpga_event callback for + * pop-attenuation) */ + twl4030->sysclk = 26000; codec->private_data = twl4030; socdev->card->codec = codec; -- cgit v1.2.3 From 9da28c7b38170882b1c43d7d133ddce34e25f161 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 22 May 2009 10:13:15 +0300 Subject: ASoC: TWL4030: Add support for platform dependent configuration twl4030_setup_data structure can be passed from platform drivers to the codec via the snd_soc_device->codec_data pointer. Currently the setup data has support for the Headset pop-removal related configuration, which differs from board to board. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 22 +++++++++++++++++++--- sound/soc/codecs/twl4030.h | 5 +++++ 2 files changed, 24 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index f554672f67c1..584507f71efb 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1997,6 +1997,8 @@ static int twl4030_resume(struct platform_device *pdev) static int twl4030_init(struct snd_soc_device *socdev) { struct snd_soc_codec *codec = socdev->card->codec; + struct twl4030_setup_data *setup = socdev->codec_data; + struct twl4030_priv *twl4030 = codec->private_data; int ret = 0; printk(KERN_INFO "TWL4030 Audio Codec init \n"); @@ -2014,6 +2016,23 @@ static int twl4030_init(struct snd_soc_device *socdev) if (codec->reg_cache == NULL) return -ENOMEM; + /* Configuration for headset ramp delay from setup data */ + if (setup) { + unsigned char hs_pop; + + if (setup->sysclk) + twl4030->sysclk = setup->sysclk; + else + twl4030->sysclk = 26000; + + hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); + hs_pop &= ~TWL4030_RAMP_DELAY; + hs_pop |= (setup->ramp_delay_value << 2); + twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, hs_pop); + } else { + twl4030->sysclk = 26000; + } + /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { @@ -2063,9 +2082,6 @@ static int twl4030_probe(struct platform_device *pdev) kfree(codec); return -ENOMEM; } - /* Set default sysclk (used by the headsetl/rpga_event callback for - * pop-attenuation) */ - twl4030->sysclk = 26000; codec->private_data = twl4030; socdev->card->codec = codec; diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 9668bdf430fb..48326e2bd9de 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -266,4 +266,9 @@ extern struct snd_soc_dai twl4030_dai[2]; extern struct snd_soc_codec_device soc_codec_dev_twl4030; +struct twl4030_setup_data { + unsigned int ramp_delay_value; + unsigned int sysclk; +}; + #endif /* End of __TWL4030_AUDIO_H__ */ -- cgit v1.2.3 From b4852b793a1dd74ccde5572d8a8f73e948a5b1a1 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 22 May 2009 15:12:15 +0300 Subject: ASoC: TWL4030: Differentiate the playback streams Give unique stream names for the two playback streams so DAPM can figure out which codec_dai is in use. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 584507f71efb..9197fdd0a29d 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1092,13 +1092,13 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("VIBRA"), /* DACs */ - SND_SOC_DAPM_DAC("DAC Right1", "Right Front Playback", + SND_SOC_DAPM_DAC("DAC Right1", "Right Front HiFi Playback", SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("DAC Left1", "Left Front Playback", + SND_SOC_DAPM_DAC("DAC Left1", "Left Front HiFi Playback", SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("DAC Right2", "Right Rear Playback", + SND_SOC_DAPM_DAC("DAC Right2", "Right Rear HiFi Playback", SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback", + SND_SOC_DAPM_DAC("DAC Left2", "Left Rear HiFi Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC Voice", "Voice Playback", SND_SOC_NOPM, 0, 0), @@ -1937,7 +1937,7 @@ struct snd_soc_dai twl4030_dai[] = { { .name = "twl4030", .playback = { - .stream_name = "Playback", + .stream_name = "HiFi Playback", .channels_min = 2, .channels_max = 4, .rates = TWL4030_RATES | SNDRV_PCM_RATE_96000, @@ -1953,7 +1953,7 @@ struct snd_soc_dai twl4030_dai[] = { { .name = "twl4030 Voice", .playback = { - .stream_name = "Playback", + .stream_name = "Voice Playback", .channels_min = 1, .channels_max = 1, .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, -- cgit v1.2.3 From 86ed3669f068b514ab85ffd548456a342b3fb8d3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 May 2009 15:01:19 +0100 Subject: ASoC: WM9081 mono DAC with integrated 2.6W class AB/D amplifier driver The WM9081 is designed to provide high power output at low distortion levels in space-constrained portable applications. Signed-off-by: Mark Brown --- include/sound/wm9081.h | 25 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm9081.c | 1532 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm9081.h | 787 +++++++++++++++++++++++ 5 files changed, 2350 insertions(+) create mode 100644 include/sound/wm9081.h create mode 100644 sound/soc/codecs/wm9081.c create mode 100644 sound/soc/codecs/wm9081.h (limited to 'sound/soc/codecs') diff --git a/include/sound/wm9081.h b/include/sound/wm9081.h new file mode 100644 index 000000000000..e173ddbf6bd4 --- /dev/null +++ b/include/sound/wm9081.h @@ -0,0 +1,25 @@ +/* + * linux/sound/wm9081.h -- Platform data for WM9081 + * + * Copyright 2009 Wolfson Microelectronics. PLC. + * + * 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 __LINUX_SND_WM_9081_H +#define __LINUX_SND_WM_9081_H + +struct wm9081_retune_mobile_setting { + const char *name; + unsigned int rate; + u16 config[20]; +}; + +struct wm9081_retune_mobile_config { + struct wm9081_retune_mobile_setting *configs; + int num_configs; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1c19ad54a9f9..7f78b65fc4e3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -40,6 +40,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8971 if I2C select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8990 if I2C + select SND_SOC_WM9081 if I2C select SND_SOC_WM9705 if SND_SOC_AC97_BUS select SND_SOC_WM9712 if SND_SOC_AC97_BUS select SND_SOC_WM9713 if SND_SOC_AC97_BUS @@ -156,6 +157,9 @@ config SND_SOC_WM8988 config SND_SOC_WM8990 tristate +config SND_SOC_WM9081 + tristate + config SND_SOC_WM9705 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3d31b6bea834..70c55fa2c436 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -28,6 +28,7 @@ snd-soc-wm8960-objs := wm8960.o snd-soc-wm8971-objs := wm8971.o snd-soc-wm8988-objs := wm8988.o snd-soc-wm8990-objs := wm8990.o +snd-soc-wm9081-objs := wm9081.o snd-soc-wm9705-objs := wm9705.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o @@ -62,6 +63,7 @@ obj-$(CONFIG_SND_SOC_WM8940) += snd-soc-wm8940.o obj-$(CONFIG_SND_SOC_WM8960) += snd-soc-wm8960.o obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o +obj-$(CONFIG_SND_SOC_WM9081) += snd-soc-wm9081.o obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c new file mode 100644 index 000000000000..83e3148c258b --- /dev/null +++ b/sound/soc/codecs/wm9081.c @@ -0,0 +1,1532 @@ +/* + * wm9081.c -- WM9081 ALSA SoC Audio driver + * + * Author: Mark Brown + * + * Copyright 2009 Wolfson Microelectronics plc + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include "wm9081.h" + +static u16 wm9081_reg_defaults[] = { + 0x0000, /* R0 - Software Reset */ + 0x0000, /* R1 */ + 0x00B9, /* R2 - Analogue Lineout */ + 0x00B9, /* R3 - Analogue Speaker PGA */ + 0x0001, /* R4 - VMID Control */ + 0x0068, /* R5 - Bias Control 1 */ + 0x0000, /* R6 */ + 0x0000, /* R7 - Analogue Mixer */ + 0x0000, /* R8 - Anti Pop Control */ + 0x01DB, /* R9 - Analogue Speaker 1 */ + 0x0018, /* R10 - Analogue Speaker 2 */ + 0x0180, /* R11 - Power Management */ + 0x0000, /* R12 - Clock Control 1 */ + 0x0038, /* R13 - Clock Control 2 */ + 0x4000, /* R14 - Clock Control 3 */ + 0x0000, /* R15 */ + 0x0000, /* R16 - FLL Control 1 */ + 0x0200, /* R17 - FLL Control 2 */ + 0x0000, /* R18 - FLL Control 3 */ + 0x0204, /* R19 - FLL Control 4 */ + 0x0000, /* R20 - FLL Control 5 */ + 0x0000, /* R21 */ + 0x0000, /* R22 - Audio Interface 1 */ + 0x0002, /* R23 - Audio Interface 2 */ + 0x0008, /* R24 - Audio Interface 3 */ + 0x0022, /* R25 - Audio Interface 4 */ + 0x0000, /* R26 - Interrupt Status */ + 0x0006, /* R27 - Interrupt Status Mask */ + 0x0000, /* R28 - Interrupt Polarity */ + 0x0000, /* R29 - Interrupt Control */ + 0x00C0, /* R30 - DAC Digital 1 */ + 0x0008, /* R31 - DAC Digital 2 */ + 0x09AF, /* R32 - DRC 1 */ + 0x4201, /* R33 - DRC 2 */ + 0x0000, /* R34 - DRC 3 */ + 0x0000, /* R35 - DRC 4 */ + 0x0000, /* R36 */ + 0x0000, /* R37 */ + 0x0000, /* R38 - Write Sequencer 1 */ + 0x0000, /* R39 - Write Sequencer 2 */ + 0x0002, /* R40 - MW Slave 1 */ + 0x0000, /* R41 */ + 0x0000, /* R42 - EQ 1 */ + 0x0000, /* R43 - EQ 2 */ + 0x0FCA, /* R44 - EQ 3 */ + 0x0400, /* R45 - EQ 4 */ + 0x00B8, /* R46 - EQ 5 */ + 0x1EB5, /* R47 - EQ 6 */ + 0xF145, /* R48 - EQ 7 */ + 0x0B75, /* R49 - EQ 8 */ + 0x01C5, /* R50 - EQ 9 */ + 0x169E, /* R51 - EQ 10 */ + 0xF829, /* R52 - EQ 11 */ + 0x07AD, /* R53 - EQ 12 */ + 0x1103, /* R54 - EQ 13 */ + 0x1C58, /* R55 - EQ 14 */ + 0xF373, /* R56 - EQ 15 */ + 0x0A54, /* R57 - EQ 16 */ + 0x0558, /* R58 - EQ 17 */ + 0x0564, /* R59 - EQ 18 */ + 0x0559, /* R60 - EQ 19 */ + 0x4000, /* R61 - EQ 20 */ +}; + +static struct { + int ratio; + int clk_sys_rate; +} clk_sys_rates[] = { + { 64, 0 }, + { 128, 1 }, + { 192, 2 }, + { 256, 3 }, + { 384, 4 }, + { 512, 5 }, + { 768, 6 }, + { 1024, 7 }, + { 1408, 8 }, + { 1536, 9 }, +}; + +static struct { + int rate; + int sample_rate; +} sample_rates[] = { + { 8000, 0 }, + { 11025, 1 }, + { 12000, 2 }, + { 16000, 3 }, + { 22050, 4 }, + { 24000, 5 }, + { 32000, 6 }, + { 44100, 7 }, + { 48000, 8 }, + { 88200, 9 }, + { 96000, 10 }, +}; + +static struct { + int div; /* *10 due to .5s */ + int bclk_div; +} bclk_divs[] = { + { 10, 0 }, + { 15, 1 }, + { 20, 2 }, + { 30, 3 }, + { 40, 4 }, + { 50, 5 }, + { 55, 6 }, + { 60, 7 }, + { 80, 8 }, + { 100, 9 }, + { 110, 10 }, + { 120, 11 }, + { 160, 12 }, + { 200, 13 }, + { 220, 14 }, + { 240, 15 }, + { 250, 16 }, + { 300, 17 }, + { 320, 18 }, + { 440, 19 }, + { 480, 20 }, +}; + +struct wm9081_priv { + struct snd_soc_codec codec; + u16 reg_cache[WM9081_MAX_REGISTER + 1]; + int sysclk_source; + int mclk_rate; + int sysclk_rate; + int fs; + int bclk; + int master; + int fll_fref; + int fll_fout; + struct wm9081_retune_mobile_config *retune; +}; + +static int wm9081_reg_is_volatile(int reg) +{ + switch (reg) { + default: + return 0; + } +} + +static unsigned int wm9081_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + BUG_ON(reg > WM9081_MAX_REGISTER); + return cache[reg]; +} + +static unsigned int wm9081_read_hw(struct snd_soc_codec *codec, u8 reg) +{ + struct i2c_msg xfer[2]; + u16 data; + int ret; + struct i2c_client *client = codec->control_data; + + BUG_ON(reg > WM9081_MAX_REGISTER); + + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 1; + xfer[0].buf = ® + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 2; + xfer[1].buf = (u8 *)&data; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret != 2) { + dev_err(&client->dev, "i2c_transfer() returned %d\n", ret); + return 0; + } + + return (data >> 8) | ((data & 0xff) << 8); +} + +static unsigned int wm9081_read(struct snd_soc_codec *codec, unsigned int reg) +{ + if (wm9081_reg_is_volatile(reg)) + return wm9081_read_hw(codec, reg); + else + return wm9081_read_reg_cache(codec, reg); +} + +static int wm9081_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u16 *cache = codec->reg_cache; + u8 data[3]; + + BUG_ON(reg > WM9081_MAX_REGISTER); + + if (!wm9081_reg_is_volatile(reg)) + cache[reg] = value; + + data[0] = reg; + data[1] = value >> 8; + data[2] = value & 0x00ff; + + if (codec->hw_write(codec->control_data, data, 3) == 3) + return 0; + else + return -EIO; +} + +static int wm9081_reset(struct snd_soc_codec *codec) +{ + return wm9081_write(codec, WM9081_SOFTWARE_RESET, 0); +} + +static const DECLARE_TLV_DB_SCALE(drc_in_tlv, -4500, 75, 0); +static const DECLARE_TLV_DB_SCALE(drc_out_tlv, -2250, 75, 0); +static const DECLARE_TLV_DB_SCALE(drc_min_tlv, -1800, 600, 0); +static unsigned int drc_max_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 0, TLV_DB_SCALE_ITEM(1200, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(1800, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), + 3, 3, TLV_DB_SCALE_ITEM(3600, 0, 0), +}; +static const DECLARE_TLV_DB_SCALE(drc_qr_tlv, 1200, 600, 0); +static const DECLARE_TLV_DB_SCALE(drc_startup_tlv, -300, 50, 0); + +static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); + +static const DECLARE_TLV_DB_SCALE(in_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(dac_tlv, -7200, 75, 1); +static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0); + +static const char *drc_high_text[] = { + "1", + "1/2", + "1/4", + "1/8", + "1/16", + "0", +}; + +static const struct soc_enum drc_high = + SOC_ENUM_SINGLE(WM9081_DRC_3, 3, 6, drc_high_text); + +static const char *drc_low_text[] = { + "1", + "1/2", + "1/4", + "1/8", + "0", +}; + +static const struct soc_enum drc_low = + SOC_ENUM_SINGLE(WM9081_DRC_3, 0, 5, drc_low_text); + +static const char *drc_atk_text[] = { + "181us", + "181us", + "363us", + "726us", + "1.45ms", + "2.9ms", + "5.8ms", + "11.6ms", + "23.2ms", + "46.4ms", + "92.8ms", + "185.6ms", +}; + +static const struct soc_enum drc_atk = + SOC_ENUM_SINGLE(WM9081_DRC_2, 12, 12, drc_atk_text); + +static const char *drc_dcy_text[] = { + "186ms", + "372ms", + "743ms", + "1.49s", + "2.97s", + "5.94s", + "11.89s", + "23.78s", + "47.56s", +}; + +static const struct soc_enum drc_dcy = + SOC_ENUM_SINGLE(WM9081_DRC_2, 8, 9, drc_dcy_text); + +static const char *drc_qr_dcy_text[] = { + "0.725ms", + "1.45ms", + "5.8ms", +}; + +static const struct soc_enum drc_qr_dcy = + SOC_ENUM_SINGLE(WM9081_DRC_2, 4, 3, drc_qr_dcy_text); + +static const char *dac_deemph_text[] = { + "None", + "32kHz", + "44.1kHz", + "48kHz", +}; + +static const struct soc_enum dac_deemph = + SOC_ENUM_SINGLE(WM9081_DAC_DIGITAL_2, 1, 4, dac_deemph_text); + +static const char *speaker_mode_text[] = { + "Class D", + "Class AB", +}; + +static const struct soc_enum speaker_mode = + SOC_ENUM_SINGLE(WM9081_ANALOGUE_SPEAKER_2, 6, 2, speaker_mode_text); + +static int speaker_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg; + + reg = wm9081_read(codec, WM9081_ANALOGUE_SPEAKER_2); + if (reg & WM9081_SPK_MODE) + ucontrol->value.integer.value[0] = 1; + else + ucontrol->value.integer.value[0] = 0; + + return 0; +} + +/* + * Stop any attempts to change speaker mode while the speaker is enabled. + * + * We also have some special anti-pop controls dependant on speaker + * mode which must be changed along with the mode. + */ +static int speaker_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg_pwr = wm9081_read(codec, WM9081_POWER_MANAGEMENT); + unsigned int reg2 = wm9081_read(codec, WM9081_ANALOGUE_SPEAKER_2); + + /* Are we changing anything? */ + if (ucontrol->value.integer.value[0] == + ((reg2 & WM9081_SPK_MODE) != 0)) + return 0; + + /* Don't try to change modes while enabled */ + if (reg_pwr & WM9081_SPK_ENA) + return -EINVAL; + + if (ucontrol->value.integer.value[0]) { + /* Class AB */ + reg2 &= ~(WM9081_SPK_INV_MUTE | WM9081_OUT_SPK_CTRL); + reg2 |= WM9081_SPK_MODE; + } else { + /* Class D */ + reg2 |= WM9081_SPK_INV_MUTE | WM9081_OUT_SPK_CTRL; + reg2 &= ~WM9081_SPK_MODE; + } + + wm9081_write(codec, WM9081_ANALOGUE_SPEAKER_2, reg2); + + return 0; +} + +static const struct snd_kcontrol_new wm9081_snd_controls[] = { +SOC_SINGLE_TLV("IN1 Volume", WM9081_ANALOGUE_MIXER, 1, 1, 1, in_tlv), +SOC_SINGLE_TLV("IN2 Volume", WM9081_ANALOGUE_MIXER, 3, 1, 1, in_tlv), + +SOC_SINGLE_TLV("Playback Volume", WM9081_DAC_DIGITAL_1, 1, 96, 0, dac_tlv), + +SOC_SINGLE("LINEOUT Switch", WM9081_ANALOGUE_LINEOUT, 7, 1, 1), +SOC_SINGLE("LINEOUT ZC Switch", WM9081_ANALOGUE_LINEOUT, 6, 1, 0), +SOC_SINGLE_TLV("LINEOUT