* ASoC: Patches for an STA32X and WM8782
@ 2011-06-14 19:27 Daniel Mack
2011-06-14 19:27 ` [PATCH 1/2] ALSA: ASoC: add STA32X codec driver Daniel Mack
` (2 more replies)
0 siblings, 3 replies; 38+ messages in thread
From: Daniel Mack @ 2011-06-14 19:27 UTC (permalink / raw)
To: alsa-devel; +Cc: broonie, lrg
Hi,
I'm sending two patches to support the ST STA32x codec and the WM8782 ADC.
The code was originally written by Johannes Stezenbach for 2.6.36.
As the ASoC internals changed quite a bit due to the multi-component
reconstruction, I took care for the code transition to make these drivers
compatible to more recent versions. If there's anything left to care for
in these drivers, please let me know.
Daniel
[PATCH 1/2] ALSA: ASoC: add STA32X codec driver
[PATCH 2/2] ALSA: ASoC: add WM8782 ADC Codec Driver
^ permalink raw reply [flat|nested] 38+ messages in thread* [PATCH 1/2] ALSA: ASoC: add STA32X codec driver 2011-06-14 19:27 ASoC: Patches for an STA32X and WM8782 Daniel Mack @ 2011-06-14 19:27 ` Daniel Mack 2011-06-15 15:05 ` Mark Brown 2011-06-14 19:27 ` [PATCH 2/2] ALSA: ASoC: add WM8782 ADC Codec Driver Daniel Mack 2011-06-15 10:00 ` ASoC: Patches for an STA32X and WM8782 Johannes Stezenbach 2 siblings, 1 reply; 38+ messages in thread From: Daniel Mack @ 2011-06-14 19:27 UTC (permalink / raw) To: alsa-devel; +Cc: broonie, lrg From: Johannes Stezenbach <js@sig21.net> Signed-off-by: Johannes Stezenbach <js@sig21.net> [zonque@gmail.com: transform to new ASoC structure] --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sta32x.c | 786 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/sta32x.h | 210 ++++++++++++ 4 files changed, 1002 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/sta32x.c create mode 100644 sound/soc/codecs/sta32x.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 98175a0..dd075f2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -42,6 +42,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI + select SND_SOC_STA32X if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER @@ -216,6 +217,9 @@ config SND_SOC_SPDIF config SND_SOC_SSM2602 tristate +config SND_SOC_STA32X + tristate + config SND_SOC_STAC9766 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fd85584..2ad1310 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -28,6 +28,7 @@ snd-soc-alc5623-objs := alc5623.o snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-sta32x-objs := sta32x.o snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o @@ -120,6 +121,7 @@ obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(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 diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c new file mode 100644 index 0000000..997ea7c --- /dev/null +++ b/sound/soc/codecs/sta32x.c @@ -0,0 +1,786 @@ +/* + * Codec driver for ST STA32x 2.1-channel high-efficiency digital audio system + * + * Copyright: 2011 Raumfeld GmbH + * Author: Johannes Stezenbach <js@sig21.net> + * + * based on code from: + * Wolfson Microelectronics PLC. + * Mark Brown <broonie@opensource.wolfsonmicro.com> + * Freescale Semiconductor, Inc. + * Timur Tabi <timur@freescale.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#define DEBUG +#define pr_fmt(fmt) KBUILD_MODNAME ":%s:%d: " fmt, __func__, __LINE__ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "sta32x.h" + +#define STA32X_RATES (SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000) + +/* + * The codec isn't really big-endian or little-endian, since the I2S + * interface requires data to be sent serially with the MSbit first. + * However, to support BE and LE I2S devices, we specify both here. That + * way, ALSA will always match the bit patterns. + */ +#define STA32X_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE) + + +/* regulator power supply names */ +static const char *sta32x_supply_names[] = { + "Vdda", /* analog supply, 3.3VV */ + "Vdd3", /* digital supply, 3.3V */ + "Vcc" /* power amp spply, 10V - 36V */ +}; + + +/* codec private data */ +struct sta32x_priv { + struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)]; + struct snd_soc_codec *codec; + + unsigned int mclk; + unsigned int format; +}; + + +static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12700, 50, 1); +static const DECLARE_TLV_DB_SCALE(chvol_tlv, -7950, 50, 1); +static const DECLARE_TLV_DB_SCALE(tone_tlv, -120, 200, 0); + +static const char *sta32x_drc_ac[] = { + "Anti-Clipping", "Dynamic Range Compression" }; +static const char *sta32x_auto_eq_mode[] = { + "User", "Preset", "Loudness" }; +static const char *sta32x_auto_gc_mode[] = { + "User", "AC no clipping", "AC limited clipping (10%)", + "DRC nighttime listening mode" }; +static const char *sta32x_auto_xo_mode[] = { + "User", "80Hz", "100Hz", "120Hz", "140Hz", "160Hz", "180Hz", "200Hz", + "220Hz", "240Hz", "260Hz", "280Hz", "300Hz", "320Hz", "340Hz", "360Hz" }; +static const char *sta32x_preset_eq_mode[] = { + "Flat", "Rock", "Soft Rock", "Jazz", "Classical", "Dance", "Pop", "Soft", + "Hard", "Party", "Vocal", "Hip-Hop", "Dialog", "Bass-boost #1", + "Bass-boost #2", "Bass-boost #3", "Loudness 1", "Loudness 2", + "Loudness 3", "Loudness 4", "Loudness 5", "Loudness 6", "Loudness 7", + "Loudness 8", "Loudness 9", "Loudness 10", "Loudness 11", "Loudness 12", + "Loudness 13", "Loudness 14", "Loudness 15", "Loudness 16" }; +static const char *sta32x_limiter_select[] = { + "Limiter Disabled", "Limiter #1", "Limiter #2" }; +#if 0 +static const char *sta32x_pwm_output_mapping[] = { + "Channel 1", "Channel 2", "Channel 3" }; +#endif +static const char *sta32x_limiter_attack_rate[] = { + "3.1584", "2.7072", "2.2560", "1.8048", "1.3536", "0.9024", + "0.4512", "0.2256", "0.1504", "0.1123", "0.0902", "0.0752", + "0.0645", "0.0564", "0.0501", "0.0451" }; +static const char *sta32x_limiter_release_rate[] = { + "0.5116", "0.1370", "0.0744", "0.0499", "0.0360", "0.0299", + "0.0264", "0.0208", "0.0198", "0.0172", "0.0147", "0.0137", + "0.0134", "0.0117", "0.0110", "0.0104" }; +static const char *sta32x_limiter_ac_attack_thr[] = { + "-12dB", "-10dB", "-8dB", "-6dB", "-4dB", "-2dB", "0dB", "+2dB", + "+3dB", "+4dB", "+5dB", "+6dB", "+7dB", "+8dB", "+9dB", "+10dB" }; +static const char *sta32x_limiter_ac_release_thr[] = { + "-inf", "-29dB", "-20dB", "-16dB", "-14dB", "-12dB", "-10dB", "-8dB", + "-7dB", "-6dB", "-5dB", "-4dB", "-3dB", "-2dB", "-1dB", "0dB" }; +static const char *sta32x_limiter_drc_attack_thr[] = { + "-31dB", "-29dB", "-27dB", "-25dB", "-23dB", "-21dB", "-19dB", "-17dB", + "-16dB", "-15dB", "-14dB", "-13dB", "-12dB", "-10dB", "-7dB", "-4dB" }; +static const char *sta32x_limiter_drc_release_thr[] = { + "-inf", "-38dB", "-36dB", "-33dB", "-31dB", "-30dB", "-28dB", "-26dB", + "-24dB", "-22dB", "-20dB", "-18dB", "-15dB", "-12dB", "-9dB", "-6dB" }; + + +static const struct soc_enum sta32x_drc_ac_enum = + SOC_ENUM_SINGLE(STA32X_CONFD, STA32X_CONFD_DRC_SHIFT, + 2, sta32x_drc_ac); +static const struct soc_enum sta32x_auto_eq_enum = + SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMEQ_SHIFT, + 3, sta32x_auto_eq_mode); +static const struct soc_enum sta32x_auto_gc_enum = + SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMGC_SHIFT, + 4, sta32x_auto_gc_mode); +static const struct soc_enum sta32x_auto_xo_enum = + SOC_ENUM_SINGLE(STA32X_AUTO2, STA32X_AUTO2_XO_SHIFT, + 16, sta32x_auto_xo_mode); +static const struct soc_enum sta32x_preset_eq_enum = + SOC_ENUM_SINGLE(STA32X_AUTO3, STA32X_AUTO3_PEQ_SHIFT, + 32, sta32x_preset_eq_mode); +static const struct soc_enum sta32x_limiter_ch1_enum = + SOC_ENUM_SINGLE(STA32X_C1CFG, STA32X_CxCFG_LS_SHIFT, + 3, sta32x_limiter_select); +static const struct soc_enum sta32x_limiter_ch2_enum = + SOC_ENUM_SINGLE(STA32X_C2CFG, STA32X_CxCFG_LS_SHIFT, + 3, sta32x_limiter_select); +static const struct soc_enum sta32x_limiter_ch3_enum = + SOC_ENUM_SINGLE(STA32X_C3CFG, STA32X_CxCFG_LS_SHIFT, + 3, sta32x_limiter_select); +#if 0 +static const struct soc_enum sta32x_pwm_out_ch1_enum = + SOC_ENUM_SINGLE(STA32X_C1CFG, STA32X_CxCFG_OM_SHIFT, + 2, sta32x_pwm_output_mapping); +static const struct soc_enum sta32x_pwm_out_ch2_enum = + SOC_ENUM_SINGLE(STA32X_C2CFG, STA32X_CxCFG_OM_SHIFT, + 2, sta32x_pwm_output_mapping); +static const struct soc_enum sta32x_pwm_out_ch3_enum = + SOC_ENUM_SINGLE(STA32X_C3CFG, STA32X_CxCFG_OM_SHIFT, + 2, sta32x_pwm_output_mapping); +#endif +static const struct soc_enum sta32x_limiter1_attack_rate_enum = + SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxA_SHIFT, + 16, sta32x_limiter_attack_rate); +static const struct soc_enum sta32x_limiter2_attack_rate_enum = + SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxA_SHIFT, + 16, sta32x_limiter_attack_rate); +static const struct soc_enum sta32x_limiter1_release_rate_enum = + SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxR_SHIFT, + 16, sta32x_limiter_release_rate); +static const struct soc_enum sta32x_limiter2_release_rate_enum = + SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxR_SHIFT, + 16, sta32x_limiter_release_rate); +/* depending on mode, the attack/release thresholds have + * two different enum definitions; provide both + */ +static const struct soc_enum sta32x_limiter1_ac_attack_thr_enum = + SOC_ENUM_SINGLE(STA32X_L1ATRT, STA32X_LxA_SHIFT, + 16, sta32x_limiter_ac_attack_thr); +static const struct soc_enum sta32x_limiter2_ac_attack_thr_enum = + SOC_ENUM_SINGLE(STA32X_L2ATRT, STA32X_LxA_SHIFT, + 16, sta32x_limiter_ac_attack_thr); +static const struct soc_enum sta32x_limiter1_ac_release_thr_enum = + SOC_ENUM_SINGLE(STA32X_L1ATRT, STA32X_LxR_SHIFT, + 16, sta32x_limiter_ac_release_thr); +static const struct soc_enum sta32x_limiter2_ac_release_thr_enum = + SOC_ENUM_SINGLE(STA32X_L2ATRT, STA32X_LxR_SHIFT, + 16, sta32x_limiter_ac_release_thr); +static const struct soc_enum sta32x_limiter1_drc_attack_thr_enum = + SOC_ENUM_SINGLE(STA32X_L1ATRT, STA32X_LxA_SHIFT, + 16, sta32x_limiter_drc_attack_thr); +static const struct soc_enum sta32x_limiter2_drc_attack_thr_enum = + SOC_ENUM_SINGLE(STA32X_L2ATRT, STA32X_LxA_SHIFT, + 16, sta32x_limiter_drc_attack_thr); +static const struct soc_enum sta32x_limiter1_drc_release_thr_enum = + SOC_ENUM_SINGLE(STA32X_L1ATRT, STA32X_LxR_SHIFT, + 16, sta32x_limiter_drc_release_thr); +static const struct soc_enum sta32x_limiter2_drc_release_thr_enum = + SOC_ENUM_SINGLE(STA32X_L2ATRT, STA32X_LxR_SHIFT, + 16, sta32x_limiter_drc_release_thr); + +static const struct snd_kcontrol_new sta32x_snd_controls[] = { +SOC_SINGLE_TLV("Master Volume", STA32X_MVOL, 0, 0xff, 1, mvol_tlv), +SOC_SINGLE("Master Mute", STA32X_MMUTE, 0, 1, 0), +SOC_SINGLE("Ch1 Mute", STA32X_MMUTE, 1, 1, 0), +SOC_SINGLE("Ch2 Mute", STA32X_MMUTE, 2, 1, 0), +SOC_SINGLE("Ch3 Mute", STA32X_MMUTE, 3, 1, 0), +SOC_SINGLE_TLV("Ch1 Volume", STA32X_C1VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE_TLV("Ch2 Volume", STA32X_C2VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE_TLV("Ch3 Volume", STA32X_C3VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE("De-emphasis Filter Switch", STA32X_CONFD, STA32X_CONFD_DEMP_SHIFT, 1, 0), +SOC_ENUM("Compressor/Limiter Switch", sta32x_drc_ac_enum), +SOC_SINGLE("Miami Mode Switch", STA32X_CONFD, STA32X_CONFD_MME_SHIFT, 1, 0), +SOC_SINGLE("Zero Cross Switch", STA32X_CONFE, STA32X_CONFE_ZCE_SHIFT, 1, 0), +SOC_SINGLE("Soft Ramp Switch", STA32X_CONFE, STA32X_CONFE_SVE_SHIFT, 1, 0), +SOC_SINGLE("Auto-Mute Switch", STA32X_CONFF, STA32X_CONFF_IDE_SHIFT, 1, 0), +SOC_ENUM("Automode EQ", sta32x_auto_eq_enum), +SOC_ENUM("Automode GC", sta32x_auto_gc_enum), +SOC_ENUM("Automode XO", sta32x_auto_xo_enum), +SOC_ENUM("Preset EQ", sta32x_preset_eq_enum), +SOC_SINGLE("Ch1 Tone Control Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_TCB_SHIFT, 1, 0), +SOC_SINGLE("Ch2 Tone Control Bypass Switch", STA32X_C2CFG, STA32X_CxCFG_TCB_SHIFT, 1, 0), +SOC_SINGLE("Ch1 EQ Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_EQBP_SHIFT, 1, 0), +SOC_SINGLE("Ch2 EQ Bypass Switch", STA32X_C2CFG, STA32X_CxCFG_EQBP_SHIFT, 1, 0), +SOC_SINGLE("Ch1 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0), +SOC_SINGLE("Ch2 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0), +SOC_SINGLE("Ch3 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0), +SOC_ENUM("Ch1 Limiter Select", sta32x_limiter_ch1_enum), +SOC_ENUM("Ch2 Limiter Select", sta32x_limiter_ch2_enum), +SOC_ENUM("Ch3 Limiter Select", sta32x_limiter_ch3_enum), +#if 0 +SOC_ENUM("Ch1 PWM Output Mapping", sta32x_pwm_out_ch1_enum), +SOC_ENUM("Ch2 PWM Output Mapping", sta32x_pwm_out_ch2_enum), +SOC_ENUM("Ch3 PWM Output Mapping", sta32x_pwm_out_ch3_enum), +#endif +SOC_SINGLE_TLV("Bass Tone Control", STA32X_TONE, STA32X_TONE_BTC_SHIFT, 15, 0, tone_tlv), +SOC_SINGLE_TLV("Treble Tone Control", STA32X_TONE, STA32X_TONE_TTC_SHIFT, 15, 0, tone_tlv), +SOC_ENUM("Limiter1 Attack Rate (dB/ms)", sta32x_limiter1_attack_rate_enum), +SOC_ENUM("Limiter2 Attack Rate (dB/ms)", sta32x_limiter2_attack_rate_enum), +SOC_ENUM("Limiter1 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum), +SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum), +SOC_ENUM("Limiter1 Attack Threshold (AC Mode)", sta32x_limiter1_ac_attack_thr_enum), +SOC_ENUM("Limiter2 Attack Threshold (AC Mode)", sta32x_limiter2_ac_attack_thr_enum), +SOC_ENUM("Limiter1 Release Threshold (AC Mode)", sta32x_limiter1_ac_release_thr_enum), +SOC_ENUM("Limiter2 Release Threshold (AC Mode)", sta32x_limiter2_ac_release_thr_enum), +SOC_ENUM("Limiter1 Attack Threshold (DRC Mode)", sta32x_limiter1_drc_attack_thr_enum), +SOC_ENUM("Limiter2 Attack Threshold (DRC Mode)", sta32x_limiter2_drc_attack_thr_enum), +SOC_ENUM("Limiter1 Release Threshold (DRC Mode)", sta32x_limiter1_drc_release_thr_enum), +SOC_ENUM("Limiter2 Release Threshold (DRC Mode)", sta32x_limiter2_drc_release_thr_enum), +}; + +static const struct snd_soc_dapm_widget sta32x_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_OUTPUT("LEFT"), +SND_SOC_DAPM_OUTPUT("RIGHT"), +SND_SOC_DAPM_OUTPUT("SUB"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + { "LEFT", NULL, "DAC" }, + { "RIGHT", NULL, "DAC" }, + { "SUB", NULL, "DAC" }, +}; + +/* MCLK interpolation ratio per fs */ +static struct { + int fs; + int ir; +} interpolation_ratios[] = { + { 32000, 0 }, + { 44100, 0 }, + { 48000, 0 }, + { 88200, 1 }, + { 96000, 1 }, + { 176400, 2 }, + { 192000, 2 }, +}; + +/* MCLK to fs clock ratios */ +static struct { + int ratio; + int mcs; +} mclk_ratios[3][7] = { + { { 768, 0 }, { 512, 1 }, { 384, 2 }, { 256, 3 }, + { 128, 4 }, { 576, 5 }, { 0, 0 } }, + { { 384, 2 }, { 256, 3 }, { 192, 4 }, { 128, 5 }, {64, 0 }, { 0, 0 } }, + { { 384, 2 }, { 256, 3 }, { 192, 4 }, { 128, 5 }, {64, 0 }, { 0, 0 } }, +}; + + +/** + * sta32x_set_dai_sysclk - configure MCLK + * @codec_dai: the codec DAI + * @clk_id: the clock ID (ignored) + * @freq: the MCLK input frequency + * @dir: the clock direction (ignored) + * + * The value of MCLK is used to determine which sample rates are supported + * by the STA32X, based on the mclk_ratios table. + * + * This function must be called by the machine driver's 'startup' function, + * otherwise the list of supported sample rates will not be available in + * time for ALSA. + * + * For setups with variable MCLKs, pass 0 as 'freq' argument. This will cause + * theoretically possible sample rates to be enabled. Call it again with a + * proper value set one the external clock is set (most probably you would do + * that from a machine's driver 'hw_param' hook. + */ +static int sta32x_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 sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + int i, j, ir, fs; + unsigned int rates = 0; + unsigned int rate_min = -1; + unsigned int rate_max = 0; + + pr_debug("mclk=%u\n", freq); + sta32x->mclk = freq; + + if (sta32x->mclk) { + for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) { + ir = interpolation_ratios[i].ir; + fs = interpolation_ratios[i].fs; + for (j = 0; mclk_ratios[ir][j].ratio; j++) { + if (mclk_ratios[ir][j].ratio * fs == freq) { + rates |= snd_pcm_rate_to_rate_bit(fs); + if (fs < rate_min) + rate_min = fs; + if (fs > rate_max) + rate_max = fs; + } + } + } + /* FIXME: soc should support a rate list */ + rates &= ~SNDRV_PCM_RATE_KNOT; + + if (!rates) { + dev_err(codec->dev, "could not find a valid sample rate\n"); + return -EINVAL; + } + } else { + /* enable all possible rates */ + rates = STA32X_RATES; + rate_min = 32000; + rate_max = 192000; + } + + codec_dai->driver->playback.rates = rates; + codec_dai->driver->playback.rate_min = rate_min; + codec_dai->driver->playback.rate_max = rate_max; + return 0; +} + +/** + * sta32x_set_dai_fmt - configure the codec for the selected audio format + * @codec_dai: the codec DAI + * @fmt: a SND_SOC_DAIFMT_x value indicating the data format + * + * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the + * codec accordingly. + */ +static int sta32x_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + u8 confb = snd_soc_read(codec, STA32X_CONFB); + + pr_debug("\n"); + confb &= ~(STA32X_CONFB_C1IM | STA32X_CONFB_C2IM); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + sta32x->format = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + confb |= STA32X_CONFB_C2IM; + break; + case SND_SOC_DAIFMT_NB_IF: + confb |= STA32X_CONFB_C1IM; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, STA32X_CONFB, confb); + return 0; +} + +/** + * sta32x_hw_params - program the STA32X with the given hardware parameters. + * @substream: the audio stream + * @params: the hardware parameters to set + * @dai: the SOC DAI (ignored) + * + * This function programs the hardware with the values provided. + * Specifically, the sample rate and the data format. + */ +static int sta32x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + unsigned int rate; + int i, mcs = -1, ir = -1; + u8 confa, confb; + + rate = params_rate(params); + pr_debug("rate: %u\n", rate); + for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) + if (interpolation_ratios[i].fs == rate) + ir = interpolation_ratios[i].ir; + if (ir < 0) + return -EINVAL; + for (i = 0; mclk_ratios[ir][i].ratio; i++) + if (mclk_ratios[ir][i].ratio * rate == sta32x->mclk) + mcs = mclk_ratios[ir][i].mcs; + if (mcs < 0) + return -EINVAL; + + confa = snd_soc_read(codec, STA32X_CONFA); + confa &= ~(STA32X_CONFA_MCS_MASK | STA32X_CONFA_IR_MASK); + confa |= (ir << STA32X_CONFA_IR_SHIFT) | (mcs << STA32X_CONFA_MCS_SHIFT); + + confb = snd_soc_read(codec, STA32X_CONFB); + confb &= ~(STA32X_CONFB_SAI_MASK | STA32X_CONFB_SAIFB); + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S24_BE: + case SNDRV_PCM_FORMAT_S24_3LE: + case SNDRV_PCM_FORMAT_S24_3BE: + pr_debug("24bit\n"); + case SNDRV_PCM_FORMAT_S32_LE: + case SNDRV_PCM_FORMAT_S32_BE: + pr_debug("24bit or 32bit\n"); + if (sta32x->format == SND_SOC_DAIFMT_I2S) + confb |= 0x0; + else if (sta32x->format == SND_SOC_DAIFMT_LEFT_J) + confb |= 0x1; + else if (sta32x->format == SND_SOC_DAIFMT_RIGHT_J) + confb |= 0x2; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + case SNDRV_PCM_FORMAT_S20_3BE: + pr_debug("20bit\n"); + if (sta32x->format == SND_SOC_DAIFMT_I2S) + confb |= 0x4; + else if (sta32x->format == SND_SOC_DAIFMT_LEFT_J) + confb |= 0x5; + else if (sta32x->format == SND_SOC_DAIFMT_RIGHT_J) + confb |= 0x6; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + case SNDRV_PCM_FORMAT_S18_3BE: + pr_debug("18bit\n"); + if (sta32x->format == SND_SOC_DAIFMT_I2S) + confb |= 0x8; + else if (sta32x->format == SND_SOC_DAIFMT_LEFT_J) + confb |= 0x9; + else if (sta32x->format == SND_SOC_DAIFMT_RIGHT_J) + confb |= 0xa; + break; + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S16_BE: + pr_debug("16bit\n"); + if (sta32x->format == SND_SOC_DAIFMT_I2S) + confb |= 0x0; + else if (sta32x->format == SND_SOC_DAIFMT_LEFT_J) + confb |= 0xd; + else if (sta32x->format == SND_SOC_DAIFMT_RIGHT_J) + confb |= 0xe; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, STA32X_CONFA, confa); + snd_soc_write(codec, STA32X_CONFB, confb); + return 0; +} + +/** + * sta32x_set_bias_level - DAPM callback + * @codec: the codec device + * @level: DAPM power level + * + * This is called by ALSA to put the codec into low power mode + * or to wake it up. If the codec is powered off completely + * all registers must be restored after power on. + */ +static int sta32x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int ret; + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + + pr_debug("level = %d\n", level); + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* Full power on */ + snd_soc_update_bits(codec, STA32X_CONFF, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), + sta32x->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", ret); + return ret; + } + + snd_soc_cache_sync(codec); + } + + /* Power up to mute */ + /* FIXME */ + snd_soc_update_bits(codec, STA32X_CONFF, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD); + + break; + + case SND_SOC_BIAS_OFF: + /* The chip runs through the power down sequence for us. */ + snd_soc_update_bits(codec, STA32X_CONFF, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD, + STA32X_CONFF_PWDN); + msleep(300); + + regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), + sta32x->supplies); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static struct snd_soc_dai_ops sta32x_dai_ops = { + .hw_params = sta32x_hw_params, + .set_sysclk = sta32x_set_dai_sysclk, + .set_fmt = sta32x_set_dai_fmt, +}; + +static struct snd_soc_dai_driver sta32x_dai = { + .name = "STA32X", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = STA32X_RATES, + .formats = STA32X_FORMATS, + }, + .ops = &sta32x_dai_ops, +}; + +#ifdef CONFIG_PM +static int sta32x_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int sta32x_resume(struct snd_soc_codec *codec) +{ + sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} +#else +#define sta32x_suspend NULL +#define sta32x_resume NULL +#endif + +static int sta32x_probe(struct snd_soc_codec *codec) +{ + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; + int i, ret = 0; + + sta32x->codec = codec; + + /* regulators */ + for (i = 0; i < ARRAY_SIZE(sta32x->supplies); i++) + sta32x->supplies[i].supply = sta32x_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sta32x->supplies), + sta32x->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), + sta32x->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_get; + } + + /* Tell ASoC what kind of I/O to use to read the registers. ASoC will + * then do the I2C transactions itself. + */ + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret); + return ret; + } + + /* read reg reset values into cache */ + for (i = 0; i < STA32X_REGISTER_COUNT; i++) { + unsigned int val = codec->hw_read(codec, i); + + switch (i) { + case STA32X_CONFA: + /* FIXME enable thermal warning adjustment and recovery */ + val &= ~(STA32X_CONFA_TWAB | STA32X_CONFA_TWRB); + snd_soc_write(codec, i, val); + break; + case STA32X_CONFF: + /* FIXME select 2.1 mode */ + val &= ~STA32X_CONFF_OCFG_MASK; + val |= 1 << STA32X_CONFF_OCFG_SHIFT; + snd_soc_write(codec, i, val); + break; + case STA32X_C1CFG: + /* FIXME channel to output mapping */ + val &= ~STA32X_CxCFG_OM_MASK; + val |= 0 << STA32X_CxCFG_OM_SHIFT; + snd_soc_write(codec, i, val); + break; + case STA32X_C2CFG: + /* FIXME channel to output mapping */ + val &= ~STA32X_CxCFG_OM_MASK; + val |= 1 << STA32X_CxCFG_OM_SHIFT; + snd_soc_write(codec, i, val); + break; + case STA32X_C3CFG: + /* FIXME channel to output mapping */ + val &= ~STA32X_CxCFG_OM_MASK; + val |= 2 << STA32X_CxCFG_OM_SHIFT; + snd_soc_write(codec, i, val); + break; + default: + snd_soc_cache_write(codec, i, val); + } + } + + sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Bias level configuration will have done an extra enable */ + regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + snd_soc_add_controls(codec, sta32x_snd_controls, + ARRAY_SIZE(sta32x_snd_controls)); + + snd_soc_dapm_new_controls(dapm, sta32x_dapm_widgets, + ARRAY_SIZE(sta32x_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); + + return 0; + +err_get: + regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); +err: + return ret; +} + +static int sta32x_remove(struct snd_soc_codec *codec) +{ + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + + regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + return 0; +} + +static const struct snd_soc_codec_driver sta32x_codec = { + .probe = sta32x_probe, + .remove = sta32x_remove, + .suspend = sta32x_suspend, + .resume = sta32x_resume, + .reg_cache_size = STA32X_REGISTER_COUNT, + .reg_word_size = sizeof(u8), + .set_bias_level = sta32x_set_bias_level, +}; + +static __devinit int sta32x_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct sta32x_priv *sta32x; + int ret; + + sta32x = kzalloc(sizeof(struct sta32x_priv), GFP_KERNEL); + if (!sta32x) + return -ENOMEM; + + i2c_set_clientdata(i2c, sta32x); + + ret = snd_soc_register_codec(&i2c->dev, &sta32x_codec, &sta32x_dai, 1); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret); + return ret; + } + + return 0; +} + +static __devexit int sta32x_i2c_remove(struct i2c_client *client) +{ + struct sta32x_priv *sta32x = i2c_get_clientdata(client); + struct snd_soc_codec *codec = sta32x->codec; + + if (codec) + sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); + + regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + if (codec) { + snd_soc_unregister_codec(&client->dev); + snd_soc_codec_set_drvdata(codec, NULL); + } + + kfree(sta32x); + return 0; +} + +static const struct i2c_device_id sta32x_i2c_id[] = { + { "sta32x", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, sta32x_i2c_id); + +static struct i2c_driver sta32x_i2c_driver = { + .driver = { + .name = "sta32x-codec", + .owner = THIS_MODULE, + }, + .probe = sta32x_i2c_probe, + .remove = __devexit_p(sta32x_i2c_remove), + .id_table = sta32x_i2c_id, +}; + +static int __init sta32x_init(void) +{ + return i2c_add_driver(&sta32x_i2c_driver); +} +module_init(sta32x_init); + +static void __exit sta32x_exit(void) +{ + i2c_del_driver(&sta32x_i2c_driver); +} +module_exit(sta32x_exit); + +MODULE_DESCRIPTION("ASoC STA32X driver"); +MODULE_AUTHOR("Johannes Stezenbach <js@sig21.net>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sta32x.h b/sound/soc/codecs/sta32x.h new file mode 100644 index 0000000..b97ee5a --- /dev/null +++ b/sound/soc/codecs/sta32x.h @@ -0,0 +1,210 @@ +/* + * Codec driver for ST STA32x 2.1-channel high-efficiency digital audio system + * + * Copyright: 2011 Raumfeld GmbH + * Author: Johannes Stezenbach <js@sig21.net> + * + * based on code from: + * Wolfson Microelectronics PLC. + * Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#ifndef _ASOC_STA_32X_H +#define _ASOC_STA_32X_H + +/* STA326 register addresses */ + +#define STA32X_REGISTER_COUNT 0x2d + +#define STA32X_CONFA 0x00 +#define STA32X_CONFB 0x01 +#define STA32X_CONFC 0x02 +#define STA32X_CONFD 0x03 +#define STA32X_CONFE 0x04 +#define STA32X_CONFF 0x05 +#define STA32X_MMUTE 0x06 +#define STA32X_MVOL 0x07 +#define STA32X_C1VOL 0x08 +#define STA32X_C2VOL 0x09 +#define STA32X_C3VOL 0x0a +#define STA32X_AUTO1 0x0b +#define STA32X_AUTO2 0x0c +#define STA32X_AUTO3 0x0d +#define STA32X_C1CFG 0x0e +#define STA32X_C2CFG 0x0f +#define STA32X_C3CFG 0x10 +#define STA32X_TONE 0x11 +#define STA32X_L1AR 0x12 +#define STA32X_L1ATRT 0x13 +#define STA32X_L2AR 0x14 +#define STA32X_L2ATRT 0x15 +#define STA32X_CFADDR2 0x16 +#define STA32X_B1CF1 0x17 +#define STA32X_B1CF2 0x18 +#define STA32X_B1CF3 0x19 +#define STA32X_B2CF1 0x1a +#define STA32X_B2CF2 0x1b +#define STA32X_B2CF3 0x1c +#define STA32X_A1CF1 0x1d +#define STA32X_A1CF2 0x1e +#define STA32X_A1CF3 0x1f +#define STA32X_A2CF1 0x20 +#define STA32X_A2CF2 0x21 +#define STA32X_A2CF3 0x22 +#define STA32X_B0CF1 0x23 +#define STA32X_B0CF2 0x24 +#define STA32X_B0CF3 0x25 +#define STA32X_CFUD 0x26 +#define STA32X_MPCC1 0x27 +#define STA32X_MPCC2 0x28 +/* Reserved 0x29 */ +/* Reserved 0x2a */ +#define STA32X_Reserved 0x2a +#define STA32X_FDRC1 0x2b +#define STA32X_FDRC2 0x2c +/* Reserved 0x2d */ + + +/* STA326 register field definitions */ + +/* 0x00 CONFA */ +#define STA32X_CONFA_MCS_MASK 0x03 +#define STA32X_CONFA_MCS_SHIFT 0 +#define STA32X_CONFA_IR_MASK 0x18 +#define STA32X_CONFA_IR_SHIFT 3 +#define STA32X_CONFA_TWRB 0x20 +#define STA32X_CONFA_TWAB 0x40 +#define STA32X_CONFA_FDRB 0x80 + +/* 0x01 CONFB */ +#define STA32X_CONFB_SAI_MASK 0x0f +#define STA32X_CONFB_SAI_SHIFT 0 +#define STA32X_CONFB_SAIFB 0x10 +#define STA32X_CONFB_DSCKE 0x20 +#define STA32X_CONFB_C1IM 0x40 +#define STA32X_CONFB_C2IM 0x80 + +/* 0x02 CONFC */ +#define STA32X_CONFC_OM_MASK 0x03 +#define STA32X_CONFC_OM_SHIFT 0 +#define STA32X_CONFC_CSZ_MASK 0x7c +#define STA32X_CONFC_CSZ_SHIFT 2 + +/* 0x03 CONFD */ +#define STA32X_CONFD_HPB 0x01 +#define STA32X_CONFD_HPB_SHIFT 0 +#define STA32X_CONFD_DEMP 0x02 +#define STA32X_CONFD_DEMP_SHIFT 1 +#define STA32X_CONFD_DSPB 0x04 +#define STA32X_CONFD_DSPB_SHIFT 2 +#define STA32X_CONFD_PSL 0x08 +#define STA32X_CONFD_PSL_SHIFT 3 +#define STA32X_CONFD_BQL 0x10 +#define STA32X_CONFD_BQL_SHIFT 4 +#define STA32X_CONFD_DRC 0x20 +#define STA32X_CONFD_DRC_SHIFT 5 +#define STA32X_CONFD_ZDE 0x40 +#define STA32X_CONFD_ZDE_SHIFT 6 +#define STA32X_CONFD_MME 0x80 +#define STA32X_CONFD_MME_SHIFT 7 + +/* 0x04 CONFE */ +#define STA32X_CONFE_MPCV 0x01 +#define STA32X_CONFE_MPCV_SHIFT 0 +#define STA32X_CONFE_MPC 0x02 +#define STA32X_CONFE_MPC_SHIFT 1 +#define STA32X_CONFE_AME 0x08 +#define STA32X_CONFE_AME_SHIFT 3 +#define STA32X_CONFE_PWMS 0x10 +#define STA32X_CONFE_PWMS_SHIFT 4 +#define STA32X_CONFE_ZCE 0x40 +#define STA32X_CONFE_ZCE_SHIFT 6 +#define STA32X_CONFE_SVE 0x80 +#define STA32X_CONFE_SVE_SHIFT 7 + +/* 0x05 CONFF */ +#define STA32X_CONFF_OCFG_MASK 0x03 +#define STA32X_CONFF_OCFG_SHIFT 0 +#define STA32X_CONFF_IDE 0x04 +#define STA32X_CONFF_IDE_SHIFT 3 +#define STA32X_CONFF_BCLE 0x08 +#define STA32X_CONFF_ECLE 0x20 +#define STA32X_CONFF_PWDN 0x40 +#define STA32X_CONFF_EAPD 0x80 + +/* 0x06 MMUTE */ +#define STA32X_MMUTE_MMUTE 0x01 + +/* 0x0b AUTO1 */ +#define STA32X_AUTO1_AMEQ_MASK 0x03 +#define STA32X_AUTO1_AMEQ_SHIFT 0 +#define STA32X_AUTO1_AMV_MASK 0xc0 +#define STA32X_AUTO1_AMV_SHIFT 2 +#define STA32X_AUTO1_AMGC_MASK 0x30 +#define STA32X_AUTO1_AMGC_SHIFT 4 +#define STA32X_AUTO1_AMPS 0x80 + +/* 0x0c AUTO2 */ +#define STA32X_AUTO2_AMAME 0x01 +#define STA32X_AUTO2_AMAM_MASK 0x0e +#define STA32X_AUTO2_AMAM_SHIFT 1 +#define STA32X_AUTO2_XO_MASK 0xf0 +#define STA32X_AUTO2_XO_SHIFT 4 + +/* 0x0d AUTO3 */ +#define STA32X_AUTO3_PEQ_MASK 0x1f +#define STA32X_AUTO3_PEQ_SHIFT 0 + +/* 0x0e 0x0f 0x10 CxCFG */ +#define STA32X_CxCFG_TCB 0x01 /* only C1 and C2 */ +#define STA32X_CxCFG_TCB_SHIFT 0 +#define STA32X_CxCFG_EQBP 0x02 /* only C1 and C2 */ +#define STA32X_CxCFG_EQBP_SHIFT 1 +#define STA32X_CxCFG_VBP 0x03 +#define STA32X_CxCFG_VBP_SHIFT 2 +#define STA32X_CxCFG_BO 0x04 +#define STA32X_CxCFG_LS_MASK 0x30 +#define STA32X_CxCFG_LS_SHIFT 4 +#define STA32X_CxCFG_OM_MASK 0xc0 +#define STA32X_CxCFG_OM_SHIFT 6 + +/* 0x11 TONE */ +#define STA32X_TONE_BTC_SHIFT 0 +#define STA32X_TONE_TTC_SHIFT 4 + +/* 0x12 0x13 0x14 0x15 limiter attack/release */ +#define STA32X_LxA_SHIFT 0 +#define STA32X_LxR_SHIFT 4 + +/* 0x26 CFUD */ +#define STA32X_CFUD_W1 0x01 +#define STA32X_CFUD_WA 0x02 +#define STA32X_CFUD_R1 0x04 +#define STA32X_CFUD_RA 0x08 + + +/* biquad filter coefficient table offsets */ +#define STA32X_C1_BQ_BASE 0 +#define STA32X_C2_BQ_BASE 20 +#define STA32X_CH_BQ_NUM 4 +#define STA32X_BQ_NUM_COEF 5 +#define STA32X_XO_HP_BQ_BASE 40 +#define STA32X_XO_LP_BQ_BASE 45 +#define STA32X_C1_PRESCALE 50 +#define STA32X_C2_PRESCALE 51 +#define STA32X_C1_POSTSCALE 52 +#define STA32X_C2_POSTSCALE 53 +#define STA32X_C3_POSTSCALE 54 +#define STA32X_TW_POSTSCALE 55 +#define STA32X_C1_MIX1 56 +#define STA32X_C1_MIX2 57 +#define STA32X_C2_MIX1 58 +#define STA32X_C2_MIX2 59 +#define STA32X_C3_MIX1 60 +#define STA32X_C3_MIX2 61 + +#endif /* _ASOC_STA_32X_H */ -- 1.7.5.1 ^ permalink raw reply related [flat|nested] 38+ messages in thread
* Re: [PATCH 1/2] ALSA: ASoC: add STA32X codec driver 2011-06-14 19:27 ` [PATCH 1/2] ALSA: ASoC: add STA32X codec driver Daniel Mack @ 2011-06-15 15:05 ` Mark Brown 2011-06-15 20:53 ` Johannes Stezenbach 2011-06-16 8:19 ` Daniel Mack 0 siblings, 2 replies; 38+ messages in thread From: Mark Brown @ 2011-06-15 15:05 UTC (permalink / raw) To: Daniel Mack; +Cc: alsa-devel, lrg On Tue, Jun 14, 2011 at 09:27:05PM +0200, Daniel Mack wrote: > +/* > + * The codec isn't really big-endian or little-endian, since the I2S > + * interface requires data to be sent serially with the MSbit first. > + * However, to support BE and LE I2S devices, we specify both here. That > + * way, ALSA will always match the bit patterns. > + */ The core is supposed to be fixing this up for you... > +#if 0 > +static const char *sta32x_pwm_output_mapping[] = { > + "Channel 1", "Channel 2", "Channel 3" }; > +#endif Remove all the if 0 blocks in the driver or enable them. > +static const char *sta32x_limiter_ac_release_thr[] = { > + "-inf", "-29dB", "-20dB", "-16dB", "-14dB", "-12dB", "-10dB", "-8dB", > + "-7dB", "-6dB", "-5dB", "-4dB", "-3dB", "-2dB", "-1dB", "0dB" }; > +static const char *sta32x_limiter_drc_attack_thr[] = { > + "-31dB", "-29dB", "-27dB", "-25dB", "-23dB", "-21dB", "-19dB", "-17dB", > + "-16dB", "-15dB", "-14dB", "-13dB", "-12dB", "-10dB", "-7dB", "-4dB" }; > +static const char *sta32x_limiter_drc_release_thr[] = { > + "-inf", "-38dB", "-36dB", "-33dB", "-31dB", "-30dB", "-28dB", "-26dB", > + "-24dB", "-22dB", "-20dB", "-18dB", "-15dB", "-12dB", "-9dB", "-6dB" }; > + Doing these as regular volume TLVs will tend to work better in UIs. > +SOC_SINGLE("Master Mute", STA32X_MMUTE, 0, 1, 0), > +SOC_SINGLE("Ch1 Mute", STA32X_MMUTE, 1, 1, 0), > +SOC_SINGLE("Ch2 Mute", STA32X_MMUTE, 2, 1, 0), > +SOC_SINGLE("Ch3 Mute", STA32X_MMUTE, 3, 1, 0), All Mutes should be replaced with Switch in control names. > + if (sta32x->format == SND_SOC_DAIFMT_I2S) > + confb |= 0x8; > + else if (sta32x->format == SND_SOC_DAIFMT_LEFT_J) > + confb |= 0x9; > + else if (sta32x->format == SND_SOC_DAIFMT_RIGHT_J) > + confb |= 0xa; Looks like you want to use switch statements for these? > + ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), > + sta32x->supplies); > + if (ret != 0) { > + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); > + goto err_get; > + } If you enable the supplies when you go to standby you can just punt this to when you set the bias level. > + /* read reg reset values into cache */ > + for (i = 0; i < STA32X_REGISTER_COUNT; i++) { > + unsigned int val = codec->hw_read(codec, i); > + Does the chip not have a specified hardware state when it comes out of reset? > + case STA32X_CONFA: > + /* FIXME enable thermal warning adjustment and recovery */ > + val &= ~(STA32X_CONFA_TWAB | STA32X_CONFA_TWRB); > + snd_soc_write(codec, i, val); > + break; This would be much more legible if you just did an update bits to set the values independantly of the cache init, both from the point of view of the open coding and also for splitting out the default changes from the cache init. > + snd_soc_add_controls(codec, sta32x_snd_controls, > + ARRAY_SIZE(sta32x_snd_controls)); > + > + snd_soc_dapm_new_controls(dapm, sta32x_dapm_widgets, > + ARRAY_SIZE(sta32x_dapm_widgets)); > + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); You should just add these arrays to the driver data structure and then the core will add the controls and routes for you. > +static const struct i2c_device_id sta32x_i2c_id[] = { > + { "sta32x", 0 }, > + { } > +}; > +MODULE_DEVICE_TABLE(i2c, sta32x_i2c_id); If you support a range of different devices then just list them here. > +static struct i2c_driver sta32x_i2c_driver = { > + .driver = { > + .name = "sta32x-codec", Drop the -codec from the name, it's redundant. ^ permalink raw reply [flat|nested] 38+ messages in thread
* Re: [PATCH 1/2] ALSA: ASoC: add STA32X codec driver 2011-06-15 15:05 ` Mark Brown @ 2011-06-15 20:53 ` Johannes Stezenbach 2011-06-16 8:39 ` Lars-Peter Clausen 2011-06-16 8:19 ` Daniel Mack 1 sibling, 1 reply; 38+ messages in thread From: Johannes Stezenbach @ 2011-06-15 20:53 UTC (permalink / raw) To: Mark Brown; +Cc: alsa-devel, lrg, Daniel Mack On Wed, Jun 15, 2011 at 04:05:16PM +0100, Mark Brown wrote: > On Tue, Jun 14, 2011 at 09:27:05PM +0200, Daniel Mack wrote: > > > +static const char *sta32x_limiter_ac_release_thr[] = { > > + "-inf", "-29dB", "-20dB", "-16dB", "-14dB", "-12dB", "-10dB", "-8dB", > > + "-7dB", "-6dB", "-5dB", "-4dB", "-3dB", "-2dB", "-1dB", "0dB" }; > > +static const char *sta32x_limiter_drc_attack_thr[] = { > > + "-31dB", "-29dB", "-27dB", "-25dB", "-23dB", "-21dB", "-19dB", "-17dB", > > + "-16dB", "-15dB", "-14dB", "-13dB", "-12dB", "-10dB", "-7dB", "-4dB" }; > > +static const char *sta32x_limiter_drc_release_thr[] = { > > + "-inf", "-38dB", "-36dB", "-33dB", "-31dB", "-30dB", "-28dB", "-26dB", > > + "-24dB", "-22dB", "-20dB", "-18dB", "-15dB", "-12dB", "-9dB", "-6dB" }; > > + > > Doing these as regular volume TLVs will tend to work better in UIs. The steps are not evenly distributed so I thought this is the only way to describe the hardware correctly. Or is there a way to do controls which look like a slider in alsamixer with steps as shown above? Thanks for your review. Johannes ^ permalink raw reply [flat|nested] 38+ messages in thread
* Re: [PATCH 1/2] ALSA: ASoC: add STA32X codec driver 2011-06-15 20:53 ` Johannes Stezenbach @ 2011-06-16 8:39 ` Lars-Peter Clausen 2011-06-16 9:12 ` Johannes Stezenbach 2011-06-16 9:51 ` Daniel Mack 0 siblings, 2 replies; 38+ messages in thread From: Lars-Peter Clausen @ 2011-06-16 8:39 UTC (permalink / raw) To: Johannes Stezenbach; +Cc: alsa-devel, Mark Brown, Daniel Mack, lrg On 06/15/2011 10:53 PM, Johannes Stezenbach wrote: > On Wed, Jun 15, 2011 at 04:05:16PM +0100, Mark Brown wrote: >> On Tue, Jun 14, 2011 at 09:27:05PM +0200, Daniel Mack wrote: >> >>> +static const char *sta32x_limiter_ac_release_thr[] = { >>> + "-inf", "-29dB", "-20dB", "-16dB", "-14dB", "-12dB", "-10dB", "-8dB", >>> + "-7dB", "-6dB", "-5dB", "-4dB", "-3dB", "-2dB", "-1dB", "0dB" }; >>> +static const char *sta32x_limiter_drc_attack_thr[] = { >>> + "-31dB", "-29dB", "-27dB", "-25dB", "-23dB", "-21dB", "-19dB", "-17dB", >>> + "-16dB", "-15dB", "-14dB", "-13dB", "-12dB", "-10dB", "-7dB", "-4dB" }; >>> +static const char *sta32x_limiter_drc_release_thr[] = { >>> + "-inf", "-38dB", "-36dB", "-33dB", "-31dB", "-30dB", "-28dB", "-26dB", >>> + "-24dB", "-22dB", "-20dB", "-18dB", "-15dB", "-12dB", "-9dB", "-6dB" }; >>> + >> >> Doing these as regular volume TLVs will tend to work better in UIs. > > The steps are not evenly distributed so I thought this is the > only way to describe the hardware correctly. Or is there a > way to do controls which look like a slider in alsamixer > with steps as shown above? > You can put multiple scale items each with their own min/max value into a TLV. For example for DRC attack in your case this would look like: static const unsigned int drc_attack_tlv[] = { TLV_DB_RANGE_HEAD(3), 0, 7, TLV_DB_SCALE_ITEM(-3100, 200, 0), 8, 13, TLV_DB_SCALE_ITEM(-1600, 100, 0), 14, 16, TLV_DB_SCALE_ITEM(-1000, 300, 0), }; - Lars ^ permalink raw reply [flat|nested] 38+ messages in thread
* Re: [PATCH 1/2] ALSA: ASoC: add STA32X codec driver 2011-06-16 8:39 ` Lars-Peter Clausen @ 2011-06-16 9:12 ` Johannes Stezenbach 2011-06-16 9:51 ` Daniel Mack 1 sibling, 0 replies; 38+ messages in thread From: Johannes Stezenbach @ 2011-06-16 9:12 UTC (permalink / raw) To: Lars-Peter Clausen; +Cc: alsa-devel, Mark Brown, Daniel Mack, lrg On Thu, Jun 16, 2011 at 10:39:22AM +0200, Lars-Peter Clausen wrote: > >> On Tue, Jun 14, 2011 at 09:27:05PM +0200, Daniel Mack wrote: > >> > >>> +static const char *sta32x_limiter_drc_attack_thr[] = { > >>> + "-31dB", "-29dB", "-27dB", "-25dB", "-23dB", "-21dB", "-19dB", "-17dB", > >>> + "-16dB", "-15dB", "-14dB", "-13dB", "-12dB", "-10dB", "-7dB", "-4dB" }; > > You can put multiple scale items each with their own min/max value into a TLV. > For example for DRC attack in your case this would look like: > > static const unsigned int drc_attack_tlv[] = { > TLV_DB_RANGE_HEAD(3), > 0, 7, TLV_DB_SCALE_ITEM(-3100, 200, 0), > 8, 13, TLV_DB_SCALE_ITEM(-1600, 100, 0), > 14, 16, TLV_DB_SCALE_ITEM(-1000, 300, 0), > }; That looks indeed much better than using enums. Thanks, Johannes ^ permalink raw reply [flat|nested] 38+ messages in thread
* Re: [PATCH 1/2] ALSA: ASoC: add STA32X codec driver 2011-06-16 8:39 ` Lars-Peter Clausen 2011-06-16 9:12 ` Johannes Stezenbach @ 2011-06-16 9:51 ` Daniel Mack 2011-06-16 10:13 ` Mark Brown 1 sibling, 1 reply; 38+ messages in thread From: Daniel Mack @ 2011-06-16 9:51 UTC (permalink / raw) To: Lars-Peter Clausen; +Cc: Mark Brown, alsa-devel, lrg On Thu, Jun 16, 2011 at 10:39 AM, Lars-Peter Clausen <lars@metafoo.de> wrote: > You can put multiple scale items each with their own min/max value into a TLV. > For example for DRC attack in your case this would look like: > > static const unsigned int drc_attack_tlv[] = { > TLV_DB_RANGE_HEAD(3), > 0, 7, TLV_DB_SCALE_ITEM(-3100, 200, 0), > 8, 13, TLV_DB_SCALE_ITEM(-1600, 100, 0), > 14, 16, TLV_DB_SCALE_ITEM(-1000, 300, 0), > }; Using this results in strange values in userspace: # amixer cget numid=39 numid=39,iface=MIXER,name='Limiter1 Attack Threshold (DRC Mode)' ; type=INTEGER,access=rw---R--,values=1,min=0,max=16,step=0 : values=9 | dBrange- 0x00000000,0x00000007,0x00000001,0x00000008,0xfffff3e4,0x000000c8,0x00000008,0x0000000d,0x00000001,0x00000008,0xfffff9c0,0x00000064,0x0000000e,0x00000010,0x00000001,0x00000008,0xfffffc18,0x0000012c, Is there a way to get more descriptive information about possible values for such controls? Thanks, Daniel ^ permalink raw reply [flat|nested] 38+ messages in thread
* Re: [PATCH 1/2] ALSA: ASoC: add STA32X codec driver 2011-06-16 9:51 ` Daniel Mack @ 2011-06-16 10:13 ` Mark Brown 2011-06-16 11:03 ` Daniel Mack 0 siblings, 1 reply; 38+ messages in thread From: Mark Brown @ 2011-06-16 10:13 UTC (permalink / raw) To: Daniel Mack; +Cc: Lars-Peter Clausen, alsa-devel, lrg On Thu, Jun 16, 2011 at 11:51:20AM +0200, Daniel Mack wrote: > # amixer cget numid=39 > numid=39,iface=MIXER,name='Limiter1 Attack Threshold (DRC Mode)' > ; type=INTEGER,access=rw---R--,values=1,min=0,max=16,step=0 > : values=9 > | dBrange- > 0x00000000,0x00000007,0x00000001,0x00000008,0xfffff3e4,0x000000c8,0x00000008,0x0000000d,0x00000001,0x00000008,0xfffff9c0,0x00000064,0x0000000e,0x00000010,0x00000001,0x00000008,0xfffffc18,0x0000012c, > Is there a way to get more descriptive information about possible > values for such controls? This looks rather like a buggy userspace - are you sure you're using a current version? ^ permalink raw reply [flat|nested] 38+ messages in thread
* Re: [PATCH 1/2] ALSA: ASoC: add STA32X codec driver 2011-06-16 10:13 ` Mark Brown @ 2011-06-16 11:03 ` Daniel Mack 2011-06-16 11:11 ` Lars-Peter Clausen 0 siblings, 1 reply; 38+ messages in thread From: Daniel Mack @ 2011-06-16 11:03 UTC (permalink / raw) To: Mark Brown; +Cc: Lars-Peter Clausen, alsa-devel, lrg On Thu, Jun 16, 2011 at 12:13 PM, Mark Brown <broonie@opensource.wolfsonmicro.com> wrote: > On Thu, Jun 16, 2011 at 11:51:20AM +0200, Daniel Mack wrote: > >> # amixer cget numid=39 >> numid=39,iface=MIXER,name='Limiter1 Attack Threshold (DRC Mode)' >> ; type=INTEGER,access=rw---R--,values=1,min=0,max=16,step=0 >> : values=9 >> | dBrange- >> 0x00000000,0x00000007,0x00000001,0x00000008,0xfffff3e4,0x000000c8,0x00000008,0x0000000d,0x00000001,0x00000008,0xfffff9c0,0x00000064,0x0000000e,0x00000010,0x00000001,0x00000008,0xfffffc18,0x0000012c, > >> Is there a way to get more descriptive information about possible >> values for such controls? > > This looks rather like a buggy userspace - are you sure you're using a > current version? # amixer --version amixer version 1.0.24.2 This is indeed strange, and I can't see how I use the API in any illegal way. Any other way to debug this? Thanks, Daniel ^ permalink raw reply [flat|nested] 38+ messages in thread
* Re: [PATCH 1/2] ALSA: ASoC: add STA32X codec driver 2011-06-16 11:03 ` Daniel Mack @ 2011-06-16 11:11 ` Lars-Peter Clausen 2011-06-16 11:26 ` Daniel Mack 0 siblings, 1 reply; 38+ messages in thread From: Lars-Peter Clausen @ 2011-06-16 11:11 UTC (permalink / raw) To: Daniel Mack; +Cc: Mark Brown, alsa-devel, lrg On 06/16/2011 01:03 PM, Daniel Mack wrote: > On Thu, Jun 16, 2011 at 12:13 PM, Mark Brown > <broonie@opensource.wolfsonmicro.com> wrote: >> On Thu, Jun 16, 2011 at 11:51:20AM +0200, Daniel Mack wrote: >> >>> # amixer cget numid=39 >>> numid=39,iface=MIXER,name='Limiter1 Attack Threshold (DRC Mode)' >>> ; type=INTEGER,access=rw---R--,values=1,min=0,max=16,step=0 >>> : values=9 >>> | dBrange- >>> 0x00000000,0x00000007,0x00000001,0x00000008,0xfffff3e4,0x000000c8,0x00000008,0x0000000d,0x00000001,0x00000008,0xfffff9c0,0x00000064,0x0000000e,0x00000010,0x00000001,0x00000008,0xfffffc18,0x0000012c, >> >>> Is there a way to get more descriptive information about possible >>> values for such controls? >> >> This looks rather like a buggy userspace - are you sure you're using a >> current version? > > # amixer --version > amixer version 1.0.24.2 > > This is indeed strange, and I can't see how I use the API in any > illegal way. Any other way to debug this? > There seems to be a bug in amixer, please try: diff --git a/amixer/amixer.c b/amixer/amixer.c index a177288..c7b93b7 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -517,7 +517,7 @@ static void decode_tlv(unsigned int spaces, unsigned int *tlv, unsigned int tlv_ #ifdef SND_CTL_TLVT_DB_RANGE case SND_CTL_TLVT_DB_RANGE: printf("dBrange-\n"); - if ((size / (6 * sizeof(unsigned int))) != 0) { + if ((size % (6 * sizeof(unsigned int))) != 0) { while (size > 0) { printf("0x%08x,", tlv[idx++]); size -= sizeof(unsigned int); @@ -529,7 +529,7 @@ static void decode_tlv(unsigned int spaces, unsigned int *tlv, unsigned int tlv_ print_spaces(spaces + 2); printf("rangemin=%i,", tlv[0]); printf(",rangemax=%i\n", tlv[1]); - decode_tlv(spaces + 4, tlv + 2, 6 * sizeof(unsigned int)); + decode_tlv(spaces + 4, tlv + 2, 4 * sizeof(unsigned int)); idx += 6 * sizeof(unsigned int); } break; ^ permalink raw reply related [flat|nested] 38+ messages in thread
* Re: [PATCH 1/2] ALSA: ASoC: add STA32X codec driver 2011-06-16 11:11 ` Lars-Peter Clausen @ 2011-06-16 11:26 ` Daniel Mack 2011-06-16 11:38 ` Lars-Peter Clausen 0 siblings, 1 reply; 38+ messages in thread From: Daniel Mack @ 2011-06-16 11:26 UTC (permalink / raw) To: Lars-Peter Clausen; +Cc: Mark Brown, alsa-devel, lrg On Thu, Jun 16, 2011 at 1:11 PM, Lars-Peter Clausen <lars@metafoo.de> wrote: > On 06/16/2011 01:03 PM, Daniel Mack wrote: >> On Thu, Jun 16, 2011 at 12:13 PM, Mark Brown >> <broonie@opensource.wolfsonmicro.com> wrote: >>> On Thu, Jun 16, 2011 at 11:51:20AM +0200, Daniel Mack wrote: >>> >>>> # amixer cget numid=39 >>>> numid=39,iface=MIXER,name='Limiter1 Attack Threshold (DRC Mode)' >>>> ; type=INTEGER,access=rw---R--,values=1,min=0,max=16,step=0 >>>> : values=9 >>>> | dBrange- >>>> 0x00000000,0x00000007,0x00000001,0x00000008,0xfffff3e4,0x000000c8,0x00000008,0x0000000d,0x00000001,0x00000008,0xfffff9c0,0x00000064,0x0000000e,0x00000010,0x00000001,0x00000008,0xfffffc18,0x0000012c, >>> >>>> Is there a way to get more descriptive information about possible >>>> values for such controls? >>> >>> This looks rather like a buggy userspace - are you sure you're using a >>> current version? >> >> # amixer --version >> amixer version 1.0.24.2 >> >> This is indeed strange, and I can't see how I use the API in any >> illegal way. Any other way to debug this? >> > There seems to be a bug in amixer, please try: > > diff --git a/amixer/amixer.c b/amixer/amixer.c > index a177288..c7b93b7 100644 > --- a/amixer/amixer.c > +++ b/amixer/amixer.c > @@ -517,7 +517,7 @@ static void decode_tlv(unsigned int spaces, unsigned int *tlv, unsigned int tlv_ > #ifdef SND_CTL_TLVT_DB_RANGE > case SND_CTL_TLVT_DB_RANGE: > printf("dBrange-\n"); > - if ((size / (6 * sizeof(unsigned int))) != 0) { > + if ((size % (6 * sizeof(unsigned int))) != 0) { > while (size > 0) { > printf("0x%08x,", tlv[idx++]); > size -= sizeof(unsigned int); > @@ -529,7 +529,7 @@ static void decode_tlv(unsigned int spaces, unsigned int *tlv, unsigned int tlv_ > print_spaces(spaces + 2); > printf("rangemin=%i,", tlv[0]); > printf(",rangemax=%i\n", tlv[1]); > - decode_tlv(spaces + 4, tlv + 2, 6 * sizeof(unsigned int)); > + decode_tlv(spaces + 4, tlv + 2, 4 * sizeof(unsigned int)); > idx += 6 * sizeof(unsigned int); > } > break; > Looks different now, but I'd say that's still not what you expected: # amixer cget numid=39 numid=39,iface=MIXER,name='Limiter1 Attack Threshold (DRC Mode)' ; type=INTEGER,access=rw---R--,values=1,min=0,max=16,step=0 : values=9 | dBrange- rangemin=3,,rangemax=72 | rangemin=3,,rangemax=72 | rangemin=3,,rangemax=72 | ^ permalink raw reply [flat|nested] 38+ messages in thread
* Re: [PATCH 1/2] ALSA: ASoC: add STA32X codec driver 2011-06-16 11:26 ` Daniel Mack @ 2011-06-16 11:38 ` Lars-Peter Clausen 2011-06-16 11:47 ` Daniel Mack 0 siblings, 1 reply; 38+ messages in thread From: Lars-Peter Clausen @ 2011-06-16 11:38 UTC (permalink / raw) To: Daniel Mack; +Cc: Mark Brown, alsa-devel, lrg On 06/16/2011 01:26 PM, Daniel Mack wrote: > [...] > Looks different now, but I'd say that's still not what you expected: > > # amixer cget numid=39 > numid=39,iface=MIXER,name='Limiter1 Attack Threshold (DRC Mode)' > ; type=INTEGER,access=rw---R--,values=1,min=0,max=16,step=0 > : values=9 > | dBrange- > rangemin=3,,rangemax=72 > | > rangemin=3,,rangemax=72 > | > rangemin=3,,rangemax=72 > | Ok, looks like the current implementation is completely broken... diff --git a/amixer/amixer.c b/amixer/amixer.c index a177288..3bc6743 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -517,20 +517,20 @@ static void decode_tlv(unsigned int spaces, unsigned int *tlv, unsigned int tlv_ #ifdef SND_CTL_TLVT_DB_RANGE case SND_CTL_TLVT_DB_RANGE: printf("dBrange-\n"); - if ((size / (6 * sizeof(unsigned int))) != 0) { + if ((size % (6 * sizeof(unsigned int))) != 0) { while (size > 0) { printf("0x%08x,", tlv[idx++]); size -= sizeof(unsigned int); } break; } - idx = 0; - while (idx < size) { + while (size >= 0) { print_spaces(spaces + 2); - printf("rangemin=%i,", tlv[0]); - printf(",rangemax=%i\n", tlv[1]); - decode_tlv(spaces + 4, tlv + 2, 6 * sizeof(unsigned int)); - idx += 6 * sizeof(unsigned int); + printf("rangemin=%i,", tlv[idx++]); + printf(",rangemax=%i\n", tlv[idx++]); + decode_tlv(spaces + 4, tlv + idx, 4 * sizeof(unsigned int)); + idx += 4 * sizeof(unsigned int); + size -= 6 * sizeof(unsigned int); } break; #endif ^ permalink raw reply related [flat|nested] 38+ messages in thread
* Re: [PATCH 1/2] ALSA: ASoC: add STA32X codec driver 2011-06-16 11:38 ` Lars-Peter Clausen @ 2011-06-16 11:47 ` Daniel Mack 2011-06-16 11:51 ` Lars-Peter Clausen 0 siblings, 1 reply; 38+ messages in thread From: Daniel Mack @ 2011-06-16 11:47 UTC (permalink / raw) To: Lars-Peter Clausen; +Cc: alsa-devel, Mark Brown, lrg On Thu, Jun 16, 2011 at 1:38 PM, Lars-Peter Clausen <lars@metafoo.de> wrote: > On 06/16/2011 01:26 PM, Daniel Mack wrote: >> [...] >> Looks different now, but I'd say that's still not what you expected: >> >> # amixer cget numid=39 >> numid=39,iface=MIXER,name='Limiter1 Attack Threshold (DRC Mode)' >> ; type=INTEGER,access=rw---R--,values=1,min=0,max=16,step=0 >> : values=9 >> | dBrange- >> rangemin=3,,rangemax=72 >> | >> rangemin=3,,rangemax=72 >> | >> rangemin=3,,rangemax=72 >> | > > Ok, looks like the current implementation is completely broken... > > diff --git a/amixer/amixer.c b/amixer/amixer.c > index a177288..3bc6743 100644 > --- a/amixer/amixer.c > +++ b/amixer/amixer.c > @@ -517,20 +517,20 @@ static void decode_tlv(unsigned int spaces, unsigned int *tlv, unsigned int tlv_ > #ifdef SND_CTL_TLVT_DB_RANGE > case SND_CTL_TLVT_DB_RANGE: > printf("dBrange-\n"); > - if ((size / (6 * sizeof(unsigned int))) != 0) { > + if ((size % (6 * sizeof(unsigned int))) != 0) { > while (size > 0) { > printf("0x%08x,", tlv[idx++]); > size -= sizeof(unsigned int); > } > break; > } > - idx = 0; > - while (idx < size) { > + while (size >= 0) { > print_spaces(spaces + 2); > - printf("rangemin=%i,", tlv[0]); > - printf(",rangemax=%i\n", tlv[1]); > - decode_tlv(spaces + 4, tlv + 2, 6 * sizeof(unsigned int)); > - idx += 6 * sizeof(unsigned int); > + printf("rangemin=%i,", tlv[idx++]); > + printf(",rangemax=%i\n", tlv[idx++]); > + decode_tlv(spaces + 4, tlv + idx, 4 * sizeof(unsigned int)); > + idx += 4 * sizeof(unsigned int); > + size -= 6 * sizeof(unsigned int); > } > break; > #endif Nope, that's not yet it ... # amixer cget numid=39 numid=39,iface=MIXER,name='Limiter1 Attack Threshold (DRC Mode)' ; type=INTEGER,access=rw---R--,values=1,min=0,max=16,step=0 : values=9 | dBrange- rangemin=0,,rangemax=7 | dBscale-min=-31.00dB,step=2.00dB,mute=0 rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=3,,rangemax=72 | rangemin=-1000,,rangemax=300 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | rangemin=0,,rangemax=0 | Segmentation fault ^ permalink raw reply [flat|nested] 38+ messages in thread
* Re: [PATCH 1/2] ALSA: ASoC: add STA32X codec driver 2011-06-16 11:47 ` Daniel Mack @ 2011-06-16 11:51 ` Lars-Peter Clausen 2011-06-16 11:54 ` Daniel Mack 0 siblings, 1 reply; 38+ messages in thread From: Lars-Peter Clausen @ 2011-06-16 11:51 UTC (permalink / raw) To: Daniel Mack; +Cc: alsa-devel, Mark Brown, lrg On 06/16/2011 01:47 PM, Daniel Mack wrote: > On Thu, Jun 16, 2011 at 1:38 PM, Lars-Peter Clausen <lars@metafoo.de> wrote: >> On 06/16/2011 01:26 PM, Daniel Mack wrote: >>> [...] >>> Looks different now, but I'd say that's still not what you expected: >>> >>> # amixer cget numid=39 >>> numid=39,iface=MIXER,name='Limiter1 Attack Threshold (DRC Mode)' >>> ; type=INTEGER,access=rw---R--,values=1,min=0,max=16,step=0 >>> : values=9 >>> | dBrange- >>> rangemin=3,,rangemax=72 >>> | >>> rangemin=3,,rangemax=72 >>> | >>> rangemin=3,,rangemax=72 >>> | >> >> Ok, looks like the current implementation is completely broken... >> > > Nope, that's not yet it ... > >> - while (idx < size) { >> + while (size >= 0) { Should have been while(size > 0) { >> print_spaces(spaces + 2); >> - printf("rangemin=%i,", tlv[0]); >> - printf(",rangemax=%i\n", tlv[1]); >> - decode_tlv(spaces + 4, tlv + 2, 6 * sizeof(unsigned int)); >> - idx += 6 * sizeof(unsigned int); >> + printf("rangemin=%i,", tlv[idx++]); >> + printf(",rangemax=%i\n", tlv[idx++]); >> + decode_tlv(spaces + 4, tlv + idx, 4 * sizeof(unsigned int)); >> + idx += 4 * sizeof(unsigned int); Should have been idx += 4; >> + size -= 6 * sizeof(unsigned int); >> } >> break; >> #endif ^ permalink raw reply [flat|nested] 38+ messages in thread
* Re: [PATCH 1/2] ALSA: ASoC: add STA32X codec driver 2011-06-16 11:51 ` Lars-Peter Clausen @ 2011-06-16 11:54 ` Daniel Mack 0 siblings, 0 replies; 38+ messages in thread From: Daniel Mack @ 2011-06-16 11:54 UTC (permalink / raw) To: Lars-Peter Clausen; +Cc: alsa-devel, Mark Brown, lrg On Thu, Jun 16, 2011 at 1:51 PM, Lars-Peter Clausen <lars@metafoo.de> wrote: > On 06/16/2011 01:47 PM, Daniel Mack wrote: >> On Thu, Jun 16, 2011 at 1:38 PM, Lars-Peter Clausen <lars@metafoo.de> wrote: >>> On 06/16/2011 01:26 PM, Daniel Mack wrote: >>>> [...] >>>> Looks different now, but I'd say that's still not what you expected: >>>> >>>> # amixer cget numid=39 >>>> numid=39,iface=MIXER,name='Limiter1 Attack Threshold (DRC Mode)' >>>> ; type=INTEGER,access=rw---R--,values=1,min=0,max=16,step=0 >>>> : values=9 >>>> | dBrange- >>>> rangemin=3,,rangemax=72 >>>> | >>>> rangemin=3,,rangemax=72 >>>> | >>>> rangemin=3,,rangemax=72 >>>> | >>> >>> Ok, looks like the current implementation is completely broken... >>> >> >> Nope, that's not yet it ... >> >>> - while (idx < size) { >>> + while (size >= 0) { > Should have been while(size > 0) { >>> print_spaces(spaces + 2); >>> - printf("rangemin=%i,", tlv[0]); >>> - printf(",rangemax=%i\n", tlv[1]); >>> - decode_tlv(spaces + 4, tlv + 2, 6 * sizeof(unsigned int)); >>> - idx += 6 * sizeof(unsigned int); >>> + printf("rangemin=%i,", tlv[idx++]); >>> + printf(",rangemax=%i\n", tlv[idx++]); >>> + decode_tlv(spaces + 4, tlv + idx, 4 * sizeof(unsigned int)); >>> + idx += 4 * sizeof(unsigned int); > Should have been idx += 4; >>> + size -= 6 * sizeof(unsigned int); >>> } >>> break; >>> #endif # amixer cget numid=39 numid=39,iface=MIXER,name='Limiter1 Attack Threshold (DRC Mode)' ; type=INTEGER,access=rw---R--,values=1,min=0,max=16,step=0 : values=9 | dBrange- rangemin=0,,rangemax=7 | dBscale-min=-31.00dB,step=2.00dB,mute=0 rangemin=8,,rangemax=13 | dBscale-min=-16.00dB,step=1.00dB,mute=0 rangemin=14,,rangemax=16 | dBscale-min=-10.00dB,step=3.00dB,mute=0 Great. Thanks a lot! Daniel ^ permalink raw reply [flat|nested] 38+ messages in thread
* [PATCH 1/2] ALSA: ASoC: add STA32X codec driver 2011-06-15 15:05 ` Mark Brown 2011-06-15 20:53 ` Johannes Stezenbach @ 2011-06-16 8:19 ` Daniel Mack 2011-06-16 8:49 ` Liam Girdwood 1 sibling, 1 reply; 38+ messages in thread From: Daniel Mack @ 2011-06-16 8:19 UTC (permalink / raw) To: alsa-devel; +Cc: broonie, lrg From: Johannes Stezenbach <js@sig21.net> Signed-off-by: Johannes Stezenbach <js@sig21.net> [zonque@gmail.com: transform to new ASoC structure] --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sta32x.c | 796 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/sta32x.h | 210 ++++++++++++ 4 files changed, 1012 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/sta32x.c create mode 100644 sound/soc/codecs/sta32x.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 98175a0..dd075f2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -42,6 +42,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI + select SND_SOC_STA32X if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER @@ -216,6 +217,9 @@ config SND_SOC_SPDIF config SND_SOC_SSM2602 tristate +config SND_SOC_STA32X + tristate + config SND_SOC_STAC9766 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fd85584..2ad1310 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -28,6 +28,7 @@ snd-soc-alc5623-objs := alc5623.o snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-sta32x-objs := sta32x.o snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o @@ -120,6 +121,7 @@ obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(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 diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c new file mode 100644 index 0000000..599a58b --- /dev/null +++ b/sound/soc/codecs/sta32x.c @@ -0,0 +1,796 @@ +/* + * Codec driver for ST STA32x 2.1-channel high-efficiency digital audio system + * + * Copyright: 2011 Raumfeld GmbH + * Author: Johannes Stezenbach <js@sig21.net> + * + * based on code from: + * Wolfson Microelectronics PLC. + * Mark Brown <broonie@opensource.wolfsonmicro.com> + * Freescale Semiconductor, Inc. + * Timur Tabi <timur@freescale.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#define DEBUG +#define pr_fmt(fmt) KBUILD_MODNAME ":%s:%d: " fmt, __func__, __LINE__ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "sta32x.h" + +#define STA32X_RATES (SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000) + +#define STA32X_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE) + +/* Power-up register defaults */ +static const u8 sta32x_regs[STA32X_REGISTER_COUNT] = { + 0x63, 0x80, 0xc2, 0x40, 0xc2, 0x5c, 0x10, 0xff, 0x60, 0x60, + 0x60, 0x80, 0x00, 0x00, 0x00, 0x40, 0x80, 0x77, 0x6a, 0x69, + 0x6a, 0x69, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x2d, + 0xc0, 0xf3, 0x33, 0x00, 0x0c, +}; + +/* regulator power supply names */ +static const char *sta32x_supply_names[] = { + "Vdda", /* analog supply, 3.3VV */ + "Vdd3", /* digital supply, 3.3V */ + "Vcc" /* power amp spply, 10V - 36V */ +}; + +/* codec private data */ +struct sta32x_priv { + struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)]; + struct snd_soc_codec *codec; + + unsigned int mclk; + unsigned int format; +}; + +static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12700, 50, 1); +static const DECLARE_TLV_DB_SCALE(chvol_tlv, -7950, 50, 1); +static const DECLARE_TLV_DB_SCALE(tone_tlv, -120, 200, 0); + +static const char *sta32x_drc_ac[] = { + "Anti-Clipping", "Dynamic Range Compression" }; +static const char *sta32x_auto_eq_mode[] = { + "User", "Preset", "Loudness" }; +static const char *sta32x_auto_gc_mode[] = { + "User", "AC no clipping", "AC limited clipping (10%)", + "DRC nighttime listening mode" }; +static const char *sta32x_auto_xo_mode[] = { + "User", "80Hz", "100Hz", "120Hz", "140Hz", "160Hz", "180Hz", "200Hz", + "220Hz", "240Hz", "260Hz", "280Hz", "300Hz", "320Hz", "340Hz", "360Hz" }; +static const char *sta32x_preset_eq_mode[] = { + "Flat", "Rock", "Soft Rock", "Jazz", "Classical", "Dance", "Pop", "Soft", + "Hard", "Party", "Vocal", "Hip-Hop", "Dialog", "Bass-boost #1", + "Bass-boost #2", "Bass-boost #3", "Loudness 1", "Loudness 2", + "Loudness 3", "Loudness 4", "Loudness 5", "Loudness 6", "Loudness 7", + "Loudness 8", "Loudness 9", "Loudness 10", "Loudness 11", "Loudness 12", + "Loudness 13", "Loudness 14", "Loudness 15", "Loudness 16" }; +static const char *sta32x_limiter_select[] = { + "Limiter Disabled", "Limiter #1", "Limiter #2" }; +#if 0 +static const char *sta32x_pwm_output_mapping[] = { + "Channel 1", "Channel 2", "Channel 3" }; +#endif +static const char *sta32x_limiter_attack_rate[] = { + "3.1584", "2.7072", "2.2560", "1.8048", "1.3536", "0.9024", + "0.4512", "0.2256", "0.1504", "0.1123", "0.0902", "0.0752", + "0.0645", "0.0564", "0.0501", "0.0451" }; +static const char *sta32x_limiter_release_rate[] = { + "0.5116", "0.1370", "0.0744", "0.0499", "0.0360", "0.0299", + "0.0264", "0.0208", "0.0198", "0.0172", "0.0147", "0.0137", + "0.0134", "0.0117", "0.0110", "0.0104" }; +static const char *sta32x_limiter_ac_attack_thr[] = { + "-12dB", "-10dB", "-8dB", "-6dB", "-4dB", "-2dB", "0dB", "+2dB", + "+3dB", "+4dB", "+5dB", "+6dB", "+7dB", "+8dB", "+9dB", "+10dB" }; +static const char *sta32x_limiter_ac_release_thr[] = { + "-inf", "-29dB", "-20dB", "-16dB", "-14dB", "-12dB", "-10dB", "-8dB", + "-7dB", "-6dB", "-5dB", "-4dB", "-3dB", "-2dB", "-1dB", "0dB" }; +static const char *sta32x_limiter_drc_attack_thr[] = { + "-31dB", "-29dB", "-27dB", "-25dB", "-23dB", "-21dB", "-19dB", "-17dB", + "-16dB", "-15dB", "-14dB", "-13dB", "-12dB", "-10dB", "-7dB", "-4dB" }; +static const char *sta32x_limiter_drc_release_thr[] = { + "-inf", "-38dB", "-36dB", "-33dB", "-31dB", "-30dB", "-28dB", "-26dB", + "-24dB", "-22dB", "-20dB", "-18dB", "-15dB", "-12dB", "-9dB", "-6dB" }; + + +static const struct soc_enum sta32x_drc_ac_enum = + SOC_ENUM_SINGLE(STA32X_CONFD, STA32X_CONFD_DRC_SHIFT, + 2, sta32x_drc_ac); +static const struct soc_enum sta32x_auto_eq_enum = + SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMEQ_SHIFT, + 3, sta32x_auto_eq_mode); +static const struct soc_enum sta32x_auto_gc_enum = + SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMGC_SHIFT, + 4, sta32x_auto_gc_mode); +static const struct soc_enum sta32x_auto_xo_enum = + SOC_ENUM_SINGLE(STA32X_AUTO2, STA32X_AUTO2_XO_SHIFT, + 16, sta32x_auto_xo_mode); +static const struct soc_enum sta32x_preset_eq_enum = + SOC_ENUM_SINGLE(STA32X_AUTO3, STA32X_AUTO3_PEQ_SHIFT, + 32, sta32x_preset_eq_mode); +static const struct soc_enum sta32x_limiter_ch1_enum = + SOC_ENUM_SINGLE(STA32X_C1CFG, STA32X_CxCFG_LS_SHIFT, + 3, sta32x_limiter_select); +static const struct soc_enum sta32x_limiter_ch2_enum = + SOC_ENUM_SINGLE(STA32X_C2CFG, STA32X_CxCFG_LS_SHIFT, + 3, sta32x_limiter_select); +static const struct soc_enum sta32x_limiter_ch3_enum = + SOC_ENUM_SINGLE(STA32X_C3CFG, STA32X_CxCFG_LS_SHIFT, + 3, sta32x_limiter_select); +#if 0 +static const struct soc_enum sta32x_pwm_out_ch1_enum = + SOC_ENUM_SINGLE(STA32X_C1CFG, STA32X_CxCFG_OM_SHIFT, + 2, sta32x_pwm_output_mapping); +static const struct soc_enum sta32x_pwm_out_ch2_enum = + SOC_ENUM_SINGLE(STA32X_C2CFG, STA32X_CxCFG_OM_SHIFT, + 2, sta32x_pwm_output_mapping); +static const struct soc_enum sta32x_pwm_out_ch3_enum = + SOC_ENUM_SINGLE(STA32X_C3CFG, STA32X_CxCFG_OM_SHIFT, + 2, sta32x_pwm_output_mapping); +#endif +static const struct soc_enum sta32x_limiter1_attack_rate_enum = + SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxA_SHIFT, + 16, sta32x_limiter_attack_rate); +static const struct soc_enum sta32x_limiter2_attack_rate_enum = + SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxA_SHIFT, + 16, sta32x_limiter_attack_rate); +static const struct soc_enum sta32x_limiter1_release_rate_enum = + SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxR_SHIFT, + 16, sta32x_limiter_release_rate); +static const struct soc_enum sta32x_limiter2_release_rate_enum = + SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxR_SHIFT, + 16, sta32x_limiter_release_rate); +/* depending on mode, the attack/release thresholds have + * two different enum definitions; provide both + */ +static const struct soc_enum sta32x_limiter1_ac_attack_thr_enum = + SOC_ENUM_SINGLE(STA32X_L1ATRT, STA32X_LxA_SHIFT, + 16, sta32x_limiter_ac_attack_thr); +static const struct soc_enum sta32x_limiter2_ac_attack_thr_enum = + SOC_ENUM_SINGLE(STA32X_L2ATRT, STA32X_LxA_SHIFT, + 16, sta32x_limiter_ac_attack_thr); +static const struct soc_enum sta32x_limiter1_ac_release_thr_enum = + SOC_ENUM_SINGLE(STA32X_L1ATRT, STA32X_LxR_SHIFT, + 16, sta32x_limiter_ac_release_thr); +static const struct soc_enum sta32x_limiter2_ac_release_thr_enum = + SOC_ENUM_SINGLE(STA32X_L2ATRT, STA32X_LxR_SHIFT, + 16, sta32x_limiter_ac_release_thr); +static const struct soc_enum sta32x_limiter1_drc_attack_thr_enum = + SOC_ENUM_SINGLE(STA32X_L1ATRT, STA32X_LxA_SHIFT, + 16, sta32x_limiter_drc_attack_thr); +static const struct soc_enum sta32x_limiter2_drc_attack_thr_enum = + SOC_ENUM_SINGLE(STA32X_L2ATRT, STA32X_LxA_SHIFT, + 16, sta32x_limiter_drc_attack_thr); +static const struct soc_enum sta32x_limiter1_drc_release_thr_enum = + SOC_ENUM_SINGLE(STA32X_L1ATRT, STA32X_LxR_SHIFT, + 16, sta32x_limiter_drc_release_thr); +static const struct soc_enum sta32x_limiter2_drc_release_thr_enum = + SOC_ENUM_SINGLE(STA32X_L2ATRT, STA32X_LxR_SHIFT, + 16, sta32x_limiter_drc_release_thr); + +static const struct snd_kcontrol_new sta32x_snd_controls[] = { +SOC_SINGLE_TLV("Master Volume", STA32X_MVOL, 0, 0xff, 1, mvol_tlv), +SOC_SINGLE("Master Playback Switch", STA32X_MMUTE, 0, 1, 1), +SOC_SINGLE("Ch1 Playback Switch", STA32X_MMUTE, 1, 1, 1), +SOC_SINGLE("Ch2 Playback Switch", STA32X_MMUTE, 2, 1, 1), +SOC_SINGLE("Ch3 Playback Switch", STA32X_MMUTE, 3, 1, 1), +SOC_SINGLE_TLV("Ch1 Volume", STA32X_C1VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE_TLV("Ch2 Volume", STA32X_C2VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE_TLV("Ch3 Volume", STA32X_C3VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE("De-emphasis Filter Switch", STA32X_CONFD, STA32X_CONFD_DEMP_SHIFT, 1, 0), +SOC_ENUM("Compressor/Limiter Switch", sta32x_drc_ac_enum), +SOC_SINGLE("Miami Mode Switch", STA32X_CONFD, STA32X_CONFD_MME_SHIFT, 1, 0), +SOC_SINGLE("Zero Cross Switch", STA32X_CONFE, STA32X_CONFE_ZCE_SHIFT, 1, 0), +SOC_SINGLE("Soft Ramp Switch", STA32X_CONFE, STA32X_CONFE_SVE_SHIFT, 1, 0), +SOC_SINGLE("Auto-Mute Switch", STA32X_CONFF, STA32X_CONFF_IDE_SHIFT, 1, 0), +SOC_ENUM("Automode EQ", sta32x_auto_eq_enum), +SOC_ENUM("Automode GC", sta32x_auto_gc_enum), +SOC_ENUM("Automode XO", sta32x_auto_xo_enum), +SOC_ENUM("Preset EQ", sta32x_preset_eq_enum), +SOC_SINGLE("Ch1 Tone Control Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_TCB_SHIFT, 1, 0), +SOC_SINGLE("Ch2 Tone Control Bypass Switch", STA32X_C2CFG, STA32X_CxCFG_TCB_SHIFT, 1, 0), +SOC_SINGLE("Ch1 EQ Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_EQBP_SHIFT, 1, 0), +SOC_SINGLE("Ch2 EQ Bypass Switch", STA32X_C2CFG, STA32X_CxCFG_EQBP_SHIFT, 1, 0), +SOC_SINGLE("Ch1 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0), +SOC_SINGLE("Ch2 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0), +SOC_SINGLE("Ch3 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0), +SOC_ENUM("Ch1 Limiter Select", sta32x_limiter_ch1_enum), +SOC_ENUM("Ch2 Limiter Select", sta32x_limiter_ch2_enum), +SOC_ENUM("Ch3 Limiter Select", sta32x_limiter_ch3_enum), +#if 0 +SOC_ENUM("Ch1 PWM Output Mapping", sta32x_pwm_out_ch1_enum), +SOC_ENUM("Ch2 PWM Output Mapping", sta32x_pwm_out_ch2_enum), +SOC_ENUM("Ch3 PWM Output Mapping", sta32x_pwm_out_ch3_enum), +#endif +SOC_SINGLE_TLV("Bass Tone Control", STA32X_TONE, STA32X_TONE_BTC_SHIFT, 15, 0, tone_tlv), +SOC_SINGLE_TLV("Treble Tone Control", STA32X_TONE, STA32X_TONE_TTC_SHIFT, 15, 0, tone_tlv), +SOC_ENUM("Limiter1 Attack Rate (dB/ms)", sta32x_limiter1_attack_rate_enum), +SOC_ENUM("Limiter2 Attack Rate (dB/ms)", sta32x_limiter2_attack_rate_enum), +SOC_ENUM("Limiter1 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum), +SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum), +SOC_ENUM("Limiter1 Attack Threshold (AC Mode)", sta32x_limiter1_ac_attack_thr_enum), +SOC_ENUM("Limiter2 Attack Threshold (AC Mode)", sta32x_limiter2_ac_attack_thr_enum), +SOC_ENUM("Limiter1 Release Threshold (AC Mode)", sta32x_limiter1_ac_release_thr_enum), +SOC_ENUM("Limiter2 Release Threshold (AC Mode)", sta32x_limiter2_ac_release_thr_enum), +SOC_ENUM("Limiter1 Attack Threshold (DRC Mode)", sta32x_limiter1_drc_attack_thr_enum), +SOC_ENUM("Limiter2 Attack Threshold (DRC Mode)", sta32x_limiter2_drc_attack_thr_enum), +SOC_ENUM("Limiter1 Release Threshold (DRC Mode)", sta32x_limiter1_drc_release_thr_enum), +SOC_ENUM("Limiter2 Release Threshold (DRC Mode)", sta32x_limiter2_drc_release_thr_enum), +}; + +static const struct snd_soc_dapm_widget sta32x_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_OUTPUT("LEFT"), +SND_SOC_DAPM_OUTPUT("RIGHT"), +SND_SOC_DAPM_OUTPUT("SUB"), +}; + +static const struct snd_soc_dapm_route sta32x_dapm_routes[] = { + { "LEFT", NULL, "DAC" }, + { "RIGHT", NULL, "DAC" }, + { "SUB", NULL, "DAC" }, +}; + +/* MCLK interpolation ratio per fs */ +static struct { + int fs; + int ir; +} interpolation_ratios[] = { + { 32000, 0 }, + { 44100, 0 }, + { 48000, 0 }, + { 88200, 1 }, + { 96000, 1 }, + { 176400, 2 }, + { 192000, 2 }, +}; + +/* MCLK to fs clock ratios */ +static struct { + int ratio; + int mcs; +} mclk_ratios[3][7] = { + { { 768, 0 }, { 512, 1 }, { 384, 2 }, { 256, 3 }, + { 128, 4 }, { 576, 5 }, { 0, 0 } }, + { { 384, 2 }, { 256, 3 }, { 192, 4 }, { 128, 5 }, {64, 0 }, { 0, 0 } }, + { { 384, 2 }, { 256, 3 }, { 192, 4 }, { 128, 5 }, {64, 0 }, { 0, 0 } }, +}; + + +/** + * sta32x_set_dai_sysclk - configure MCLK + * @codec_dai: the codec DAI + * @clk_id: the clock ID (ignored) + * @freq: the MCLK input frequency + * @dir: the clock direction (ignored) + * + * The value of MCLK is used to determine which sample rates are supported + * by the STA32X, based on the mclk_ratios table. + * + * This function must be called by the machine driver's 'startup' function, + * otherwise the list of supported sample rates will not be available in + * time for ALSA. + * + * For setups with variable MCLKs, pass 0 as 'freq' argument. This will cause + * theoretically possible sample rates to be enabled. Call it again with a + * proper value set one the external clock is set (most probably you would do + * that from a machine's driver 'hw_param' hook. + */ +static int sta32x_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 sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + int i, j, ir, fs; + unsigned int rates = 0; + unsigned int rate_min = -1; + unsigned int rate_max = 0; + + pr_debug("mclk=%u\n", freq); + sta32x->mclk = freq; + + if (sta32x->mclk) { + for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) { + ir = interpolation_ratios[i].ir; + fs = interpolation_ratios[i].fs; + for (j = 0; mclk_ratios[ir][j].ratio; j++) { + if (mclk_ratios[ir][j].ratio * fs == freq) { + rates |= snd_pcm_rate_to_rate_bit(fs); + if (fs < rate_min) + rate_min = fs; + if (fs > rate_max) + rate_max = fs; + } + } + } + /* FIXME: soc should support a rate list */ + rates &= ~SNDRV_PCM_RATE_KNOT; + + if (!rates) { + dev_err(codec->dev, "could not find a valid sample rate\n"); + return -EINVAL; + } + } else { + /* enable all possible rates */ + rates = STA32X_RATES; + rate_min = 32000; + rate_max = 192000; + } + + codec_dai->driver->playback.rates = rates; + codec_dai->driver->playback.rate_min = rate_min; + codec_dai->driver->playback.rate_max = rate_max; + return 0; +} + +/** + * sta32x_set_dai_fmt - configure the codec for the selected audio format + * @codec_dai: the codec DAI + * @fmt: a SND_SOC_DAIFMT_x value indicating the data format + * + * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the + * codec accordingly. + */ +static int sta32x_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + u8 confb = snd_soc_read(codec, STA32X_CONFB); + + pr_debug("\n"); + confb &= ~(STA32X_CONFB_C1IM | STA32X_CONFB_C2IM); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + sta32x->format = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + confb |= STA32X_CONFB_C2IM; + break; + case SND_SOC_DAIFMT_NB_IF: + confb |= STA32X_CONFB_C1IM; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, STA32X_CONFB, confb); + return 0; +} + +/** + * sta32x_hw_params - program the STA32X with the given hardware parameters. + * @substream: the audio stream + * @params: the hardware parameters to set + * @dai: the SOC DAI (ignored) + * + * This function programs the hardware with the values provided. + * Specifically, the