Volume", WM9081_ANALOGUE_LINEOUT, 0, 63, 0, out_tlv), + +SOC_SINGLE("DRC Switch", WM9081_DRC_1, 15, 1, 0), +SOC_ENUM("DRC High Slope", drc_high), +SOC_ENUM("DRC Low Slope", drc_low), +SOC_SINGLE_TLV("DRC Input Volume", WM9081_DRC_4, 5, 60, 1, drc_in_tlv), +SOC_SINGLE_TLV("DRC Output Volume", WM9081_DRC_4, 0, 30, 1, drc_out_tlv), +SOC_SINGLE_TLV("DRC Minimum Volume", WM9081_DRC_2, 2, 3, 1, drc_min_tlv), +SOC_SINGLE_TLV("DRC Maximum Volume", WM9081_DRC_2, 0, 3, 0, drc_max_tlv), +SOC_ENUM("DRC Attack", drc_atk), +SOC_ENUM("DRC Decay", drc_dcy), +SOC_SINGLE("DRC Quick Release Switch", WM9081_DRC_1, 2, 1, 0), +SOC_SINGLE_TLV("DRC Quick Release Volume", WM9081_DRC_2, 6, 3, 0, drc_qr_tlv), +SOC_ENUM("DRC Quick Release Decay", drc_qr_dcy), +SOC_SINGLE_TLV("DRC Startup Volume", WM9081_DRC_1, 6, 18, 0, drc_startup_tlv), + +SOC_SINGLE("EQ Switch", WM9081_EQ_1, 0, 1, 0), + +SOC_SINGLE("Speaker DC Volume", WM9081_ANALOGUE_SPEAKER_1, 3, 5, 0), +SOC_SINGLE("Speaker AC Volume", WM9081_ANALOGUE_SPEAKER_1, 0, 5, 0), +SOC_SINGLE("Speaker Switch", WM9081_ANALOGUE_SPEAKER_PGA, 7, 1, 1), +SOC_SINGLE("Speaker ZC Switch", WM9081_ANALOGUE_SPEAKER_PGA, 6, 1, 0), +SOC_SINGLE_TLV("Speaker Volume", WM9081_ANALOGUE_SPEAKER_PGA, 0, 63, 0, + out_tlv), +SOC_ENUM("DAC Deemphasis", dac_deemph), +SOC_ENUM_EXT("Speaker Mode", speaker_mode, speaker_mode_get, speaker_mode_put), +}; + +static const struct snd_kcontrol_new wm9081_eq_controls[] = { +SOC_SINGLE_TLV("EQ1 Volume", WM9081_EQ_1, 11, 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 Volume", WM9081_EQ_1, 6, 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 Volume", WM9081_EQ_1, 1, 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 Volume", WM9081_EQ_2, 11, 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ5 Volume", WM9081_EQ_2, 6, 24, 0, eq_tlv), +}; + +static const struct snd_kcontrol_new mixer[] = { +SOC_DAPM_SINGLE("IN1 Switch", WM9081_ANALOGUE_MIXER, 0, 1, 0), +SOC_DAPM_SINGLE("IN2 Switch", WM9081_ANALOGUE_MIXER, 2, 1, 0), +SOC_DAPM_SINGLE("Playback Switch", WM9081_ANALOGUE_MIXER, 4, 1, 0), +}; + +static int speaker_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int reg = wm9081_read(codec, WM9081_POWER_MANAGEMENT); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + reg |= WM9081_SPK_ENA; + break; + + case SND_SOC_DAPM_PRE_PMD: + reg &= ~WM9081_SPK_ENA; + break; + } + + wm9081_write(codec, WM9081_POWER_MANAGEMENT, reg); + + return 0; +} + +struct _fll_div { + u16 fll_fratio; + u16 fll_outdiv; + u16 fll_clk_ref_div; + u16 n; + u16 k; +}; + +/* The size in bits of the FLL divide multiplied by 10 + * to allow rounding later */ +#define FIXED_FLL_SIZE ((1 << 16) * 10) + +static struct { + unsigned int min; + unsigned int max; + u16 fll_fratio; + int ratio; +} fll_fratios[] = { + { 0, 64000, 4, 16 }, + { 64000, 128000, 3, 8 }, + { 128000, 256000, 2, 4 }, + { 256000, 1000000, 1, 2 }, + { 1000000, 13500000, 0, 1 }, +}; + +static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, + unsigned int Fout) +{ + u64 Kpart; + unsigned int K, Ndiv, Nmod, target; + unsigned int div; + int i; + + /* Fref must be <=13.5MHz */ + div = 1; + while ((Fref / div) > 13500000) { + div *= 2; + + if (div > 8) { + pr_err("Can't scale %dMHz input down to <=13.5MHz\n", + Fref); + return -EINVAL; + } + } + fll_div->fll_clk_ref_div = div / 2; + + pr_debug("Fref=%u Fout=%u\n", Fref, Fout); + + /* Apply the division for our remaining calculations */ + Fref /= div; + + /* Fvco should be 90-100MHz; don't check the upper bound */ + div = 0; + target = Fout * 2; + while (target < 90000000) { + div++; + target *= 2; + if (div > 7) { + pr_err("Unable to find FLL_OUTDIV for Fout=%uHz\n", + Fout); + return -EINVAL; + } + } + fll_div->fll_outdiv = div; + + pr_debug("Fvco=%dHz\n", target); + + /* Find an appropraite FLL_FRATIO and factor it out of the target */ + for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { + if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { + fll_div->fll_fratio = fll_fratios[i].fll_fratio; + target /= fll_fratios[i].ratio; + break; + } + } + if (i == ARRAY_SIZE(fll_fratios)) { + pr_err("Unable to find FLL_FRATIO for Fref=%uHz\n", Fref); + return -EINVAL; + } + + /* Now, calculate N.K */ + Ndiv = target / Fref; + + fll_div->n = Ndiv; + Nmod = target % Fref; + pr_debug("Nmod=%d\n", Nmod); + + /* Calculate fractional part - scale up so we can round. */ + Kpart = FIXED_FLL_SIZE * (long long)Nmod; + + do_div(Kpart, Fref); + + K = Kpart & 0xFFFFFFFF; + + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + fll_div->k = K / 10; + + pr_debug("N=%x K=%x FLL_FRATIO=%x FLL_OUTDIV=%x FLL_CLK_REF_DIV=%x\n", + fll_div->n, fll_div->k, + fll_div->fll_fratio, fll_div->fll_outdiv, + fll_div->fll_clk_ref_div); + + return 0; +} + +static int wm9081_set_fll(struct snd_soc_codec *codec, int fll_id, + unsigned int Fref, unsigned int Fout) +{ + struct wm9081_priv *wm9081 = codec->private_data; + u16 reg1, reg4, reg5; + struct _fll_div fll_div; + int ret; + int clk_sys_reg; + + /* Any change? */ + if (Fref == wm9081->fll_fref && Fout == wm9081->fll_fout) + return 0; + + /* Disable the FLL */ + if (Fout == 0) { + dev_dbg(codec->dev, "FLL disabled\n"); + wm9081->fll_fref = 0; + wm9081->fll_fout = 0; + + return 0; + } + + ret = fll_factors(&fll_div, Fref, Fout); + if (ret != 0) + return ret; + + reg5 = wm9081_read(codec, WM9081_FLL_CONTROL_5); + reg5 &= ~WM9081_FLL_CLK_SRC_MASK; + + switch (fll_id) { + case WM9081_SYSCLK_FLL_MCLK: + reg5 |= 0x1; + break; + + default: + dev_err(codec->dev, "Unknown FLL ID %d\n", fll_id); + return -EINVAL; + } + + /* Disable CLK_SYS while we reconfigure */ + clk_sys_reg = wm9081_read(codec, WM9081_CLOCK_CONTROL_3); + if (clk_sys_reg & WM9081_CLK_SYS_ENA) + wm9081_write(codec, WM9081_CLOCK_CONTROL_3, + clk_sys_reg & ~WM9081_CLK_SYS_ENA); + + /* Any FLL configuration change requires that the FLL be + * disabled first. */ + reg1 = wm9081_read(codec, WM9081_FLL_CONTROL_1); + reg1 &= ~WM9081_FLL_ENA; + wm9081_write(codec, WM9081_FLL_CONTROL_1, reg1); + + /* Apply the configuration */ + if (fll_div.k) + reg1 |= WM9081_FLL_FRAC_MASK; + else + reg1 &= ~WM9081_FLL_FRAC_MASK; + wm9081_write(codec, WM9081_FLL_CONTROL_1, reg1); + + wm9081_write(codec, WM9081_FLL_CONTROL_2, + (fll_div.fll_outdiv << WM9081_FLL_OUTDIV_SHIFT) | + (fll_div.fll_fratio << WM9081_FLL_FRATIO_SHIFT)); + wm9081_write(codec, WM9081_FLL_CONTROL_3, fll_div.k); + + reg4 = wm9081_read(codec, WM9081_FLL_CONTROL_4); + reg4 &= ~WM9081_FLL_N_MASK; + reg4 |= fll_div.n << WM9081_FLL_N_SHIFT; + wm9081_write(codec, WM9081_FLL_CONTROL_4, reg4); + + reg5 &= ~WM9081_FLL_CLK_REF_DIV_MASK; + reg5 |= fll_div.fll_clk_ref_div << WM9081_FLL_CLK_REF_DIV_SHIFT; + wm9081_write(codec, WM9081_FLL_CONTROL_5, reg5); + + /* Enable the FLL */ + wm9081_write(codec, WM9081_FLL_CONTROL_1, reg1 | WM9081_FLL_ENA); + + /* Then bring CLK_SYS up again if it was disabled */ + if (clk_sys_reg & WM9081_CLK_SYS_ENA) + wm9081_write(codec, WM9081_CLOCK_CONTROL_3, clk_sys_reg); + + dev_dbg(codec->dev, "FLL enabled at %dHz->%dHz\n", Fref, Fout); + + wm9081->fll_fref = Fref; + wm9081->fll_fout = Fout; + + return 0; +} + +static int configure_clock(struct snd_soc_codec *codec) +{ + struct wm9081_priv *wm9081 = codec->private_data; + int new_sysclk, i, target; + unsigned int reg; + int ret = 0; + int mclkdiv = 0; + int fll = 0; + + switch (wm9081->sysclk_source) { + case WM9081_SYSCLK_MCLK: + if (wm9081->mclk_rate > 12225000) { + mclkdiv = 1; + wm9081->sysclk_rate = wm9081->mclk_rate / 2; + } else { + wm9081->sysclk_rate = wm9081->mclk_rate; + } + wm9081_set_fll(codec, WM9081_SYSCLK_FLL_MCLK, 0, 0); + break; + + case WM9081_SYSCLK_FLL_MCLK: + /* If we have a sample rate calculate a CLK_SYS that + * gives us a suitable DAC configuration, plus BCLK. + * Ideally we would check to see if we can clock + * directly from MCLK and only use the FLL if this is + * not the case, though care must be taken with free + * running mode. + */ + if (wm9081->master && wm9081->bclk) { + /* Make sure we can generate CLK_SYS and BCLK + * and that we've got 3MHz for optimal + * performance. */ + for (i = 0; i < ARRAY_SIZE(clk_sys_rates); i++) { + target = wm9081->fs * clk_sys_rates[i].ratio; + if (target >= wm9081->bclk && + target > 3000000) + new_sysclk = target; + } + } else if (wm9081->fs) { + for (i = 0; i < ARRAY_SIZE(clk_sys_rates); i++) { + new_sysclk = clk_sys_rates[i].ratio + * wm9081->fs; + if (new_sysclk > 3000000) + break; + } + } else { + new_sysclk = 12288000; + } + + ret = wm9081_set_fll(codec, WM9081_SYSCLK_FLL_MCLK, + wm9081->mclk_rate, new_sysclk); + if (ret == 0) { + wm9081->sysclk_rate = new_sysclk; + + /* Switch SYSCLK over to FLL */ + fll = 1; + } else { + wm9081->sysclk_rate = wm9081->mclk_rate; + } + break; + + default: + return -EINVAL; + } + + reg = wm9081_read(codec, WM9081_CLOCK_CONTROL_1); + if (mclkdiv) + reg |= WM9081_MCLKDIV2; + else + reg &= ~WM9081_MCLKDIV2; + wm9081_write(codec, WM9081_CLOCK_CONTROL_1, reg); + + reg = wm9081_read(codec, WM9081_CLOCK_CONTROL_3); + if (fll) + reg |= WM9081_CLK_SRC_SEL; + else + reg &= ~WM9081_CLK_SRC_SEL; + wm9081_write(codec, WM9081_CLOCK_CONTROL_3, reg); + + dev_dbg(codec->dev, "CLK_SYS is %dHz\n", wm9081->sysclk_rate); + + return ret; +} + +static int clk_sys_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm9081_priv *wm9081 = codec->private_data; + + /* This should be done on init() for bypass paths */ + switch (wm9081->sysclk_source) { + case WM9081_SYSCLK_MCLK: + dev_dbg(codec->dev, "Using %dHz MCLK\n", wm9081->mclk_rate); + break; + case WM9081_SYSCLK_FLL_MCLK: + dev_dbg(codec->dev, "Using %dHz MCLK with FLL\n", + wm9081->mclk_rate); + break; + default: + dev_err(codec->dev, "System clock not configured\n"); + return -EINVAL; + } + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + configure_clock(codec); + break; + + case SND_SOC_DAPM_POST_PMD: + /* Disable the FLL if it's running */ + wm9081_set_fll(codec, 0, 0, 0); + break; + } + + return 0; +} + +static const struct snd_soc_dapm_widget wm9081_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("IN1"), +SND_SOC_DAPM_INPUT("IN2"), + +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM9081_POWER_MANAGEMENT, 0, 0), + +SND_SOC_DAPM_MIXER_NAMED_CTL("Mixer", SND_SOC_NOPM, 0, 0, + mixer, ARRAY_SIZE(mixer)), + +SND_SOC_DAPM_PGA("LINEOUT PGA", WM9081_POWER_MANAGEMENT, 4, 0, NULL, 0), + +SND_SOC_DAPM_PGA_E("Speaker PGA", WM9081_POWER_MANAGEMENT, 2, 0, NULL, 0, + speaker_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + +SND_SOC_DAPM_OUTPUT("LINEOUT"), +SND_SOC_DAPM_OUTPUT("SPKN"), +SND_SOC_DAPM_OUTPUT("SPKP"), + +SND_SOC_DAPM_SUPPLY("CLK_SYS", WM9081_CLOCK_CONTROL_3, 0, 0, clk_sys_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("CLK_DSP", WM9081_CLOCK_CONTROL_3, 1, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("TOCLK", WM9081_CLOCK_CONTROL_3, 2, 0, NULL, 0), +}; + + +static const struct snd_soc_dapm_route audio_paths[] = { + { "DAC", NULL, "CLK_SYS" }, + { "DAC", NULL, "CLK_DSP" }, + + { "Mixer", "IN1 Switch", "IN1" }, + { "Mixer", "IN2 Switch", "IN2" }, + { "Mixer", "Playback Switch", "DAC" }, + + { "LINEOUT PGA", NULL, "Mixer" }, + { "LINEOUT PGA", NULL, "TOCLK" }, + { "LINEOUT PGA", NULL, "CLK_SYS" }, + + { "LINEOUT", NULL, "LINEOUT PGA" }, + + { "Speaker PGA", NULL, "Mixer" }, + { "Speaker PGA", NULL, "TOCLK" }, + { "Speaker PGA", NULL, "CLK_SYS" }, + + { "SPKN", NULL, "Speaker PGA" }, + { "SPKP", NULL, "Speaker PGA" }, +}; + +static int wm9081_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* VMID=2*40k */ + reg = wm9081_read(codec, WM9081_VMID_CONTROL); + reg &= ~WM9081_VMID_SEL_MASK; + reg |= 0x2; + wm9081_write(codec, WM9081_VMID_CONTROL, reg); + + /* Normal bias current */ + reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1); + reg &= ~WM9081_STBY_BIAS_ENA; + wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg); + break; + + case SND_SOC_BIAS_STANDBY: + /* Initial cold start */ + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Disable LINEOUT discharge */ + reg = wm9081_read(codec, WM9081_ANTI_POP_CONTROL); + reg &= ~WM9081_LINEOUT_DISCH; + wm9081_write(codec, WM9081_ANTI_POP_CONTROL, reg); + + /* Select startup bias source */ + reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1); + reg |= WM9081_BIAS_SRC | WM9081_BIAS_ENA; + wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg); + + /* VMID 2*4k; Soft VMID ramp enable */ + reg = wm9081_read(codec, WM9081_VMID_CONTROL); + reg |= WM9081_VMID_RAMP | 0x6; + wm9081_write(codec, WM9081_VMID_CONTROL, reg); + + mdelay(100); + + /* Normal bias enable & soft start off */ + reg |= WM9081_BIAS_ENA; + reg &= ~WM9081_VMID_RAMP; + wm9081_write(codec, WM9081_VMID_CONTROL, reg); + + /* Standard bias source */ + reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1); + reg &= ~WM9081_BIAS_SRC; + wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg); + } + + /* VMID 2*240k */ + reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1); + reg &= ~WM9081_VMID_SEL_MASK; + reg |= 0x40; + wm9081_write(codec, WM9081_VMID_CONTROL, reg); + + /* Standby bias current on */ + reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1); + reg |= WM9081_STBY_BIAS_ENA; + wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg); + break; + + case SND_SOC_BIAS_OFF: + /* Startup bias source */ + reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1); + reg |= WM9081_BIAS_SRC; + wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg); + + /* Disable VMID and biases with soft ramping */ + reg = wm9081_read(codec, WM9081_VMID_CONTROL); + reg &= ~(WM9081_VMID_SEL_MASK | WM9081_BIAS_ENA); + reg |= WM9081_VMID_RAMP; + wm9081_write(codec, WM9081_VMID_CONTROL, reg); + + /* Actively discharge LINEOUT */ + reg = wm9081_read(codec, WM9081_ANTI_POP_CONTROL); + reg |= WM9081_LINEOUT_DISCH; + wm9081_write(codec, WM9081_ANTI_POP_CONTROL, reg); + break; + } + + codec->bias_level = level; + + return 0; +} + +static int wm9081_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm9081_priv *wm9081 = codec->private_data; + unsigned int aif2 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_2); + + aif2 &= ~(WM9081_AIF_BCLK_INV | WM9081_AIF_LRCLK_INV | + WM9081_BCLK_DIR | WM9081_LRCLK_DIR | WM9081_AIF_FMT_MASK); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + wm9081->master = 0; + break; + case SND_SOC_DAIFMT_CBS_CFM: + aif2 |= WM9081_LRCLK_DIR; + wm9081->master = 1; + break; + case SND_SOC_DAIFMT_CBM_CFS: + aif2 |= WM9081_BCLK_DIR; + wm9081->master = 1; + break; + case SND_SOC_DAIFMT_CBM_CFM: + aif2 |= WM9081_LRCLK_DIR | WM9081_BCLK_DIR; + wm9081->master = 1; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_B: + aif2 |= WM9081_AIF_LRCLK_INV; + case SND_SOC_DAIFMT_DSP_A: + aif2 |= 0x3; + break; + case SND_SOC_DAIFMT_I2S: + aif2 |= 0x2; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + aif2 |= 0x1; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + /* frame inversion not valid for DSP modes */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + aif2 |= WM9081_AIF_BCLK_INV; + break; + default: + return -EINVAL; + } + break; + + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + aif2 |= WM9081_AIF_BCLK_INV | WM9081_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + aif2 |= WM9081_AIF_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + aif2 |= WM9081_AIF_LRCLK_INV; + break; + default: + return -EINVAL; + } + break; + default: + return -EINVAL; + } + + wm9081_write(codec, WM9081_AUDIO_INTERFACE_2, aif2); + + return 0; +} + +static int wm9081_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm9081_priv *wm9081 = codec->private_data; + int ret, i, best, best_val, cur_val; + unsigned int clk_ctrl2, aif1, aif2, aif3, aif4; + + clk_ctrl2 = wm9081_read(codec, WM9081_CLOCK_CONTROL_2); + clk_ctrl2 &= ~(WM9081_CLK_SYS_RATE_MASK | WM9081_SAMPLE_RATE_MASK); + + aif1 