sample rate and the data format. + */ +static int sta32x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + unsigned int rate; + int i, mcs = -1, ir = -1; + u8 confa, confb; + + rate = params_rate(params); + pr_debug("rate: %u\n", rate); + for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) + if (interpolation_ratios[i].fs == rate) + ir = interpolation_ratios[i].ir; + if (ir < 0) + return -EINVAL; + for (i = 0; mclk_ratios[ir][i].ratio; i++) + if (mclk_ratios[ir][i].ratio * rate == sta32x->mclk) + mcs = mclk_ratios[ir][i].mcs; + if (mcs < 0) + return -EINVAL; + + confa = snd_soc_read(codec, STA32X_CONFA); + confa &= ~(STA32X_CONFA_MCS_MASK | STA32X_CONFA_IR_MASK); + confa |= (ir << STA32X_CONFA_IR_SHIFT) | (mcs << STA32X_CONFA_MCS_SHIFT); + + confb = snd_soc_read(codec, STA32X_CONFB); + confb &= ~(STA32X_CONFB_SAI_MASK | STA32X_CONFB_SAIFB); + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S24_BE: + case SNDRV_PCM_FORMAT_S24_3LE: + case SNDRV_PCM_FORMAT_S24_3BE: + pr_debug("24bit\n"); + /* fall through */ + case SNDRV_PCM_FORMAT_S32_LE: + case SNDRV_PCM_FORMAT_S32_BE: + pr_debug("24bit or 32bit\n"); + switch (sta32x->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x0; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x1; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0x2; + break; + } + + break; + case SNDRV_PCM_FORMAT_S20_3LE: + case SNDRV_PCM_FORMAT_S20_3BE: + pr_debug("20bit\n"); + switch (sta32x->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x4; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x5; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0x6; + break; + } + + break; + case SNDRV_PCM_FORMAT_S18_3LE: + case SNDRV_PCM_FORMAT_S18_3BE: + pr_debug("18bit\n"); + switch (sta32x->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x8; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x9; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0xa; + break; + } + + break; + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S16_BE: + pr_debug("16bit\n"); + switch (sta32x->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x0; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0xd; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0xe; + break; + } + + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, STA32X_CONFA, confa); + snd_soc_write(codec, STA32X_CONFB, confb); + return 0; +} + +/** + * sta32x_set_bias_level - DAPM callback + * @codec: the codec device + * @level: DAPM power level + * + * This is called by ALSA to put the codec into low power mode + * or to wake it up. If the codec is powered off completely + * all registers must be restored after power on. + */ +static int sta32x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int ret; + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + + pr_debug("level = %d\n", level); + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* Full power on */ + snd_soc_update_bits(codec, STA32X_CONFF, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), + sta32x->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", ret); + return ret; + } + + snd_soc_cache_sync(codec); + } + + /* Power up to mute */ + /* FIXME */ + snd_soc_update_bits(codec, STA32X_CONFF, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD); + + break; + + case SND_SOC_BIAS_OFF: + /* The chip runs through the power down sequence for us. */ + snd_soc_update_bits(codec, STA32X_CONFF, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD, + STA32X_CONFF_PWDN); + msleep(300); + + regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), + sta32x->supplies); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static struct snd_soc_dai_ops sta32x_dai_ops = { + .hw_params = sta32x_hw_params, + .set_sysclk = sta32x_set_dai_sysclk, + .set_fmt = sta32x_set_dai_fmt, +}; + +static struct snd_soc_dai_driver sta32x_dai = { + .name = "STA32X", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = STA32X_RATES, + .formats = STA32X_FORMATS, + }, + .ops = &sta32x_dai_ops, +}; + +#ifdef CONFIG_PM +static int sta32x_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int sta32x_resume(struct snd_soc_codec *codec) +{ + sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} +#else +#define sta32x_suspend NULL +#define sta32x_resume NULL +#endif + +static int sta32x_probe(struct snd_soc_codec *codec) +{ + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + int i, ret = 0; + + sta32x->codec = codec; + + /* regulators */ + for (i = 0; i < ARRAY_SIZE(sta32x->supplies); i++) + sta32x->supplies[i].supply = sta32x_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sta32x->supplies), + sta32x->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), + sta32x->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_get; + } + + /* Tell ASoC what kind of I/O to use to read the registers. ASoC will + * then do the I2C transactions itself. + */ + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret); + return ret; + } + + /* read reg reset values into cache */ + for (i = 0; i < STA32X_REGISTER_COUNT; i++) + snd_soc_cache_write(codec, i, sta32x_regs[i]); + + /* FIXME enable thermal warning adjustment and recovery */ + snd_soc_update_bits(codec, STA32X_CONFA, + STA32X_CONFA_TWAB | STA32X_CONFA_TWRB, 0); + + /* FIXME select 2.1 mode */ + snd_soc_update_bits(codec, STA32X_CONFF, + STA32X_CONFF_OCFG_MASK, + 1 << STA32X_CONFF_OCFG_SHIFT); + + /* FIXME channel to output mapping */ + snd_soc_update_bits(codec, STA32X_C1CFG, + STA32X_CxCFG_OM_MASK, + 0 << STA32X_CxCFG_OM_SHIFT); + snd_soc_update_bits(codec, STA32X_C2CFG, + STA32X_CxCFG_OM_MASK, + 1 << STA32X_CxCFG_OM_SHIFT); + snd_soc_update_bits(codec, STA32X_C3CFG, + STA32X_CxCFG_OM_MASK, + 2 << STA32X_CxCFG_OM_SHIFT); + + sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Bias level configuration will have done an extra enable */ + regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + return 0; + +err_get: + regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); +err: + return ret; +} + +static int sta32x_remove(struct snd_soc_codec *codec) +{ + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + + regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + return 0; +} + +static const struct snd_soc_codec_driver sta32x_codec = { + .probe = sta32x_probe, + .remove = sta32x_remove, + .suspend = sta32x_suspend, + .resume = sta32x_resume, + .reg_cache_size = STA32X_REGISTER_COUNT, + .reg_word_size = sizeof(u8), + .set_bias_level = sta32x_set_bias_level, + .controls = sta32x_snd_controls, + .num_controls = ARRAY_SIZE(sta32x_snd_controls), + .dapm_widgets = sta32x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sta32x_dapm_widgets), + .dapm_routes = sta32x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(sta32x_dapm_routes), +}; + +static __devinit int sta32x_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct sta32x_priv *sta32x; + int ret; + + sta32x = kzalloc(sizeof(struct sta32x_priv), GFP_KERNEL); + if (!sta32x) + return -ENOMEM; + + i2c_set_clientdata(i2c, sta32x); + + ret = snd_soc_register_codec(&i2c->dev, &sta32x_codec, &sta32x_dai, 1); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret); + return ret; + } + + return 0; +} + +static __devexit int sta32x_i2c_remove(struct i2c_client *client) +{ + struct sta32x_priv *sta32x = i2c_get_clientdata(client); + struct snd_soc_codec *codec = sta32x->codec; + + if (codec) + sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); + + regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + if (codec) { + snd_soc_unregister_codec(&client->dev); + snd_soc_codec_set_drvdata(codec, NULL); + } + + kfree(sta32x); + return 0; +} + +static const struct i2c_device_id sta32x_i2c_id[] = { + { "sta326", 0 }, + { "sta328", 0 }, + { "sta329", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, sta32x_i2c_id); + +static struct i2c_driver sta32x_i2c_driver = { + .driver = { + .name = "sta32x", + .owner = THIS_MODULE, + }, + .probe = sta32x_i2c_probe, + .remove = __devexit_p(sta32x_i2c_remove), + .id_table = sta32x_i2c_id, +}; + +static int __init sta32x_init(void) +{ + return i2c_add_driver(&sta32x_i2c_driver); +} +module_init(sta32x_init); + +static void __exit sta32x_exit(void) +{ + i2c_del_driver(&sta32x_i2c_driver); +} +module_exit(sta32x_exit); + +MODULE_DESCRIPTION("ASoC STA32X driver"); +MODULE_AUTHOR("Johannes Stezenbach <js@sig21.net>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sta32x.h b/sound/soc/codecs/sta32x.h new file mode 100644 index 0000000..b97ee5a --- /dev/null +++ b/sound/soc/codecs/sta32x.h @@ -0,0 +1,210 @@ +/* + * Codec driver for ST STA32x 2.1-channel high-efficiency digital audio system + * + * Copyright: 2011 Raumfeld GmbH + * Author: Johannes Stezenbach <js@sig21.net> + * + * based on code from: + * Wolfson Microelectronics PLC. + * Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#ifndef _ASOC_STA_32X_H +#define _ASOC_STA_32X_H + +/* STA326 register addresses */ + +#define STA32X_REGISTER_COUNT 0x2d + +#define STA32X_CONFA 0x00 +#define STA32X_CONFB 0x01 +#define STA32X_CONFC 0x02 +#define STA32X_CONFD 0x03 +#define STA32X_CONFE 0x04 +#define STA32X_CONFF 0x05 +#define STA32X_MMUTE 0x06 +#define STA32X_MVOL 0x07 +#define STA32X_C1VOL 0x08 +#define STA32X_C2VOL 0x09 +#define STA32X_C3VOL 0x0a +#define STA32X_AUTO1 0x0b +#define STA32X_AUTO2 0x0c +#define STA32X_AUTO3 0x0d +#define STA32X_C1CFG 0x0e +#define STA32X_C2CFG 0x0f +#define STA32X_C3CFG 0x10 +#define STA32X_TONE 0x11 +#define STA32X_L1AR 0x12 +#define STA32X_L1ATRT 0x13 +#define STA32X_L2AR 0x14 +#define STA32X_L2ATRT 0x15 +#define STA32X_CFADDR2 0x16 +#define STA32X_B1CF1 0x17 +#define STA32X_B1CF2 0x18 +#define STA32X_B1CF3 0x19 +#define STA32X_B2CF1 0x1a +#define STA32X_B2CF2 0x1b +#define STA32X_B2CF3 0x1c +#define STA32X_A1CF1 0x1d +#define STA32X_A1CF2 0x1e +#define STA32X_A1CF3 0x1f +#define STA32X_A2CF1 0x20 +#define STA32X_A2CF2 0x21 +#define STA32X_A2CF3 0x22 +#define STA32X_B0CF1 0x23 +#define STA32X_B0CF2 0x24 +#define STA32X_B0CF3 0x25 +#define STA32X_CFUD 0x26 +#define STA32X_MPCC1 0x27 +#define STA32X_MPCC2 0x28 +/* Reserved 0x29 */ +/* Reserved 0x2a */ +#define STA32X_Reserved 0x2a +#define STA32X_FDRC1 0x2b +#define STA32X_FDRC2 0x2c +/* Reserved 0x2d */ + + +/* STA326 register field definitions */ + +/* 0x00 CONFA */ +#define STA32X_CONFA_MCS_MASK 0x03 +#define STA32X_CONFA_MCS_SHIFT 0 +#define STA32X_CONFA_IR_MASK 0x18 +#define STA32X_CONFA_IR_SHIFT 3 +#define STA32X_CONFA_TWRB 0x20 +#define STA32X_CONFA_TWAB 0x40 +#define STA32X_CONFA_FDRB 0x80 + +/* 0x01 CONFB */ +#define STA32X_CONFB_SAI_MASK 0x0f +#define STA32X_CONFB_SAI_SHIFT 0 +#define STA32X_CONFB_SAIFB 0x10 +#define STA32X_CONFB_DSCKE 0x20 +#define STA32X_CONFB_C1IM 0x40 +#define STA32X_CONFB_C2IM 0x80 + +/* 0x02 CONFC */ +#define STA32X_CONFC_OM_MASK 0x03 +#define STA32X_CONFC_OM_SHIFT 0 +#define STA32X_CONFC_CSZ_MASK 0x7c +#define STA32X_CONFC_CSZ_SHIFT 2 + +/* 0x03 CONFD */ +#define STA32X_CONFD_HPB 0x01 +#define STA32X_CONFD_HPB_SHIFT 0 +#define STA32X_CONFD_DEMP 0x02 +#define STA32X_CONFD_DEMP_SHIFT 1 +#define STA32X_CONFD_DSPB 0x04 +#define STA32X_CONFD_DSPB_SHIFT 2 +#define STA32X_CONFD_PSL 0x08 +#define STA32X_CONFD_PSL_SHIFT 3 +#define STA32X_CONFD_BQL 0x10 +#define STA32X_CONFD_BQL_SHIFT 4 +#define STA32X_CONFD_DRC 0x20 +#define STA32X_CONFD_DRC_SHIFT 5 +#define STA32X_CONFD_ZDE 0x40 +#define STA32X_CONFD_ZDE_SHIFT 6 +#define STA32X_CONFD_MME 0x80 +#define STA32X_CONFD_MME_SHIFT 7 + +/* 0x04 CONFE */ +#define STA32X_CONFE_MPCV 0x01 +#define STA32X_CONFE_MPCV_SHIFT 0 +#define STA32X_CONFE_MPC 0x02 +#define STA32X_CONFE_MPC_SHIFT 1 +#define STA32X_CONFE_AME 0x08 +#define STA32X_CONFE_AME_SHIFT 3 +#define STA32X_CONFE_PWMS 0x10 +#define STA32X_CONFE_PWMS_SHIFT 4 +#define STA32X_CONFE_ZCE 0x40 +#define STA32X_CONFE_ZCE_SHIFT 6 +#define STA32X_CONFE_SVE 0x80 +#define STA32X_CONFE_SVE_SHIFT 7 + +/* 0x05 CONFF */ +#define STA32X_CONFF_OCFG_MASK 0x03 +#define STA32X_CONFF_OCFG_SHIFT 0 +#define STA32X_CONFF_IDE 0x04 +#define STA32X_CONFF_IDE_SHIFT 3 +#define STA32X_CONFF_BCLE 0x08 +#define STA32X_CONFF_ECLE 0x20 +#define STA32X_CONFF_PWDN 0x40 +#define STA32X_CONFF_EAPD 0x80 + +/* 0x06 MMUTE */ +#define STA32X_MMUTE_MMUTE 0x01 + +/* 0x0b AUTO1 */ +#define STA32X_AUTO1_AMEQ_MASK 0x03 +#define STA32X_AUTO1_AMEQ_SHIFT 0 +#define STA32X_AUTO1_AMV_MASK 0xc0 +#define STA32X_AUTO1_AMV_SHIFT 2 +#define STA32X_AUTO1_AMGC_MASK 0x30 +#define STA32X_AUTO1_AMGC_SHIFT 4 +#define STA32X_AUTO1_AMPS 0x80 + +/* 0x0c AUTO2 */ +#define STA32X_AUTO2_AMAME 0x01 +#define STA32X_AUTO2_AMAM_MASK 0x0e +#define STA32X_AUTO2_AMAM_SHIFT 1 +#define STA32X_AUTO2_XO_MASK 0xf0 +#define STA32X_AUTO2_XO_SHIFT 4 + +/* 0x0d AUTO3 */ +#define STA32X_AUTO3_PEQ_MASK 0x1f +#define STA32X_AUTO3_PEQ_SHIFT 0 + +/* 0x0e 0x0f 0x10 CxCFG */ +#define STA32X_CxCFG_TCB 0x01 /* only C1 and C2 */ +#define STA32X_CxCFG_TCB_SHIFT 0 +#define STA32X_CxCFG_EQBP 0x02 /* only C1 and C2 */ +#define STA32X_CxCFG_EQBP_SHIFT 1 +#define STA32X_CxCFG_VBP 0x03 +#define STA32X_CxCFG_VBP_SHIFT 2 +#define STA32X_CxCFG_BO 0x04 +#define STA32X_CxCFG_LS_MASK 0x30 +#define STA32X_CxCFG_LS_SHIFT 4 +#define STA32X_CxCFG_OM_MASK 0xc0 +#define STA32X_CxCFG_OM_SHIFT 6 + +/* 0x11 TONE */ +#define STA32X_TONE_BTC_SHIFT 0 +#define STA32X_TONE_TTC_SHIFT 4 + +/* 0x12 0x13 0x14 0x15 limiter attack/release */ +#define STA32X_LxA_SHIFT 0 +#define STA32X_LxR_SHIFT 4 + +/* 0x26 CFUD */ +#define STA32X_CFUD_W1 0x01 +#define STA32X_CFUD_WA 0x02 +#define STA32X_CFUD_R1 0x04 +#define STA32X_CFUD_RA 0x08 + + +/* biquad filter coefficient table offsets */ +#define STA32X_C1_BQ_BASE 0 +#define STA32X_C2_BQ_BASE 20 +#define STA32X_CH_BQ_NUM 4 +#define STA32X_BQ_NUM_COEF 5 +#define STA32X_XO_HP_BQ_BASE 40 +#define STA32X_XO_LP_BQ_BASE 45 +#define STA32X_C1_PRESCALE 50 +#define STA32X_C2_PRESCALE 51 +#define STA32X_C1_POSTSCALE 52 +#define STA32X_C2_POSTSCALE 53 +#define STA32X_C3_POSTSCALE 54 +#define STA32X_TW_POSTSCALE 55 +#define STA32X_C1_MIX1 56 +#define STA32X_C1_MIX2 57 +#define STA32X_C2_MIX1 58 +#define STA32X_C2_MIX2 59 +#define STA32X_C3_MIX1 60 +#define STA32X_C3_MIX2 61 + +#endif /* _ASOC_STA_32X_H */ -- 1.7.5.1 ^ permalink raw reply related [flat|nested] 38+ messages in thread
* Re: [PATCH 1/2] ALSA: ASoC: add STA32X codec driver 2011-06-16 8:19 ` Daniel Mack @ 2011-06-16 8:49 ` Liam Girdwood 2011-06-16 8:53 ` Daniel Mack 0 siblings, 1 reply; 38+ messages in thread From: Liam Girdwood @ 2011-06-16 8:49 UTC (permalink / raw) To: Daniel Mack; +Cc: alsa-devel, broonie, lrg On 16/06/11 09:19, Daniel Mack wrote: > From: Johannes Stezenbach <js@sig21.net> > > Signed-off-by: Johannes Stezenbach <js@sig21.net> > [zonque@gmail.com: transform to new ASoC structure] > --- > sound/soc/codecs/Kconfig | 4 + > sound/soc/codecs/Makefile | 2 + > sound/soc/codecs/sta32x.c | 796 +++++++++++++++++++++++++++++++++++++++++++++ > sound/soc/codecs/sta32x.h | 210 ++++++++++++ > 4 files changed, 1012 insertions(+), 0 deletions(-) > create mode 100644 sound/soc/codecs/sta32x.c > create mode 100644 sound/soc/codecs/sta32x.h Is this intended for review only or upstream as there is a lot of code behind #if 0 and a lots of FIXMEs ? Liam ^ permalink raw reply [flat|nested] 38+ messages in thread
* Re: [PATCH 1/2] ALSA: ASoC: add STA32X codec driver 2011-06-16 8:49 ` Liam Girdwood @ 2011-06-16 8:53 ` Daniel Mack 2011-06-16 12:07 ` [PATCH] " Daniel Mack 0 siblings, 1 reply; 38+ messages in thread From: Daniel Mack @ 2011-06-16 8:53 UTC (permalink / raw) To: Liam Girdwood; +Cc: alsa-devel, broonie, lrg On Thu, Jun 16, 2011 at 10:49 AM, Liam Girdwood <lrg@ti.com> wrote: > On 16/06/11 09:19, Daniel Mack wrote: >> From: Johannes Stezenbach <js@sig21.net> >> >> Signed-off-by: Johannes Stezenbach <js@sig21.net> >> [zonque@gmail.com: transform to new ASoC structure] >> --- >> sound/soc/codecs/Kconfig | 4 + >> sound/soc/codecs/Makefile | 2 + >> sound/soc/codecs/sta32x.c | 796 +++++++++++++++++++++++++++++++++++++++++++++ >> sound/soc/codecs/sta32x.h | 210 ++++++++++++ >> 4 files changed, 1012 insertions(+), 0 deletions(-) >> create mode 100644 sound/soc/codecs/sta32x.c >> create mode 100644 sound/soc/codecs/sta32x.h > > Is this intended for review only or upstream as there is a lot of code behind #if 0 and a lots of FIXMEs ? It is sure intended for upstream. I'll fix the TLV issues and remove the #if0. The code runs fine on a board here, and the FIXMEs are just markers for future improvements :) As I said in my first email, the code was not written by me. I just care to bring it mainline. Thanks for the reviews, everyone. Daniel ^ permalink raw reply [flat|nested] 38+ messages in thread
* [PATCH] ALSA: ASoC: add STA32X codec driver 2011-06-16 8:53 ` Daniel Mack @ 2011-06-16 12:07 ` Daniel Mack 2011-06-17 8:30 ` Daniel Mack 2011-06-17 9:45 ` Mark Brown 0 siblings, 2 replies; 38+ messages in thread From: Daniel Mack @ 2011-06-16 12:07 UTC (permalink / raw) To: alsa-devel; +Cc: broonie, lrg From: Johannes Stezenbach <js@sig21.net> Signed-off-by: Johannes Stezenbach <js@sig21.net> [zonque@gmail.com: transform to new ASoC structure] --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sta32x.c | 778 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/sta32x.h | 210 ++++++++++++ 4 files changed, 994 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/sta32x.c create mode 100644 sound/soc/codecs/sta32x.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 98175a0..dd075f2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -42,6 +42,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI + select SND_SOC_STA32X if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER @@ -216,6 +217,9 @@ config SND_SOC_SPDIF config SND_SOC_SSM2602 tristate +config SND_SOC_STA32X + tristate + config SND_SOC_STAC9766 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fd85584..2ad1310 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -28,6 +28,7 @@ snd-soc-alc5623-objs := alc5623.o snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-sta32x-objs := sta32x.o snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o @@ -120,6 +121,7 @@ obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(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 diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c new file mode 100644 index 0000000..0fe38975 --- /dev/null +++ b/sound/soc/codecs/sta32x.c @@ -0,0 +1,778 @@ +/* + * Codec driver for ST STA32x 2.1-channel high-efficiency digital audio system + * + * Copyright: 2011 Raumfeld GmbH + * Author: Johannes Stezenbach <js@sig21.net> + * + * based on code from: + * Wolfson Microelectronics PLC. + * Mark Brown <broonie@opensource.wolfsonmicro.com> + * Freescale Semiconductor, Inc. + * Timur Tabi <timur@freescale.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#define DEBUG +#define pr_fmt(fmt) KBUILD_MODNAME ":%s:%d: " fmt, __func__, __LINE__ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "sta32x.h" + +#define STA32X_RATES (SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000) + +#define STA32X_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE) + +/* Power-up register defaults */ +static const u8 sta32x_regs[STA32X_REGISTER_COUNT] = { + 0x63, 0x80, 0xc2, 0x40, 0xc2, 0x5c, 0x10, 0xff, 0x60, 0x60, + 0x60, 0x80, 0x00, 0x00, 0x00, 0x40, 0x80, 0x77, 0x6a, 0x69, + 0x6a, 0x69, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x2d, + 0xc0, 0xf3, 0x33, 0x00, 0x0c, +}; + +/* regulator power supply names */ +static const char *sta32x_supply_names[] = { + "Vdda", /* analog supply, 3.3VV */ + "Vdd3", /* digital supply, 3.3V */ + "Vcc" /* power amp spply, 10V - 36V */ +}; + +/* codec private data */ +struct sta32x_priv { + struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)]; + struct snd_soc_codec *codec; + + unsigned int mclk; + unsigned int format; +}; + +static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12700, 50, 1); +static const DECLARE_TLV_DB_SCALE(chvol_tlv, -7950, 50, 1); +static const DECLARE_TLV_DB_SCALE(tone_tlv, -120, 200, 0); + +static const char *sta32x_drc_ac[] = { + "Anti-Clipping", "Dynamic Range Compression" }; +static const char *sta32x_auto_eq_mode[] = { + "User", "Preset", "Loudness" }; +static const char *sta32x_auto_gc_mode[] = { + "User", "AC no clipping", "AC limited clipping (10%)", + "DRC nighttime listening mode" }; +static const char *sta32x_auto_xo_mode[] = { + "User", "80Hz", "100Hz", "120Hz", "140Hz", "160Hz", "180Hz", "200Hz", + "220Hz", "240Hz", "260Hz", "280Hz", "300Hz", "320Hz", "340Hz", "360Hz" }; +static const char *sta32x_preset_eq_mode[] = { + "Flat", "Rock", "Soft Rock", "Jazz", "Classical", "Dance", "Pop", "Soft", + "Hard", "Party", "Vocal", "Hip-Hop", "Dialog", "Bass-boost #1", + "Bass-boost #2", "Bass-boost #3", "Loudness 1", "Loudness 2", + "Loudness 3", "Loudness 4", "Loudness 5", "Loudness 6", "Loudness 7", + "Loudness 8", "Loudness 9", "Loudness 10", "Loudness 11", "Loudness 12", + "Loudness 13", "Loudness 14", "Loudness 15", "Loudness 16" }; +static const char *sta32x_limiter_select[] = { + "Limiter Disabled", "Limiter #1", "Limiter #2" }; +static const char *sta32x_limiter_attack_rate[] = { + "3.1584", "2.7072", "2.2560", "1.8048", "1.3536", "0.9024", + "0.4512", "0.2256", "0.1504", "0.1123", "0.0902", "0.0752", + "0.0645", "0.0564", "0.0501", "0.0451" }; +static const char *sta32x_limiter_release_rate[] = { + "0.5116", "0.1370", "0.0744", "0.0499", "0.0360", "0.0299", + "0.0264", "0.0208", "0.0198", "0.0172", "0.0147", "0.0137", + "0.0134", "0.0117", "0.0110", "0.0104" }; + +static const unsigned int sta32x_limiter_ac_attack_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 7, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 8, 16, TLV_DB_SCALE_ITEM(300, 100, 0), +}; + +static const unsigned int sta32x_limiter_ac_release_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(-2900, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(-2000, 0, 0), + 3, 8, TLV_DB_SCALE_ITEM(-1400, 200, 0), + 8, 16, TLV_DB_SCALE_ITEM(-700, 100, 0), +}; + +static const unsigned int sta32x_limiter_drc_attack_tlv[] = { + TLV_DB_RANGE_HEAD(3), + 0, 7, TLV_DB_SCALE_ITEM(-3100, 200, 0), + 8, 13, TLV_DB_SCALE_ITEM(-1600, 100, 0), + 14, 16, TLV_DB_SCALE_ITEM(-1000, 300, 0), +}; + +static const unsigned int sta32x_limiter_drc_release_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0), + 1, 2, TLV_DB_SCALE_ITEM(-3800, 200, 0), + 3, 4, TLV_DB_SCALE_ITEM(-3300, 200, 0), + 5, 12, TLV_DB_SCALE_ITEM(-3000, 200, 0), + 13, 16, TLV_DB_SCALE_ITEM(-1500, 300, 0), +}; + +static const struct soc_enum sta32x_drc_ac_enum = + SOC_ENUM_SINGLE(STA32X_CONFD, STA32X_CONFD_DRC_SHIFT, + 2, sta32x_drc_ac); +static const struct soc_enum sta32x_auto_eq_enum = + SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMEQ_SHIFT, + 3, sta32x_auto_eq_mode); +static const struct soc_enum sta32x_auto_gc_enum = + SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMGC_SHIFT, + 4, sta32x_auto_gc_mode); +static const struct soc_enum sta32x_auto_xo_enum = + SOC_ENUM_SINGLE(STA32X_AUTO2, STA32X_AUTO2_XO_SHIFT, + 16, sta32x_auto_xo_mode); +static const struct soc_enum sta32x_preset_eq_enum = + SOC_ENUM_SINGLE(STA32X_AUTO3, STA32X_AUTO3_PEQ_SHIFT, + 32, sta32x_preset_eq_mode); +static const struct soc_enum sta32x_limiter_ch1_enum = + SOC_ENUM_SINGLE(STA32X_C1CFG, STA32X_CxCFG_LS_SHIFT, + 3, sta32x_limiter_select); +static const struct soc_enum sta32x_limiter_ch2_enum = + SOC_ENUM_SINGLE(STA32X_C2CFG, STA32X_CxCFG_LS_SHIFT, + 3, sta32x_limiter_select); +static const struct soc_enum sta32x_limiter_ch3_enum = + SOC_ENUM_SINGLE(STA32X_C3CFG, STA32X_CxCFG_LS_SHIFT, + 3, sta32x_limiter_select); +static const struct soc_enum sta32x_limiter1_attack_rate_enum = + SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxA_SHIFT, + 16, sta32x_limiter_attack_rate); +static const struct soc_enum sta32x_limiter2_attack_rate_enum = + SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxA_SHIFT, + 16, sta32x_limiter_attack_rate); +static const struct soc_enum sta32x_limiter1_release_rate_enum = + SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxR_SHIFT, + 16, sta32x_limiter_release_rate); +static const struct soc_enum sta32x_limiter2_release_rate_enum = + SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxR_SHIFT, + 16, sta32x_limiter_release_rate); +static const struct snd_kcontrol_new sta32x_snd_controls[] = { +SOC_SINGLE_TLV("Master Volume", STA32X_MVOL, 0, 0xff, 1, mvol_tlv), +SOC_SINGLE("Master Playback Switch", STA32X_MMUTE, 0, 1, 1), +SOC_SINGLE("Ch1 Playback Switch", STA32X_MMUTE, 1, 1, 1), +SOC_SINGLE("Ch2 Playback Switch", STA32X_MMUTE, 2, 1, 1), +SOC_SINGLE("Ch3 Playback Switch", STA32X_MMUTE, 3, 1, 1), +SOC_SINGLE_TLV("Ch1 Volume", STA32X_C1VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE_TLV("Ch2 Volume", STA32X_C2VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE_TLV("Ch3 Volume", STA32X_C3VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE("De-emphasis Filter Switch", STA32X_CONFD, STA32X_CONFD_DEMP_SHIFT, 1, 0), +SOC_ENUM("Compressor/Limiter Switch", sta32x_drc_ac_enum), +SOC_SINGLE("Miami Mode Switch", STA32X_CONFD, STA32X_CONFD_MME_SHIFT, 1, 0), +SOC_SINGLE("Zero Cross Switch", STA32X_CONFE, STA32X_CONFE_ZCE_SHIFT, 1, 0), +SOC_SINGLE("Soft Ramp Switch", STA32X_CONFE, STA32X_CONFE_SVE_SHIFT, 1, 0), +SOC_SINGLE("Auto-Mute Switch", STA32X_CONFF, STA32X_CONFF_IDE_SHIFT, 1, 0), +SOC_ENUM("Automode EQ", sta32x_auto_eq_enum), +SOC_ENUM("Automode GC", sta32x_auto_gc_enum), +SOC_ENUM("Automode XO", sta32x_auto_xo_enum), +SOC_ENUM("Preset EQ", sta32x_preset_eq_enum), +SOC_SINGLE("Ch1 Tone Control Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_TCB_SHIFT, 1, 0), +SOC_SINGLE("Ch2 Tone Control Bypass Switch", STA32X_C2CFG, STA32X_CxCFG_TCB_SHIFT, 1, 0), +SOC_SINGLE("Ch1 EQ Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_EQBP_SHIFT, 1, 0), +SOC_SINGLE("Ch2 EQ Bypass Switch", STA32X_C2CFG, STA32X_CxCFG_EQBP_SHIFT, 1, 0), +SOC_SINGLE("Ch1 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0), +SOC_SINGLE("Ch2 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0), +SOC_SINGLE("Ch3 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0), +SOC_ENUM("Ch1 Limiter Select", sta32x_limiter_ch1_enum), +SOC_ENUM("Ch2 Limiter Select", sta32x_limiter_ch2_enum), +SOC_ENUM("Ch3 Limiter Select", sta32x_limiter_ch3_enum), +SOC_SINGLE_TLV("Bass Tone Control", STA32X_TONE, STA32X_TONE_BTC_SHIFT, 15, 0, tone_tlv), +SOC_SINGLE_TLV("Treble Tone Control", STA32X_TONE, STA32X_TONE_TTC_SHIFT, 15, 0, tone_tlv), +SOC_ENUM("Limiter1 Attack Rate (dB/ms)", sta32x_limiter1_attack_rate_enum), +SOC_ENUM("Limiter2 Attack Rate (dB/ms)", sta32x_limiter2_attack_rate_enum), +SOC_ENUM("Limiter1 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum), +SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum), + +/* depending on mode, the attack/release thresholds have + * two different enum definitions; provide both + */ +SOC_SINGLE_TLV("Limiter1 Attack Threshold (AC Mode)", STA32X_L1ATRT, STA32X_LxA_SHIFT, + 16, 0, sta32x_limiter_ac_attack_tlv), +SOC_SINGLE_TLV("Limiter2 Attack Threshold (AC Mode)", STA32X_L2ATRT, STA32X_LxA_SHIFT, + 16, 0, sta32x_limiter_ac_attack_tlv), +SOC_SINGLE_TLV("Limiter1 Release Threshold (AC Mode)", STA32X_L1ATRT, STA32X_LxR_SHIFT, + 16, 0, sta32x_limiter_ac_release_tlv), +SOC_SINGLE_TLV("Limiter2 Release Threshold (AC Mode)", STA32X_L2ATRT, STA32X_LxR_SHIFT, + 16, 0, sta32x_limiter_ac_release_tlv), +SOC_SINGLE_TLV("Limiter1 Attack Threshold (DRC Mode)", STA32X_L1ATRT, STA32X_LxA_SHIFT, + 16, 0, sta32x_limiter_drc_attack_tlv), +SOC_SINGLE_TLV("Limiter2 Attack Threshold (DRC Mode)", STA32X_L2ATRT, STA32X_LxA_SHIFT, + 16, 0, sta32x_limiter_drc_attack_tlv), +SOC_SINGLE_TLV("Limiter1 Release Threshold (DRC Mode)", STA32X_L1ATRT, STA32X_LxR_SHIFT, + 16, 0, sta32x_limiter_drc_release_tlv), +SOC_SINGLE_TLV("Limiter2 Release Threshold (DRC Mode)", STA32X_L2ATRT, STA32X_LxR_SHIFT, + 16, 0, sta32x_limiter_drc_release_tlv), +}; + +static const struct snd_soc_dapm_widget sta32x_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_OUTPUT("LEFT"), +SND_SOC_DAPM_OUTPUT("RIGHT"), +SND_SOC_DAPM_OUTPUT("SUB"), +}; + +static const struct snd_soc_dapm_route sta32x_dapm_routes[] = { + { "LEFT", NULL, "DAC" }, + { "RIGHT", NULL, "DAC" }, + { "SUB", NULL, "DAC" }, +}; + +/* MCLK interpolation ratio per fs */ +static struct { + int fs; + int ir; +} interpolation_ratios[] = { + { 32000, 0 }, + { 44100, 0 }, + { 48000, 0 }, + { 88200, 1 }, + { 96000, 1 }, + { 176400, 2 }, + { 192000, 2 }, +}; + +/* MCLK to fs clock ratios */ +static struct { + int ratio; + int mcs; +} mclk_ratios[3][7] = { + { { 768, 0 }, { 512, 1 }, { 384, 2 }, { 256, 3 }, + { 128, 4 }, { 576, 5 }, { 0, 0 } }, + { { 384, 2 }, { 256, 3 }, { 192, 4 }, { 128, 5 }, {64, 0 }, { 0, 0 } }, + { { 384, 2 }, { 256, 3 }, { 192, 4 }, { 128, 5 }, {64, 0 }, { 0, 0 } }, +}; + + +/** + * sta32x_set_dai_sysclk - configure MCLK + * @codec_dai: the codec DAI + * @clk_id: the clock ID (ignored) + * @freq: the MCLK input frequency + * @dir: the clock direction (ignored) + * + * The value of MCLK is used to determine which sample rates are supported + * by the STA32X, based on the mclk_ratios table. + * + * This function must be called by the machine driver's 'startup' function, + * otherwise the list of supported sample rates will not be available in + * time for ALSA. + * + * For setups with variable MCLKs, pass 0 as 'freq' argument. This will cause + * theoretically possible sample rates to be enabled. Call it again with a + * proper value set one the external clock is set (most probably you would do + * that from a machine's driver 'hw_param' hook. + */ +static int sta32x_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 sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + int i, j, ir, fs; + unsigned int rates = 0; + unsigned int rate_min = -1; + unsigned int rate_max = 0; + + pr_debug("mclk=%u\n", freq); + sta32x->mclk = freq; + + if (sta32x->mclk) { + for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) { + ir = interpolation_ratios[i].ir; + fs = interpolation_ratios[i].fs; + for (j = 0; mclk_ratios[ir][j].ratio; j++) { + if (mclk_ratios[ir][j].ratio * fs == freq) { + rates |= snd_pcm_rate_to_rate_bit(fs); + if (fs < rate_min) + rate_min = fs; + if (fs > rate_max) + rate_max = fs; + } + } + } + /* FIXME: soc should support a rate list */ + rates &= ~SNDRV_PCM_RATE_KNOT; + + if (!rates) { + dev_err(codec->dev, "could not find a valid sample rate\n"); + return -EINVAL; + } + } else { + /* enable all possible rates */ + rates = STA32X_RATES; + rate_min = 32000; + rate_max = 192000; + } + + codec_dai->driver->playback.rates = rates; + codec_dai->driver->playback.rate_min = rate_min; + codec_dai->driver->playback.rate_max = rate_max; + return 0; +} + +/** + * sta32x_set_dai_fmt - configure the codec for the selected audio format + * @codec_dai: the codec DAI + * @fmt: a SND_SOC_DAIFMT_x value indicating the data format + * + * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the + * codec accordingly. + */ +static int sta32x_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + u8 confb = snd_soc_read(codec, STA32X_CONFB); + + pr_debug("\n"); + confb &= ~(STA32X_CONFB_C1IM | STA32X_CONFB_C2IM); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + sta32x->format = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + confb |= STA32X_CONFB_C2IM; + break; + case SND_SOC_DAIFMT_NB_IF: + confb |= STA32X_CONFB_C1IM; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, STA32X_CONFB, confb); + return 0; +} + +/** + * sta32x_hw_params - program the STA32X with the given hardware parameters. + * @substream: the audio stream + * @params: the hardware parameters to set + * @dai: the SOC DAI (ignored) + * + * This function programs the hardware with the values provided. + * Specifically, the sample rate and the data format. + */ +static int sta32x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + unsigned int rate; + int i, mcs = -1, ir = -1; + u8 confa, confb; + + rate = params_rate(params); + pr_debug("rate: %u\n", rate); + for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) + if (interpolation_ratios[i].fs == rate) + ir = interpolation_ratios[i].ir; + if (ir < 0) + return -EINVAL; + for (i = 0; mclk_ratios[ir][i].ratio; i++) + if (mclk_ratios[ir][i].ratio * rate == sta32x->mclk) + mcs = mclk_ratios[ir][i].mcs; + if (mcs < 0) + return -EINVAL; + + confa = snd_soc_read(codec, STA32X_CONFA); + confa &= ~(STA32X_CONFA_MCS_MASK | STA32X_CONFA_IR_MASK); + confa |= (ir << STA32X_CONFA_IR_SHIFT) | (mcs << STA32X_CONFA_MCS_SHIFT); + + confb = snd_soc_read(codec, STA32X_CONFB); + confb &= ~(STA32X_CONFB_SAI_MASK | STA32X_CONFB_SAIFB); + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S24_BE: + case SNDRV_PCM_FORMAT_S24_3LE: + case SNDRV_PCM_FORMAT_S24_3BE: + pr_debug("24bit\n"); + /* fall through */ + case SNDRV_PCM_FORMAT_S32_LE: + case SNDRV_PCM_FORMAT_S32_BE: + pr_debug("24bit or 32bit\n"); + switch (sta32x->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x0; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x1; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0x2; + break; + } + + break; + case SNDRV_PCM_FORMAT_S20_3LE: + case SNDRV_PCM_FORMAT_S20_3BE: + pr_debug("20bit\n"); + switch (sta32x->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x4; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x5; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0x6; + break; + } + + break; + case SNDRV_PCM_FORMAT_S18_3LE: + case SNDRV_PCM_FORMAT_S18_3BE: + pr_debug("18bit\n"); + switch (sta32x->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x8; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x9; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0xa; + break; + } + + break; + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S16_BE: + pr_debug("16bit\n"); + switch (sta32x->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x0; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0xd; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0xe; + break; + } + + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, STA32X_CONFA, confa); + snd_soc_write(codec, STA32X_CONFB, confb); + return 0; +} + +/** + * sta32x_set_bias_level - DAPM callback + * @codec: the codec device + * @level: DAPM power level + * + * This is called by ALSA to put the codec into low power mode + * or to wake it up. If the codec is powered off completely + * all registers must be restored after power on. + */ +static int sta32x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int ret; + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + + pr_debug("level = %d\n", level); + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* Full power on */ + snd_soc_update_bits(codec, STA32X_CONFF, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), + sta32x->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", ret); + return ret; + } + + snd_soc_cache_sync(codec); + } + + /* Power up to mute */ + /* FIXME */ + snd_soc_update_bits(codec, STA32X_CONFF, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD); + + break; + + case SND_SOC_BIAS_OFF: + /* The chip runs through the power down sequence for us. */ + snd_soc_update_bits(codec, STA32X_CONFF, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD, + STA32X_CONFF_PWDN); + msleep(300); + + regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), + sta32x->supplies); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static struct snd_soc_dai_ops sta32x_dai_ops = { + .hw_params = sta32x_hw_params, + .set_sysclk = sta32x_set_dai_sysclk, + .set_fmt = sta32x_set_dai_fmt, +}; + +static struct snd_soc_dai_driver sta32x_dai = { + .name = "STA32X", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = STA32X_RATES, + .formats = STA32X_FORMATS, + }, + .ops = &sta32x_dai_ops, +}; + +#ifdef CONFIG_PM +static int sta32x_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int sta32x_resume(struct snd_soc_codec *codec) +{ + sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} +#else +#define sta32x_suspend NULL +#define sta32x_resume NULL +#endif + +static int sta32x_probe(struct snd_soc_codec *codec) +{ + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + int i, ret = 0; + + sta32x->codec = codec; + + /* regulators */ + for (i = 0; i < ARRAY_SIZE(sta32x->supplies); i++) + sta32x->supplies[i].supply = sta32x_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sta32x->supplies), + sta32x->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), + sta32x->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_get; + } + + /* Tell ASoC what kind of I/O to use to read the registers. ASoC will + * then do the I2C transactions itself. + */ + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret); + return ret; + } + + /* read reg reset values into cache */ + for (i = 0; i < STA32X_REGISTER_COUNT; i++) + snd_soc_cache_write(codec, i, sta32x_regs[i]); + + /* FIXME enable thermal warning adjustment and recovery */ + snd_soc_update_bits(codec, STA32X_CONFA, + STA32X_CONFA_TWAB | STA32X_CONFA_TWRB, 0); + + /* FIXME select 2.1 mode */ + snd_soc_update_bits(codec, STA32X_CONFF, + STA32X_CONFF_OCFG_MASK, + 1 << STA32X_CONFF_OCFG_SHIFT); + + /* FIXME channel to output mapping */ + snd_soc_update_bits(codec, STA32X_C1CFG, + STA32X_CxCFG_OM_MASK, + 0 << STA32X_CxCFG_OM_SHIFT); + snd_soc_update_bits(codec, STA32X_C2CFG, + STA32X_CxCFG_OM_MASK, + 1 << STA32X_CxCFG_OM_SHIFT); + snd_soc_update_bits(codec, STA32X_C3CFG, + STA32X_CxCFG_OM_MASK, + 2 << STA32X_CxCFG_OM_SHIFT); + + sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Bias level configuration will have done an extra enable */ + regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + return 0; + +err_get: + regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); +err: + return ret; +} + +static int sta32x_remove(struct snd_soc_codec *codec) +{ + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + + regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + return 0; +} + +static const struct snd_soc_codec_driver sta32x_codec = { + .probe = sta32x_probe, + .remove = sta32x_remove, + .suspend = sta32x_suspend, + .resume = sta32x_resume, + .reg_cache_size = STA32X_REGISTER_COUNT, + .reg_word_size = sizeof(u8), + .set_bias_level = sta32x_set_bias_level, + .controls = sta32x_snd_controls, + .num_controls = ARRAY_SIZE(sta32x_snd_controls), + .dapm_widgets = sta32x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sta32x_dapm_widgets), + .dapm_routes = sta32x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(sta32x_dapm_routes), +}; + +static __devinit int sta32x_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct sta32x_priv *sta32x; + int ret; + + sta32x = kzalloc(sizeof(struct sta32x_priv), GFP_KERNEL); + if (!sta32x) + return -ENOMEM; + + i2c_set_clientdata(i2c, sta32x); + + ret = snd_soc_register_codec(&i2c->dev, &sta32x_codec, &sta32x_dai, 1); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret); + return ret; + } + + return 0; +} + +static __devexit int sta32x_i2c_remove(struct i2c_client *client) +{ + struct sta32x_priv *sta32x = i2c_get_clientdata(client); + struct snd_soc_codec *codec = sta32x->codec; + + if (codec) + sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); + + regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + if (codec) { + snd_soc_unregister_codec(&client->dev); + snd_soc_codec_set_drvdata(codec, NULL); + } + + kfree(sta32x); + return 0; +} + +static const struct i2c_device_id sta32x_i2c_id[] = { + { "sta326", 0 }, + { "sta328", 0 }, + { "sta329", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, sta32x_i2c_id); + +static struct i2c_driver sta32x_i2c_driver = { + .driver = { + .name = "sta32x", + .owner = THIS_MODULE, + }, + .probe = sta32x_i2c_probe, + .remove = __devexit_p(sta32x_i2c_remove), + .id_table = sta32x_i2c_id, +}; + +static int __init sta32x_init(void) +{ + return i2c_add_driver(&sta32x_i2c_driver); +} +module_init(sta32x_init); + +static void __exit sta32x_exit(void) +{ + i2c_del_driver(&sta32x_i2c_driver); +} +module_exit(sta32x_exit); + +MODULE_DESCRIPTION("ASoC STA32X driver"); +MODULE_AUTHOR("Johannes Stezenbach <js@sig21.net>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sta32x.h b/sound/soc/codecs/sta32x.h new file mode 100644 index 0000000..b97ee5a --- /dev/null +++ b/sound/soc/codecs/sta32x.h @@ -0,0 +1,210 @@ +/* + * Codec driver for ST STA32x 2.1-channel high-efficiency digital audio system + * + * Copyright: 2011 Raumfeld GmbH + * Author: Johannes Stezenbach <js@sig21.net> + * + * based on code from: + * Wolfson Microelectronics PLC. + * Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#ifndef _ASOC_STA_32X_H +#define _ASOC_STA_32X_H + +/* STA326 register addresses */ + +#define STA32X_REGISTER_COUNT 0x2d + +#define STA32X_CONFA 0x00 +#define STA32X_CONFB 0x01 +#define STA32X_CONFC 0x02 +#define STA32X_CONFD 0x03 +#define STA32X_CONFE 0x04 +#define STA32X_CONFF 0x05 +#define STA32X_MMUTE 0x06 +#define STA32X_MVOL 0x07 +#define STA32X_C1VOL 0x08 +#define STA32X_C2VOL 0x09 +#define STA32X_C3VOL 0x0a +#define STA32X_AUTO1 0x0b +#define STA32X_AUTO2 0x0c +#define STA32X_AUTO3 0x0d +#define STA32X_C1CFG 0x0e +#define STA32X_C2CFG 0x0f +#define STA32X_C3CFG 0x10 +#define STA32X_TONE 0x11 +#define STA32X_L1AR 0x12 +#define STA32X_L1ATRT 0x13 +#define STA32X_L2AR 0x14 +#define STA32X_L2ATRT 0x15 +#define STA32X_CFADDR2 0x16 +#define STA32X_B1CF1 0x17 +#define STA32X_B1CF2 0x18 +#define STA32X_B1CF3 0x19 +#define STA32X_B2CF1 0x1a +#define STA32X_B2CF2 0x1b +#define STA32X_B2CF3 0x1c +#define STA32X_A1CF1 0x1d +#define STA32X_A1CF2 0x1e +#define STA32X_A1CF3 0x1f +#define STA32X_A2CF1 0x20 +#define STA32X_A2CF2 0x21 +#define STA32X_A2CF3 0x22 +#define STA32X_B0CF1 0x23 +#define STA32X_B0CF2 0x24 +#define STA32X_B0CF3 0x25 +#define STA32X_CFUD 0x26 +#define STA32X_MPCC1 0x27 +#define STA32X_MPCC2 0x28 +/* Reserved 0x29 */ +/* Reserved 0x2a */ +#define STA32X_Reserved 0x2a +#define STA32X_FDRC1 0x2b +#define STA32X_FDRC2 0x2c +/* Reserved 0x2d */ + + +/* STA326 register field definitions */ + +/* 0x00 CONFA */ +#define STA32X_CONFA_MCS_MASK 0x03 +#define STA32X_CONFA_MCS_SHIFT 0 +#define STA32X_CONFA_IR_MASK 0x18 +#define STA32X_CONFA_IR_SHIFT 3 +#define STA32X_CONFA_TWRB 0x20 +#define STA32X_CONFA_TWAB 0x40 +#define STA32X_CONFA_FDRB 0x80 + +/* 0x01 CONFB */ +#define STA32X_CONFB_SAI_MASK 0x0f +#define STA32X_CONFB_SAI_SHIFT 0 +#define STA32X_CONFB_SAIFB 0x10 +#define STA32X_CONFB_DSCKE 0x20 +#define STA32X_CONFB_C1IM 0x40 +#define STA32X_CONFB_C2IM 0x80 + +/* 0x02 CONFC */ +#define STA32X_CONFC_OM_MASK 0x03 +#define STA32X_CONFC_OM_SHIFT 0 +#define STA32X_CONFC_CSZ_MASK 0x7c +#define STA32X_CONFC_CSZ_SHIFT 2 + +/* 0x03 CONFD */ +#define STA32X_CONFD_HPB 0x01 +#define STA32X_CONFD_HPB_SHIFT 0 +#define STA32X_CONFD_DEMP 0x02 +#define STA32X_CONFD_DEMP_SHIFT 1 +#define STA32X_CONFD_DSPB 0x04 +#define STA32X_CONFD_DSPB_SHIFT 2 +#define STA32X_CONFD_PSL 0x08 +#define STA32X_CONFD_PSL_SHIFT 3 +#define STA32X_CONFD_BQL 0x10 +#define STA32X_CONFD_BQL_SHIFT 4 +#define STA32X_CONFD_DRC 0x20 +#define STA32X_CONFD_DRC_SHIFT 5 +#define STA32X_CONFD_ZDE 0x40 +#define STA32X_CONFD_ZDE_SHIFT 6 +#define STA32X_CONFD_MME 0x80 +#define STA32X_CONFD_MME_SHIFT 7 + +/* 0x04 CONFE */ +#define STA32X_CONFE_MPCV 0x01 +#define STA32X_CONFE_MPCV_SHIFT 0 +#define STA32X_CONFE_MPC 0x02 +#define STA32X_CONFE_MPC_SHIFT 1 +#define STA32X_CONFE_AME 0x08 +#define STA32X_CONFE_AME_SHIFT 3 +#define STA32X_CONFE_PWMS 0x10 +#define STA32X_CONFE_PWMS_SHIFT 4 +#define STA32X_CONFE_ZCE 0x40 +#define STA32X_CONFE_ZCE_SHIFT 6 +#define STA32X_CONFE_SVE 0x80 +#define STA32X_CONFE_SVE_SHIFT 7 + +/* 0x05 CONFF */ +#define STA32X_CONFF_OCFG_MASK 0x03 +#define STA32X_CONFF_OCFG_SHIFT 0 +#define STA32X_CONFF_IDE 0x04 +#define STA32X_CONFF_IDE_SHIFT 3 +#define STA32X_CONFF_BCLE 0x08 +#define STA32X_CONFF_ECLE 0x20 +#define STA32X_CONFF_PWDN 0x40 +#define STA32X_CONFF_EAPD 0x80 + +/* 0x06 MMUTE */ +#define STA32X_MMUTE_MMUTE 0x01 + +/* 0x0b AUTO1 */ +#define STA32X_AUTO1_AMEQ_MASK 0x03 +#define STA32X_AUTO1_AMEQ_SHIFT 0 +#define STA32X_AUTO1_AMV_MASK 0xc0 +#define STA32X_AUTO1_AMV_SHIFT 2 +#define STA32X_AUTO1_AMGC_MASK 0x30 +#define STA32X_AUTO1_AMGC_SHIFT 4 +#define STA32X_AUTO1_AMPS 0x80 + +/* 0x0c AUTO2 */ +#define STA32X_AUTO2_AMAME 0x01 +#define STA32X_AUTO2_AMAM_MASK 0x0e +#define STA32X_AUTO2_AMAM_SHIFT 1 +#define STA32X_AUTO2_XO_MASK 0xf0 +#define STA32X_AUTO2_XO_SHIFT 4 + +/* 0x0d AUTO3 */ +#define STA32X_AUTO3_PEQ_MASK 0x1f +#define STA32X_AUTO3_PEQ_SHIFT 0 + +/* 0x0e 0x0f 0x10 CxCFG */ +#define STA32X_CxCFG_TCB 0x01 /* only C1 and C2 */ +#define STA32X_CxCFG_TCB_SHIFT 0 +#define STA32X_CxCFG_EQBP 0x02 /* only C1 and C2 */ +#define STA32X_CxCFG_EQBP_SHIFT 1 +#define STA32X_CxCFG_VBP 0x03 +#define STA32X_CxCFG_VBP_SHIFT 2 +#define STA32X_CxCFG_BO 0x04 +#define STA32X_CxCFG_LS_MASK 0x30 +#define STA32X_CxCFG_LS_SHIFT 4 +#define STA32X_CxCFG_OM_MASK 0xc0 +#define STA32X_CxCFG_OM_SHIFT 6 + +/* 0x11 TONE */ +#define STA32X_TONE_BTC_SHIFT 0 +#define STA32X_TONE_TTC_SHIFT 4 + +/* 0x12 0x13 0x14 0x15 limiter attack/release */ +#define STA32X_LxA_SHIFT 0 +#define STA32X_LxR_SHIFT 4 + +/* 0x26 CFUD */ +#define STA32X_CFUD_W1 0x01 +#define STA32X_CFUD_WA 0x02 +#define STA32X_CFUD_R1 0x04 +#define STA32X_CFUD_RA 0x08 + + +/* biquad filter coefficient table offsets */ +#define STA32X_C1_BQ_BASE 0 +#define STA32X_C2_BQ_BASE 20 +#define STA32X_CH_BQ_NUM 4 +#define STA32X_BQ_NUM_COEF 5 +#define STA32X_XO_HP_BQ_BASE 40 +#define STA32X_XO_LP_BQ_BASE 45 +#define STA32X_C1_PRESCALE 50 +#define STA32X_C2_PRESCALE 51 +#define STA32X_C1_POSTSCALE 52 +#define STA32X_C2_POSTSCALE 53 +#define STA32X_C3_POSTSCALE 54 +#define STA32X_TW_POSTSCALE 55 +#define STA32X_C1_MIX1 56 +#define STA32X_C1_MIX2 57 +#define STA32X_C2_MIX1 58 +#define STA32X_C2_MIX2 59 +#define STA32X_C3_MIX1 60 +#define STA32X_C3_MIX2 61 + +#endif /* _ASOC_STA_32X_H */ -- 1.7.5.1 ^ permalink raw reply related [flat|nested] 38+ messages in thread