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_1); + + aif2 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_2); + aif2 &= ~WM9081_AIF_WL_MASK; + + aif3 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_3); + aif3 &= ~WM9081_BCLK_DIV_MASK; + + aif4 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_4); + aif4 &= ~WM9081_LRCLK_RATE_MASK; + + /* What BCLK do we need? */ + wm9081->fs = params_rate(params); + wm9081->bclk = 2 * wm9081->fs; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + wm9081->bclk *= 16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + wm9081->bclk *= 20; + aif2 |= 0x4; + break; + case SNDRV_PCM_FORMAT_S24_LE: + wm9081->bclk *= 24; + aif2 |= 0x8; + break; + case SNDRV_PCM_FORMAT_S32_LE: + wm9081->bclk *= 32; + aif2 |= 0xc; + break; + default: + return -EINVAL; + } + + if (aif1 & WM9081_AIFDAC_TDM_MODE_MASK) { + int slots = ((aif1 & WM9081_AIFDAC_TDM_MODE_MASK) >> + WM9081_AIFDAC_TDM_MODE_SHIFT) + 1; + wm9081->bclk *= slots; + } + + dev_dbg(codec->dev, "Target BCLK is %dHz\n", wm9081->bclk); + + ret = configure_clock(codec); + if (ret != 0) + return ret; + + /* Select nearest CLK_SYS_RATE */ + best = 0; + best_val = abs((wm9081->sysclk_rate / clk_sys_rates[0].ratio) + - wm9081->fs); + for (i = 1; i < ARRAY_SIZE(clk_sys_rates); i++) { + cur_val = abs((wm9081->sysclk_rate / + clk_sys_rates[i].ratio) - wm9081->fs);; + if (cur_val < best_val) { + best = i; + best_val = cur_val; + } + } + dev_dbg(codec->dev, "Selected CLK_SYS_RATIO of %d\n", + clk_sys_rates[best].ratio); + clk_ctrl2 |= (clk_sys_rates[best].clk_sys_rate + << WM9081_CLK_SYS_RATE_SHIFT); + + /* SAMPLE_RATE */ + best = 0; + best_val = abs(wm9081->fs - sample_rates[0].rate); + for (i = 1; i < ARRAY_SIZE(sample_rates); i++) { + /* Closest match */ + cur_val = abs(wm9081->fs - sample_rates[i].rate); + if (cur_val < best_val) { + best = i; + best_val = cur_val; + } + } + dev_dbg(codec->dev, "Selected SAMPLE_RATE of %dHz\n", + sample_rates[best].rate); + clk_ctrl2 |= (sample_rates[i].sample_rate << WM9081_SAMPLE_RATE_SHIFT); + + /* BCLK_DIV */ + best = 0; + best_val = INT_MAX; + for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) { + cur_val = ((wm9081->sysclk_rate * 10) / bclk_divs[i].div) + - wm9081->bclk; + if (cur_val < 0) /* Table is sorted */ + break; + if (cur_val < best_val) { + best = i; + best_val = cur_val; + } + } + wm9081->bclk = (wm9081->sysclk_rate * 10) / bclk_divs[best].div; + dev_dbg(codec->dev, "Selected BCLK_DIV of %d for %dHz BCLK\n", + bclk_divs[best].div, wm9081->bclk); + aif3 |= bclk_divs[best].bclk_div; + + /* LRCLK is a simple fraction of BCLK */ + dev_dbg(codec->dev, "LRCLK_RATE is %d\n", wm9081->bclk / wm9081->fs); + aif4 |= wm9081->bclk / wm9081->fs; + + /* Apply a ReTune Mobile configuration if it's in use */ + if (wm9081->retune) { + struct wm9081_retune_mobile_config *retune = wm9081->retune; + struct wm9081_retune_mobile_setting *s; + int eq1; + + best = 0; + best_val = abs(retune->configs[0].rate - wm9081->fs); + for (i = 0; i < retune->num_configs; i++) { + cur_val = abs(retune->configs[i].rate - wm9081->fs); + if (cur_val < best_val) { + best_val = cur_val; + best = i; + } + } + s = &retune->configs[best]; + + dev_dbg(codec->dev, "ReTune Mobile %s tuned for %dHz\n", + s->name, s->rate); + + /* If the EQ is enabled then disable it while we write out */ + eq1 = wm9081_read(codec, WM9081_EQ_1) & WM9081_EQ_ENA; + if (eq1 & WM9081_EQ_ENA) + wm9081_write(codec, WM9081_EQ_1, 0); + + /* Write out the other values */ + for (i = 1; i < ARRAY_SIZE(s->config); i++) + wm9081_write(codec, WM9081_EQ_1 + i, s->config[i]); + + eq1 |= (s->config[0] & ~WM9081_EQ_ENA); + wm9081_write(codec, WM9081_EQ_1, eq1); + } + + wm9081_write(codec, WM9081_CLOCK_CONTROL_2, clk_ctrl2); + wm9081_write(codec, WM9081_AUDIO_INTERFACE_2, aif2); + wm9081_write(codec, WM9081_AUDIO_INTERFACE_3, aif3); + wm9081_write(codec, WM9081_AUDIO_INTERFACE_4, aif4); + + return 0; +} + +static int wm9081_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + unsigned int reg; + + reg = wm9081_read(codec, WM9081_DAC_DIGITAL_2); + + if (mute) + reg |= WM9081_DAC_MUTE; + else + reg &= ~WM9081_DAC_MUTE; + + wm9081_write(codec, WM9081_DAC_DIGITAL_2, reg); + + return 0; +} + +static int wm9081_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm9081_priv *wm9081 = codec->private_data; + + switch (clk_id) { + case WM9081_SYSCLK_MCLK: + case WM9081_SYSCLK_FLL_MCLK: + wm9081->sysclk_source = clk_id; + wm9081->mclk_rate = freq; + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int mask, int slots) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int aif1 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_1); + + aif1 &= ~(WM9081_AIFDAC_TDM_SLOT_MASK | WM9081_AIFDAC_TDM_MODE_MASK); + + if (slots < 1 || slots > 4) + return -EINVAL; + + aif1 |= (slots - 1) << WM9081_AIFDAC_TDM_MODE_SHIFT; + + switch (mask) { + case 1: + break; + case 2: + aif1 |= 0x10; + break; + case 4: + aif1 |= 0x20; + break; + case 8: + aif1 |= 0x30; + break; + default: + return -EINVAL; + } + + wm9081_write(codec, WM9081_AUDIO_INTERFACE_1, aif1); + + return 0; +} + +#define WM9081_RATES SNDRV_PCM_RATE_8000_96000 + +#define WM9081_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 wm9081_dai_ops = { + .hw_params = wm9081_hw_params, + .set_sysclk = wm9081_set_sysclk, + .set_fmt = wm9081_set_dai_fmt, + .digital_mute = wm9081_digital_mute, + .set_tdm_slot = wm9081_set_tdm_slot, +}; + +/* We report two channels because the CODEC processes a stereo signal, even + * though it is only capable of handling a mono output. + */ +struct snd_soc_dai wm9081_dai = { + .name = "WM9081", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM9081_RATES, + .formats = WM9081_FORMATS, + }, + .ops = &wm9081_dai_ops, +}; +EXPORT_SYMBOL_GPL(wm9081_dai); + + +static struct snd_soc_codec *wm9081_codec; + +static int wm9081_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct wm9081_priv *wm9081; + int ret = 0; + + if (wm9081_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm9081_codec; + codec = wm9081_codec; + wm9081 = codec->private_data; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, wm9081_snd_controls, + ARRAY_SIZE(wm9081_snd_controls)); + if (!wm9081->retune) { + dev_dbg(codec->dev, + "No ReTune Mobile data, using normal EQ\n"); + snd_soc_add_controls(codec, wm9081_eq_controls, + ARRAY_SIZE(wm9081_eq_controls)); + } + + snd_soc_dapm_new_controls(codec, wm9081_dapm_widgets, + ARRAY_SIZE(wm9081_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); + snd_soc_dapm_new_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +static int wm9081_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +#ifdef CONFIG_PM +static int wm9081_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm9081_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int wm9081_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + u16 *reg_cache = codec->reg_cache; + int i; + + for (i = 0; i < codec->reg_cache_size; i++) { + if (i == WM9081_SOFTWARE_RESET) + continue; + + wm9081_write(codec, i, reg_cache[i]); + } + + wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define wm9081_suspend NULL +#define wm9081_resume NULL +#endif + +struct snd_soc_codec_device soc_codec_dev_wm9081 = { + .probe = wm9081_probe, + .remove = wm9081_remove, + .suspend = wm9081_suspend, + .resume = wm9081_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm9081); + +static int wm9081_register(struct wm9081_priv *wm9081) +{ + struct snd_soc_codec *codec = &wm9081->codec; + int ret; + u16 reg; + + if (wm9081_codec) { + dev_err(codec->dev, "Another WM9081 is registered\n"); + ret = -EINVAL; + goto err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm9081; + codec->name = "WM9081"; + codec->owner = THIS_MODULE; + codec->read = wm9081_read; + codec->write = wm9081_write; + codec->dai = &wm9081_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(wm9081->reg_cache); + codec->reg_cache = &wm9081->reg_cache; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm9081_set_bias_level; + + memcpy(codec->reg_cache, wm9081_reg_defaults, + sizeof(wm9081_reg_defaults)); + + reg = wm9081_read_hw(codec, WM9081_SOFTWARE_RESET); + if (reg != 0x9081) { + dev_err(codec->dev, "Device is not a WM9081: ID=0x%x\n", reg); + ret = -EINVAL; + goto err; + } + + ret = wm9081_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + return ret; + } + + wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Enable zero cross by default */ + reg = wm9081_read(codec, WM9081_ANALOGUE_LINEOUT); + wm9081_write(codec, WM9081_ANALOGUE_LINEOUT, reg | WM9081_LINEOUTZC); + reg = wm9081_read(codec, WM9081_ANALOGUE_SPEAKER_PGA); + wm9081_write(codec, WM9081_ANALOGUE_SPEAKER_PGA, + reg | WM9081_SPKPGAZC); + + wm9081_dai.dev = codec->dev; + + wm9081_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + ret = snd_soc_register_dai(&wm9081_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + return ret; + } + + return 0; + +err: + kfree(wm9081); + return ret; +} + +static void wm9081_unregister(struct wm9081_priv *wm9081) +{ + wm9081_set_bias_level(&wm9081->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&wm9081_dai); + snd_soc_unregister_codec(&wm9081->codec); + kfree(wm9081); + wm9081_codec = NULL; +} + +static __devinit int wm9081_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm9081_priv *wm9081; + struct snd_soc_codec *codec; + + wm9081 = kzalloc(sizeof(struct wm9081_priv), GFP_KERNEL); + if (wm9081 == NULL) + return -ENOMEM; + + codec = &wm9081->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + wm9081->retune = i2c->dev.platform_data; + + i2c_set_clientdata(i2c, wm9081); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + + return wm9081_register(wm9081); +} + +static __devexit int wm9081_i2c_remove(struct i2c_client *client) +{ + struct wm9081_priv *wm9081 = i2c_get_clientdata(client); + wm9081_unregister(wm9081); + return 0; +} + +static const struct i2c_device_id wm9081_i2c_id[] = { + { "wm9081", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm9081_i2c_id); + +static struct i2c_driver wm9081_i2c_driver = { + .driver = { + .name = "wm9081", + .owner = THIS_MODULE, + }, + .probe = wm9081_i2c_probe, + .remove = __devexit_p(wm9081_i2c_remove), + .id_table = wm9081_i2c_id, +}; + +static int __init wm9081_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&wm9081_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM9081 I2C driver: %d\n", + ret); + } + + return ret; +} +module_init(wm9081_modinit); + +static void __exit wm9081_exit(void) +{ + i2c_del_driver(&wm9081_i2c_driver); +} +module_exit(wm9081_exit); + + +MODULE_DESCRIPTION("ASoC WM9081 driver"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm9081.h b/sound/soc/codecs/wm9081.h new file mode 100644 index 000000000000..42d3bc757021 --- /dev/null +++ b/sound/soc/codecs/wm9081.h @@ -0,0 +1,787 @@ +#ifndef WM9081_H +#define WM9081_H + +/* + * wm9081.c -- WM9081 ALSA SoC Audio driver + * + * Author: Mark Brown + * + * Copyright 2009 Wolfson Microelectronics plc + * + * 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 + +extern struct snd_soc_dai wm9081_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm9081; + +/* + * SYSCLK sources + */ +#define WM9081_SYSCLK_MCLK 1 /* Use MCLK without FLL */ +#define WM9081_SYSCLK_FLL_MCLK 2 /* Use MCLK, enabling FLL if required */ + +/* + * Register values. + */ +#define WM9081_SOFTWARE_RESET 0x00 +#define WM9081_ANALOGUE_LINEOUT 0x02 +#define WM9081_ANALOGUE_SPEAKER_PGA 0x03 +#define WM9081_VMID_CONTROL 0x04 +#define WM9081_BIAS_CONTROL_1 0x05 +#define WM9081_ANALOGUE_MIXER 0x07 +#define WM9081_ANTI_POP_CONTROL 0x08 +#define WM9081_ANALOGUE_SPEAKER_1 0x09 +#define WM9081_ANALOGUE_SPEAKER_2 0x0A +#define WM9081_POWER_MANAGEMENT 0x0B +#define WM9081_CLOCK_CONTROL_1 0x0C +#define WM9081_CLOCK_CONTROL_2 0x0D +#define WM9081_CLOCK_CONTROL_3 0x0E +#define WM9081_FLL_CONTROL_1 0x10 +#define WM9081_FLL_CONTROL_2 0x11 +#define WM9081_FLL_CONTROL_3 0x12 +#define WM9081_FLL_CONTROL_4 0x13 +#define WM9081_FLL_CONTROL_5 0x14 +#define WM9081_AUDIO_INTERFACE_1 0x16 +#define WM9081_AUDIO_INTERFACE_2 0x17 +#define WM9081_AUDIO_INTERFACE_3 0x18 +#define WM9081_AUDIO_INTERFACE_4 0x19 +#define WM9081_INTERRUPT_STATUS 0x1A +#define WM9081_INTERRUPT_STATUS_MASK 0x1B +#define WM9081_INTERRUPT_POLARITY 0x1C +#define WM9081_INTERRUPT_CONTROL 0x1D +#define WM9081_DAC_DIGITAL_1 0x1E +#define WM9081_DAC_DIGITAL_2 0x1F +#define WM9081_DRC_1 0x20 +#define WM9081_DRC_2 0x21 +#define WM9081_DRC_3 0x22 +#define WM9081_DRC_4 0x23 +#define WM9081_WRITE_SEQUENCER_1 0x26 +#define WM9081_WRITE_SEQUENCER_2 0x27 +#define WM9081_MW_SLAVE_1 0x28 +#define WM9081_EQ_1 0x2A +#define WM9081_EQ_2 0x2B +#define WM9081_EQ_3 0x2C +#define WM9081_EQ_4 0x2D +#define WM9081_EQ_5 0x2E +#define WM9081_EQ_6 0x2F +#define WM9081_EQ_7 0x30 +#define WM9081_EQ_8 0x31 +#define WM9081_EQ_9 0x32 +#define WM9081_EQ_10 0x33 +#define WM9081_EQ_11 0x34 +#define WM9081_EQ_12 0x35 +#define WM9081_EQ_13 0x36 +#define WM9081_EQ_14 0x37 +#define WM9081_EQ_15 0x38 +#define WM9081_EQ_16 0x39 +#define WM9081_EQ_17 0x3A +#define WM9081_EQ_18 0x3B +#define WM9081_EQ_19 0x3C +#define WM9081_EQ_20 0x3D + +#define WM9081_REGISTER_COUNT 55 +#define WM9081_MAX_REGISTER 0x3D + +/* + * Field Definitions. + */ + +/* + * R0 (0x00) - Software Reset + */ +#define WM9081_SW_RST_DEV_ID1_MASK 0xFFFF /* SW_RST_DEV_ID1 - [15:0] */ +#define WM9081_SW_RST_DEV_ID1_SHIFT 0 /* SW_RST_DEV_ID1 - [15:0] */ +#define WM9081_SW_RST_DEV_ID1_WIDTH 16 /* SW_RST_DEV_ID1 - [15:0] */ + +/* + * R2 (0x02) - Analogue Lineout + */ +#define WM9081_LINEOUT_MUTE 0x0080 /* LINEOUT_MUTE */ +#define WM9081_LINEOUT_MUTE_MASK 0x0080 /* LINEOUT_MUTE */ +#define WM9081_LINEOUT_MUTE_SHIFT 7 /* LINEOUT_MUTE */ +#define WM9081_LINEOUT_MUTE_WIDTH 1 /* LINEOUT_MUTE */ +#define WM9081_LINEOUTZC 0x0040 /* LINEOUTZC */ +#define WM9081_LINEOUTZC_MASK 0x0040 /* LINEOUTZC */ +#define WM9081_LINEOUTZC_SHIFT 6 /* LINEOUTZC */ +#define WM9081_LINEOUTZC_WIDTH 1 /* LINEOUTZC */ +#define WM9081_LINEOUT_VOL_MASK 0x003F /* LINEOUT_VOL - [5:0] */ +#define WM9081_LINEOUT_VOL_SHIFT 0 /* LINEOUT_VOL - [5:0] */ +#define WM9081_LINEOUT_VOL_WIDTH 6 /* LINEOUT_VOL - [5:0] */ + +/* + * R3 (0x03) - Analogue Speaker PGA + */ +#define WM9081_SPKPGA_MUTE 0x0080 /* SPKPGA_MUTE */ +#define WM9081_SPKPGA_MUTE_MASK 0x0080 /* SPKPGA_MUTE */ +#define WM9081_SPKPGA_MUTE_SHIFT 7 /* SPKPGA_MUTE */ +#define WM9081_SPKPGA_MUTE_WIDTH 1 /* SPKPGA_MUTE */ +#define WM9081_SPKPGAZC 0x0040 /* SPKPGAZC */ +#define WM9081_SPKPGAZC_MASK 0x0040 /* SPKPGAZC */ +#define WM9081_SPKPGAZC_SHIFT 6 /* SPKPGAZC */ +#define WM9081_SPKPGAZC_WIDTH 1 /* SPKPGAZC */ +#define WM9081_SPKPGA_VOL_MASK 0x003F /* SPKPGA_VOL - [5:0] */ +#define WM9081_SPKPGA_VOL_SHIFT 0 /* SPKPGA_VOL - [5:0] */ +#define WM9081_SPKPGA_VOL_WIDTH 6 /* SPKPGA_VOL - [5:0] */ + +/* + * R4 (0x04) - VMID Control + */ +#define WM9081_VMID_BUF_ENA 0x0020 /* VMID_BUF_ENA */ +#define WM9081_VMID_BUF_ENA_MASK 0x0020 /* VMID_BUF_ENA */ +#define WM9081_VMID_BUF_ENA_SHIFT 5 /* VMID_BUF_ENA */ +#define WM9081_VMID_BUF_ENA_WIDTH 1 /* VMID_BUF_ENA */ +#define WM9081_VMID_RAMP 0x0008 /* VMID_RAMP */ +#define WM9081_VMID_RAMP_MASK 0x0008 /* VMID_RAMP */ +#define WM9081_VMID_RAMP_SHIFT 3 /* VMID_RAMP */ +#define WM9081_VMID_RAMP_WIDTH 1 /* VMID_RAMP */ +#define WM9081_VMID_SEL_MASK 0x0006 /* VMID_SEL - [2:1] */ +#define WM9081_VMID_SEL_SHIFT 1 /* VMID_SEL - [2:1] */ +#define WM9081_VMID_SEL_WIDTH 2 /* VMID_SEL - [2:1] */ +#define WM9081_VMID_FAST_ST 0x0001 /* VMID_FAST_ST */ +#define WM9081_VMID_FAST_ST_MASK 0x0001 /* VMID_FAST_ST */ +#define WM9081_VMID_FAST_ST_SHIFT 0 /* VMID_FAST_ST */ +#define WM9081_VMID_FAST_ST_WIDTH 1 /* VMID_FAST_ST */ + +/* + * R5 (0x05) - Bias Control 1 + */ +#define WM9081_BIAS_SRC 0x0040 /* BIAS_SRC */ +#define WM9081_BIAS_SRC_MASK 0x0040 /* BIAS_SRC */ +#define WM9081_BIAS_SRC_SHIFT 6 /* BIAS_SRC */ +#define WM9081_BIAS_SRC_WIDTH 1 /* BIAS_SRC */ +#define WM9081_STBY_BIAS_LVL 0x0020 /* STBY_BIAS_LVL */ +#define WM9081_STBY_BIAS_LVL_MASK 0x0020 /* STBY_BIAS_LVL */ +#define WM9081_STBY_BIAS_LVL_SHIFT 5 /* STBY_BIAS_LVL */ +#define WM9081_STBY_BIAS_LVL_WIDTH 1 /* STBY_BIAS_LVL */ +#define WM9081_STBY_BIAS_ENA 0x0010 /* STBY_BIAS_ENA */ +#define WM9081_STBY_BIAS_ENA_MASK 0x0010 /* STBY_BIAS_ENA */ +#define WM9081_STBY_BIAS_ENA_SHIFT 4 /* STBY_BIAS_ENA */ +#define WM9081_STBY_BIAS_ENA_WIDTH 1 /* STBY_BIAS_ENA */ +#define WM9081_BIAS_LVL_MASK 0x000C /* BIAS_LVL - [3:2] */ +#define WM9081_BIAS_LVL_SHIFT 2 /* BIAS_LVL - [3:2] */ +#define WM9081_BIAS_LVL_WIDTH 2 /* BIAS_LVL - [3:2] */ +#define WM9081_BIAS_ENA 0x0002 /* BIAS_ENA */ +#define WM9081_BIAS_ENA_MASK 0x0002 /* BIAS_ENA */ +#define WM9081_BIAS_ENA_SHIFT 1 /* BIAS_ENA */ +#define WM9081_BIAS_ENA_WIDTH 1 /* BIAS_ENA */ +#define WM9081_STARTUP_BIAS_ENA 0x0001 /* STARTUP_BIAS_ENA */ +#define WM9081_STARTUP_BIAS_ENA_MASK 0x0001 /* STARTUP_BIAS_ENA */ +#define WM9081_STARTUP_BIAS_ENA_SHIFT 0 /* STARTUP_BIAS_ENA */ +#define WM9081_STARTUP_BIAS_ENA_WIDTH 1 /* STARTUP_BIAS_ENA */ + +/* + * R7 (0x07) - Analogue Mixer + */ +#define WM9081_DAC_SEL 0x0010 /* DAC_SEL */ +#define WM9081_DAC_SEL_MASK 0x0010 /* DAC_SEL */ +#define WM9081_DAC_SEL_SHIFT 4 /* DAC_SEL */ +#define WM9081_DAC_SEL_WIDTH 1 /* DAC_SEL */ +#define WM9081_IN2_VOL 0x0008 /* IN2_VOL */ +#define WM9081_IN2_VOL_MASK 0x0008 /* IN2_VOL */ +#define WM9081_IN2_VOL_SHIFT 3 /* IN2_VOL */ +#define WM9081_IN2_VOL_WIDTH 1 /* IN2_VOL */ +#define WM9081_IN2_ENA 0x0004 /* IN2_ENA */ +#define WM9081_IN2_ENA_MASK 0x0004 /* IN2_ENA */ +#define WM9081_IN2_ENA_SHIFT 2 /* IN2_ENA */ +#define WM9081_IN2_ENA_WIDTH 1 /* IN2_ENA */ +#define WM9081_IN1_VOL 0x0002 /* IN1_VOL */ +#define WM9081_IN1_VOL_MASK 0x0002 /* IN1_VOL */ +#define WM9081_IN1_VOL_SHIFT 1 /* IN1_VOL */ +#define WM9081_IN1_VOL_WIDTH 1 /* IN1_VOL */ +#define WM9081_IN1_ENA 0x0001 /* IN1_ENA */ +#define WM9081_IN1_ENA_MASK 0x0001 /* IN1_ENA */ +#define WM9081_IN1_ENA_SHIFT 0 /* IN1_ENA */ +#define WM9081_IN1_ENA_WIDTH 1 /* IN1_ENA */ + +/* + * R8 (0x08) - Anti Pop Control + */ +#define WM9081_LINEOUT_DISCH 0x0004 /* LINEOUT_DISCH */ +#define WM9081_LINEOUT_DISCH_MASK 0x0004 /* LINEOUT_DISCH */ +#define WM9081_LINEOUT_DISCH_SHIFT 2 /* LINEOUT_DISCH */ +#define WM9081_LINEOUT_DISCH_WIDTH 1 /* LINEOUT_DISCH */ +#define WM9081_LINEOUT_VROI 0x0002 /* LINEOUT_VROI */ +#define WM9081_LINEOUT_VROI_MASK 0x0002 /* LINEOUT_VROI */ +#define WM9081_LINEOUT_VROI_SHIFT 1 /* LINEOUT_VROI */ +#define WM9081_LINEOUT_VROI_WIDTH 1 /* LINEOUT_VROI */ +#define WM9081_LINEOUT_CLAMP 0x0001 /* LINEOUT_CLAMP */ +#define WM9081_LINEOUT_CLAMP_MASK 0x0001 /* LINEOUT_CLAMP */ +#define WM9081_LINEOUT_CLAMP_SHIFT 0 /* LINEOUT_CLAMP */ +#define WM9081_LINEOUT_CLAMP_WIDTH 1 /* LINEOUT_CLAMP */ + +/* + * R9 (0x09) - Analogue Speaker 1 + */ +#define WM9081_SPK_DCGAIN_MASK 0x0038 /* SPK_DCGAIN - [5:3] */ +#define WM9081_SPK_DCGAIN_SHIFT 3 /* SPK_DCGAIN - [5:3] */ +#define WM9081_SPK_DCGAIN_WIDTH 3 /* SPK_DCGAIN - [5:3] */ +#define WM9081_SPK_ACGAIN_MASK 0x0007 /* SPK_ACGAIN - [2:0] */ +#define WM9081_SPK_ACGAIN_SHIFT 0 /* SPK_ACGAIN - [2:0] */ +#define WM9081_SPK_ACGAIN_WIDTH 3 /* SPK_ACGAIN - [2:0] */ + +/* + * R10 (0x0A) - Analogue Speaker 2 + */ +#define WM9081_SPK_MODE 0x0040 /* SPK_MODE */ +#define WM9081_SPK_MODE_MASK 0x0040 /* SPK_MODE */ +#define WM9081_SPK_MODE_SHIFT 6 /* SPK_MODE */ +#define WM9081_SPK_MODE_WIDTH 1 /* SPK_MODE */ +#define WM9081_SPK_INV_MUTE 0x0010 /* SPK_INV_MUTE */ +#define WM9081_SPK_INV_MUTE_MASK 0x0010 /* SPK_INV_MUTE */ +#define WM9081_SPK_INV_MUTE_SHIFT 4 /* SPK_INV_MUTE */ +#define WM9081_SPK_INV_MUTE_WIDTH 1 /* SPK_INV_MUTE */ +#define WM9081_OUT_SPK_CTRL 0x0008 /* OUT_SPK_CTRL */ +#define WM9081_OUT_SPK_CTRL_MASK 0x0008 /* OUT_SPK_CTRL */ +#define WM9081_OUT_SPK_CTRL_SHIFT 3 /* OUT_SPK_CTRL */ +#define WM9081_OUT_SPK_CTRL_WIDTH 1 /* OUT_SPK_CTRL */ + +/* + * R11 (0x0B) - Power Management + */ +#define WM9081_TSHUT_ENA 0x0100 /* TSHUT_ENA */ +#define WM9081_TSHUT_ENA_MASK 0x0100 /* TSHUT_ENA */ +#define WM9081_TSHUT_ENA_SHIFT 8 /* TSHUT_ENA */ +#define WM9081_TSHUT_ENA_WIDTH 1 /* TSHUT_ENA */ +#define WM9081_TSENSE_ENA 0x0080 /* TSENSE_ENA */ +#define WM9081_TSENSE_ENA_MASK 0x0080 /* TSENSE_ENA */ +#define WM9081_TSENSE_ENA_SHIFT 7 /* TSENSE_ENA */ +#define WM9081_TSENSE_ENA_WIDTH 1 /* TSENSE_ENA */ +#define WM9081_TEMP_SHUT 0x0040 /* TEMP_SHUT */ +#define WM9081_TEMP_SHUT_MASK 0x0040 /* TEMP_SHUT */ +#define WM9081_TEMP_SHUT_SHIFT 6 /* TEMP_SHUT */ +#define WM9081_TEMP_SHUT_WIDTH 1 /* TEMP_SHUT */ +#define WM9081_LINEOUT_ENA 0x0010 /* LINEOUT_ENA */ +#define WM9081_LINEOUT_ENA_MASK 0x0010 /* LINEOUT_ENA */ +#define WM9081_LINEOUT_ENA_SHIFT 4 /* LINEOUT_ENA */ +#define WM9081_LINEOUT_ENA_WIDTH 1 /* LINEOUT_ENA */ +#define WM9081_SPKPGA_ENA 0x0004 /* SPKPGA_ENA */ +#define WM9081_SPKPGA_ENA_MASK 0x0004 /* SPKPGA_ENA */ +#define WM9081_SPKPGA_ENA_SHIFT 2 /* SPKPGA_ENA */ +#define WM9081_SPKPGA_ENA_WIDTH 1 /* SPKPGA_ENA */ +#define WM9081_SPK_ENA 0x0002 /* SPK_ENA */ +#define WM9081_SPK_ENA_MASK 0x0002 /* SPK_ENA */ +#define WM9081_SPK_ENA_SHIFT 1 /* SPK_ENA */ +#define WM9081_SPK_ENA_WIDTH 1 /* SPK_ENA */ +#define WM9081_DAC_ENA 0x0001 /* DAC_ENA */ +#define WM9081_DAC_ENA_MASK 0x0001 /* DAC_ENA */ +#define WM9081_DAC_ENA_SHIFT 0 /* DAC_ENA */ +#define WM9081_DAC_ENA_WIDTH 1 /* DAC_ENA */ + +/* + * R12 (0x0C) - Clock Control 1 + */ +#define WM9081_CLK_OP_DIV_MASK 0x1C00 /* CLK_OP_DIV - [12:10] */ +#define WM9081_CLK_OP_DIV_SHIFT 10 /* CLK_OP_DIV - [12:10] */ +#define WM9081_CLK_OP_DIV_WIDTH 3 /* CLK_OP_DIV - [12:10] */ +#define WM9081_CLK_TO_DIV_MASK 0x0300 /* CLK_TO_DIV - [9:8] */ +#define WM9081_CLK_TO_DIV_SHIFT 8 /* CLK_TO_DIV - [9:8] */ +#define WM9081_CLK_TO_DIV_WIDTH 2 /* CLK_TO_DIV - [9:8] */ +#define WM9081_MCLKDIV2 0x0080 /* MCLKDIV2 */ +#define WM9081_MCLKDIV2_MASK 0x0080 /* MCLKDIV2 */ +#define WM9081_MCLKDIV2_SHIFT 7 /* MCLKDIV2 */ +#define WM9081_MCLKDIV2_WIDTH 1 /* MCLKDIV2 */ + +/* + * R13 (0x0D) - Clock Control 2 + */ +#define WM9081_CLK_SYS_RATE_MASK 0x00F0 /* CLK_SYS_RATE - [7:4] */ +#define WM9081_CLK_SYS_RATE_SHIFT 4 /* CLK_SYS_RATE - [7:4] */ +#define WM9081_CLK_SYS_RATE_WIDTH 4 /* CLK_SYS_RATE - [7:4] */ +#define WM9081_SAMPLE_RATE_MASK 0x000F /* SAMPLE_RATE - [3:0] */ +#define WM9081_SAMPLE_RATE_SHIFT 0 /* SAMPLE_RATE - [3:0] */ +#define WM9081_SAMPLE_RATE_WIDTH 4 /* SAMPLE_RATE - [3:0] */ + +/* + * R14 (0x0E) - Clock Control 3 + */ +#define WM9081_CLK_SRC_SEL 0x2000 /* CLK_SRC_SEL */ +#define WM9081_CLK_SRC_SEL_MASK 0x2000 /* CLK_SRC_SEL */ +#define WM9081_CLK_SRC_SEL_SHIFT 13 /* CLK_SRC_SEL */ +#define WM9081_CLK_SRC_SEL_WIDTH 1 /* CLK_SRC_SEL */ +#define WM9081_CLK_OP_ENA 0x0020 /* CLK_OP_ENA */ +#define WM9081_CLK_OP_ENA_MASK 0x0020 /* CLK_OP_ENA */ +#define WM9081_CLK_OP_ENA_SHIFT 5 /* CLK_OP_ENA */ +#define WM9081_CLK_OP_ENA_WIDTH 1 /* CLK_OP_ENA */ +#define WM9081_CLK_TO_ENA 0x0004 /* CLK_TO_ENA */ +#define WM9081_CLK_TO_ENA_MASK 0x0004 /* CLK_TO_ENA */ +#define WM9081_CLK_TO_ENA_SHIFT 2 /* CLK_TO_ENA */ +#define WM9081_CLK_TO_ENA_WIDTH 1 /* CLK_TO_ENA */ +#define WM9081_CLK_DSP_ENA 0x0002 /* CLK_DSP_ENA */ +#define WM9081_CLK_DSP_ENA_MASK 0x0002 /* CLK_DSP_ENA */ +#define WM9081_CLK_DSP_ENA_SHIFT 1 /* CLK_DSP_ENA */ +#define WM9081_CLK_DSP_ENA_WIDTH 1 /* CLK_DSP_ENA */ +#define WM9081_CLK_SYS_ENA 0x0001 /* CLK_SYS_ENA */ +#define WM9081_CLK_SYS_ENA_MASK 0x0001 /* CLK_SYS_ENA */ +#define WM9081_CLK_SYS_ENA_SHIFT 0 /* CLK_SYS_ENA */ +#define WM9081_CLK_SYS_ENA_WIDTH 1 /* CLK_SYS_ENA */ + +/* + * R16 (0x10) - FLL Control 1 + */ +#define WM9081_FLL_HOLD 0x0008 /* FLL_HOLD */ +#define WM9081_FLL_HOLD_MASK 0x0008 /* FLL_HOLD */ +#define WM9081_FLL_HOLD_SHIFT 3 /* FLL_HOLD */ +#define WM9081_FLL_HOLD_WIDTH 1 /* FLL_HOLD */ +#define WM9081_FLL_FRAC 0x0004 /* FLL_FRAC */ +#define WM9081_FLL_FRAC_MASK 0x0004 /* FLL_FRAC */ +#define WM9081_FLL_FRAC_SHIFT 2 /* FLL_FRAC */ +#define WM9081_FLL_FRAC_WIDTH 1 /* FLL_FRAC */ +#define WM9081_FLL_ENA 0x0001 /* FLL_ENA */ +#define WM9081_FLL_ENA_MASK 0x0001 /* FLL_ENA */ +#define WM9081_FLL_ENA_SHIFT 0 /* FLL_ENA */ +#define WM9081_FLL_ENA_WIDTH 1 /* FLL_ENA */ + +/* + * R17 (0x11) - FLL Control 2 + */ +#define WM9081_FLL_OUTDIV_MASK 0x0700 /* FLL_OUTDIV - [10:8] */ +#define WM9081_FLL_OUTDIV_SHIFT 8 /* FLL_OUTDIV - [10:8] */ +#define WM9081_FLL_OUTDIV_WIDTH 3 /* FLL_OUTDIV - [10:8] */ +#define WM9081_FLL_CTRL_RATE_MASK 0x0070 /* FLL_CTRL_RATE - [6:4] */ +#define WM9081_FLL_CTRL_RATE_SHIFT 4 /* FLL_CTRL_RATE - [6:4] */ +#define WM9081_FLL_CTRL_RATE_WIDTH 3 /* FLL_CTRL_RATE - [6:4] */ +#define WM9081_FLL_FRATIO_MASK 0x0007 /* FLL_FRATIO - [2:0] */ +#define WM9081_FLL_FRATIO_SHIFT 0 /* FLL_FRATIO - [2:0] */ +#define WM9081_FLL_FRATIO_WIDTH 3 /* FLL_FRATIO - [2:0] */ + +/* + * R18 (0x12) - FLL Control 3 + */ +#define WM9081_FLL_K_MASK 0xFFFF /* FLL_K - [15:0] */ +#define WM9081_FLL_K_SHIFT 0 /* FLL_K - [15:0] */ +#define WM9081_FLL_K_WIDTH 16 /* FLL_K - [15:0] */ + +/* + * R19 (0x13) - FLL Control 4 + */ +#define WM9081_FLL_N_MASK 0x7FE0 /* FLL_N - [14:5] */ +#define WM9081_FLL_N_SHIFT 5 /* FLL_N - [14:5] */ +#define WM9081_FLL_N_WIDTH 10 /* FLL_N - [14:5] */ +#define WM9081_FLL_GAIN_MASK 0x000F /* FLL_GAIN - [3:0] */ +#define WM9081_FLL_GAIN_SHIFT 0 /* FLL_GAIN - [3:0] */ +#define WM9081_FLL_GAIN_WIDTH 4 /* FLL_GAIN - [3:0] */ + +/* + * R20 (0x14) - FLL Control 5 + */ +#define WM9081_FLL_CLK_REF_DIV_MASK 0x0018 /* FLL_CLK_REF_DIV - [4:3] */ +#define WM9081_FLL_CLK_REF_DIV_SHIFT 3 /* FLL_CLK_REF_DIV - [4:3] */ +#define WM9081_FLL_CLK_REF_DIV_WIDTH 2 /* FLL_CLK_REF_DIV - [4:3] */ +#define WM9081_FLL_CLK_SRC_MASK 0x0003 /* FLL_CLK_SRC - [1:0] */ +#define WM9081_FLL_CLK_SRC_SHIFT 0 /* FLL_CLK_SRC - [1:0] */ +#define WM9081_FLL_CLK_SRC_WIDTH 2 /* FLL_CLK_SRC - [1:0] */ + +/* + * R22 (0x16) - Audio Interface 1 + */ +#define WM9081_AIFDAC_CHAN 0x0040 /* AIFDAC_CHAN */ +#define WM9081_AIFDAC_CHAN_MASK 0x0040 /* AIFDAC_CHAN */ +#define WM9081_AIFDAC_CHAN_SHIFT 6 /* AIFDAC_CHAN */ +#define WM9081_AIFDAC_CHAN_WIDTH 1 /* AIFDAC_CHAN */ +#define WM9081_AIFDAC_TDM_SLOT_MASK 0x0030 /* AIFDAC_TDM_SLOT - [5:4] */ +#define WM9081_AIFDAC_TDM_SLOT_SHIFT 4 /* AIFDAC_TDM_SLOT - [5:4] */ +#define WM9081_AIFDAC_TDM_SLOT_WIDTH 2 /* AIFDAC_TDM_SLOT - [5:4] */ +#define WM9081_AIFDAC_TDM_MODE_MASK 0x000C /* AIFDAC_TDM_MODE - [3:2] */ +#define WM9081_AIFDAC_TDM_MODE_SHIFT 2 /* AIFDAC_TDM_MODE - [3:2] */ +#define WM9081_AIFDAC_TDM_MODE_WIDTH 2 /* AIFDAC_TDM_MODE - [3:2] */ +#define WM9081_DAC_COMP 0x0002 /* DAC_COMP */ +#define WM9081_DAC_COMP_MASK 0x0002 /* DAC_COMP */ +#define WM9081_DAC_COMP_SHIFT 1 /* DAC_COMP */ +#define WM9081_DAC_COMP_WIDTH 1 /* DAC_COMP */ +#define WM9081_DAC_COMPMODE 0x0001 /* DAC_COMPMODE */ +#define WM9081_DAC_COMPMODE_MASK 0x0001 /* DAC_COMPMODE */ +#define WM9081_DAC_COMPMODE_SHIFT 0 /* DAC_COMPMODE */ +#define WM9081_DAC_COMPMODE_WIDTH 1 /* DAC_COMPMODE */ + +/* + * R23 (0x17) - Audio Interface 2 + */ +#define WM9081_AIF_TRIS 0x0200 /* AIF_TRIS */ +#define WM9081_AIF_TRIS_MASK 0x0200 /* AIF_TRIS */ +#define WM9081_AIF_TRIS_SHIFT 9 /* AIF_TRIS */ +#define WM9081_AIF_TRIS_WIDTH 1 /* AIF_TRIS */ +#define WM9081_DAC_DAT_INV 0x0100 /* DAC_DAT_INV */ +#define WM9081_DAC_DAT_INV_MASK 0x0100 /* DAC_DAT_INV */ +#define WM9081_DAC_DAT_INV_SHIFT 8 /* DAC_DAT_INV */ +#define WM9081_DAC_DAT_INV_WIDTH 1 /* DAC_DAT_INV */ +#define WM9081_AIF_BCLK_INV 0x0080 /* AIF_BCLK_INV */ +#define WM9081_AIF_BCLK_INV_MASK 0x0080 /* AIF_BCLK_INV */ +#define WM9081_AIF_BCLK_INV_SHIFT 7 /* AIF_BCLK_INV */ +#define WM9081_AIF_BCLK_INV_WIDTH 1 /* AIF_BCLK_INV */ +#define WM9081_BCLK_DIR 0x0040 /* BCLK_DIR */ +#define WM9081_BCLK_DIR_MASK 0x0040 /* BCLK_DIR */ +#define WM9081_BCLK_DIR_SHIFT 6 /* BCLK_DIR */ +#define WM9081_BCLK_DIR_WIDTH 1 /* BCLK_DIR */ +#define WM9081_LRCLK_DIR 0x0020 /* LRCLK_DIR */ +#define WM9081_LRCLK_DIR_MASK 0x0020 /* LRCLK_DIR */ +#define WM9081_LRCLK_DIR_SHIFT 5 /* LRCLK_DIR */ +#define WM9081_LRCLK_DIR_WIDTH 1 /* LRCLK_DIR */ +#define WM9081_AIF_LRCLK_INV 0x0010 /* AIF_LRCLK_INV */ +#define WM9081_AIF_LRCLK_INV_MASK 0x0010 /* AIF_LRCLK_INV */ +#define WM9081_AIF_LRCLK_INV_SHIFT 4 /* AIF_LRCLK_INV */ +#define WM9081_AIF_LRCLK_INV_WIDTH 1 /* AIF_LRCLK_INV */ +#define WM9081_AIF_WL_MASK 0x000C /* AIF_WL - [3:2] */ +#define WM9081_AIF_WL_SHIFT 2 /* AIF_WL - [3:2] */ +#define WM9081_AIF_WL_WIDTH 2 /* AIF_WL - [3:2] */ +#define WM9081_AIF_FMT_MASK 0x0003 /* AIF_FMT - [1:0] */ +#define WM9081_AIF_FMT_SHIFT 0 /* AIF_FMT - [1:0] */ +#define WM9081_AIF_FMT_WIDTH 2 /* AIF_FMT - [1:0] */ + +/* + * R24 (0x18) - Audio Interface 3 + */ +#define WM9081_BCLK_DIV_MASK 0x001F /* BCLK_DIV - [4:0] */ +#define WM9081_BCLK_DIV_SHIFT 0 /* BCLK_DIV - [4:0] */ +#define WM9081_BCLK_DIV_WIDTH 5 /* BCLK_DIV - [4:0] */ + +/* + * R25 (0x19) - Audio Interface 4 + */ +#define WM9081_LRCLK_RATE_MASK 0x07FF /* LRCLK_RATE - [10:0] */ +#define WM9081_LRCLK_RATE_SHIFT 0 /* LRCLK_RATE - [10:0] */ +#define WM9081_LRCLK_RATE_WIDTH 11 /* LRCLK_RATE - [10:0] */ + +/* + * R26 (0x1A) - Interrupt Status + */ +#define WM9081_WSEQ_BUSY_EINT 0x0004 /* WSEQ_BUSY_EINT */ +#define WM9081_WSEQ_BUSY_EINT_MASK 0x0004 /* WSEQ_BUSY_EINT */ +#define WM9081_WSEQ_BUSY_EINT_SHIFT 2 /* WSEQ_BUSY_EINT */ +#define WM9081_WSEQ_BUSY_EINT_WIDTH 1 /* WSEQ_BUSY_EINT */ +#define WM9081_TSHUT_EINT 0x0001 /* TSHUT_EINT */ +#define WM9081_TSHUT_EINT_MASK 0x0001 /* TSHUT_EINT */ +#define WM9081_TSHUT_EINT_SHIFT 0 /* TSHUT_EINT */ +#define WM9081_TSHUT_EINT_WIDTH 1 /* TSHUT_EINT */ + +/* + * R27 (0x1B) - Interrupt Status Mask + */ +#define WM9081_IM_WSEQ_BUSY_EINT 0x0004 /* IM_WSEQ_BUSY_EINT */ +#define WM9081_IM_WSEQ_BUSY_EINT_MASK 0x0004 /* IM_WSEQ_BUSY_EINT */ +#define WM9081_IM_WSEQ_BUSY_EINT_SHIFT 2 /* IM_WSEQ_BUSY_EINT */ +#define WM9081_IM_WSEQ_BUSY_EINT_WIDTH 1 /* IM_WSEQ_BUSY_EINT */ +#define WM9081_IM_TSHUT_EINT 0x0001 /* IM_TSHUT_EINT */ +#define WM9081_IM_TSHUT_EINT_MASK 0x0001 /* IM_TSHUT_EINT */ +#define WM9081_IM_TSHUT_EINT_SHIFT 0 /* IM_TSHUT_EINT */ +#define WM9081_IM_TSHUT_EINT_WIDTH 1 /* IM_TSHUT_EINT */ + +/* + * R28 (0x1C) - Interrupt Polarity + */ +#define WM9081_TSHUT_INV 0x0001 /* TSHUT_INV */ +#define WM9081_TSHUT_INV_MASK 0x0001 /* TSHUT_INV */ +#define WM9081_TSHUT_INV_SHIFT 0 /* TSHUT_INV */ +#define WM9081_TSHUT_INV_WIDTH 1 /* TSHUT_INV */ + +/* + * R29 (0x1D) - Interrupt Control + */ +#define WM9081_IRQ_POL 0x8000 /* IRQ_POL */ +#define WM9081_IRQ_POL_MASK 0x8000 /* IRQ_POL */ +#define WM9081_IRQ_POL_SHIFT 15 /* IRQ_POL */ +#define WM9081_IRQ_POL_WIDTH 1 /* IRQ_POL */ +#define WM9081_IRQ_OP_CTRL 0x0001 /* IRQ_OP_CTRL */ +#define WM9081_IRQ_OP_CTRL_MASK 0x0001 /* IRQ_OP_CTRL */ +#define WM9081_IRQ_OP_CTRL_SHIFT 0 /* IRQ_OP_CTRL */ +#define WM9081_IRQ_OP_CTRL_WIDTH 1 /* IRQ_OP_CTRL */ + +/* + * R30 (0x1E) - DAC Digital 1 + */ +#define WM9081_DAC_VOL_MASK 0x00FF /* DAC_VOL - [7:0] */ +#define WM9081_DAC_VOL_SHIFT 0 /* DAC_VOL - [7:0] */ +#define WM9081_DAC_VOL_WIDTH 8 /* DAC_VOL - [7:0] */ + +/* + * R31 (0x1F) - DAC Digital 2 + */ +#define WM9081_DAC_MUTERATE 0x0400 /* DAC_MUTERATE */ +#define WM9081_DAC_MUTERATE_MASK 0x0400 /* DAC_MUTERATE */ +#define WM9081_DAC_MUTERATE_SHIFT 10 /* DAC_MUTERATE */ +#define WM9081_DAC_MUTERATE_WIDTH 1 /* DAC_MUTERATE */ +#define WM9081_DAC_MUTEMODE 0x0200 /* DAC_MUTEMODE */ +#define WM9081_DAC_MUTEMODE_MASK 0x0200 /* DAC_MUTEMODE */ +#define WM9081_DAC_MUTEMODE_SHIFT 9 /* DAC_MUTEMODE */ +#define WM9081_DAC_MUTEMODE_WIDTH 1 /* DAC_MUTEMODE */ +#define WM9081_DAC_MUTE 0x0008 /* DAC_MUTE */ +#define WM9081_DAC_MUTE_MASK 0x0008 /* DAC_MUTE */ +#define WM9081_DAC_MUTE_SHIFT 3 /* DAC_MUTE */ +#define