* Re: [PATCH] ALSA: ASoC: add STA32X codec driver 2011-06-16 12:07 ` [PATCH] " Daniel Mack @ 2011-06-17 8:30 ` Daniel Mack 2011-06-17 9:45 ` Mark Brown 1 sibling, 0 replies; 38+ messages in thread From: Daniel Mack @ 2011-06-17 8:30 UTC (permalink / raw) To: alsa-devel; +Cc: broonie, lrg Is that version good to go in or is there anything left you want me correct? Thanks, Daniel On Thu, Jun 16, 2011 at 2:07 PM, Daniel Mack <zonque@gmail.com> wrote: > From: Johannes Stezenbach <js@sig21.net> > > Signed-off-by: Johannes Stezenbach <js@sig21.net> > [zonque@gmail.com: transform to new ASoC structure] > --- > sound/soc/codecs/Kconfig | 4 + > sound/soc/codecs/Makefile | 2 + > sound/soc/codecs/sta32x.c | 778 +++++++++++++++++++++++++++++++++++++++++++++ > sound/soc/codecs/sta32x.h | 210 ++++++++++++ > 4 files changed, 994 insertions(+), 0 deletions(-) > create mode 100644 sound/soc/codecs/sta32x.c > create mode 100644 sound/soc/codecs/sta32x.h > > diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig > index 98175a0..dd075f2 100644 > --- a/sound/soc/codecs/Kconfig > +++ b/sound/soc/codecs/Kconfig > @@ -42,6 +42,7 @@ config SND_SOC_ALL_CODECS > select SND_SOC_SN95031 if INTEL_SCU_IPC > select SND_SOC_SPDIF > select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI > + select SND_SOC_STA32X if I2C > select SND_SOC_STAC9766 if SND_SOC_AC97_BUS > select SND_SOC_TLV320AIC23 if I2C > select SND_SOC_TLV320AIC26 if SPI_MASTER > @@ -216,6 +217,9 @@ config SND_SOC_SPDIF > config SND_SOC_SSM2602 > tristate > > +config SND_SOC_STA32X > + tristate > + > config SND_SOC_STAC9766 > tristate > > diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile > index fd85584..2ad1310 100644 > --- a/sound/soc/codecs/Makefile > +++ b/sound/soc/codecs/Makefile > @@ -28,6 +28,7 @@ snd-soc-alc5623-objs := alc5623.o > snd-soc-sn95031-objs := sn95031.o > snd-soc-spdif-objs := spdif_transciever.o > snd-soc-ssm2602-objs := ssm2602.o > +snd-soc-sta32x-objs := sta32x.o > snd-soc-stac9766-objs := stac9766.o > snd-soc-tlv320aic23-objs := tlv320aic23.o > snd-soc-tlv320aic26-objs := tlv320aic26.o > @@ -120,6 +121,7 @@ obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o > obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o > obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o > obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o > +obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o > obj-$(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 > diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c > new file mode 100644 > index 0000000..0fe38975 > --- /dev/null > +++ b/sound/soc/codecs/sta32x.c > @@ -0,0 +1,778 @@ > +/* > + * Codec driver for ST STA32x 2.1-channel high-efficiency digital audio system > + * > + * Copyright: 2011 Raumfeld GmbH > + * Author: Johannes Stezenbach <js@sig21.net> > + * > + * based on code from: > + * Wolfson Microelectronics PLC. > + * Mark Brown <broonie@opensource.wolfsonmicro.com> > + * Freescale Semiconductor, Inc. > + * Timur Tabi <timur@freescale.com> > + * > + * This program is free software; you can redistribute it and/or modify it > + * under the terms of the GNU General Public License as published by the > + * Free Software Foundation; either version 2 of the License, or (at your > + * option) any later version. > + */ > + > +#define DEBUG > +#define pr_fmt(fmt) KBUILD_MODNAME ":%s:%d: " fmt, __func__, __LINE__ > + > +#include <linux/module.h> > +#include <linux/moduleparam.h> > +#include <linux/init.h> > +#include <linux/delay.h> > +#include <linux/pm.h> > +#include <linux/i2c.h> > +#include <linux/platform_device.h> > +#include <linux/regulator/consumer.h> > +#include <linux/slab.h> > +#include <sound/core.h> > +#include <sound/pcm.h> > +#include <sound/pcm_params.h> > +#include <sound/soc.h> > +#include <sound/soc-dapm.h> > +#include <sound/initval.h> > +#include <sound/tlv.h> > + > +#include "sta32x.h" > + > +#define STA32X_RATES (SNDRV_PCM_RATE_32000 | \ > + SNDRV_PCM_RATE_44100 | \ > + SNDRV_PCM_RATE_48000 | \ > + SNDRV_PCM_RATE_88200 | \ > + SNDRV_PCM_RATE_96000 | \ > + SNDRV_PCM_RATE_176400 | \ > + SNDRV_PCM_RATE_192000) > + > +#define STA32X_FORMATS \ > + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ > + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ > + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ > + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ > + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | \ > + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE) > + > +/* Power-up register defaults */ > +static const u8 sta32x_regs[STA32X_REGISTER_COUNT] = { > + 0x63, 0x80, 0xc2, 0x40, 0xc2, 0x5c, 0x10, 0xff, 0x60, 0x60, > + 0x60, 0x80, 0x00, 0x00, 0x00, 0x40, 0x80, 0x77, 0x6a, 0x69, > + 0x6a, 0x69, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, > + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x2d, > + 0xc0, 0xf3, 0x33, 0x00, 0x0c, > +}; > + > +/* regulator power supply names */ > +static const char *sta32x_supply_names[] = { > + "Vdda", /* analog supply, 3.3VV */ > + "Vdd3", /* digital supply, 3.3V */ > + "Vcc" /* power amp spply, 10V - 36V */ > +}; > + > +/* codec private data */ > +struct sta32x_priv { > + struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)]; > + struct snd_soc_codec *codec; > + > + unsigned int mclk; > + unsigned int format; > +}; > + > +static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12700, 50, 1); > +static const DECLARE_TLV_DB_SCALE(chvol_tlv, -7950, 50, 1); > +static const DECLARE_TLV_DB_SCALE(tone_tlv, -120, 200, 0); > + > +static const char *sta32x_drc_ac[] = { > + "Anti-Clipping", "Dynamic Range Compression" }; > +static const char *sta32x_auto_eq_mode[] = { > + "User", "Preset", "Loudness" }; > +static const char *sta32x_auto_gc_mode[] = { > + "User", "AC no clipping", "AC limited clipping (10%)", > + "DRC nighttime listening mode" }; > +static const char *sta32x_auto_xo_mode[] = { > + "User", "80Hz", "100Hz", "120Hz", "140Hz", "160Hz", "180Hz", "200Hz", > + "220Hz", "240Hz", "260Hz", "280Hz", "300Hz", "320Hz", "340Hz", "360Hz" }; > +static const char *sta32x_preset_eq_mode[] = { > + "Flat", "Rock", "Soft Rock", "Jazz", "Classical", "Dance", "Pop", "Soft", > + "Hard", "Party", "Vocal", "Hip-Hop", "Dialog", "Bass-boost #1", > + "Bass-boost #2", "Bass-boost #3", "Loudness 1", "Loudness 2", > + "Loudness 3", "Loudness 4", "Loudness 5", "Loudness 6", "Loudness 7", > + "Loudness 8", "Loudness 9", "Loudness 10", "Loudness 11", "Loudness 12", > + "Loudness 13", "Loudness 14", "Loudness 15", "Loudness 16" }; > +static const char *sta32x_limiter_select[] = { > + "Limiter Disabled", "Limiter #1", "Limiter #2" }; > +static const char *sta32x_limiter_attack_rate[] = { > + "3.1584", "2.7072", "2.2560", "1.8048", "1.3536", "0.9024", > + "0.4512", "0.2256", "0.1504", "0.1123", "0.0902", "0.0752", > + "0.0645", "0.0564", "0.0501", "0.0451" }; > +static const char *sta32x_limiter_release_rate[] = { > + "0.5116", "0.1370", "0.0744", "0.0499", "0.0360", "0.0299", > + "0.0264", "0.0208", "0.0198", "0.0172", "0.0147", "0.0137", > + "0.0134", "0.0117", "0.0110", "0.0104" }; > + > +static const unsigned int sta32x_limiter_ac_attack_tlv[] = { > + TLV_DB_RANGE_HEAD(2), > + 0, 7, TLV_DB_SCALE_ITEM(-1200, 200, 0), > + 8, 16, TLV_DB_SCALE_ITEM(300, 100, 0), > +}; > + > +static const unsigned int sta32x_limiter_ac_release_tlv[] = { > + TLV_DB_RANGE_HEAD(5), > + 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0), > + 1, 1, TLV_DB_SCALE_ITEM(-2900, 0, 0), > + 2, 2, TLV_DB_SCALE_ITEM(-2000, 0, 0), > + 3, 8, TLV_DB_SCALE_ITEM(-1400, 200, 0), > + 8, 16, TLV_DB_SCALE_ITEM(-700, 100, 0), > +}; > + > +static const unsigned int sta32x_limiter_drc_attack_tlv[] = { > + TLV_DB_RANGE_HEAD(3), > + 0, 7, TLV_DB_SCALE_ITEM(-3100, 200, 0), > + 8, 13, TLV_DB_SCALE_ITEM(-1600, 100, 0), > + 14, 16, TLV_DB_SCALE_ITEM(-1000, 300, 0), > +}; > + > +static const unsigned int sta32x_limiter_drc_release_tlv[] = { > + TLV_DB_RANGE_HEAD(5), > + 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0), > + 1, 2, TLV_DB_SCALE_ITEM(-3800, 200, 0), > + 3, 4, TLV_DB_SCALE_ITEM(-3300, 200, 0), > + 5, 12, TLV_DB_SCALE_ITEM(-3000, 200, 0), > + 13, 16, TLV_DB_SCALE_ITEM(-1500, 300, 0), > +}; > + > +static const struct soc_enum sta32x_drc_ac_enum = > + SOC_ENUM_SINGLE(STA32X_CONFD, STA32X_CONFD_DRC_SHIFT, > + 2, sta32x_drc_ac); > +static const struct soc_enum sta32x_auto_eq_enum = > + SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMEQ_SHIFT, > + 3, sta32x_auto_eq_mode); > +static const struct soc_enum sta32x_auto_gc_enum = > + SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMGC_SHIFT, > + 4, sta32x_auto_gc_mode); > +static const struct soc_enum sta32x_auto_xo_enum = > + SOC_ENUM_SINGLE(STA32X_AUTO2, STA32X_AUTO2_XO_SHIFT, > + 16, sta32x_auto_xo_mode); > +static const struct soc_enum sta32x_preset_eq_enum = > + SOC_ENUM_SINGLE(STA32X_AUTO3, STA32X_AUTO3_PEQ_SHIFT, > + 32, sta32x_preset_eq_mode); > +static const struct soc_enum sta32x_limiter_ch1_enum = > + SOC_ENUM_SINGLE(STA32X_C1CFG, STA32X_CxCFG_LS_SHIFT, > + 3, sta32x_limiter_select); > +static const struct soc_enum sta32x_limiter_ch2_enum = > + SOC_ENUM_SINGLE(STA32X_C2CFG, STA32X_CxCFG_LS_SHIFT, > + 3, sta32x_limiter_select); > +static const struct soc_enum sta32x_limiter_ch3_enum = > + SOC_ENUM_SINGLE(STA32X_C3CFG, STA32X_CxCFG_LS_SHIFT, > + 3, sta32x_limiter_select); > +static const struct soc_enum sta32x_limiter1_attack_rate_enum = > + SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxA_SHIFT, > + 16, sta32x_limiter_attack_rate); > +static const struct soc_enum sta32x_limiter2_attack_rate_enum = > + SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxA_SHIFT, > + 16, sta32x_limiter_attack_rate); > +static const struct soc_enum sta32x_limiter1_release_rate_enum = > + SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxR_SHIFT, > + 16, sta32x_limiter_release_rate); > +static const struct soc_enum sta32x_limiter2_release_rate_enum = > + SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxR_SHIFT, > + 16, sta32x_limiter_release_rate); > +static const struct snd_kcontrol_new sta32x_snd_controls[] = { > +SOC_SINGLE_TLV("Master Volume", STA32X_MVOL, 0, 0xff, 1, mvol_tlv), > +SOC_SINGLE("Master Playback Switch", STA32X_MMUTE, 0, 1, 1), > +SOC_SINGLE("Ch1 Playback Switch", STA32X_MMUTE, 1, 1, 1), > +SOC_SINGLE("Ch2 Playback Switch", STA32X_MMUTE, 2, 1, 1), > +SOC_SINGLE("Ch3 Playback Switch", STA32X_MMUTE, 3, 1, 1), > +SOC_SINGLE_TLV("Ch1 Volume", STA32X_C1VOL, 0, 0xff, 1, chvol_tlv), > +SOC_SINGLE_TLV("Ch2 Volume", STA32X_C2VOL, 0, 0xff, 1, chvol_tlv), > +SOC_SINGLE_TLV("Ch3 Volume", STA32X_C3VOL, 0, 0xff, 1, chvol_tlv), > +SOC_SINGLE("De-emphasis Filter Switch", STA32X_CONFD, STA32X_CONFD_DEMP_SHIFT, 1, 0), > +SOC_ENUM("Compressor/Limiter Switch", sta32x_drc_ac_enum), > +SOC_SINGLE("Miami Mode Switch", STA32X_CONFD, STA32X_CONFD_MME_SHIFT, 1, 0), > +SOC_SINGLE("Zero Cross Switch", STA32X_CONFE, STA32X_CONFE_ZCE_SHIFT, 1, 0), > +SOC_SINGLE("Soft Ramp Switch", STA32X_CONFE, STA32X_CONFE_SVE_SHIFT, 1, 0), > +SOC_SINGLE("Auto-Mute Switch", STA32X_CONFF, STA32X_CONFF_IDE_SHIFT, 1, 0), > +SOC_ENUM("Automode EQ", sta32x_auto_eq_enum), > +SOC_ENUM("Automode GC", sta32x_auto_gc_enum), > +SOC_ENUM("Automode XO", sta32x_auto_xo_enum), > +SOC_ENUM("Preset EQ", sta32x_preset_eq_enum), > +SOC_SINGLE("Ch1 Tone Control Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_TCB_SHIFT, 1, 0), > +SOC_SINGLE("Ch2 Tone Control Bypass Switch", STA32X_C2CFG, STA32X_CxCFG_TCB_SHIFT, 1, 0), > +SOC_SINGLE("Ch1 EQ Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_EQBP_SHIFT, 1, 0), > +SOC_SINGLE("Ch2 EQ Bypass Switch", STA32X_C2CFG, STA32X_CxCFG_EQBP_SHIFT, 1, 0), > +SOC_SINGLE("Ch1 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0), > +SOC_SINGLE("Ch2 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0), > +SOC_SINGLE("Ch3 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0), > +SOC_ENUM("Ch1 Limiter Select", sta32x_limiter_ch1_enum), > +SOC_ENUM("Ch2 Limiter Select", sta32x_limiter_ch2_enum), > +SOC_ENUM("Ch3 Limiter Select", sta32x_limiter_ch3_enum), > +SOC_SINGLE_TLV("Bass Tone Control", STA32X_TONE, STA32X_TONE_BTC_SHIFT, 15, 0, tone_tlv), > +SOC_SINGLE_TLV("Treble Tone Control", STA32X_TONE, STA32X_TONE_TTC_SHIFT, 15, 0, tone_tlv), > +SOC_ENUM("Limiter1 Attack Rate (dB/ms)", sta32x_limiter1_attack_rate_enum), > +SOC_ENUM("Limiter2 Attack Rate (dB/ms)", sta32x_limiter2_attack_rate_enum), > +SOC_ENUM("Limiter1 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum), > +SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum), > + > +/* depending on mode, the attack/release thresholds have > + * two different enum definitions; provide both > + */ > +SOC_SINGLE_TLV("Limiter1 Attack Threshold (AC Mode)", STA32X_L1ATRT, STA32X_LxA_SHIFT, > + 16, 0, sta32x_limiter_ac_attack_tlv), > +SOC_SINGLE_TLV("Limiter2 Attack Threshold (AC Mode)", STA32X_L2ATRT, STA32X_LxA_SHIFT, > + 16, 0, sta32x_limiter_ac_attack_tlv), > +SOC_SINGLE_TLV("Limiter1 Release Threshold (AC Mode)", STA32X_L1ATRT, STA32X_LxR_SHIFT, > + 16, 0, sta32x_limiter_ac_release_tlv), > +SOC_SINGLE_TLV("Limiter2 Release Threshold (AC Mode)", STA32X_L2ATRT, STA32X_LxR_SHIFT, > + 16, 0, sta32x_limiter_ac_release_tlv), > +SOC_SINGLE_TLV("Limiter1 Attack Threshold (DRC Mode)", STA32X_L1ATRT, STA32X_LxA_SHIFT, > + 16, 0, sta32x_limiter_drc_attack_tlv), > +SOC_SINGLE_TLV("Limiter2 Attack Threshold (DRC Mode)", STA32X_L2ATRT, STA32X_LxA_SHIFT, > + 16, 0, sta32x_limiter_drc_attack_tlv), > +SOC_SINGLE_TLV("Limiter1 Release Threshold (DRC Mode)", STA32X_L1ATRT, STA32X_LxR_SHIFT, > + 16, 0, sta32x_limiter_drc_release_tlv), > +SOC_SINGLE_TLV("Limiter2 Release Threshold (DRC Mode)", STA32X_L2ATRT, STA32X_LxR_SHIFT, > + 16, 0, sta32x_limiter_drc_release_tlv), > +}; > + > +static const struct snd_soc_dapm_widget sta32x_dapm_widgets[] = { > +SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), > +SND_SOC_DAPM_OUTPUT("LEFT"), > +SND_SOC_DAPM_OUTPUT("RIGHT"), > +SND_SOC_DAPM_OUTPUT("SUB"), > +}; > + > +static const struct snd_soc_dapm_route sta32x_dapm_routes[] = { > + { "LEFT", NULL, "DAC" }, > + { "RIGHT", NULL, "DAC" }, > + { "SUB", NULL, "DAC" }, > +}; > + > +/* MCLK interpolation ratio per fs */ > +static struct { > + int fs; > + int ir; > +} interpolation_ratios[] = { > + { 32000, 0 }, > + { 44100, 0 }, > + { 48000, 0 }, > + { 88200, 1 }, > + { 96000, 1 }, > + { 176400, 2 }, > + { 192000, 2 }, > +}; > + > +/* MCLK to fs clock ratios */ > +static struct { > + int ratio; > + int mcs; > +} mclk_ratios[3][7] = { > + { { 768, 0 }, { 512, 1 }, { 384, 2 }, { 256, 3 }, > + { 128, 4 }, { 576, 5 }, { 0, 0 } }, > + { { 384, 2 }, { 256, 3 }, { 192, 4 }, { 128, 5 }, {64, 0 }, { 0, 0 } }, > + { { 384, 2 }, { 256, 3 }, { 192, 4 }, { 128, 5 }, {64, 0 }, { 0, 0 } }, > +}; > + > + > +/** > + * sta32x_set_dai_sysclk - configure MCLK > + * @codec_dai: the codec DAI > + * @clk_id: the clock ID (ignored) > + * @freq: the MCLK input frequency > + * @dir: the clock direction (ignored) > + * > + * The value of MCLK is used to determine which sample rates are supported > + * by the STA32X, based on the mclk_ratios table. > + * > + * This function must be called by the machine driver's 'startup' function, > + * otherwise the list of supported sample rates will not be available in > + * time for ALSA. > + * > + * For setups with variable MCLKs, pass 0 as 'freq' argument. This will cause > + * theoretically possible sample rates to be enabled. Call it again with a > + * proper value set one the external clock is set (most probably you would do > + * that from a machine's driver 'hw_param' hook. > + */ > +static int sta32x_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 sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); > + int i, j, ir, fs; > + unsigned int rates = 0; > + unsigned int rate_min = -1; > + unsigned int rate_max = 0; > + > + pr_debug("mclk=%u\n", freq); > + sta32x->mclk = freq; > + > + if (sta32x->mclk) { > + for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) { > + ir = interpolation_ratios[i].ir; > + fs = interpolation_ratios[i].fs; > + for (j = 0; mclk_ratios[ir][j].ratio; j++) { > + if (mclk_ratios[ir][j].ratio * fs == freq) { > + rates |= snd_pcm_rate_to_rate_bit(fs); > + if (fs < rate_min) > + rate_min = fs; > + if (fs > rate_max) > + rate_max = fs; > + } > + } > + } > + /* FIXME: soc should support a rate list */ > + rates &= ~SNDRV_PCM_RATE_KNOT; > + > + if (!rates) { > + dev_err(codec->dev, "could not find a valid sample rate\n"); > + return -EINVAL; > + } > + } else { > + /* enable all possible rates */ > + rates = STA32X_RATES; > + rate_min = 32000; > + rate_max = 192000; > + } > + > + codec_dai->driver->playback.rates = rates; > + codec_dai->driver->playback.rate_min = rate_min; > + codec_dai->driver->playback.rate_max = rate_max; > + return 0; > +} > + > +/** > + * sta32x_set_dai_fmt - configure the codec for the selected audio format > + * @codec_dai: the codec DAI > + * @fmt: a SND_SOC_DAIFMT_x value indicating the data format > + * > + * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the > + * codec accordingly. > + */ > +static int sta32x_set_dai_fmt(struct snd_soc_dai *codec_dai, > + unsigned int fmt) > +{ > + struct snd_soc_codec *codec = codec_dai->codec; > + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); > + u8 confb = snd_soc_read(codec, STA32X_CONFB); > + > + pr_debug("\n"); > + confb &= ~(STA32X_CONFB_C1IM | STA32X_CONFB_C2IM); > + > + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { > + case SND_SOC_DAIFMT_CBS_CFS: > + break; > + default: > + return -EINVAL; > + } > + > + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { > + case SND_SOC_DAIFMT_I2S: > + case SND_SOC_DAIFMT_RIGHT_J: > + case SND_SOC_DAIFMT_LEFT_J: > + sta32x->format = fmt & SND_SOC_DAIFMT_FORMAT_MASK; > + break; > + default: > + return -EINVAL; > + } > + > + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { > + case SND_SOC_DAIFMT_NB_NF: > + confb |= STA32X_CONFB_C2IM; > + break; > + case SND_SOC_DAIFMT_NB_IF: > + confb |= STA32X_CONFB_C1IM; > + break; > + default: > + return -EINVAL; > + } > + > + snd_soc_write(codec, STA32X_CONFB, confb); > + return 0; > +} > + > +/** > + * sta32x_hw_params - program the STA32X with the given hardware parameters. > + * @substream: the audio stream > + * @params: the hardware parameters to set > + * @dai: the SOC DAI (ignored) > + * > + * This function programs the hardware with the values provided. > + * Specifically, the sample rate and the data format. > + */ > +static int sta32x_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params, > + struct snd_soc_dai *dai) > +{ > + struct snd_soc_pcm_runtime *rtd = substream->private_data; > + struct snd_soc_codec *codec = rtd->codec; > + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); > + unsigned int rate; > + int i, mcs = -1, ir = -1; > + u8 confa, confb; > + > + rate = params_rate(params); > + pr_debug("rate: %u\n", rate); > + for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) > + if (interpolation_ratios[i].fs == rate) > + ir = interpolation_ratios[i].ir; > + if (ir < 0) > + return -EINVAL; > + for (i = 0; mclk_ratios[ir][i].ratio; i++) > + if (mclk_ratios[ir][i].ratio * rate == sta32x->mclk) > + mcs = mclk_ratios[ir][i].mcs; > + if (mcs < 0) > + return -EINVAL; > + > + confa = snd_soc_read(codec, STA32X_CONFA); > + confa &= ~(STA32X_CONFA_MCS_MASK | STA32X_CONFA_IR_MASK); > + confa |= (ir << STA32X_CONFA_IR_SHIFT) | (mcs << STA32X_CONFA_MCS_SHIFT); > + > + confb = snd_soc_read(codec, STA32X_CONFB); > + confb &= ~(STA32X_CONFB_SAI_MASK | STA32X_CONFB_SAIFB); > + switch (params_format(params)) { > + case SNDRV_PCM_FORMAT_S24_LE: > + case SNDRV_PCM_FORMAT_S24_BE: > + case SNDRV_PCM_FORMAT_S24_3LE: > + case SNDRV_PCM_FORMAT_S24_3BE: > + pr_debug("24bit\n"); > + /* fall through */ > + case SNDRV_PCM_FORMAT_S32_LE: > + case SNDRV_PCM_FORMAT_S32_BE: > + pr_debug("24bit or 32bit\n"); > + switch (sta32x->format) { > + case SND_SOC_DAIFMT_I2S: > + confb |= 0x0; > + break; > + case SND_SOC_DAIFMT_LEFT_J: > + confb |= 0x1; > + break; > + case SND_SOC_DAIFMT_RIGHT_J: > + confb |= 0x2; > + break; > + } > + > + break; > + case SNDRV_PCM_FORMAT_S20_3LE: > + case SNDRV_PCM_FORMAT_S20_3BE: > + pr_debug("20bit\n"); > + switch (sta32x->format) { > + case SND_SOC_DAIFMT_I2S: > + confb |= 0x4; > + break; > + case SND_SOC_DAIFMT_LEFT_J: > + confb |= 0x5; > + break; > + case SND_SOC_DAIFMT_RIGHT_J: > + confb |= 0x6; > + break; > + } > + > + break; > + case SNDRV_PCM_FORMAT_S18_3LE: > + case SNDRV_PCM_FORMAT_S18_3BE: > + pr_debug("18bit\n"); > + switch (sta32x->format) { > + case SND_SOC_DAIFMT_I2S: > + confb |= 0x8; > + break; > + case SND_SOC_DAIFMT_LEFT_J: > + confb |= 0x9; > + break; > + case SND_SOC_DAIFMT_RIGHT_J: > + confb |= 0xa; > + break; > + } > + > + break; > + case SNDRV_PCM_FORMAT_S16_LE: > + case SNDRV_PCM_FORMAT_S16_BE: > + pr_debug("16bit\n"); > + switch (sta32x->format) { > + case SND_SOC_DAIFMT_I2S: > + confb |= 0x0; > + break; > + case SND_SOC_DAIFMT_LEFT_J: > + confb |= 0xd; > + break; > + case SND_SOC_DAIFMT_RIGHT_J: > + confb |= 0xe; > + break; > + } > + > + break; > + default: > + return -EINVAL; > + } > + > + snd_soc_write(codec, STA32X_CONFA, confa); > + snd_soc_write(codec, STA32X_CONFB, confb); > + return 0; > +} > + > +/** > + * sta32x_set_bias_level - DAPM callback > + * @codec: the codec device > + * @level: DAPM power level > + * > + * This is called by ALSA to put the codec into low power mode > + * or to wake it up. If the codec is powered off completely > + * all registers must be restored after power on. > + */ > +static int sta32x_set_bias_level(struct snd_soc_codec *codec, > + enum snd_soc_bias_level level) > +{ > + int ret; > + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); > + > + pr_debug("level = %d\n", level); > + switch (level) { > + case SND_SOC_BIAS_ON: > + break; > + > + case SND_SOC_BIAS_PREPARE: > + /* Full power on */ > + snd_soc_update_bits(codec, STA32X_CONFF, > + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD, > + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD); > + break; > + > + case SND_SOC_BIAS_STANDBY: > + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { > + ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), > + sta32x->supplies); > + if (ret != 0) { > + dev_err(codec->dev, > + "Failed to enable supplies: %d\n", ret); > + return ret; > + } > + > + snd_soc_cache_sync(codec); > + } > + > + /* Power up to mute */ > + /* FIXME */ > + snd_soc_update_bits(codec, STA32X_CONFF, > + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD, > + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD); > + > + break; > + > + case SND_SOC_BIAS_OFF: > + /* The chip runs through the power down sequence for us. */ > + snd_soc_update_bits(codec, STA32X_CONFF, > + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD, > + STA32X_CONFF_PWDN); > + msleep(300); > + > + regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), > + sta32x->supplies); > + break; > + } > + codec->dapm.bias_level = level; > + return 0; > +} > + > +static struct snd_soc_dai_ops sta32x_dai_ops = { > + .hw_params = sta32x_hw_params, > + .set_sysclk = sta32x_set_dai_sysclk, > + .set_fmt = sta32x_set_dai_fmt, > +}; > + > +static struct snd_soc_dai_driver sta32x_dai = { > + .name = "STA32X", > + .playback = { > + .stream_name = "Playback", > + .channels_min = 2, > + .channels_max = 2, > + .rates = STA32X_RATES, > + .formats = STA32X_FORMATS, > + }, > + .ops = &sta32x_dai_ops, > +}; > + > +#ifdef CONFIG_PM > +static int sta32x_suspend(struct snd_soc_codec *codec, pm_message_t state) > +{ > + sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); > + return 0; > +} > + > +static int sta32x_resume(struct snd_soc_codec *codec) > +{ > + sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); > + return 0; > +} > +#else > +#define sta32x_suspend NULL > +#define sta32x_resume NULL > +#endif > + > +static int sta32x_probe(struct snd_soc_codec *codec) > +{ > + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); > + int i, ret = 0; > + > + sta32x->codec = codec; > + > + /* regulators */ > + for (i = 0; i < ARRAY_SIZE(sta32x->supplies); i++) > + sta32x->supplies[i].supply = sta32x_supply_names[i]; > + > + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sta32x->supplies), > + sta32x->supplies); > + if (ret != 0) { > + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); > + goto err; > + } > + > + ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), > + sta32x->supplies); > + if (ret != 0) { > + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); > + goto err_get; > + } > + > + /* Tell ASoC what kind of I/O to use to read the registers. ASoC will > + * then do the I2C transactions itself. > + */ > + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); > + if (ret < 0) { > + dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret); > + return ret; > + } > + > + /* read reg reset values into cache */ > + for (i = 0; i < STA32X_REGISTER_COUNT; i++) > + snd_soc_cache_write(codec, i, sta32x_regs[i]); > + > + /* FIXME enable thermal warning adjustment and recovery */ > + snd_soc_update_bits(codec, STA32X_CONFA, > + STA32X_CONFA_TWAB | STA32X_CONFA_TWRB, 0); > + > + /* FIXME select 2.1 mode */ > + snd_soc_update_bits(codec, STA32X_CONFF, > + STA32X_CONFF_OCFG_MASK, > + 1 << STA32X_CONFF_OCFG_SHIFT); > + > + /* FIXME channel to output mapping */ > + snd_soc_update_bits(codec, STA32X_C1CFG, > + STA32X_CxCFG_OM_MASK, > + 0 << STA32X_CxCFG_OM_SHIFT); > + snd_soc_update_bits(codec, STA32X_C2CFG, > + STA32X_CxCFG_OM_MASK, > + 1 << STA32X_CxCFG_OM_SHIFT); > + snd_soc_update_bits(codec, STA32X_C3CFG, > + STA32X_CxCFG_OM_MASK, > + 2 << STA32X_CxCFG_OM_SHIFT); > + > + sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); > + /* Bias level configuration will have done an extra enable */ > + regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); > + > + return 0; > + > +err_get: > + regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); > +err: > + return ret; > +} > + > +static int sta32x_remove(struct snd_soc_codec *codec) > +{ > + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); > + > + regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); > + regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); > + > + return 0; > +} > + > +static const struct snd_soc_codec_driver sta32x_codec = { > + .probe = sta32x_probe, > + .remove = sta32x_remove, > + .suspend = sta32x_suspend, > + .resume = sta32x_resume, > + .reg_cache_size = STA32X_REGISTER_COUNT, > + .reg_word_size = sizeof(u8), > + .set_bias_level = sta32x_set_bias_level, > + .controls = sta32x_snd_controls, > + .num_controls = ARRAY_SIZE(sta32x_snd_controls), > + .dapm_widgets = sta32x_dapm_widgets, > + .num_dapm_widgets = ARRAY_SIZE(sta32x_dapm_widgets), > + .dapm_routes = sta32x_dapm_routes, > + .num_dapm_routes = ARRAY_SIZE(sta32x_dapm_routes), > +}; > + > +static __devinit int sta32x_i2c_probe(struct i2c_client *i2c, > + const struct i2c_device_id *id) > +{ > + struct sta32x_priv *sta32x; > + int ret; > + > + sta32x = kzalloc(sizeof(struct sta32x_priv), GFP_KERNEL); > + if (!sta32x) > + return -ENOMEM; > + > + i2c_set_clientdata(i2c, sta32x); > + > + ret = snd_soc_register_codec(&i2c->dev, &sta32x_codec, &sta32x_dai, 1); > + if (ret != 0) { > + dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret); > + return ret; > + } > + > + return 0; > +} > + > +static __devexit int sta32x_i2c_remove(struct i2c_client *client) > +{ > + struct sta32x_priv *sta32x = i2c_get_clientdata(client); > + struct