WM9081_DAC_MUTE_WIDTH 1 /* DAC_MUTE */ +#define WM9081_DEEMPH_MASK 0x0006 /* DEEMPH - [2:1] */ +#define WM9081_DEEMPH_SHIFT 1 /* DEEMPH - [2:1] */ +#define WM9081_DEEMPH_WIDTH 2 /* DEEMPH - [2:1] */ + +/* + * R32 (0x20) - DRC 1 + */ +#define WM9081_DRC_ENA 0x8000 /* DRC_ENA */ +#define WM9081_DRC_ENA_MASK 0x8000 /* DRC_ENA */ +#define WM9081_DRC_ENA_SHIFT 15 /* DRC_ENA */ +#define WM9081_DRC_ENA_WIDTH 1 /* DRC_ENA */ +#define WM9081_DRC_STARTUP_GAIN_MASK 0x07C0 /* DRC_STARTUP_GAIN - [10:6] */ +#define WM9081_DRC_STARTUP_GAIN_SHIFT 6 /* DRC_STARTUP_GAIN - [10:6] */ +#define WM9081_DRC_STARTUP_GAIN_WIDTH 5 /* DRC_STARTUP_GAIN - [10:6] */ +#define WM9081_DRC_FF_DLY 0x0020 /* DRC_FF_DLY */ +#define WM9081_DRC_FF_DLY_MASK 0x0020 /* DRC_FF_DLY */ +#define WM9081_DRC_FF_DLY_SHIFT 5 /* DRC_FF_DLY */ +#define WM9081_DRC_FF_DLY_WIDTH 1 /* DRC_FF_DLY */ +#define WM9081_DRC_QR 0x0004 /* DRC_QR */ +#define WM9081_DRC_QR_MASK 0x0004 /* DRC_QR */ +#define WM9081_DRC_QR_SHIFT 2 /* DRC_QR */ +#define WM9081_DRC_QR_WIDTH 1 /* DRC_QR */ +#define WM9081_DRC_ANTICLIP 0x0002 /* DRC_ANTICLIP */ +#define WM9081_DRC_ANTICLIP_MASK 0x0002 /* DRC_ANTICLIP */ +#define WM9081_DRC_ANTICLIP_SHIFT 1 /* DRC_ANTICLIP */ +#define WM9081_DRC_ANTICLIP_WIDTH 1 /* DRC_ANTICLIP */ + +/* + * R33 (0x21) - DRC 2 + */ +#define WM9081_DRC_ATK_MASK 0xF000 /* DRC_ATK - [15:12] */ +#define WM9081_DRC_ATK_SHIFT 12 /* DRC_ATK - [15:12] */ +#define WM9081_DRC_ATK_WIDTH 4 /* DRC_ATK - [15:12] */ +#define WM9081_DRC_DCY_MASK 0x0F00 /* DRC_DCY - [11:8] */ +#define WM9081_DRC_DCY_SHIFT 8 /* DRC_DCY - [11:8] */ +#define WM9081_DRC_DCY_WIDTH 4 /* DRC_DCY - [11:8] */ +#define WM9081_DRC_QR_THR_MASK 0x00C0 /* DRC_QR_THR - [7:6] */ +#define WM9081_DRC_QR_THR_SHIFT 6 /* DRC_QR_THR - [7:6] */ +#define WM9081_DRC_QR_THR_WIDTH 2 /* DRC_QR_THR - [7:6] */ +#define WM9081_DRC_QR_DCY_MASK 0x0030 /* DRC_QR_DCY - [5:4] */ +#define WM9081_DRC_QR_DCY_SHIFT 4 /* DRC_QR_DCY - [5:4] */ +#define WM9081_DRC_QR_DCY_WIDTH 2 /* DRC_QR_DCY - [5:4] */ +#define WM9081_DRC_MINGAIN_MASK 0x000C /* DRC_MINGAIN - [3:2] */ +#define WM9081_DRC_MINGAIN_SHIFT 2 /* DRC_MINGAIN - [3:2] */ +#define WM9081_DRC_MINGAIN_WIDTH 2 /* DRC_MINGAIN - [3:2] */ +#define WM9081_DRC_MAXGAIN_MASK 0x0003 /* DRC_MAXGAIN - [1:0] */ +#define WM9081_DRC_MAXGAIN_SHIFT 0 /* DRC_MAXGAIN - [1:0] */ +#define WM9081_DRC_MAXGAIN_WIDTH 2 /* DRC_MAXGAIN - [1:0] */ + +/* + * R34 (0x22) - DRC 3 + */ +#define WM9081_DRC_HI_COMP_MASK 0x0038 /* DRC_HI_COMP - [5:3] */ +#define WM9081_DRC_HI_COMP_SHIFT 3 /* DRC_HI_COMP - [5:3] */ +#define WM9081_DRC_HI_COMP_WIDTH 3 /* DRC_HI_COMP - [5:3] */ +#define WM9081_DRC_LO_COMP_MASK 0x0007 /* DRC_LO_COMP - [2:0] */ +#define WM9081_DRC_LO_COMP_SHIFT 0 /* DRC_LO_COMP - [2:0] */ +#define WM9081_DRC_LO_COMP_WIDTH 3 /* DRC_LO_COMP - [2:0] */ + +/* + * R35 (0x23) - DRC 4 + */ +#define WM9081_DRC_KNEE_IP_MASK 0x07E0 /* DRC_KNEE_IP - [10:5] */ +#define WM9081_DRC_KNEE_IP_SHIFT 5 /* DRC_KNEE_IP - [10:5] */ +#define WM9081_DRC_KNEE_IP_WIDTH 6 /* DRC_KNEE_IP - [10:5] */ +#define WM9081_DRC_KNEE_OP_MASK 0x001F /* DRC_KNEE_OP - [4:0] */ +#define WM9081_DRC_KNEE_OP_SHIFT 0 /* DRC_KNEE_OP - [4:0] */ +#define WM9081_DRC_KNEE_OP_WIDTH 5 /* DRC_KNEE_OP - [4:0] */ + +/* + * R38 (0x26) - Write Sequencer 1 + */ +#define WM9081_WSEQ_ENA 0x8000 /* WSEQ_ENA */ +#define WM9081_WSEQ_ENA_MASK 0x8000 /* WSEQ_ENA */ +#define WM9081_WSEQ_ENA_SHIFT 15 /* WSEQ_ENA */ +#define WM9081_WSEQ_ENA_WIDTH 1 /* WSEQ_ENA */ +#define WM9081_WSEQ_ABORT 0x0200 /* WSEQ_ABORT */ +#define WM9081_WSEQ_ABORT_MASK 0x0200 /* WSEQ_ABORT */ +#define WM9081_WSEQ_ABORT_SHIFT 9 /* WSEQ_ABORT */ +#define WM9081_WSEQ_ABORT_WIDTH 1 /* WSEQ_ABORT */ +#define WM9081_WSEQ_START 0x0100 /* WSEQ_START */ +#define WM9081_WSEQ_START_MASK 0x0100 /* WSEQ_START */ +#define WM9081_WSEQ_START_SHIFT 8 /* WSEQ_START */ +#define WM9081_WSEQ_START_WIDTH 1 /* WSEQ_START */ +#define WM9081_WSEQ_START_INDEX_MASK 0x007F /* WSEQ_START_INDEX - [6:0] */ +#define WM9081_WSEQ_START_INDEX_SHIFT 0 /* WSEQ_START_INDEX - [6:0] */ +#define WM9081_WSEQ_START_INDEX_WIDTH 7 /* WSEQ_START_INDEX - [6:0] */ + +/* + * R39 (0x27) - Write Sequencer 2 + */ +#define WM9081_WSEQ_CURRENT_INDEX_MASK 0x07F0 /* WSEQ_CURRENT_INDEX - [10:4] */ +#define WM9081_WSEQ_CURRENT_INDEX_SHIFT 4 /* WSEQ_CURRENT_INDEX - [10:4] */ +#define WM9081_WSEQ_CURRENT_INDEX_WIDTH 7 /* WSEQ_CURRENT_INDEX - [10:4] */ +#define WM9081_WSEQ_BUSY 0x0001 /* WSEQ_BUSY */ +#define WM9081_WSEQ_BUSY_MASK 0x0001 /* WSEQ_BUSY */ +#define WM9081_WSEQ_BUSY_SHIFT 0 /* WSEQ_BUSY */ +#define WM9081_WSEQ_BUSY_WIDTH 1 /* WSEQ_BUSY */ + +/* + * R40 (0x28) - MW Slave 1 + */ +#define WM9081_SPI_CFG 0x0020 /* SPI_CFG */ +#define WM9081_SPI_CFG_MASK 0x0020 /* SPI_CFG */ +#define WM9081_SPI_CFG_SHIFT 5 /* SPI_CFG */ +#define WM9081_SPI_CFG_WIDTH 1 /* SPI_CFG */ +#define WM9081_SPI_4WIRE 0x0010 /* SPI_4WIRE */ +#define WM9081_SPI_4WIRE_MASK 0x0010 /* SPI_4WIRE */ +#define WM9081_SPI_4WIRE_SHIFT 4 /* SPI_4WIRE */ +#define WM9081_SPI_4WIRE_WIDTH 1 /* SPI_4WIRE */ +#define WM9081_ARA_ENA 0x0008 /* ARA_ENA */ +#define WM9081_ARA_ENA_MASK 0x0008 /* ARA_ENA */ +#define WM9081_ARA_ENA_SHIFT 3 /* ARA_ENA */ +#define WM9081_ARA_ENA_WIDTH 1 /* ARA_ENA */ +#define WM9081_AUTO_INC 0x0002 /* AUTO_INC */ +#define WM9081_AUTO_INC_MASK 0x0002 /* AUTO_INC */ +#define WM9081_AUTO_INC_SHIFT 1 /* AUTO_INC */ +#define WM9081_AUTO_INC_WIDTH 1 /* AUTO_INC */ + +/* + * R42 (0x2A) - EQ 1 + */ +#define WM9081_EQ_B1_GAIN_MASK 0xF800 /* EQ_B1_GAIN - [15:11] */ +#define WM9081_EQ_B1_GAIN_SHIFT 11 /* EQ_B1_GAIN - [15:11] */ +#define WM9081_EQ_B1_GAIN_WIDTH 5 /* EQ_B1_GAIN - [15:11] */ +#define WM9081_EQ_B2_GAIN_MASK 0x07C0 /* EQ_B2_GAIN - [10:6] */ +#define WM9081_EQ_B2_GAIN_SHIFT 6 /* EQ_B2_GAIN - [10:6] */ +#define WM9081_EQ_B2_GAIN_WIDTH 5 /* EQ_B2_GAIN - [10:6] */ +#define WM9081_EQ_B4_GAIN_MASK 0x003E /* EQ_B4_GAIN - [5:1] */ +#define WM9081_EQ_B4_GAIN_SHIFT 1 /* EQ_B4_GAIN - [5:1] */ +#define WM9081_EQ_B4_GAIN_WIDTH 5 /* EQ_B4_GAIN - [5:1] */ +#define WM9081_EQ_ENA 0x0001 /* EQ_ENA */ +#define WM9081_EQ_ENA_MASK 0x0001 /* EQ_ENA */ +#define WM9081_EQ_ENA_SHIFT 0 /* EQ_ENA */ +#define WM9081_EQ_ENA_WIDTH 1 /* EQ_ENA */ + +/* + * R43 (0x2B) - EQ 2 + */ +#define WM9081_EQ_B3_GAIN_MASK 0xF800 /* EQ_B3_GAIN - [15:11] */ +#define WM9081_EQ_B3_GAIN_SHIFT 11 /* EQ_B3_GAIN - [15:11] */ +#define WM9081_EQ_B3_GAIN_WIDTH 5 /* EQ_B3_GAIN - [15:11] */ +#define WM9081_EQ_B5_GAIN_MASK 0x07C0 /* EQ_B5_GAIN - [10:6] */ +#define WM9081_EQ_B5_GAIN_SHIFT 6 /* EQ_B5_GAIN - [10:6] */ +#define WM9081_EQ_B5_GAIN_WIDTH 5 /* EQ_B5_GAIN - [10:6] */ + +/* + * R44 (0x2C) - EQ 3 + */ +#define WM9081_EQ_B1_A_MASK 0xFFFF /* EQ_B1_A - [15:0] */ +#define WM9081_EQ_B1_A_SHIFT 0 /* EQ_B1_A - [15:0] */ +#define WM9081_EQ_B1_A_WIDTH 16 /* EQ_B1_A - [15:0] */ + +/* + * R45 (0x2D) - EQ 4 + */ +#define WM9081_EQ_B1_B_MASK 0xFFFF /* EQ_B1_B - [15:0] */ +#define WM9081_EQ_B1_B_SHIFT 0 /* EQ_B1_B - [15:0] */ +#define WM9081_EQ_B1_B_WIDTH 16 /* EQ_B1_B - [15:0] */ + +/* + * R46 (0x2E) - EQ 5 + */ +#define WM9081_EQ_B1_PG_MASK 0xFFFF /* EQ_B1_PG - [15:0] */ +#define WM9081_EQ_B1_PG_SHIFT 0 /* EQ_B1_PG - [15:0] */ +#define WM9081_EQ_B1_PG_WIDTH 16 /* EQ_B1_PG - [15:0] */ + +/* + * R47 (0x2F) - EQ 6 + */ +#define WM9081_EQ_B2_A_MASK 0xFFFF /* EQ_B2_A - [15:0] */ +#define WM9081_EQ_B2_A_SHIFT 0 /* EQ_B2_A - [15:0] */ +#define WM9081_EQ_B2_A_WIDTH 16 /* EQ_B2_A - [15:0] */ + +/* + * R48 (0x30) - EQ 7 + */ +#define WM9081_EQ_B2_B_MASK 0xFFFF /* EQ_B2_B - [15:0] */ +#define WM9081_EQ_B2_B_SHIFT 0 /* EQ_B2_B - [15:0] */ +#define WM9081_EQ_B2_B_WIDTH 16 /* EQ_B2_B - [15:0] */ + +/* + * R49 (0x31) - EQ 8 + */ +#define WM9081_EQ_B2_C_MASK 0xFFFF /* EQ_B2_C - [15:0] */ +#define WM9081_EQ_B2_C_SHIFT 0 /* EQ_B2_C - [15:0] */ +#define WM9081_EQ_B2_C_WIDTH 16 /* EQ_B2_C - [15:0] */ + +/* + * R50 (0x32) - EQ 9 + */ +#define WM9081_EQ_B2_PG_MASK 0xFFFF /* EQ_B2_PG - [15:0] */ +#define WM9081_EQ_B2_PG_SHIFT 0 /* EQ_B2_PG - [15:0] */ +#define WM9081_EQ_B2_PG_WIDTH 16 /* EQ_B2_PG - [15:0] */ + +/* + * R51 (0x33) - EQ 10 + */ +#define WM9081_EQ_B4_A_MASK 0xFFFF /* EQ_B4_A - [15:0] */ +#define WM9081_EQ_B4_A_SHIFT 0 /* EQ_B4_A - [15:0] */ +#define WM9081_EQ_B4_A_WIDTH 16 /* EQ_B4_A - [15:0] */ + +/* + * R52 (0x34) - EQ 11 + */ +#define WM9081_EQ_B4_B_MASK 0xFFFF /* EQ_B4_B - [15:0] */ +#define WM9081_EQ_B4_B_SHIFT 0 /* EQ_B4_B - [15:0] */ +#define WM9081_EQ_B4_B_WIDTH 16 /* EQ_B4_B - [15:0] */ + +/* + * R53 (0x35) - EQ 12 + */ +#define WM9081_EQ_B4_C_MASK 0xFFFF /* EQ_B4_C - [15:0] */ +#define WM9081_EQ_B4_C_SHIFT 0 /* EQ_B4_C - [15:0] */ +#define WM9081_EQ_B4_C_WIDTH 16 /* EQ_B4_C - [15:0] */ + +/* + * R54 (0x36) - EQ 13 + */ +#define WM9081_EQ_B4_PG_MASK 0xFFFF /* EQ_B4_PG - [15:0] */ +#define WM9081_EQ_B4_PG_SHIFT 0 /* EQ_B4_PG - [15:0] */ +#define WM9081_EQ_B4_PG_WIDTH 16 /* EQ_B4_PG - [15:0] */ + +/* + * R55 (0x37) - EQ 14 + */ +#define WM9081_EQ_B3_A_MASK 0xFFFF /* EQ_B3_A - [15:0] */ +#define WM9081_EQ_B3_A_SHIFT 0 /* EQ_B3_A - [15:0] */ +#define WM9081_EQ_B3_A_WIDTH 16 /* EQ_B3_A - [15:0] */ + +/* + * R56 (0x38) - EQ 15 + */ +#define WM9081_EQ_B3_B_MASK 0xFFFF /* EQ_B3_B - [15:0] */ +#define WM9081_EQ_B3_B_SHIFT 0 /* EQ_B3_B - [15:0] */ +#define WM9081_EQ_B3_B_WIDTH 16 /* EQ_B3_B - [15:0] */ + +/* + * R57 (0x39) - EQ 16 + */ +#define WM9081_EQ_B3_C_MASK 0xFFFF /* EQ_B3_C - [15:0] */ +#define WM9081_EQ_B3_C_SHIFT 0 /* EQ_B3_C - [15:0] */ +#define WM9081_EQ_B3_C_WIDTH 16 /* EQ_B3_C - [15:0] */ + +/* + * R58 (0x3A) - EQ 17 + */ +#define WM9081_EQ_B3_PG_MASK 0xFFFF /* EQ_B3_PG - [15:0] */ +#define WM9081_EQ_B3_PG_SHIFT 0 /* EQ_B3_PG - [15:0] */ +#define WM9081_EQ_B3_PG_WIDTH 16 /* EQ_B3_PG - [15:0] */ + +/* + * R59 (0x3B) - EQ 18 + */ +#define WM9081_EQ_B5_A_MASK 0xFFFF /* EQ_B5_A - [15:0] */ +#define WM9081_EQ_B5_A_SHIFT 0 /* EQ_B5_A - [15:0] */ +#define WM9081_EQ_B5_A_WIDTH 16 /* EQ_B5_A - [15:0] */ + +/* + * R60 (0x3C) - EQ 19 + */ +#define WM9081_EQ_B5_B_MASK 0xFFFF /* EQ_B5_B - [15:0] */ +#define WM9081_EQ_B5_B_SHIFT 0 /* EQ_B5_B - [15:0] */ +#define WM9081_EQ_B5_B_WIDTH 16 /* EQ_B5_B - [15:0] */ + +/* + * R61 (0x3D) - EQ 20 + */ +#define WM9081_EQ_B5_PG_MASK 0xFFFF /* EQ_B5_PG - [15:0] */ +#define WM9081_EQ_B5_PG_SHIFT 0 /* EQ_B5_PG - [15:0] */ +#define WM9081_EQ_B5_PG_WIDTH 16 /* EQ_B5_PG - [15:0] */ + + +#endif -- cgit v1.2.3 From 0154724d487586241c1ad57cfd348ed2ff2274e2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 23 May 2009 00:01:05 +0100 Subject: ASoC: Fix WM9081 PowerPC compiler issues Ensure that we always set a new sysclk when using the FLL in master mode and pick out the correct value for the sample rate in hw_params(). Signed-off-by: Mark Brown --- sound/soc/codecs/wm9081.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 83e3148c258b..86fc57e25f97 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -702,9 +702,10 @@ static int configure_clock(struct snd_soc_codec *codec) * performance. */ for (i = 0; i < ARRAY_SIZE(clk_sys_rates); i++) { target = wm9081->fs * clk_sys_rates[i].ratio; + new_sysclk = target; if (target >= wm9081->bclk && target > 3000000) - new_sysclk = target; + break; } } else if (wm9081->fs) { for (i = 0; i < ARRAY_SIZE(clk_sys_rates); i++) { @@ -1102,7 +1103,8 @@ static int wm9081_hw_params(struct snd_pcm_substream *substream, } dev_dbg(codec->dev, "Selected SAMPLE_RATE of %dHz\n", sample_rates[best].rate); - clk_ctrl2 |= (sample_rates[i].sample_rate << WM9081_SAMPLE_RATE_SHIFT); + clk_ctrl2 |= (sample_rates[best].sample_rate + << WM9081_SAMPLE_RATE_SHIFT); /* BCLK_DIV */ best = 0; -- cgit v1.2.3 From 3c166c7f1828f226c7f478758bf6c8ce8be1623f Mon Sep 17 00:00:00 2001 From: Jon Smirl Date: Sat, 23 May 2009 19:13:07 -0400 Subject: ASoC: Codec for STAC9766 used on the Efika Datasheet: http://www.idt.com/products/getDoc.cfm?docID=13134007 Signed-off-by: Jon Smirl Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/stac9766.c | 470 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/stac9766.h | 21 ++ 4 files changed, 497 insertions(+) create mode 100644 sound/soc/codecs/stac9766.c create mode 100644 sound/soc/codecs/stac9766.h (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7f78b65fc4e3..cb07d9b51b61 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -19,6 +19,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4270 if I2C select SND_SOC_PCM3008 select SND_SOC_SSM2602 if I2C + select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C @@ -93,6 +94,9 @@ config SND_SOC_PCM3008 config SND_SOC_SSM2602 tristate +config SND_SOC_STAC9766 + tristate + config SND_SOC_TLV320AIC23 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70c55fa2c436..46c007cb5625 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -7,6 +7,7 @@ snd-soc-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o @@ -42,6 +43,7 @@ obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c new file mode 100644 index 000000000000..7740cd5a7604 --- /dev/null +++ b/sound/soc/codecs/stac9766.c @@ -0,0 +1,470 @@ +/* + * stac9766.c -- ALSA SoC STAC9766 codec support + * + * Copyright 2009 Jon Smirl, Digispeaker + * Author: Jon Smirl + * + * 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. + * + * Features:- + * + * o Support for AC97 Codec, S/PDIF + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "stac9766.h" + +#define STAC9766_VERSION "0.10" + +/* + * STAC9766 register cache + */ +static const u16 stac9766_reg[] = { + 0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */ + 0x0000, 0x0000, 0x8008, 0x8008, /* e */ + 0x8808, 0x8808, 0x8808, 0x8808, /* 16 */ + 0x8808, 0x0000, 0x8000, 0x0000, /* 1e */ + 0x0000, 0x0000, 0x0000, 0x000f, /* 26 */ + 0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */ + 0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */ + 0x0000, 0x2000, 0x0000, 0x0100, /* 3e */ + 0x0000, 0x0000, 0x0080, 0x0000, /* 46 */ + 0x0000, 0x0000, 0x0003, 0xffff, /* 4e */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */ + 0x4000, 0x0000, 0x0000, 0x0000, /* 5e */ + 0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */ + 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 7e */ +}; + +static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX", "Line", "Stereo Mix", "Mono Mix", "Phone"}; +static const char *stac9766_mono_mux[] = {"Mix", "Mic"}; +static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"}; +static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"}; +static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"}; +static const char *stac9766_record_all_mux[] = {"All analog", "Analog plus DAC"}; +static const char *stac9766_boost1[] = {"0dB", "10dB"}; +static const char *stac9766_boost2[] = {"0dB", "20dB"}; +static const char *stac9766_stereo_mic[] = {"Off", "On"}; + +static const struct soc_enum stac9766_record_enum = + SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux); +static const struct soc_enum stac9766_mono_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux); +static const struct soc_enum stac9766_mic_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux); +static const struct soc_enum stac9766_SPDIF_enum = + SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux); +static const struct soc_enum stac9766_popbypass_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux); +static const struct soc_enum stac9766_record_all_enum = + SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2, stac9766_record_all_mux); +static const struct soc_enum stac9766_boost1_enum = + SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */ +static const struct soc_enum stac9766_boost2_enum = + SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */ +static const struct soc_enum stac9766_stereo_mic_enum = + SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, stac9766_stereo_mic); + +static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0); +static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250); +static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0); +static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200); + +static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = { + SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv), + SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1), + SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1, master_tlv), + SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1), + SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1, master_tlv), + SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1), + + SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv), + SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1), + + + SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv), + SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1), + SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1), + SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv), + SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1), + + SOC_ENUM("Mic Boost1", stac9766_boost1_enum), + SOC_ENUM("Mic Boost2", stac9766_boost2_enum), + SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv), + SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1), + SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum), + + SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1), + SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1), + SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1), + SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1), + + SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1), + SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0), + SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1), + SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0), + + SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum), + SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum), + SOC_ENUM("Record All Mux", stac9766_record_all_enum), + SOC_ENUM("Record Mux", stac9766_record_enum), + SOC_ENUM("Mono Mux", stac9766_mono_enum), + SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum), +}; + +int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + u16 *cache = codec->reg_cache; + + if (reg > AC97_STAC_PAGE0) { + stac9766_ac97_write(codec, AC97_INT_PAGING, 0); + soc_ac97_ops.write(codec->ac97, reg, val); + stac9766_ac97_write(codec, AC97_INT_PAGING, 1); + return 0; + } + if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + return -EIO; + + soc_ac97_ops.write(codec->ac97, reg, val); + cache[reg / 2] = val; + return 0; +} + +unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, unsigned int reg) +{ + u16 val = 0, *cache = codec->reg_cache; + + if (reg > AC97_STAC_PAGE0) { + stac9766_ac97_write(codec, AC97_INT_PAGING, 0); + val = soc_ac97_ops.read(codec->ac97, reg - AC97_STAC_PAGE0); + stac9766_ac97_write(codec, AC97_INT_PAGING, 1); + return val; + } + if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + return -EIO; + + if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || + reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 || + reg == AC97_VENDOR_ID2) { + + val = soc_ac97_ops.read(codec->ac97, reg); + return val; + } + return cache[reg / 2]; +} + +static int ac97_analog_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned short reg, vra; + + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + + vra |= 0x1; /* enable variable rate audio */ + + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = AC97_PCM_FRONT_DAC_RATE; + else + reg = AC97_PCM_LR_ADC_RATE; + + return stac9766_ac97_write(codec, reg, runtime->rate); +} + +static int ac97_digital_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned short reg, vra; + + stac9766_ac97_write(codec, AC97_SPDIF, 0x2002); + + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + vra |= 0x5; /* Enable VRA and SPDIF out */ + + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + + reg = AC97_PCM_FRONT_DAC_RATE; + + return stac9766_ac97_write(codec, reg, runtime->rate); +} + +static int ac97_digital_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned short vra; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + vra &= !0x04; + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + break; + } + return 0; +} + +static int stac9766_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: /* full On */ + case SND_SOC_BIAS_PREPARE: /* partial On */ + case SND_SOC_BIAS_STANDBY: /* Off, with power */ + stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000); + break; + case SND_SOC_BIAS_OFF: /* Off, without power */ + /* disable everything including AC link */ + stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff); + break; + } + codec->bias_level = level; + return 0; +} + +int stac9766_reset(struct snd_soc_codec *codec, int try_warm) +{ + if (try_warm && soc_ac97_ops.warm_reset) { + soc_ac97_ops.warm_reset(codec->ac97); + if (stac9766_ac97_read(codec, 0) == stac9766_reg[0]) + return 1; + } + + soc_ac97_ops.reset(codec->ac97); + if (soc_ac97_ops.warm_reset) + soc_ac97_ops.warm_reset(codec->ac97); + if (stac9766_ac97_read(codec, 0) != stac9766_reg[0]) + return -EIO; + return 0; +} + +static int stac9766_codec_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int stac9766_codec_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + u16 id, reset; + + reset = 0; + /* give the codec an AC97 warm reset to start the link */ +reset: + if (reset > 5) { + printk(KERN_ERR "stac9766 failed to resume"); + return -EIO; + } + codec->ac97->bus->ops->warm_reset(codec->ac97); + id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2); + if (id != 0x4c13) { + stac9766_reset(codec, 0); + reset++; + goto reset; + } + stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) + stac9766_set_bias_level(codec, SND_SOC_BIAS_ON); + + return 0; +} + +static struct snd_soc_dai_ops stac9766_dai_ops_analog = +{ + .prepare = ac97_analog_prepare, +}; + +static struct snd_soc_dai_ops stac9766_dai_ops_digital = +{ + .prepare = ac97_digital_prepare, + .trigger = ac97_digital_trigger, +}; + +struct snd_soc_dai stac9766_dai[] = { +{ + .name = "stac9766 analog", + .id = 0, + .ac97_control = 1, + + /* stream cababilities */ + .playback = { + .stream_name = "stac9766 analog", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SND_SOC_STD_AC97_FMTS, + }, + .capture = { + .stream_name = "stac9766 analog", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SND_SOC_STD_AC97_FMTS, + }, + /* alsa ops */ + .ops = &stac9766_dai_ops_analog, +}, +{ + .name = "stac9766 IEC958", + .id = 1, + .ac97_control = 1, + + /* stream cababilities */ + .playback = { + .stream_name = "stac9766 IEC958", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE, + }, + /* alsa ops */ + .ops = &stac9766_dai_ops_digital, +}}; +EXPORT_SYMBOL_GPL(stac9766_dai); + +static int stac9766_codec_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION); + + socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->card->codec == NULL) + return -ENOMEM; + codec = socdev->card->codec; + mutex_init(&codec->mutex); + + codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + codec->reg_cache_size = sizeof(stac9766_reg); + codec->reg_cache_step = 2; + + codec->name = "STAC9766"; + codec->owner = THIS_MODULE; + codec->dai = stac9766_dai; + codec->num_dai = ARRAY_SIZE(stac9766_dai); + codec->write = stac9766_ac97_write; + codec->read = stac9766_ac97_read; + codec->set_bias_level = stac9766_set_bias_level; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + if (ret < 0) + goto codec_err; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) + goto pcm_err; + + /* do a cold reset for the controller and then try + * a warm reset followed by an optional cold reset for codec */ + stac9766_reset(codec, 0); + ret = stac9766_reset(codec, 1); + if (ret < 0) { + printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n"); + goto reset_err; + } + + stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + snd_soc_add_controls(codec, stac9766_snd_ac97_controls, ARRAY_SIZE( + stac9766_snd_ac97_controls)); + + ret = snd_soc_init_card(socdev); + if (ret < 0) + goto reset_err; + return 0; + +reset_err: + snd_soc_free_pcms(socdev); +pcm_err: + snd_soc_free_ac97_codec(codec); +codec_err: + kfree(codec->private_data); +cache_err: + kfree(socdev->card->codec); + socdev->card->codec = NULL; + return ret; +} + +static int stac9766_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + if (codec == NULL) + return 0; + + snd_soc_free_pcms(socdev); + snd_soc_free_ac97_codec(codec); + kfree(codec->reg_cache); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_stac9766 = +{ + .probe = stac9766_codec_probe, + .remove = stac9766_codec_remove, + .suspend = stac9766_codec_suspend, + .resume = stac9766_codec_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_stac9766); + +static int __init stac9766_modinit(void) +{ + return snd_soc_register_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); +} +module_init(stac9766_modinit); + +static void __exit stac9766_exit(void) +{ + snd_soc_unregister_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); +} +module_exit(stac9766_exit); + +MODULE_DESCRIPTION("ASoC stac9766 driver"); +MODULE_AUTHOR("Jon Smirl "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/stac9766.h b/sound/soc/codecs/stac9766.h new file mode 100644 index 000000000000..65642eb8393e --- /dev/null +++ b/sound/soc/codecs/stac9766.h @@ -0,0 +1,21 @@ +/* + * stac9766.h -- STAC9766 Soc Audio driver + */ + +#ifndef _STAC9766_H +#define _STAC9766_H + +#define AC97_STAC_PAGE0 0x1000 +#define AC97_STAC_DA_CONTROL (AC97_STAC_PAGE0 | 0x6A) +#define AC97_STAC_ANALOG_SPECIAL (AC97_STAC_PAGE0 | 0x6E) +#define AC97_STAC_STEREO_MIC 0x78 + +/* STAC9766 DAI ID's */ +#define STAC9766_DAI_AC97_ANALOG 0 +#define STAC9766_DAI_AC97_DIGITAL 1 + +extern struct snd_soc_dai stac9766_dai[]; +extern struct snd_soc_codec_device soc_codec_dev_stac9766; + + +#endif -- cgit v1.2.3 From 05e1efa2deb42b1bd548208e5c43f471e2cf0da1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 24 May 2009 13:32:24 +0100 Subject: ASoC: Fix minor issues in STAC9766 driver Fairly minor issues: - Don't register the DAIs, it's not required for AC97 devices. - Make unexported functions static. - Wrap some excessively long lines. - Undo tab/space breakage. Signed-off-by: Mark Brown --- sound/soc/codecs/stac9766.c | 65 ++++++++++++++++++++------------------------- 1 file changed, 29 insertions(+), 36 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 7740cd5a7604..8ad4b7b3e3ba 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -52,12 +52,14 @@ static const u16 stac9766_reg[] = { 0x0000, 0x0000, 0x0000, 0x0000, /* 7e */ }; -static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX", "Line", "Stereo Mix", "Mono Mix", "Phone"}; +static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX", + "Line", "Stereo Mix", "Mono Mix", "Phone"}; static const char *stac9766_mono_mux[] = {"Mix", "Mic"}; static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"}; static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"}; static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"}; -static const char *stac9766_record_all_mux[] = {"All analog", "Analog plus DAC"}; +static const char *stac9766_record_all_mux[] = {"All analog", + "Analog plus DAC"}; static const char *stac9766_boost1[] = {"0dB", "10dB"}; static const char *stac9766_boost2[] = {"0dB", "20dB"}; static const char *stac9766_stereo_mic[] = {"Off", "On"}; @@ -73,7 +75,8 @@ static const struct soc_enum stac9766_SPDIF_enum = static const struct soc_enum stac9766_popbypass_enum = SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux); static const struct soc_enum stac9766_record_all_enum = - SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2, stac9766_record_all_mux); + SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2, + stac9766_record_all_mux); static const struct soc_enum stac9766_boost1_enum = SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */ static const struct soc_enum stac9766_boost2_enum = @@ -89,9 +92,11 @@ static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200); static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = { SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv), SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1), - SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1, master_tlv), + SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1, + master_tlv), SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1), - SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1, master_tlv), + SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1, + master_tlv), SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1), SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv), @@ -133,8 +138,8 @@ static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = { SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum), }; -int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int val) +static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) { u16 *cache = codec->reg_cache; @@ -152,7 +157,8 @@ int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, return 0; } -unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, unsigned int reg) +static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, + unsigned int reg) { u16 val = 0, *cache = codec->reg_cache; @@ -176,7 +182,7 @@ unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, unsigned int reg) } static int ac97_analog_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) + struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct snd_pcm_runtime *runtime = substream->runtime; @@ -197,7 +203,7 @@ static int ac97_analog_prepare(struct snd_pcm_substream *substream, } static int ac97_digital_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) + struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct snd_pcm_runtime *runtime = substream->runtime; @@ -216,7 +222,7 @@ static int ac97_digital_prepare(struct snd_pcm_substream *substream, } static int ac97_digital_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) + int cmd, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; unsigned short vra; @@ -232,7 +238,7 @@ static int ac97_digital_trigger(struct snd_pcm_substream *substream, } static int stac9766_set_bias_level(struct snd_soc_codec *codec, - enum snd_soc_bias_level level) + enum snd_soc_bias_level level) { switch (level) { case SND_SOC_BIAS_ON: /* full On */ @@ -249,7 +255,7 @@ static int stac9766_set_bias_level(struct snd_soc_codec *codec, return 0; } -int stac9766_reset(struct snd_soc_codec *codec, int try_warm) +static int stac9766_reset(struct snd_soc_codec *codec, int try_warm) { if (try_warm && soc_ac97_ops.warm_reset) { soc_ac97_ops.warm_reset(codec->ac97); @@ -266,7 +272,7 @@ int stac9766_reset(struct snd_soc_codec *codec, int try_warm) } static int stac9766_codec_suspend(struct platform_device *pdev, - pm_message_t state) + pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; @@ -303,13 +309,11 @@ reset: return 0; } -static struct snd_soc_dai_ops stac9766_dai_ops_analog = -{ +static struct snd_soc_dai_ops stac9766_dai_ops_analog = { .prepare = ac97_analog_prepare, }; -static struct snd_soc_dai_ops stac9766_dai_ops_digital = -{ +static struct snd_soc_dai_ops stac9766_dai_ops_digital = { .prepare = ac97_digital_prepare, .trigger = ac97_digital_trigger, }; @@ -354,7 +358,8 @@ struct snd_soc_dai stac9766_dai[] = { }, /* alsa ops */ .ops = &stac9766_dai_ops_digital, -}}; +} +}; EXPORT_SYMBOL_GPL(stac9766_dai); static int stac9766_codec_probe(struct platform_device *pdev) @@ -371,7 +376,8 @@ static int stac9766_codec_probe(struct platform_device *pdev) codec = socdev->card->codec; mutex_init(&codec->mutex); - codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg), GFP_KERNEL); + codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg), + GFP_KERNEL); if (codec->reg_cache == NULL) { ret = -ENOMEM; goto cache_err; @@ -409,8 +415,8 @@ static int stac9766_codec_probe(struct platform_device *pdev) stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_controls(codec, stac9766_snd_ac97_controls, ARRAY_SIZE( - stac9766_snd_ac97_controls)); + snd_soc_add_controls(codec, stac9766_snd_ac97_controls, + ARRAY_SIZE(stac9766_snd_ac97_controls)); ret = snd_soc_init_card(socdev); if (ret < 0) @@ -444,8 +450,7 @@ static int stac9766_codec_remove(struct platform_device *pdev) return 0; } -struct snd_soc_codec_device soc_codec_dev_stac9766 = -{ +struct snd_soc_codec_device soc_codec_dev_stac9766 = { .probe = stac9766_codec_probe, .remove = stac9766_codec_remove, .suspend = stac9766_codec_suspend, @@ -453,18 +458,6 @@ struct snd_soc_codec_device soc_codec_dev_stac9766 = }; EXPORT_SYMBOL_GPL(soc_codec_dev_stac9766); -static int __init stac9766_modinit(void) -{ - return snd_soc_register_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); -} -module_init(stac9766_modinit); - -static void __exit stac9766_exit(void) -{ - snd_soc_unregister_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); -} -module_exit(stac9766_exit); - MODULE_DESCRIPTION("ASoC stac9766 driver"); MODULE_AUTHOR("Jon Smirl "); MODULE_LICENSE("GPL"); -- cgit v1.2.3 From 5a2e9a48b1d6de35ae5efea35d117133c3eb30f2 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 25 May 2009 11:12:11 +0300 Subject: ASoC: TWL4030: Handsfree pop removal redesign Move the HandsfreeL/R (IHFL/R) pop removal code from the DAPM_MUX_E to a more appropriate DAPM_PGA_E widget. Also fix the power-up sequence to match with the TRM. The power-down sequence is not described in the TRM, so do it in a way, which seams like the correct sequence. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 78 +++++++++++++++++++++++++++++++++++----------- 1 file changed, 59 insertions(+), 19 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 9197fdd0a29d..17ddcb265134 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -546,27 +546,61 @@ static int micpath_event(struct snd_soc_dapm_widget *w, return 0; } -static int handsfree_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static void handsfree_ramp(struct snd_soc_codec *codec, int reg, int ramp) { - struct soc_enum *e = (struct soc_enum *)w->kcontrols->private_value; unsigned char hs_ctl; - hs_ctl = twl4030_read_reg_cache(w->codec, e->reg); + hs_ctl = twl4030_read_reg_cache(codec, reg); - if (hs_ctl & TWL4030_HF_CTL_REF_EN) { + if (ramp) { + /* HF ramp-up */ + hs_ctl |= TWL4030_HF_CTL_REF_EN; + twl4030_write(codec, reg, hs_ctl); + udelay(10); hs_ctl |= TWL4030_HF_CTL_RAMP_EN; - twl4030_write(w->codec, e->reg, hs_ctl); + twl4030_write(codec, reg, hs_ctl); + udelay(40); hs_ctl |= TWL4030_HF_CTL_LOOP_EN; - twl4030_write(w->codec, e->reg, hs_ctl); hs_ctl |= TWL4030_HF_CTL_HB_EN; - twl4030_write(w->codec, e->reg, hs_ctl); + twl4030_write(codec, reg, hs_ctl); } else { - hs_ctl &= ~(TWL4030_HF_CTL_RAMP_EN | TWL4030_HF_CTL_LOOP_EN - | TWL4030_HF_CTL_HB_EN); - twl4030_write(w->codec, e->reg, hs_ctl); + /* HF ramp-down */ + hs_ctl &= ~TWL4030_HF_CTL_LOOP_EN; + hs_ctl &= ~TWL4030_HF_CTL_HB_EN; + twl4030_write(codec, reg, hs_ctl); + hs_ctl &= ~TWL4030_HF_CTL_RAMP_EN; + twl4030_write(codec, reg, hs_ctl); + udelay(40); + hs_ctl &= ~TWL4030_HF_CTL_REF_EN; + twl4030_write(codec, reg, hs_ctl); } +} +static int handsfreelpga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + handsfree_ramp(w->codec, TWL4030_REG_HFL_CTL, 1); + break; + case SND_SOC_DAPM_POST_PMD: + handsfree_ramp(w->codec, TWL4030_REG_HFL_CTL, 0); + break; + } + return 0; +} + +static int handsfreerpga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + handsfree_ramp(w->codec, TWL4030_REG_HFR_CTL, 1); + break; + case SND_SOC_DAPM_POST_PMD: + handsfree_ramp(w->codec, TWL4030_REG_HFR_CTL, 0); + break; + } return 0; } @@ -1190,12 +1224,16 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { /* Output MUX controls */ /* HandsfreeL/R */ - SND_SOC_DAPM_MUX_E("HandsfreeL Mux", TWL4030_REG_HFL_CTL, 5, 0, - &twl4030_dapm_handsfreel_control, handsfree_event, - SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_MUX_E("HandsfreeR Mux", TWL4030_REG_HFR_CTL, 5, 0, - &twl4030_dapm_handsfreer_control, handsfree_event, - SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX("HandsfreeL Mux", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_handsfreel_control), + SND_SOC_DAPM_PGA_E("HandsfreeL PGA", SND_SOC_NOPM, + 0, 0, NULL, 0, handsfreelpga_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX("HandsfreeR Mux", SND_SOC_NOPM, 5, 0, + &twl4030_dapm_handsfreer_control), + SND_SOC_DAPM_PGA_E("HandsfreeR PGA", SND_SOC_NOPM, + 0, 0, NULL, 0, handsfreerpga_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), /* Vibra */ SND_SOC_DAPM_MUX("Vibra Mux", TWL4030_REG_VIBRA_CTL, 0, 0, &twl4030_dapm_vibra_control), @@ -1303,11 +1341,13 @@ static const struct snd_soc_dapm_route intercon[] = { {"HandsfreeL Mux", "AudioL1", "Analog L1 Playback Mixer"}, {"HandsfreeL Mux", "AudioL2", "Analog L2 Playback Mixer"}, {"HandsfreeL Mux", "AudioR2", "Analog R2 Playback Mixer"}, + {"HandsfreeL PGA", NULL, "HandsfreeL Mux"}, /* HandsfreeR */ {"HandsfreeR Mux", "Voice", "Analog Voice Playback Mixer"}, {"HandsfreeR Mux", "AudioR1", "Analog R1 Playback Mixer"}, {"HandsfreeR Mux", "AudioR2", "Analog R2 Playback Mixer"}, {"HandsfreeR Mux", "AudioL2", "Analog L2 Playback Mixer"}, + {"HandsfreeR PGA", NULL, "HandsfreeR Mux"}, /* Vibra */ {"Vibra Mux", "AudioL1", "DAC Left1"}, {"Vibra Mux", "AudioR1", "DAC Right1"}, @@ -1324,8 +1364,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"HSOR", NULL, "HeadsetR PGA"}, {"CARKITL", NULL, "CarkitL Mixer"}, {"CARKITR", NULL, "CarkitR Mixer"}, - {"HFL", NULL, "HandsfreeL Mux"}, - {"HFR", NULL, "HandsfreeR Mux"}, + {"HFL", NULL, "HandsfreeL PGA"}, + {"HFR", NULL, "HandsfreeR PGA"}, {"Vibra Route", "Audio", "Vibra Mux"}, {"VIBRA", NULL, "Vibra Route"}, -- cgit v1.2.3 From f3b5d3002d5b43d277dedc1e044d02f2a40a43c5 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 25 May 2009 11:12:12 +0300 Subject: ASoC: TWL4030: Add shadow register Shadow, non HW register for dealing with the HandsfreeL/R muting. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 7 ++++++- sound/soc/codecs/twl4030.h | 7 ++++++- 2 files changed, 12 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 17ddcb265134..989446dabcda 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -115,6 +115,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_VIBRA_PWM_SET (0x47) */ 0x00, /* REG_ANAMIC_GAIN (0x48) */ 0x00, /* REG_MISC_SET_2 (0x49) */ + 0x00, /* REG_SW_SHADOW (0x4A) - Shadow, non HW register */ }; /* codec private data */ @@ -172,7 +173,11 @@ static int twl4030_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { twl4030_write_reg_cache(codec, reg, value); - return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg); + if (likely(reg < TWL4030_REG_SW_SHADOW)) + return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, + reg); + else + return 0; } static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable) diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 48326e2bd9de..fe5f395d9e4f 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -92,8 +92,9 @@ #define TWL4030_REG_VIBRA_PWM_SET 0x47 #define TWL4030_REG_ANAMIC_GAIN 0x48 #define TWL4030_REG_MISC_SET_2 0x49 +#define TWL4030_REG_SW_SHADOW 0x4A -#define TWL4030_CACHEREGNUM (TWL4030_REG_MISC_SET_2 + 1) +#define TWL4030_CACHEREGNUM (TWL4030_REG_SW_SHADOW + 1) /* Bitfield Definitions */ @@ -260,6 +261,10 @@ #define TWL4030_SMOOTH_ANAVOL_EN 0x02 #define TWL4030_DIGMIC_LR_SWAP_EN 0x01 +/* TWL4030_REG_SW_SHADOW (0x4A) Fields */ +#define TWL4030_HFL_EN 0x01 +#define TWL4030_HFR_EN 0x02 + #define TWL4030_DAI_HIFI 0 #define TWL4030_DAI_VOICE 1 -- cgit v1.2.3 From 0f89bdcac61536c5cb2a095a514657019573afb4 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 25 May 2009 11:12:13 +0300 Subject: ASoC: TWL4030: HandsfreeL/R mute DAPM switch Add DAPM switch for HeadsetL/R mute. Since all bits are are needed for the HFL/R pop removal to work the switch is using the SW_SHADOW no HW register for the HandsfreeL/R mute. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 18 ++++++++++++++++-- 1 file changed, 16 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 989446dabcda..63ebd176fbe4 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -395,6 +395,10 @@ static const struct soc_enum twl4030_handsfreel_enum = static const struct snd_kcontrol_new twl4030_dapm_handsfreel_control = SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum); +/* Handsfree Left virtual mute */ +static const struct snd_kcontrol_new twl4030_dapm_handsfreelmute_control = + SOC_DAPM_SINGLE("Switch", TWL4030_REG_SW_SHADOW, 0, 1, 0); + /* Handsfree Right */ static const char *twl4030_handsfreer_texts[] = {"Voice", "AudioR1", "AudioR2", "AudioL2"}; @@ -407,6 +411,10 @@ static const struct soc_enum twl4030_handsfreer_enum = static const struct snd_kcontrol_new twl4030_dapm_handsfreer_control = SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum); +/* Handsfree Right virtual mute */ +static const struct snd_kcontrol_new twl4030_dapm_handsfreermute_control = + SOC_DAPM_SINGLE("Switch", TWL4030_REG_SW_SHADOW, 1, 1, 0); + /* Vibra */ /* Vibra audio path selection */ static const char *twl4030_vibra_texts[] = @@ -1231,11 +1239,15 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { /* HandsfreeL/R */ SND_SOC_DAPM_MUX("HandsfreeL Mux", SND_SOC_NOPM, 0, 0, &twl4030_dapm_handsfreel_control), + SND_SOC_DAPM_SWITCH("HandsfreeL Switch", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_handsfreelmute_control), SND_SOC_DAPM_PGA_E("HandsfreeL PGA", SND_SOC_NOPM, 0, 0, NULL, 0, handsfreelpga_event, SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MUX("HandsfreeR Mux", SND_SOC_NOPM, 5, 0, &twl4030_dapm_handsfreer_control), + SND_SOC_DAPM_SWITCH("HandsfreeR Switch", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_handsfreermute_control), SND_SOC_DAPM_PGA_E("HandsfreeR PGA", SND_SOC_NOPM, 0, 0, NULL, 0, handsfreerpga_event, SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), @@ -1346,13 +1358,15 @@ static const struct snd_soc_dapm_route intercon[] = { {"HandsfreeL Mux", "AudioL1", "Analog L1 Playback Mixer"}, {"HandsfreeL Mux", "AudioL2", "Analog L2 Playback Mixer"}, {"HandsfreeL Mux", "AudioR2", "Analog R2 Playback Mixer"}, - {"HandsfreeL PGA", NULL, "HandsfreeL Mux"}, + {"HandsfreeL Switch", "Switch", "HandsfreeL Mux"}, + {"HandsfreeL PGA", NULL, "HandsfreeL Switch"}, /* HandsfreeR */ {"HandsfreeR Mux", "Voice", "Analog Voice Playback Mixer"}, {"HandsfreeR Mux", "AudioR1", "Analog R1 Playback Mixer"}, {"HandsfreeR Mux", "AudioR2", "Analog R2 Playback Mixer"}, {"HandsfreeR Mux", "AudioL2", "Analog L2 Playback Mixer"}, - {"HandsfreeR PGA", NULL, "HandsfreeR Mux"}, + {"HandsfreeR Switch", "Switch", "HandsfreeR Mux"}, + {"HandsfreeR PGA", NULL, "HandsfreeR Switch"}, /* Vibra */ {"Vibra Mux", "AudioL1", "DAC Left1"}, {"Vibra Mux", "AudioR1", "DAC Right1"}, -- cgit v1.2.3 From 449bd54dcbd0b60070ce4129fedaf0f4ae044099 Mon Sep 17 00:00:00 2001 From: Roel Kluin Date: Wed, 27 May 2009 17:08:39 -0700 Subject: ASoC: correct print specifiers for unsigneds Unsigned variables should use `%u' rather than `%d'. Signed-off-by: Roel Kluin Signed-off-by: Andrew Morton Signed-off-by: Mark Brown --- sound/soc/atmel/playpaq_wm8510.c | 2 +- sound/soc/codecs/tlv320aic23.c | 4 ++-- sound/soc/codecs/uda134x.c | 4 ++-- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8400.c | 4 ++-- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8580.c | 4 ++-- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8900.c | 6 +++--- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm9713.c | 2 +- sound/soc/pxa/pxa-ssp.c | 4 ++-- sound/soc/s3c24xx/s3c-i2s-v2.c | 4 ++-- sound/soc/sh/ssi.c | 2 +- 14 files changed, 22 insertions(+), 22 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c index 70657534e6b1..9eb610c2ba91 100644 --- a/sound/soc/atmel/playpaq_wm8510.c +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -117,7 +117,7 @@ static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( * Find actual rate, compare to requested rate */ actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1)); - pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n", + pr_debug("playpaq_wm8510: Request rate = %u, actual rate = %u\n", rate, actual_rate); diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 21f69df9994c..9fcbb9c7766b 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -86,7 +86,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, */ if ((reg < 0 || reg > 9) && (reg != 15)) { - printk(KERN_WARNING "%s Invalid register R%d\n", __func__, reg); + printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg); return -1; } @@ -98,7 +98,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, if (codec->hw_write(codec->control_data, data, 2) == 2) return 0; - printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__, + printk(KERN_ERR "%s cannot write %03x to register R%u\n", __func__, value, reg); return -EIO; diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index ddefb8f80145..269b108e1de6 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -101,7 +101,7 @@ static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg, pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value); if (reg >= UDA134X_REGS_NUM) { - printk(KERN_ERR "%s unkown register: reg: %d", + printk(KERN_ERR "%s unkown register: reg: %u", __func__, reg); return -EINVAL; } @@ -296,7 +296,7 @@ static int uda134x_set_dai_sysclk(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; struct uda134x_priv *uda134x = codec->private_data; - pr_debug("%s clk_id: %d, freq: %d, dir: %d\n", __func__, + pr_debug("%s clk_id: %d, freq: %u, dir: %d\n", __func__, clk_id, freq, dir); /* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 0275321ff8ab..e7348d341b76 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1108,7 +1108,7 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai, if (ret < 0) return ret; dev_dbg(wm8350->dev, - "FLL in %d FLL out %d N 0x%x K 0x%x div %d ratio %d", + "FLL in %u FLL out %u N 0x%x K 0x%x div %d ratio %d", freq_in, freq_out, fll_div.n, fll_div.k, fll_div.div, fll_div.ratio); diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index e4547de8eec2..502eefac1ecd 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -954,7 +954,7 @@ static int fll_factors(struct wm8400_priv *wm8400, struct fll_factors *factors, factors->outdiv *= 2; if (factors->outdiv > 32) { dev_err(wm8400->wm8400->dev, - "Unsupported FLL output frequency %dHz\n", + "Unsupported FLL output frequency %uHz\n", Fout); return -EINVAL; } @@ -1003,7 +1003,7 @@ static int fll_factors(struct wm8400_priv *wm8400, struct fll_factors *factors, factors->k = K / 10; dev_dbg(wm8400->wm8400->dev, - "FLL: Fref=%d Fout=%d N=%x K=%x, FRATIO=%x OUTDIV=%x\n", + "FLL: Fref=%u Fout=%u N=%x K=%x, FRATIO=%x OUTDIV=%x\n", Fref, Fout, factors->n, factors->k, factors->fratio, factors->outdiv); diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 6a4cea09c45d..c8b8dba85890 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -298,7 +298,7 @@ static void pll_factors(unsigned int target, unsigned int source) if ((Ndiv < 6) || (Ndiv > 12)) printk(KERN_WARNING - "WM8510 N value %d outwith recommended range!d\n", + "WM8510 N value %u outwith recommended range!d\n", Ndiv); pll_div.n = Ndiv; diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 9f6be3d31ac0..86c4b24db817 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -415,7 +415,7 @@ static int pll_factors(struct _pll_div *pll_div, unsigned int target, unsigned int K, Ndiv, Nmod; int i; - pr_debug("wm8580: PLL %dHz->%dHz\n", source, target); + pr_debug("wm8580: PLL %uHz->%uHz\n", source, target); /* Scale the output frequency up; the PLL should run in the * region of 90-100MHz. @@ -447,7 +447,7 @@ static int pll_factors(struct _pll_div *pll_div, unsigned int target, if ((Ndiv < 5) || (Ndiv > 13)) { printk(KERN_ERR - "WM8580 N=%d outside supported range\n", Ndiv); + "WM8580 N=%u outside supported range\n", Ndiv); return -EINVAL; } diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index d121e58cae2b..d28eeaceb857 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -703,7 +703,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, if ((Ndiv < 6) || (Ndiv > 12)) printk(KERN_WARNING - "wm8753: unsupported N = %d\n", Ndiv); + "wm8753: unsupported N = %u\n", Ndiv); pll_div->n = Ndiv; Nmod = target % source; diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 46c5ea1ff921..3c78945244b8 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -778,11 +778,11 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, } if (target > 100000000) - printk(KERN_WARNING "wm8900: FLL rate %d out of range, Fref=%d" - " Fout=%d\n", target, Fref, Fout); + printk(KERN_WARNING "wm8900: FLL rate %u out of range, Fref=%u" + " Fout=%u\n", target, Fref, Fout); if (div > 32) { printk(KERN_ERR "wm8900: Invalid FLL division rate %u, " - "Fref=%d, Fout=%d, target=%d\n", + "Fref=%u, Fout=%u, target=%u\n", div, Fref, Fout, target); return -EINVAL; } diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 40cd274eb1ef..d029818350e9 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -998,7 +998,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, if ((Ndiv < 6) || (Ndiv > 12)) printk(KERN_WARNING - "WM8990 N value outwith recommended range! N = %d\n", Ndiv); + "WM8990 N value outwith recommended range! N = %u\n", Ndiv); pll_div->n = Ndiv; Nmod = target % source; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index d1744e96f303..abed37acf787 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -710,7 +710,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int source) Ndiv = target / source; if ((Ndiv < 5) || (Ndiv > 12)) printk(KERN_WARNING - "WM9713 PLL N value %d out of recommended range!\n", + "WM9713 PLL N value %u out of recommended range!\n", Ndiv); pll_div->n = Ndiv; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 6fc787610ad7..19c45409d94c 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -208,7 +208,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); dev_dbg(&ssp->pdev->dev, - "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %d\n", + "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n", cpu_dai->id, clk_id, freq); switch (clk_id) { @@ -357,7 +357,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, ssacd |= (0x6 << 4); dev_dbg(&ssp->pdev->dev, - "Using SSACDD %x to supply %dHz\n", + "Using SSACDD %x to supply %uHz\n", val, freq_out); break; } diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 972c27684198..1a283170ca92 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -547,7 +547,7 @@ int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, actual = clkrate / (fsdiv * div); deviation = actual - rate; - printk(KERN_DEBUG "%dfs: div %d => result %d, deviation %d\n", + printk(KERN_DEBUG "%ufs: div %u => result %u, deviation %d\n", fsdiv, div, actual, deviation); deviation = abs(deviation); @@ -563,7 +563,7 @@ int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, break; } - printk(KERN_DEBUG "best: fs=%d, div=%d, rate=%d\n", + printk(KERN_DEBUG "best: fs=%u, div=%u, rate=%u\n", best_fs, best_div, best_rate); info->fs_div = best_fs; diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index 56fa0872abbb..b378096cadb1 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -145,7 +145,7 @@ static int ssi_hw_params(struct snd_pcm_substream *substream, recv = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 0 : 1; pr_debug("ssi_hw_params() enter\nssicr was %08lx\n", ssicr); - pr_debug("bits: %d channels: %d\n", bits, channels); + pr_debug("bits: %u channels: %u\n", bits, channels); ssicr &= ~(CR_TRMD | CR_CHNL_MASK | CR_DWL_MASK | CR_PDTA | CR_SWL_MASK); -- cgit v1.2.3 From be461ba836770263826457624bc4a5173a1f5040 Mon Sep 17 00:00:00 2001 From: Chaithrika U S Date: Thu, 28 May 2009 05:10:50 -0400 Subject: ASoC: Add dummy S/PDIF codec support McASP on DM646x can operate in DIT (S/PDIF) where no codec is needed. This patch provides stub codec that can be used in these configurations. On DM646x EVM the McASP1 is connected to the S/PDIF out. Signed-off-by: Steve Chen Signed-off-by: Pavel Kiryukhin Signed-off-by: Naresh Medisetty Signed-off-by: Chaithrika U S Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 +++ sound/soc/codecs/Makefile | 2 ++ sound/soc/codecs/spdif_transciever.c | 69 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/spdif_transciever.h | 17 +++++++++ 4 files changed, 92 insertions(+) create mode 100644 sound/soc/codecs/spdif_transciever.c create mode 100644 sound/soc/codecs/spdif_transciever.h (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cb07d9b51b61..bbc97fd76648 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -18,6 +18,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4535 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_PCM3008 + select SND_SOC_SPDIF select SND_SOC_SSM2602 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C @@ -91,6 +92,9 @@ config SND_SOC_L3 config SND_SOC_PCM3008 tristate +config SND_SOC_SPDIF + tristate + config SND_SOC_SSM2602 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 46c007cb5625..8b7530546f4d 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -6,6 +6,7 @@ snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o +snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o @@ -42,6 +43,7 @@ obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o +obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c new file mode 100644 index 000000000000..118e976b28ac --- /dev/null +++ b/sound/soc/codecs/spdif_transciever.c @@ -0,0 +1,69 @@ +/* + * ALSA SoC SPDIF DIT driver + * + * This driver is used by controllers which can operate in DIT (SPDI/F) where + * no codec is needed. This file provides stub codec that can be used + * in these configurations. TI DaVinci Audio controller uses this driver. + * + * Author: Steve Chen, + * Copyright: (C) 2009 MontaVista Software, Inc., + * Copyright: (C) 2009 Texas Instruments, India + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include + +#define STUB_RATES SNDRV_PCM_RATE_8000_96000 +#define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE + + +struct snd_soc_dai dit_stub_dai = { + .name = "DIT", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 384, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, +}; + +static int spdif_dit_probe(struct platform_device *pdev) +{ + return snd_soc_register_dai(&dit_stub_dai); +} + +static int spdif_dit_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dai(&dit_stub_dai); + return 0; +} + +static struct platform_driver spdif_dit_driver = { + .probe = spdif_dit_probe, + .remove = spdif_dit_remove, + .driver = { + .name = "spdif-dit", + .owner = THIS_MODULE, + }, +}; + +static int __init dit_modinit(void) +{ + return platform_driver_register(&spdif_dit_driver); +} + +static void __exit dit_exit(void) +{ + platform_driver_unregister(&spdif_dit_driver); +} + +module_init(dit_modinit); +module_exit(dit_exit); + diff --git a/sound/soc/codecs/spdif_transciever.h b/sound/soc/codecs/spdif_transciever.h new file mode 100644 index 000000000000..296f2eb6c4ef --- /dev/null +++ b/sound/soc/codecs/spdif_transciever.h @@ -0,0 +1,17 @@ +/* + * ALSA SoC DIT/DIR driver header + * + * Author: Steve Chen, + * Copyright: (C) 2008 MontaVista Software, Inc., + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef CODEC_STUBS_H +#define CODEC_STUBS_H + +extern struct snd_soc_dai dit_stub_dai; + +#endif /* CODEC_STUBS_H */ -- cgit v1.2.3 From 203350c1a8e23adf17fd9a96d8bfc7adf63c1ff6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 28 May 2009 14:51:00 +0100 Subject: ASoC: Initialise dev for the dummy S/PDIF DAI Also include the header to make sure the DAI is prototyped. Signed-off-by: Mark Brown --- sound/soc/codecs/spdif_transciever.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c index 118e976b28ac..218b33adad90 100644 --- a/sound/soc/codecs/spdif_transciever.c +++ b/sound/soc/codecs/spdif_transciever.c @@ -19,10 +19,11 @@ #include #include +#include "spdif_transciever.h" + #define STUB_RATES SNDRV_PCM_RATE_8000_96000 #define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE - struct snd_soc_dai dit_stub_dai = { .name = "DIT", .playback = { @@ -36,6 +37,7 @@ struct snd_soc_dai dit_stub_dai = { static int spdif_dit_probe(struct platform_device *pdev) { + dit_stub_dai.dev = &pdev->dev; return snd_soc_register_dai(&dit_stub_dai); } -- cgit v1.2.3 From 16a30fbb0d3aa4ee829a2dd3d0e314e2b5ae96a9 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 29 May 2009 09:22:37 +0300 Subject: ASoC: TWL4030: Use reg_cache in twl4030_init_chip Use the codec->reg_cache instead of the array directly in twl4030_init_chip for setting the default values. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 63ebd176fbe4..df474a5dd357 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -145,7 +145,6 @@ struct twl4030_priv { static inline unsigned int twl4030_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) { - u8 *cache = codec->reg_cache; if (reg >= TWL4030_CACHEREGNUM) return -EIO; @@ -204,6 +203,7 @@ static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable) static void twl4030_init_chip(struct snd_soc_codec *codec) { + u8 *cache = codec->reg_cache; int i; /* clear CODECPDZ prior to setting register defaults */ @@ -211,7 +211,7 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) /* set all audio section registers to reasonable defaults */ for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++) - twl4030_write(codec, i, twl4030_reg[i]); + twl4030_write(codec, i, cache[i]); } -- cgit v1.2.3 From eaf1ac8bb58888e0773c0b81dfedb9d7c0123a1d Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 1 Jun 2009 14:06:40 +0300 Subject: ASoC: TWL4030: Check the interface format for 4 channel mode In addition to the operating mode check, also check the codec's interface format in case of four channel mode. If the codec is not in TDM (DSP_A) mode, return with error. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index df474a5dd357..c53c7cabbd27 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1608,9 +1608,15 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, /* If the substream has 4 channel, do the necessary setup */ if (params_channels(params) == 4) { - /* Safety check: are we in the correct operating mode? */ - if ((twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) & - TWL4030_OPTION_1)) + u8 format, mode; + + format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); + mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE); + + /* Safety check: are we in the correct operating mode and + * the interface is in TDM mode? */ + if ((mode & TWL4030_OPTION_1) && + ((format & TWL4030_AIF_FORMAT) == TWL4030_AIF_FORMAT_TDM)) twl4030_tdm_enable(codec, substream->stream, 1); else return -EINVAL; -- cgit v1.2.3 From 2552a710f4b991136c650bf2a6d1b81f27f6273e Mon Sep 17 00:00:00 2001 From: Cliff Cai Date: Tue, 2 Jun 2009 00:18:53 -0400 Subject: ASoC: SSM2602: remove unsupported sample rates Signed-off-by: Cliff Cai Signed-off-by: Mike Frysinger Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 87f606c76822..d6af069b7ed1 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -497,11 +497,9 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, return 0; } -#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ - SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ - SNDRV_PCM_RATE_96000) +#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) #define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -- cgit v1.2.3 From f692fce0cf8625b6cc8678e802fb0e2e657b1ca6 Mon Sep 17 00:00:00 2001 From: Cliff Cai Date: Tue, 2 Jun 2009 00:18:54 -0400 Subject: ASoC: SSM2602: assign last substream to the master when shutting down Fixes crash when shutting down. Signed-off-by: Cliff Cai Signed-off-by: Mike Frysinger Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 25 ++++++++++++++++--------- 1 file changed, 16 insertions(+), 9 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index d6af069b7ed1..1fc4c8e0899c 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -336,15 +336,17 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, master_runtime->sample_bits, master_runtime->rate); - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - master_runtime->rate, - master_runtime->rate); - - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - master_runtime->sample_bits, - master_runtime->sample_bits); + if (master_runtime->rate != 0) + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + master_runtime->rate, + master_runtime->rate); + + if (master_runtime->sample_bits != 0) + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + master_runtime->sample_bits, + master_runtime->sample_bits); ssm2602->slave_substream = substream; } else @@ -372,6 +374,11 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; struct ssm2602_priv *ssm2602 = codec->private_data; + + if (ssm2602->master_substream == substream) + ssm2602->master_substream = ssm2602->slave_substream; + + ssm2602->slave_substream = NULL; /* deactivate */ if (!codec->active) ssm2602_write(codec, SSM2602_ACTIVE, 0); -- cgit v1.2.3 From d08664fdb50795b29cf70b0269ea02f7248e76c3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 4 Jun 2009 09:58:18 +0200 Subject: ASoC: Fix build error in twl4030.c Fix the (likely cut-n-paste) error by commit 16a30fbb0d3aa4ee829a2dd3d0e314e2b5ae96a9, which causes the error below: sound/soc/codecs/twl4030.c: In function 'twl4030_read_reg_cache': sound/soc/codecs/twl4030.c:152: error: 'cache' undeclared (first use in this function) Signed-off-by: Takashi Iwai --- sound/soc/codecs/twl4030.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index c53c7cabbd27..4dbb853eef5a 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -145,6 +145,7 @@ struct twl4030_priv { static inline unsigned int twl4030_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) { + u8 *cache = codec->reg_cache; if (reg >= TWL4030_CACHEREGNUM) return -EIO; -- cgit v1.2.3 From ccff4b15e0847223de0a481f5b7fa5ef902cf3bd Mon Sep 17 00:00:00 2001 From: Troy Kisky Date: Fri, 5 Jun 2009 19:15:58 -0700 Subject: ASoC: codec tlv320aic23 fix bogus divide by 0 message Some code analyzer software mistakenly gives divide by 0 error messages for these lines. This patch will end its confusion. Signed-off-by: Troy Kisky Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 9fcbb9c7766b..0b8dcb5cd729 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -273,14 +273,14 @@ static const unsigned short sr_valid_mask[] = { * Every divisor is a factor of 11*12 */ #define SR_MULT (11*12) -#define A(x) (x) ? (SR_MULT/x) : 0 +#define A(x) (SR_MULT/x) static const unsigned char sr_adc_mult_table[] = { - A(2), A(2), A(12), A(12), A(0), A(0), A(3), A(1), - A(2), A(2), A(11), A(11), A(0), A(0), A(0), A(1) + A(2), A(2), A(12), A(12), 0, 0, A(3), A(1), + A(2), A(2), A(11), A(11), 0, 0, 0, A(1) }; static const unsigned char sr_dac_mult_table[] = { - A(2), A(12), A(2), A(12), A(0), A(0), A(3), A(1), - A(2), A(11), A(2), A(11), A(0), A(0), A(0), A(1) + A(2), A(12), A(2), A(12), 0, 0, A(3), A(1), + A(2), A(11), A(2), A(11), 0, 0, 0, A(1) }; static unsigned get_score(int adc, int adc_l, int adc_h, int need_adc, -- cgit v1.2.3