snd_soc_codec *codec = sta32x->codec; > + > + if (codec) > + sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); > + > + regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); > + > + if (codec) { > + snd_soc_unregister_codec(&client->dev); > + snd_soc_codec_set_drvdata(codec, NULL); > + } > + > + kfree(sta32x); > + return 0; > +} > + > +static const struct i2c_device_id sta32x_i2c_id[] = { > + { "sta326", 0 }, > + { "sta328", 0 }, > + { "sta329", 0 }, > + { } > +}; > +MODULE_DEVICE_TABLE(i2c, sta32x_i2c_id); > + > +static struct i2c_driver sta32x_i2c_driver = { > + .driver = { > + .name = "sta32x", > + .owner = THIS_MODULE, > + }, > + .probe = sta32x_i2c_probe, > + .remove = __devexit_p(sta32x_i2c_remove), > + .id_table = sta32x_i2c_id, > +}; > + > +static int __init sta32x_init(void) > +{ > + return i2c_add_driver(&sta32x_i2c_driver); > +} > +module_init(sta32x_init); > + > +static void __exit sta32x_exit(void) > +{ > + i2c_del_driver(&sta32x_i2c_driver); > +} > +module_exit(sta32x_exit); > + > +MODULE_DESCRIPTION("ASoC STA32X driver"); > +MODULE_AUTHOR("Johannes Stezenbach <js@sig21.net>"); > +MODULE_LICENSE("GPL"); > diff --git a/sound/soc/codecs/sta32x.h b/sound/soc/codecs/sta32x.h > new file mode 100644 > index 0000000..b97ee5a > --- /dev/null > +++ b/sound/soc/codecs/sta32x.h > @@ -0,0 +1,210 @@ > +/* > + * Codec driver for ST STA32x 2.1-channel high-efficiency digital audio system > + * > + * Copyright: 2011 Raumfeld GmbH > + * Author: Johannes Stezenbach <js@sig21.net> > + * > + * based on code from: > + * Wolfson Microelectronics PLC. > + * Mark Brown <broonie@opensource.wolfsonmicro.com> > + * > + * This program is free software; you can redistribute it and/or modify it > + * under the terms of the GNU General Public License as published by the > + * Free Software Foundation; either version 2 of the License, or (at your > + * option) any later version. > + */ > +#ifndef _ASOC_STA_32X_H > +#define _ASOC_STA_32X_H > + > +/* STA326 register addresses */ > + > +#define STA32X_REGISTER_COUNT 0x2d > + > +#define STA32X_CONFA 0x00 > +#define STA32X_CONFB 0x01 > +#define STA32X_CONFC 0x02 > +#define STA32X_CONFD 0x03 > +#define STA32X_CONFE 0x04 > +#define STA32X_CONFF 0x05 > +#define STA32X_MMUTE 0x06 > +#define STA32X_MVOL 0x07 > +#define STA32X_C1VOL 0x08 > +#define STA32X_C2VOL 0x09 > +#define STA32X_C3VOL 0x0a > +#define STA32X_AUTO1 0x0b > +#define STA32X_AUTO2 0x0c > +#define STA32X_AUTO3 0x0d > +#define STA32X_C1CFG 0x0e > +#define STA32X_C2CFG 0x0f > +#define STA32X_C3CFG 0x10 > +#define STA32X_TONE 0x11 > +#define STA32X_L1AR 0x12 > +#define STA32X_L1ATRT 0x13 > +#define STA32X_L2AR 0x14 > +#define STA32X_L2ATRT 0x15 > +#define STA32X_CFADDR2 0x16 > +#define STA32X_B1CF1 0x17 > +#define STA32X_B1CF2 0x18 > +#define STA32X_B1CF3 0x19 > +#define STA32X_B2CF1 0x1a > +#define STA32X_B2CF2 0x1b > +#define STA32X_B2CF3 0x1c > +#define STA32X_A1CF1 0x1d > +#define STA32X_A1CF2 0x1e > +#define STA32X_A1CF3 0x1f > +#define STA32X_A2CF1 0x20 > +#define STA32X_A2CF2 0x21 > +#define STA32X_A2CF3 0x22 > +#define STA32X_B0CF1 0x23 > +#define STA32X_B0CF2 0x24 > +#define STA32X_B0CF3 0x25 > +#define STA32X_CFUD 0x26 > +#define STA32X_MPCC1 0x27 > +#define STA32X_MPCC2 0x28 > +/* Reserved 0x29 */ > +/* Reserved 0x2a */ > +#define STA32X_Reserved 0x2a > +#define STA32X_FDRC1 0x2b > +#define STA32X_FDRC2 0x2c > +/* Reserved 0x2d */ > + > + > +/* STA326 register field definitions */ > + > +/* 0x00 CONFA */ > +#define STA32X_CONFA_MCS_MASK 0x03 > +#define STA32X_CONFA_MCS_SHIFT 0 > +#define STA32X_CONFA_IR_MASK 0x18 > +#define STA32X_CONFA_IR_SHIFT 3 > +#define STA32X_CONFA_TWRB 0x20 > +#define STA32X_CONFA_TWAB 0x40 > +#define STA32X_CONFA_FDRB 0x80 > + > +/* 0x01 CONFB */ > +#define STA32X_CONFB_SAI_MASK 0x0f > +#define STA32X_CONFB_SAI_SHIFT 0 > +#define STA32X_CONFB_SAIFB 0x10 > +#define STA32X_CONFB_DSCKE 0x20 > +#define STA32X_CONFB_C1IM 0x40 > +#define STA32X_CONFB_C2IM 0x80 > + > +/* 0x02 CONFC */ > +#define STA32X_CONFC_OM_MASK 0x03 > +#define STA32X_CONFC_OM_SHIFT 0 > +#define STA32X_CONFC_CSZ_MASK 0x7c > +#define STA32X_CONFC_CSZ_SHIFT 2 > + > +/* 0x03 CONFD */ > +#define STA32X_CONFD_HPB 0x01 > +#define STA32X_CONFD_HPB_SHIFT 0 > +#define STA32X_CONFD_DEMP 0x02 > +#define STA32X_CONFD_DEMP_SHIFT 1 > +#define STA32X_CONFD_DSPB 0x04 > +#define STA32X_CONFD_DSPB_SHIFT 2 > +#define STA32X_CONFD_PSL 0x08 > +#define STA32X_CONFD_PSL_SHIFT 3 > +#define STA32X_CONFD_BQL 0x10 > +#define STA32X_CONFD_BQL_SHIFT 4 > +#define STA32X_CONFD_DRC 0x20 > +#define STA32X_CONFD_DRC_SHIFT 5 > +#define STA32X_CONFD_ZDE 0x40 > +#define STA32X_CONFD_ZDE_SHIFT 6 > +#define STA32X_CONFD_MME 0x80 > +#define STA32X_CONFD_MME_SHIFT 7 > + > +/* 0x04 CONFE */ > +#define STA32X_CONFE_MPCV 0x01 > +#define STA32X_CONFE_MPCV_SHIFT 0 > +#define STA32X_CONFE_MPC 0x02 > +#define STA32X_CONFE_MPC_SHIFT 1 > +#define STA32X_CONFE_AME 0x08 > +#define STA32X_CONFE_AME_SHIFT 3 > +#define STA32X_CONFE_PWMS 0x10 > +#define STA32X_CONFE_PWMS_SHIFT 4 > +#define STA32X_CONFE_ZCE 0x40 > +#define STA32X_CONFE_ZCE_SHIFT 6 > +#define STA32X_CONFE_SVE 0x80 > +#define STA32X_CONFE_SVE_SHIFT 7 > + > +/* 0x05 CONFF */ > +#define STA32X_CONFF_OCFG_MASK 0x03 > +#define STA32X_CONFF_OCFG_SHIFT 0 > +#define STA32X_CONFF_IDE 0x04 > +#define STA32X_CONFF_IDE_SHIFT 3 > +#define STA32X_CONFF_BCLE 0x08 > +#define STA32X_CONFF_ECLE 0x20 > +#define STA32X_CONFF_PWDN 0x40 > +#define STA32X_CONFF_EAPD 0x80 > + > +/* 0x06 MMUTE */ > +#define STA32X_MMUTE_MMUTE 0x01 > + > +/* 0x0b AUTO1 */ > +#define STA32X_AUTO1_AMEQ_MASK 0x03 > +#define STA32X_AUTO1_AMEQ_SHIFT 0 > +#define STA32X_AUTO1_AMV_MASK 0xc0 > +#define STA32X_AUTO1_AMV_SHIFT 2 > +#define STA32X_AUTO1_AMGC_MASK 0x30 > +#define STA32X_AUTO1_AMGC_SHIFT 4 > +#define STA32X_AUTO1_AMPS 0x80 > + > +/* 0x0c AUTO2 */ > +#define STA32X_AUTO2_AMAME 0x01 > +#define STA32X_AUTO2_AMAM_MASK 0x0e > +#define STA32X_AUTO2_AMAM_SHIFT 1 > +#define STA32X_AUTO2_XO_MASK 0xf0 > +#define STA32X_AUTO2_XO_SHIFT 4 > + > +/* 0x0d AUTO3 */ > +#define STA32X_AUTO3_PEQ_MASK 0x1f > +#define STA32X_AUTO3_PEQ_SHIFT 0 > + > +/* 0x0e 0x0f 0x10 CxCFG */ > +#define STA32X_CxCFG_TCB 0x01 /* only C1 and C2 */ > +#define STA32X_CxCFG_TCB_SHIFT 0 > +#define STA32X_CxCFG_EQBP 0x02 /* only C1 and C2 */ > +#define STA32X_CxCFG_EQBP_SHIFT 1 > +#define STA32X_CxCFG_VBP 0x03 > +#define STA32X_CxCFG_VBP_SHIFT 2 > +#define STA32X_CxCFG_BO 0x04 > +#define STA32X_CxCFG_LS_MASK 0x30 > +#define STA32X_CxCFG_LS_SHIFT 4 > +#define STA32X_CxCFG_OM_MASK 0xc0 > +#define STA32X_CxCFG_OM_SHIFT 6 > + > +/* 0x11 TONE */ > +#define STA32X_TONE_BTC_SHIFT 0 > +#define STA32X_TONE_TTC_SHIFT 4 > + > +/* 0x12 0x13 0x14 0x15 limiter attack/release */ > +#define STA32X_LxA_SHIFT 0 > +#define STA32X_LxR_SHIFT 4 > + > +/* 0x26 CFUD */ > +#define STA32X_CFUD_W1 0x01 > +#define STA32X_CFUD_WA 0x02 > +#define STA32X_CFUD_R1 0x04 > +#define STA32X_CFUD_RA 0x08 > + > + > +/* biquad filter coefficient table offsets */ > +#define STA32X_C1_BQ_BASE 0 > +#define STA32X_C2_BQ_BASE 20 > +#define STA32X_CH_BQ_NUM 4 > +#define STA32X_BQ_NUM_COEF 5 > +#define STA32X_XO_HP_BQ_BASE 40 > +#define STA32X_XO_LP_BQ_BASE 45 > +#define STA32X_C1_PRESCALE 50 > +#define STA32X_C2_PRESCALE 51 > +#define STA32X_C1_POSTSCALE 52 > +#define STA32X_C2_POSTSCALE 53 > +#define STA32X_C3_POSTSCALE 54 > +#define STA32X_TW_POSTSCALE 55 > +#define STA32X_C1_MIX1 56 > +#define STA32X_C1_MIX2 57 > +#define STA32X_C2_MIX1 58 > +#define STA32X_C2_MIX2 59 > +#define STA32X_C3_MIX1 60 > +#define STA32X_C3_MIX2 61 > + > +#endif /* _ASOC_STA_32X_H */ > -- > 1.7.5.1 > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel@alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > ^ permalink raw reply [flat|nested] 38+ messages in thread
* Re: [PATCH] ALSA: ASoC: add STA32X codec driver 2011-06-16 12:07 ` [PATCH] " Daniel Mack 2011-06-17 8:30 ` Daniel Mack @ 2011-06-17 9:45 ` Mark Brown 2011-06-17 9:54 ` Daniel Mack 2011-06-20 6:07 ` [PATCH] " Daniel Mack 1 sibling, 2 replies; 38+ messages in thread From: Mark Brown @ 2011-06-17 9:45 UTC (permalink / raw) To: Daniel Mack; +Cc: alsa-devel, lrg On Thu, Jun 16, 2011 at 02:07:23PM +0200, Daniel Mack wrote: Looks very good, a few small nits, if Liam's OK I guess we could fix these incrementally: > +#define DEBUG This shouldn't be in the submission. > +SOC_SINGLE_TLV("Master Volume", STA32X_MVOL, 0, 0xff, 1, mvol_tlv), > +SOC_SINGLE("Master Playback Switch", STA32X_MMUTE, 0, 1, 1), > +SOC_SINGLE("Ch1 Playback Switch", STA32X_MMUTE, 1, 1, 1), > +SOC_SINGLE("Ch2 Playback Switch", STA32X_MMUTE, 2, 1, 1), > +SOC_SINGLE("Ch3 Playback Switch", STA32X_MMUTE, 3, 1, 1), > +SOC_SINGLE_TLV("Ch1 Volume", STA32X_C1VOL, 0, 0xff, 1, chvol_tlv), > +SOC_SINGLE_TLV("Ch2 Volume", STA32X_C2VOL, 0, 0xff, 1, chvol_tlv), > +SOC_SINGLE_TLV("Ch3 Volume", STA32X_C3VOL, 0, 0xff, 1, chvol_tlv), The "Playback" should be consistently either present or not present in both volume and mute controls so UIs can match them up easily. ^ permalink raw reply [flat|nested] 38+ messages in thread
* [PATCH] ALSA: ASoC: add STA32X codec driver 2011-06-17 9:45 ` Mark Brown @ 2011-06-17 9:54 ` Daniel Mack 2011-06-22 12:54 ` Mark Brown 2011-06-20 6:07 ` [PATCH] " Daniel Mack 1 sibling, 1 reply; 38+ messages in thread From: Daniel Mack @ 2011-06-17 9:54 UTC (permalink / raw) To: alsa-devel; +Cc: broonie, lrg From: Johannes Stezenbach <js@sig21.net> Signed-off-by: Johannes Stezenbach <js@sig21.net> [zonque@gmail.com: transform to new ASoC structure] --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sta32x.c | 777 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/sta32x.h | 210 ++++++++++++ 4 files changed, 993 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/sta32x.c create mode 100644 sound/soc/codecs/sta32x.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 98175a0..dd075f2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -42,6 +42,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI + select SND_SOC_STA32X if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER @@ -216,6 +217,9 @@ config SND_SOC_SPDIF config SND_SOC_SSM2602 tristate +config SND_SOC_STA32X + tristate + config SND_SOC_STAC9766 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fd85584..2ad1310 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -28,6 +28,7 @@ snd-soc-alc5623-objs := alc5623.o snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-sta32x-objs := sta32x.o snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o @@ -120,6 +121,7 @@ obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(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 diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c new file mode 100644 index 0000000..486628a --- /dev/null +++ b/sound/soc/codecs/sta32x.c @@ -0,0 +1,777 @@ +/* + * Codec driver for ST STA32x 2.1-channel high-efficiency digital audio system + * + * Copyright: 2011 Raumfeld GmbH + * Author: Johannes Stezenbach <js@sig21.net> + * + * based on code from: + * Wolfson Microelectronics PLC. + * Mark Brown <broonie@opensource.wolfsonmicro.com> + * Freescale Semiconductor, Inc. + * Timur Tabi <timur@freescale.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#define pr_fmt(fmt) KBUILD_MODNAME ":%s:%d: " fmt, __func__, __LINE__ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "sta32x.h" + +#define STA32X_RATES (SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000) + +#define STA32X_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE) + +/* Power-up register defaults */ +static const u8 sta32x_regs[STA32X_REGISTER_COUNT] = { + 0x63, 0x80, 0xc2, 0x40, 0xc2, 0x5c, 0x10, 0xff, 0x60, 0x60, + 0x60, 0x80, 0x00, 0x00, 0x00, 0x40, 0x80, 0x77, 0x6a, 0x69, + 0x6a, 0x69, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x2d, + 0xc0, 0xf3, 0x33, 0x00, 0x0c, +}; + +/* regulator power supply names */ +static const char *sta32x_supply_names[] = { + "Vdda", /* analog supply, 3.3VV */ + "Vdd3", /* digital supply, 3.3V */ + "Vcc" /* power amp spply, 10V - 36V */ +}; + +/* codec private data */ +struct sta32x_priv { + struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)]; + struct snd_soc_codec *codec; + + unsigned int mclk; + unsigned int format; +}; + +static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12700, 50, 1); +static const DECLARE_TLV_DB_SCALE(chvol_tlv, -7950, 50, 1); +static const DECLARE_TLV_DB_SCALE(tone_tlv, -120, 200, 0); + +static const char *sta32x_drc_ac[] = { + "Anti-Clipping", "Dynamic Range Compression" }; +static const char *sta32x_auto_eq_mode[] = { + "User", "Preset", "Loudness" }; +static const char *sta32x_auto_gc_mode[] = { + "User", "AC no clipping", "AC limited clipping (10%)", + "DRC nighttime listening mode" }; +static const char *sta32x_auto_xo_mode[] = { + "User", "80Hz", "100Hz", "120Hz", "140Hz", "160Hz", "180Hz", "200Hz", + "220Hz", "240Hz", "260Hz", "280Hz", "300Hz", "320Hz", "340Hz", "360Hz" }; +static const char *sta32x_preset_eq_mode[] = { + "Flat", "Rock", "Soft Rock", "Jazz", "Classical", "Dance", "Pop", "Soft", + "Hard", "Party", "Vocal", "Hip-Hop", "Dialog", "Bass-boost #1", + "Bass-boost #2", "Bass-boost #3", "Loudness 1", "Loudness 2", + "Loudness 3", "Loudness 4", "Loudness 5", "Loudness 6", "Loudness 7", + "Loudness 8", "Loudness 9", "Loudness 10", "Loudness 11", "Loudness 12", + "Loudness 13", "Loudness 14", "Loudness 15", "Loudness 16" }; +static const char *sta32x_limiter_select[] = { + "Limiter Disabled", "Limiter #1", "Limiter #2" }; +static const char *sta32x_limiter_attack_rate[] = { + "3.1584", "2.7072", "2.2560", "1.8048", "1.3536", "0.9024", + "0.4512", "0.2256", "0.1504", "0.1123", "0.0902", "0.0752", + "0.0645", "0.0564", "0.0501", "0.0451" }; +static const char *sta32x_limiter_release_rate[] = { + "0.5116", "0.1370", "0.0744", "0.0499", "0.0360", "0.0299", + "0.0264", "0.0208", "0.0198", "0.0172", "0.0147", "0.0137", + "0.0134", "0.0117", "0.0110", "0.0104" }; + +static const unsigned int sta32x_limiter_ac_attack_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 7, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 8, 16, TLV_DB_SCALE_ITEM(300, 100, 0), +}; + +static const unsigned int sta32x_limiter_ac_release_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(-2900, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(-2000, 0, 0), + 3, 8, TLV_DB_SCALE_ITEM(-1400, 200, 0), + 8, 16, TLV_DB_SCALE_ITEM(-700, 100, 0), +}; + +static const unsigned int sta32x_limiter_drc_attack_tlv[] = { + TLV_DB_RANGE_HEAD(3), + 0, 7, TLV_DB_SCALE_ITEM(-3100, 200, 0), + 8, 13, TLV_DB_SCALE_ITEM(-1600, 100, 0), + 14, 16, TLV_DB_SCALE_ITEM(-1000, 300, 0), +}; + +static const unsigned int sta32x_limiter_drc_release_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0), + 1, 2, TLV_DB_SCALE_ITEM(-3800, 200, 0), + 3, 4, TLV_DB_SCALE_ITEM(-3300, 200, 0), + 5, 12, TLV_DB_SCALE_ITEM(-3000, 200, 0), + 13, 16, TLV_DB_SCALE_ITEM(-1500, 300, 0), +}; + +static const struct soc_enum sta32x_drc_ac_enum = + SOC_ENUM_SINGLE(STA32X_CONFD, STA32X_CONFD_DRC_SHIFT, + 2, sta32x_drc_ac); +static const struct soc_enum sta32x_auto_eq_enum = + SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMEQ_SHIFT, + 3, sta32x_auto_eq_mode); +static const struct soc_enum sta32x_auto_gc_enum = + SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMGC_SHIFT, + 4, sta32x_auto_gc_mode); +static const struct soc_enum sta32x_auto_xo_enum = + SOC_ENUM_SINGLE(STA32X_AUTO2, STA32X_AUTO2_XO_SHIFT, + 16, sta32x_auto_xo_mode); +static const struct soc_enum sta32x_preset_eq_enum = + SOC_ENUM_SINGLE(STA32X_AUTO3, STA32X_AUTO3_PEQ_SHIFT, + 32, sta32x_preset_eq_mode); +static const struct soc_enum sta32x_limiter_ch1_enum = + SOC_ENUM_SINGLE(STA32X_C1CFG, STA32X_CxCFG_LS_SHIFT, + 3, sta32x_limiter_select); +static const struct soc_enum sta32x_limiter_ch2_enum = + SOC_ENUM_SINGLE(STA32X_C2CFG, STA32X_CxCFG_LS_SHIFT, + 3, sta32x_limiter_select); +static const struct soc_enum sta32x_limiter_ch3_enum = + SOC_ENUM_SINGLE(STA32X_C3CFG, STA32X_CxCFG_LS_SHIFT, + 3, sta32x_limiter_select); +static const struct soc_enum sta32x_limiter1_attack_rate_enum = + SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxA_SHIFT, + 16, sta32x_limiter_attack_rate); +static const struct soc_enum sta32x_limiter2_attack_rate_enum = + SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxA_SHIFT, + 16, sta32x_limiter_attack_rate); +static const struct soc_enum sta32x_limiter1_release_rate_enum = + SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxR_SHIFT, + 16, sta32x_limiter_release_rate); +static const struct soc_enum sta32x_limiter2_release_rate_enum = + SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxR_SHIFT, + 16, sta32x_limiter_release_rate); +static const struct snd_kcontrol_new sta32x_snd_controls[] = { +SOC_SINGLE_TLV("Master Volume", STA32X_MVOL, 0, 0xff, 1, mvol_tlv), +SOC_SINGLE("Master Switch", STA32X_MMUTE, 0, 1, 1), +SOC_SINGLE("Ch1 Switch", STA32X_MMUTE, 1, 1, 1), +SOC_SINGLE("Ch2 Switch", STA32X_MMUTE, 2, 1, 1), +SOC_SINGLE("Ch3 Switch", STA32X_MMUTE, 3, 1, 1), +SOC_SINGLE_TLV("Ch1 Volume", STA32X_C1VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE_TLV("Ch2 Volume", STA32X_C2VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE_TLV("Ch3 Volume", STA32X_C3VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE("De-emphasis Filter Switch", STA32X_CONFD, STA32X_CONFD_DEMP_SHIFT, 1, 0), +SOC_ENUM("Compressor/Limiter Switch", sta32x_drc_ac_enum), +SOC_SINGLE("Miami Mode Switch", STA32X_CONFD, STA32X_CONFD_MME_SHIFT, 1, 0), +SOC_SINGLE("Zero Cross Switch", STA32X_CONFE, STA32X_CONFE_ZCE_SHIFT, 1, 0), +SOC_SINGLE("Soft Ramp Switch", STA32X_CONFE, STA32X_CONFE_SVE_SHIFT, 1, 0), +SOC_SINGLE("Auto-Mute Switch", STA32X_CONFF, STA32X_CONFF_IDE_SHIFT, 1, 0), +SOC_ENUM("Automode EQ", sta32x_auto_eq_enum), +SOC_ENUM("Automode GC", sta32x_auto_gc_enum), +SOC_ENUM("Automode XO", sta32x_auto_xo_enum), +SOC_ENUM("Preset EQ", sta32x_preset_eq_enum), +SOC_SINGLE("Ch1 Tone Control Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_TCB_SHIFT, 1, 0), +SOC_SINGLE("Ch2 Tone Control Bypass Switch", STA32X_C2CFG, STA32X_CxCFG_TCB_SHIFT, 1, 0), +SOC_SINGLE("Ch1 EQ Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_EQBP_SHIFT, 1, 0), +SOC_SINGLE("Ch2 EQ Bypass Switch", STA32X_C2CFG, STA32X_CxCFG_EQBP_SHIFT, 1, 0), +SOC_SINGLE("Ch1 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0), +SOC_SINGLE("Ch2 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0), +SOC_SINGLE("Ch3 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0), +SOC_ENUM("Ch1 Limiter Select", sta32x_limiter_ch1_enum), +SOC_ENUM("Ch2 Limiter Select", sta32x_limiter_ch2_enum), +SOC_ENUM("Ch3 Limiter Select", sta32x_limiter_ch3_enum), +SOC_SINGLE_TLV("Bass Tone Control", STA32X_TONE, STA32X_TONE_BTC_SHIFT, 15, 0, tone_tlv), +SOC_SINGLE_TLV("Treble Tone Control", STA32X_TONE, STA32X_TONE_TTC_SHIFT, 15, 0, tone_tlv), +SOC_ENUM("Limiter1 Attack Rate (dB/ms)", sta32x_limiter1_attack_rate_enum), +SOC_ENUM("Limiter2 Attack Rate (dB/ms)", sta32x_limiter2_attack_rate_enum), +SOC_ENUM("Limiter1 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum), +SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum), + +/* depending on mode, the attack/release thresholds have + * two different enum definitions; provide both + */ +SOC_SINGLE_TLV("Limiter1 Attack Threshold (AC Mode)", STA32X_L1ATRT, STA32X_LxA_SHIFT, + 16, 0, sta32x_limiter_ac_attack_tlv), +SOC_SINGLE_TLV("Limiter2 Attack Threshold (AC Mode)", STA32X_L2ATRT, STA32X_LxA_SHIFT, + 16, 0, sta32x_limiter_ac_attack_tlv), +SOC_SINGLE_TLV("Limiter1 Release Threshold (AC Mode)", STA32X_L1ATRT, STA32X_LxR_SHIFT, + 16, 0, sta32x_limiter_ac_release_tlv), +SOC_SINGLE_TLV("Limiter2 Release Threshold (AC Mode)", STA32X_L2ATRT, STA32X_LxR_SHIFT, + 16, 0, sta32x_limiter_ac_release_tlv), +SOC_SINGLE_TLV("Limiter1 Attack Threshold (DRC Mode)", STA32X_L1ATRT, STA32X_LxA_SHIFT, + 16, 0, sta32x_limiter_drc_attack_tlv), +SOC_SINGLE_TLV("Limiter2 Attack Threshold (DRC Mode)", STA32X_L2ATRT, STA32X_LxA_SHIFT, + 16, 0, sta32x_limiter_drc_attack_tlv), +SOC_SINGLE_TLV("Limiter1 Release Threshold (DRC Mode)", STA32X_L1ATRT, STA32X_LxR_SHIFT, + 16, 0, sta32x_limiter_drc_release_tlv), +SOC_SINGLE_TLV("Limiter2 Release Threshold (DRC Mode)", STA32X_L2ATRT, STA32X_LxR_SHIFT, + 16, 0, sta32x_limiter_drc_release_tlv), +}; + +static const struct snd_soc_dapm_widget sta32x_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_OUTPUT("LEFT"), +SND_SOC_DAPM_OUTPUT("RIGHT"), +SND_SOC_DAPM_OUTPUT("SUB"), +}; + +static const struct snd_soc_dapm_route sta32x_dapm_routes[] = { + { "LEFT", NULL, "DAC" }, + { "RIGHT", NULL, "DAC" }, + { "SUB", NULL, "DAC" }, +}; + +/* MCLK interpolation ratio per fs */ +static struct { + int fs; + int ir; +} interpolation_ratios[] = { + { 32000, 0 }, + { 44100, 0 }, + { 48000, 0 }, + { 88200, 1 }, + { 96000, 1 }, + { 176400, 2 }, + { 192000, 2 }, +}; + +/* MCLK to fs clock ratios */ +static struct { + int ratio; + int mcs; +} mclk_ratios[3][7] = { + { { 768, 0 }, { 512, 1 }, { 384, 2 }, { 256, 3 }, + { 128, 4 }, { 576, 5 }, { 0, 0 } }, + { { 384, 2 }, { 256, 3 }, { 192, 4 }, { 128, 5 }, {64, 0 }, { 0, 0 } }, + { { 384, 2 }, { 256, 3 }, { 192, 4 }, { 128, 5 }, {64, 0 }, { 0, 0 } }, +}; + + +/** + * sta32x_set_dai_sysclk - configure MCLK + * @codec_dai: the codec DAI + * @clk_id: the clock ID (ignored) + * @freq: the MCLK input frequency + * @dir: the clock direction (ignored) + * + * The value of MCLK is used to determine which sample rates are supported + * by the STA32X, based on the mclk_ratios table. + * + * This function must be called by the machine driver's 'startup' function, + * otherwise the list of supported sample rates will not be available in + * time for ALSA. + * + * For setups with variable MCLKs, pass 0 as 'freq' argument. This will cause + * theoretically possible sample rates to be enabled. Call it again with a + * proper value set one the external clock is set (most probably you would do + * that from a machine's driver 'hw_param' hook. + */ +static int sta32x_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 sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + int i, j, ir, fs; + unsigned int rates = 0; + unsigned int rate_min = -1; + unsigned int rate_max = 0; + + pr_debug("mclk=%u\n", freq); + sta32x->mclk = freq; + + if (sta32x->mclk) { + for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) { + ir = interpolation_ratios[i].ir; + fs = interpolation_ratios[i].fs; + for (j = 0; mclk_ratios[ir][j].ratio; j++) { + if (mclk_ratios[ir][j].ratio * fs == freq) { + rates |= snd_pcm_rate_to_rate_bit(fs); + if (fs < rate_min) + rate_min = fs; + if (fs > rate_max) + rate_max = fs; + } + } + } + /* FIXME: soc should support a rate list */ + rates &= ~SNDRV_PCM_RATE_KNOT; + + if (!rates) { + dev_err(codec->dev, "could not find a valid sample rate\n"); + return -EINVAL; + } + } else { + /* enable all possible rates */ + rates = STA32X_RATES; + rate_min = 32000; + rate_max = 192000; + } + + codec_dai->driver->playback.rates = rates; + codec_dai->driver->playback.rate_min = rate_min; + codec_dai->driver->playback.rate_max = rate_max; + return 0; +} + +/** + * sta32x_set_dai_fmt - configure the codec for the selected audio format + * @codec_dai: the codec DAI + * @fmt: a SND_SOC_DAIFMT_x value indicating the data format + * + * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the + * codec accordingly. + */ +static int sta32x_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + u8 confb = snd_soc_read(codec, STA32X_CONFB); + + pr_debug("\n"); + confb &= ~(STA32X_CONFB_C1IM | STA32X_CONFB_C2IM); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + sta32x->format = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + confb |= STA32X_CONFB_C2IM; + break; + case SND_SOC_DAIFMT_NB_IF: + confb |= STA32X_CONFB_C1IM; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, STA32X_CONFB, confb); + return 0; +} + +/** + * sta32x_hw_params - program the STA32X with the given hardware parameters. + * @substream: the audio stream + * @params: the hardware parameters to set + * @dai: the SOC DAI (ignored) + * + * This function programs the hardware with the values provided. + * Specifically, the sample rate and the data format. + */ +static int sta32x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + unsigned int rate; + int i, mcs = -1, ir = -1; + u8 confa, confb; + + rate = params_rate(params); + pr_debug("rate: %u\n", rate); + for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) + if (interpolation_ratios[i].fs == rate) + ir = interpolation_ratios[i].ir; + if (ir < 0) + return -EINVAL; + for (i = 0; mclk_ratios[ir][i].ratio; i++) + if (mclk_ratios[ir][i].ratio * rate == sta32x->mclk) + mcs = mclk_ratios[ir][i].mcs; + if (mcs < 0) + return -EINVAL; + + confa = snd_soc_read(codec, STA32X_CONFA); + confa &= ~(STA32X_CONFA_MCS_MASK | STA32X_CONFA_IR_MASK); + confa |= (ir << STA32X_CONFA_IR_SHIFT) | (mcs << STA32X_CONFA_MCS_SHIFT); + + confb = snd_soc_read(codec, STA32X_CONFB); + confb &= ~(STA32X_CONFB_SAI_MASK | STA32X_CONFB_SAIFB); + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S24_BE: + case SNDRV_PCM_FORMAT_S24_3LE: + case SNDRV_PCM_FORMAT_S24_3BE: + pr_debug("24bit\n"); + /* fall through */ + case SNDRV_PCM_FORMAT_S32_LE: + case SNDRV_PCM_FORMAT_S32_BE: + pr_debug("24bit or 32bit\n"); + switch (sta32x->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x0; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x1; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0x2; + break; + } + + break; + case SNDRV_PCM_FORMAT_S20_3LE: + case SNDRV_PCM_FORMAT_S20_3BE: + pr_debug("20bit\n"); + switch (sta32x->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x4; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x5; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0x6; + break; + } + + break; + case SNDRV_PCM_FORMAT_S18_3LE: + case SNDRV_PCM_FORMAT_S18_3BE: + pr_debug("18bit\n"); + switch (sta32x->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x8; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x9; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0xa; + break; + } + + break; + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S16_BE: + pr_debug("16bit\n"); + switch (sta32x->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x0; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0xd; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0xe; + break; + } + + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, STA32X_CONFA, confa); + snd_soc_write(codec, STA32X_CONFB, confb); + return 0; +} + +/** + * sta32x_set_bias_level - DAPM callback + * @codec: the codec device + * @level: DAPM power level + * + * This is called by ALSA to put the codec into low power mode + * or to wake it up. If the codec is powered off completely + * all registers must be restored after power on. + */ +static int sta32x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int ret; + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + + pr_debug("level = %d\n", level); + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* Full power on */ + snd_soc_update_bits(codec, STA32X_CONFF, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), + sta32x->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", ret); + return ret; + } + + snd_soc_cache_sync(codec); + } + + /* Power up to mute */ + /* FIXME */ + snd_soc_update_bits(codec, STA32X_CONFF, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD); + + break; + + case SND_SOC_BIAS_OFF: + /* The chip runs through the power down sequence for us. */ + snd_soc_update_bits(codec, STA32X_CONFF, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD, + STA32X_CONFF_PWDN); + msleep(300); + + regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), + sta32x->supplies); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static struct snd_soc_dai_ops sta32x_dai_ops = { + .hw_params = sta32x_hw_params, + .set_sysclk = sta32x_set_dai_sysclk, + .set_fmt = sta32x_set_dai_fmt, +}; + +static struct snd_soc_dai_driver sta32x_dai = { + .name = "STA32X", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = STA32X_RATES, + .formats = STA32X_FORMATS, + }, + .ops = &sta32x_dai_ops, +}; + +#ifdef CONFIG_PM +static int sta32x_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int sta32x_resume(struct snd_soc_codec *codec) +{ + sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} +#else +#define sta32x_suspend NULL +#define sta32x_resume NULL +#endif + +static int sta32x_probe(struct snd_soc_codec *codec) +{ + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + int i, ret = 0; + + sta32x->codec = codec; + + /* regulators */ + for (i = 0; i < ARRAY_SIZE(sta32x->supplies); i++) + sta32x->supplies[i].supply = sta32x_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sta32x->supplies), + sta32x->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), + sta32x->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_get; + } + + /* Tell ASoC what kind of I/O to use to read the registers. ASoC will + * then do the I2C transactions itself. + */ + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret); + return ret; + } + + /* read reg reset values into cache */ + for (i = 0; i < STA32X_REGISTER_COUNT; i++) + snd_soc_cache_write(codec, i, sta32x_regs[i]); + + /* FIXME enable thermal warning adjustment and recovery */ + snd_soc_update_bits(codec, STA32X_CONFA, + STA32X_CONFA_TWAB | STA32X_CONFA_TWRB, 0); + + /* FIXME select 2.1 mode */ + snd_soc_update_bits(codec, STA32X_CONFF, + STA32X_CONFF_OCFG_MASK, + 1 << STA32X_CONFF_OCFG_SHIFT); + + /* FIXME channel to output mapping */ + snd_soc_update_bits(codec, STA32X_C1CFG, + STA32X_CxCFG_OM_MASK, + 0 << STA32X_CxCFG_OM_SHIFT); + snd_soc_update_bits(codec, STA32X_C2CFG, + STA32X_CxCFG_OM_MASK, + 1 << STA32X_CxCFG_OM_SHIFT); + snd_soc_update_bits(codec, STA32X_C3CFG, + STA32X_CxCFG_OM_MASK, + 2 << STA32X_CxCFG_OM_SHIFT); + + sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Bias level configuration will have done an extra enable */ + regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + return 0; + +err_get: + regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); +err: + return ret; +} + +static int sta32x_remove(struct snd_soc_codec *codec) +{ + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + + regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + return 0; +} + +static const struct snd_soc_codec_driver sta32x_codec = { + .probe = sta32x_probe, + .remove = sta32x_remove, + .suspend = sta32x_suspend, + .resume = sta32x_resume, + .reg_cache_size = STA32X_REGISTER_COUNT, + .reg_word_size = sizeof(u8), + .set_bias_level = sta32x_set_bias_level, + .controls = sta32x_snd_controls, + .num_controls = ARRAY_SIZE(sta32x_snd_controls), + .dapm_widgets = sta32x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sta32x_dapm_widgets), + .dapm_routes = sta32x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(sta32x_dapm_routes), +}; + +static __devinit int sta32x_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct sta32x_priv *sta32x; + int ret; + + sta32x = kzalloc(sizeof(struct sta32x_priv), GFP_KERNEL); + if (!sta32x) + return -ENOMEM; + + i2c_set_clientdata(i2c, sta32x); + + ret = snd_soc_register_codec(&i2c->dev, &sta32x_codec, &sta32x_dai, 1); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret); + return ret; + } + + return 0; +} + +static __devexit int sta32x_i2c_remove(struct i2c_client *client) +{ + struct sta32x_priv *sta32x = i2c_get_clientdata(client); + struct snd_soc_codec *codec = sta32x->codec; + + if (codec) + sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); + + regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + if (codec) { + snd_soc_unregister_codec(&client->dev); + snd_soc_codec_set_drvdata(codec, NULL); + } + + kfree(sta32x); + return 0; +} + +static const struct i2c_device_id sta32x_i2c_id[] = { + { "sta326", 0 }, + { "sta328", 0 }, + { "sta329", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, sta32x_i2c_id); + +static struct i2c_driver sta32x_i2c_driver = { + .driver = { + .name = "sta32x", + .owner = THIS_MODULE, + }, + .probe = sta32x_i2c_probe, + .remove = __devexit_p(sta32x_i2c_remove), + .id_table = sta32x_i2c_id, +}; + +static int __init sta32x_init(void) +{ + return i2c_add_driver(&sta32x_i2c_driver); +} +module_init(sta32x_init); + +static void __exit sta32x_exit(void) +{ + i2c_del_driver(&sta32x_i2c_driver); +} +module_exit(sta32x_exit); + +MODULE_DESCRIPTION("ASoC STA32X driver"); +MODULE_AUTHOR("Johannes Stezenbach <js@sig21.net>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sta32x.h b/sound/soc/codecs/sta32x.h new file mode 100644 index 0000000..b97ee5a --- /dev/null +++ b/sound/soc/codecs/sta32x.h @@ -0,0 +1,210 @@ +/* + * Codec driver for ST STA32x 2.1-channel high-efficiency digital audio system + * + * Copyright: 2011 Raumfeld GmbH + * Author: Johannes Stezenbach <js@sig21.net> + * + * based on code from: + * Wolfson Microelectronics PLC. + * Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#ifndef _ASOC_STA_32X_H +#define _ASOC_STA_32X_H + +/* STA326 register addresses */ + +#define STA32X_REGISTER_COUNT 0x2d + +#define STA32X_CONFA 0x00 +#define STA32X_CONFB 0x01 +#define STA32X_CONFC 0x02 +#define STA32X_CONFD 0x03 +#define STA32X_CONFE 0x04 +#define STA32X_CONFF 0x05 +#define STA32X_MMUTE 0x06 +#define STA32X_MVOL 0x07 +#define STA32X_C1VOL 0x08 +#define STA32X_C2VOL 0x09 +#define STA32X_C3VOL 0x0a +#define STA32X_AUTO1 0x0b +#define STA32X_AUTO2 0x0c +#define STA32X_AUTO3 0x0d +#define STA32X_C1CFG 0x0e +#define STA32X_C2CFG 0x0f +#define STA32X_C3CFG 0x10 +#define STA32X_TONE 0x11 +#define STA32X_L1AR 0x12 +#define STA32X_L1ATRT 0x13 +#define STA32X_L2AR 0x14 +#define STA32X_L2ATRT 0x15 +#define STA32X_CFADDR2 0x16 +#define STA32X_B1CF1 0x17 +#define STA32X_B1CF2 0x18 +#define STA32X_B1CF3 0x19 +#define STA32X_B2CF1 0x1a +#define STA32X_B2CF2 0x1b +#define STA32X_B2CF3 0x1c +#define STA32X_A1CF1 0x1d +#define STA32X_A1CF2 0x1e +#define STA32X_A1CF3 0x1f +#define STA32X_A2CF1 0x20 +#define STA32X_A2CF2 0x21 +#define STA32X_A2CF3 0x22 +#define STA32X_B0CF1 0x23 +#define STA32X_B0CF2 0x24 +#define STA32X_B0CF3 0x25 +#define STA32X_CFUD 0x26 +#define STA32X_MPCC1 0x27 +#define STA32X_MPCC2 0x28 +/* Reserved 0x29 */ +/* Reserved 0x2a */ +#define STA32X_Reserved 0x2a +#define STA32X_FDRC1 0x2b +#define STA32X_FDRC2 0x2c +/* Reserved 0x2d */ + + +/* STA326 register field definitions */ + +/* 0x00 CONFA */ +#define STA32X_CONFA_MCS_MASK 0x03 +#define STA32X_CONFA_MCS_SHIFT 0 +#define STA32X_CONFA_IR_MASK 0x18 +#define STA32X_CONFA_IR_SHIFT 3 +#define STA32X_CONFA_TWRB 0x20 +#define STA32X_CONFA_TWAB 0x40 +#define STA32X_CONFA_FDRB 0x80 + +/* 0x01 CONFB */ +#define STA32X_CONFB_SAI_MASK 0x0f +#define STA32X_CONFB_SAI_SHIFT 0 +#define STA32X_CONFB_SAIFB 0x10 +#define STA32X_CONFB_DSCKE 0x20 +#define STA32X_CONFB_C1IM 0x40 +#define STA32X_CONFB_C2IM 0x80 + +/* 0x02 CONFC */ +#define STA32X_CONFC_OM_MASK 0x03 +#define STA32X_CONFC_OM_SHIFT 0 +#define STA32X_CONFC_CSZ_MASK 0x7c +#define STA32X_CONFC_CSZ_SHIFT 2 + +/* 0x03 CONFD */ +#define STA32X_CONFD_HPB 0x01 +#define STA32X_CONFD_HPB_SHIFT 0 +#define STA32X_CONFD_DEMP 0x02 +#define STA32X_CONFD_DEMP_SHIFT 1 +#define STA32X_CONFD_DSPB 0x04 +#define STA32X_CONFD_DSPB_SHIFT 2 +#define STA32X_CONFD_PSL 0x08 +#define STA32X_CONFD_PSL_SHIFT 3 +#define STA32X_CONFD_BQL 0x10 +#define STA32X_CONFD_BQL_SHIFT 4 +#define STA32X_CONFD_DRC 0x20 +#define STA32X_CONFD_DRC_SHIFT 5 +#define STA32X_CONFD_ZDE 0x40 +#define STA32X_CONFD_ZDE_SHIFT 6 +#define STA32X_CONFD_MME 0x80 +#define STA32X_CONFD_MME_SHIFT 7 + +/* 0x04 CONFE */ +#define STA32X_CONFE_MPCV 0x01 +#define STA32X_CONFE_MPCV_SHIFT 0 +#define STA32X_CONFE_MPC 0x02 +#define STA32X_CONFE_MPC_SHIFT 1 +#define STA32X_CONFE_AME 0x08 +#define STA32X_CONFE_AME_SHIFT 3 +#define STA32X_CONFE_PWMS 0x10 +#define STA32X_CONFE_PWMS_SHIFT 4 +#define STA32X_CONFE_ZCE 0x40 +#define STA32X_CONFE_ZCE_SHIFT 6 +#define STA32X_CONFE_SVE 0x80 +#define STA32X_CONFE_SVE_SHIFT 7 + +/* 0x05 CONFF */ +#define STA32X_CONFF_OCFG_MASK 0x03 +#define STA32X_CONFF_OCFG_SHIFT 0 +#define STA32X_CONFF_IDE 0x04 +#define STA32X_CONFF_IDE_SHIFT 3 +#define STA32X_CONFF_BCLE 0x08 +#define STA32X_CONFF_ECLE 0x20 +#define STA32X_CONFF_PWDN 0x40 +#define STA32X_CONFF_EAPD 0x80 + +/* 0x06 MMUTE */ +#define STA32X_MMUTE_MMUTE 0x01 + +/* 0x0b AUTO1 */ +#define STA32X_AUTO1_AMEQ_MASK 0x03 +#define STA32X_AUTO1_AMEQ_SHIFT 0 +#define STA32X_AUTO1_AMV_MASK 0xc0 +#define STA32X_AUTO1_AMV_SHIFT 2 +#define STA32X_AUTO1_AMGC_MASK 0x30 +#define STA32X_AUTO1_AMGC_SHIFT 4 +#define STA32X_AUTO1_AMPS 0x80 + +/* 0x0c AUTO2 */ +#define STA32X_AUTO2_AMAME 0x01 +#define STA32X_AUTO2_AMAM_MASK 0x0e +#define STA32X_AUTO2_AMAM_SHIFT 1 +#define STA32X_AUTO2_XO_MASK 0xf0 +#define STA32X_AUTO2_XO_SHIFT 4 + +/* 0x0d AUTO3 */ +#define STA32X_AUTO3_PEQ_MASK 0x1f +#define STA32X_AUTO3_PEQ_SHIFT 0 + +/* 0x0e 0x0f 0x10 CxCFG */ +#define STA32X_CxCFG_TCB 0x01 /* only C1 and C2 */ +#define STA32X_CxCFG_TCB_SHIFT 0 +#define STA32X_CxCFG_EQBP 0x02 /* only C1 and C2 */ +#define STA32X_CxCFG_EQBP_SHIFT 1 +#define STA32X_CxCFG_VBP 0x03 +#define STA32X_CxCFG_VBP_SHIFT 2 +#define STA32X_CxCFG_BO 0x04 +#define STA32X_CxCFG_LS_MASK 0x30 +#define STA32X_CxCFG_LS_SHIFT 4 +#define STA32X_CxCFG_OM_MASK 0xc0 +#define STA32X_CxCFG_OM_SHIFT 6 + +/* 0x11 TONE */ +#define STA32X_TONE_BTC_SHIFT 0 +#define STA32X_TONE_TTC_SHIFT 4 + +/* 0x12 0x13 0x14 0x15 limiter attack/release */ +#define STA32X_LxA_SHIFT 0 +#define STA32X_LxR_SHIFT 4 + +/* 0x26 CFUD */ +#define STA32X_CFUD_W1 0x01 +#define STA32X_CFUD_WA 0x02 +#define STA32X_CFUD_R1 0x04 +#define STA32X_CFUD_RA 0x08 + + +/* biquad filter coefficient table offsets */ +#define STA32X_C1_BQ_BASE 0 +#define STA32X_C2_BQ_BASE 20 +#define STA32X_CH_BQ_NUM 4 +#define STA32X_BQ_NUM_COEF 5 +#define STA32X_XO_HP_BQ_BASE 40 +#define STA32X_XO_LP_BQ_BASE 45 +#define STA32X_C1_PRESCALE 50 +#define STA32X_C2_PRESCALE 51 +#define STA32X_C1_POSTSCALE 52 +#define STA32X_C2_POSTSCALE 53 +#define STA32X_C3_POSTSCALE 54 +#define STA32X_TW_POSTSCALE 55 +#define STA32X_C1_MIX1 56 +#define STA32X_C1_MIX2 57 +#define STA32X_C2_MIX1 58 +#define STA32X_C2_MIX2 59 +#define STA32X_C3_MIX1 60 +#define STA32X_C3_MIX2 61 + +#endif /* _ASOC_STA_32X_H */ -- 1.7.5.1 ^ permalink raw reply related [flat|nested] 38+ messages in thread
* Re: [PATCH] ALSA: ASoC: add STA32X codec driver 2011-06-17 9:54 ` Daniel Mack @ 2011-06-22 12:54 ` Mark Brown 2011-06-22 12:59 ` [PATCH 1/2] " Daniel Mack 0 siblings, 1 reply; 38+ messages in thread From: Mark Brown @ 2011-06-22 12:54 UTC (permalink / raw) To: Daniel Mack; +Cc: alsa-devel, lrg On Fri, Jun 17, 2011 at 11:54:30AM +0200, Daniel Mack wrote: > From: Johannes Stezenbach <js@sig21.net> > > Signed-off-by: Johannes Stezenbach <js@sig21.net> > [zonque@gmail.com: transform to new ASoC structure] Sorry, I just noticed that you didn't actually sign off either of these patches - we need that. ^ permalink raw reply [flat|nested] 38+ messages in thread
* [PATCH 1/2] ALSA: ASoC: add STA32X codec driver 2011-06-22 12:54 ` Mark Brown @ 2011-06-22 12:59 ` Daniel Mack 2011-06-22 12:59 ` [PATCH 2/2] ALSA: ASoC: add WM8782 ADC Codec Driver Daniel Mack 2011-06-22 13:26 ` [PATCH 1/2] ALSA: ASoC: add STA32X codec driver Mark Brown 0 siblings, 2 replies; 38+ messages in thread From: Daniel Mack @ 2011-06-22 12:59 UTC (permalink / raw) To: alsa-devel; +Cc: broonie, Daniel Mack From: Johannes Stezenbach <js@sig21.net> Signed-off-by: Johannes Stezenbach <js@sig21.net> [zonque@gmail.com: transform to new ASoC structure] Signed-off-by: Daniel Mack <zonque@gmail.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sta32x.c | 777 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/sta32x.h | 210 ++++++++++++ 4 files changed, 993 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/sta32x.c create mode 100644 sound/soc/codecs/sta32x.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 98175a0..dd075f2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -42,6 +42,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI + select SND_SOC_STA32X if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER @@ -216,6 +217,9 @@ config SND_SOC_SPDIF config SND_SOC_SSM2602 tristate +config SND_SOC_STA32X + tristate + config SND_SOC_STAC9766 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fd85584..2ad1310 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -28,6 +28,7 @@ snd-soc-alc5623-objs := alc5623.o snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-sta32x-objs := sta32x.o snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o @@ -120,6 +121,7 @@ obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(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 diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c new file mode 100644 index 0000000..486628a --- /dev/null +++ b/sound/soc/codecs/sta32x.c @@ -0,0 +1,777 @@ +/* + * Codec driver for ST STA32x 2.1-channel high-efficiency digital audio system + * + * Copyright: 2011 Raumfeld GmbH + * Author: Johannes Stezenbach <js@sig21.net> + * + * based on code from: + * Wolfson Microelectronics PLC. + * Mark Brown <broonie@opensource.wolfsonmicro.com> + * Freescale Semiconductor, Inc. + * Timur Tabi <timur@freescale.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#define pr_fmt(fmt) KBUILD_MODNAME ":%s:%d: " fmt, __func__, __LINE__ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "sta32x.h" + +#define STA32X_RATES (SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000) + +#define STA32X_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE) + +/* Power-up register defaults */ +static const u8 sta32x_regs[STA32X_REGISTER_COUNT] = { + 0x63, 0x80, 0xc2, 0x40, 0xc2, 0x5c, 0x10, 0xff, 0x60, 0x60, + 0x60, 0x80, 0x00, 0x00, 0x00, 0x40, 0x80, 0x77, 0x6a, 0x69, + 0x6a, 0x69, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x2d, + 0xc0, 0xf3, 0x33, 0x00, 0x0c, +}; + +/* regulator power supply names */ +static const char *sta32x_supply_names[] = { + "Vdda", /* analog supply, 3.3VV */ + "Vdd3", /* digital supply, 3.3V */ + "Vcc" /* power amp spply, 10V - 36V */ +}; + +/* codec private data */ +struct sta32x_priv { + struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)]; + struct snd_soc_codec *codec; + + unsigned int mclk; + unsigned int format; +}; + +static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12700, 50, 1); +static const DECLARE_TLV_DB_SCALE(chvol_tlv, -7950, 50, 1); +static const DECLARE_TLV_DB_SCALE(tone_tlv, -120, 200, 0); + +static const char *sta32x_drc_ac[] = { + "Anti-Clipping", "Dynamic Range Compression" }; +static const char *sta32x_auto_eq_mode[] = { + "User", "Preset", "Loudness" }; +static const char *sta32x_auto_gc_mode[] = { + "User", "AC no clipping", "AC limited clipping (10%)", + "DRC nighttime listening mode" }; +static const char *sta32x_auto_xo_mode[] = { + "User", "80Hz", "100Hz", "120Hz", "140Hz", "160Hz", "180Hz", "200Hz", + "220Hz", "240Hz", "260Hz", "280Hz", "300Hz", "320Hz", "340Hz", "360Hz" }; +static const char *sta32x_preset_eq_mode[] = { + "Flat", "Rock", "Soft Rock", "Jazz", "Classical", "Dance", "Pop", "Soft", + "Hard", "Party", "Vocal", "Hip-Hop", "Dialog", "Bass-boost #1", + "Bass-boost #2", "Bass-boost #3", "Loudness 1", "Loudness 2", + "Loudness 3", "Loudness 4", "Loudness 5", "Loudness 6", "Loudness 7", + "Loudness 8", "Loudness 9", "Loudness 10", "Loudness 11", "Loudness 12", + "Loudness 13", "Loudness 14", "Loudness 15", "Loudness 16" }; +static const char *sta32x_limiter_select[] = { + "Limiter Disabled", "Limiter #1", "Limiter #2" }; +static const char *sta32x_limiter_attack_rate[] = { + "3.1584", "2.7072", "2.2560", "1.8048", "1.3536", "0.9024", + "0.4512", "0.2256", "0.1504", "0.1123", "0.0902", "0.0752", + "0.0645", "0.0564", "0.0501", "0.0451" }; +static const char *sta32x_limiter_release_rate[] = { + "0.5116", "0.1370", "0.0744", "0.0499", "0.0360", "0.0299", + "0.0264", "0.0208", "0.0198", "0.0172", "0.0147", "0.0137", + "0.0134", "0.0117", "0.0110", "0.0104" }; + +static const unsigned int sta32x_limiter_ac_attack_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 7, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 8, 16, TLV_DB_SCALE_ITEM(300, 100, 0), +}; + +static const unsigned int sta32x_limiter_ac_release_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(-2900, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(-2000, 0, 0), + 3, 8, TLV_DB_SCALE_ITEM(-1400, 200, 0), + 8, 16, TLV_DB_SCALE_ITEM(-700, 100, 0), +}; + +static const unsigned int sta32x_limiter_drc_attack_tlv[] = { + TLV_DB_RANGE_HEAD(3), + 0, 7, TLV_DB_SCALE_ITEM(-3100, 200, 0), + 8, 13, TLV_DB_SCALE_ITEM(-1600, 100, 0), + 14, 16, TLV_DB_SCALE_ITEM(-1000, 300, 0), +}; + +static const unsigned int sta32x_limiter_drc_release_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0), + 1, 2, TLV_DB_SCALE_ITEM(-3800, 200, 0), + 3, 4, TLV_DB_SCALE_ITEM(-3300, 200, 0), + 5, 12, TLV_DB_SCALE_ITEM(-3000, 200, 0), + 13, 16, TLV_DB_SCALE_ITEM(-1500, 300, 0), +}; + +static const struct soc_enum sta32x_drc_ac_enum = + SOC_ENUM_SINGLE(STA32X_CONFD, STA32X_CONFD_DRC_SHIFT, + 2, sta32x_drc_ac); +static const struct soc_enum sta32x_auto_eq_enum = + SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMEQ_SHIFT, + 3, sta32x_auto_eq_mode); +static const struct soc_enum sta32x_auto_gc_enum = + SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMGC_SHIFT, + 4, sta32x_auto_gc_mode); +static const struct soc_enum sta32x_auto_xo_enum = + SOC_ENUM_SINGLE(STA32X_AUTO2, STA32X_AUTO2_XO_SHIFT, + 16, sta32x_auto_xo_mode); +static const struct soc_enum sta32x_preset_eq_enum = + SOC_ENUM_SINGLE(STA32X_AUTO3, STA32X_AUTO3_PEQ_SHIFT, + 32, sta32x_preset_eq_mode); +static const struct soc_enum sta32x_limiter_ch1_enum = + SOC_ENUM_SINGLE(STA32X_C1CFG, STA32X_CxCFG_LS_SHIFT, + 3, sta32x_limiter_select); +static const struct soc_enum sta32x_limiter_ch2_enum = + SOC_ENUM_SINGLE(STA32X_C2CFG, STA32X_CxCFG_LS_SHIFT, + 3, sta32x_limiter_select); +static const struct soc_enum sta32x_limiter_ch3_enum = + SOC_ENUM_SINGLE(STA32X_C3CFG, STA32X_CxCFG_LS_SHIFT, + 3, sta32x_limiter_select); +static const struct soc_enum sta32x_limiter1_attack_rate_enum = + SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxA_SHIFT, + 16, sta32x_limiter_attack_rate); +static const struct soc_enum sta32x_limiter2_attack_rate_enum = + SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxA_SHIFT, + 16, sta32x_limiter_attack_rate); +static const struct soc_enum sta32x_limiter1_release_rate_enum = + SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxR_SHIFT, + 16, sta32x_limiter_release_rate); +static const struct soc_enum sta32x_limiter2_release_rate_enum = + SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxR_SHIFT, + 16, sta32x_limiter_release_rate); +static const struct snd_kcontrol_new sta32x_snd_controls[] = { +SOC_SINGLE_TLV("Master Volume", STA32X_MVOL, 0, 0xff, 1, mvol_tlv), +SOC_SINGLE("Master Switch", STA32X_MMUTE, 0, 1, 1), +SOC_SINGLE("Ch1 Switch", STA32X_MMUTE, 1, 1, 1), +SOC_SINGLE("Ch2 Switch", STA32X_MMUTE, 2, 1, 1), +SOC_SINGLE("Ch3 Switch", STA32X_MMUTE, 3, 1, 1), +SOC_SINGLE_TLV("Ch1 Volume", STA32X_C1VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE_TLV("Ch2 Volume", STA32X_C2VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE_TLV("Ch3 Volume", STA32X_C3VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE("De-emphasis Filter Switch", STA32X_CONFD, STA32X_CONFD_DEMP_SHIFT, 1, 0), +SOC_ENUM("Compressor/Limiter Switch", sta32x_drc_ac_enum), +SOC_SINGLE("Miami Mode Switch", STA32X_CONFD, STA32X_CONFD_MME_SHIFT, 1, 0), +SOC_SINGLE("Zero Cross Switch", STA32X_CONFE, STA32X_CONFE_ZCE_SHIFT, 1, 0), +SOC_SINGLE("Soft Ramp Switch", STA32X_CONFE, STA32X_CONFE_SVE_SHIFT, 1, 0), +SOC_SINGLE("Auto-Mute Switch", STA32X_CONFF, STA32X_CONFF_IDE_SHIFT, 1, 0), +SOC_ENUM("Automode EQ", sta32x_auto_eq_enum), +SOC_ENUM("Automode GC", sta32x_auto_gc_enum), +SOC_ENUM("Automode XO", sta32x_auto_xo_enum), +SOC_ENUM("Preset EQ", sta32x_preset_eq_enum), +SOC_SINGLE("Ch1 Tone Control Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_TCB_SHIFT, 1, 0), +SOC_SINGLE("Ch2 Tone Control Bypass Switch", STA32X_C2CFG, STA32X_CxCFG_TCB_SHIFT, 1, 0), +SOC_SINGLE("Ch1 EQ Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_EQBP_SHIFT, 1, 0), +SOC_SINGLE("Ch2 EQ Bypass Switch", STA32X_C2CFG, STA32X_CxCFG_EQBP_SHIFT, 1, 0), +SOC_SINGLE("Ch1 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0), +SOC_SINGLE("Ch2 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0), +SOC_SINGLE("Ch3 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0), +SOC_ENUM("Ch1 Limiter Select", sta32x_limiter_ch1_enum), +SOC_ENUM("Ch2 Limiter Select", sta32x_limiter_ch2_enum), +SOC_ENUM("Ch3 Limiter Select", sta32x_limiter_ch3_enum), +SOC_SINGLE_TLV("Bass Tone Control", STA32X_TONE, STA32X_TONE_BTC_SHIFT, 15, 0, tone_tlv), +SOC_SINGLE_TLV("Treble Tone Control", STA32X_TONE, STA32X_TONE_TTC_SHIFT, 15, 0, tone_tlv), +SOC_ENUM("Limiter1 Attack Rate (dB/ms)", sta32x_limiter1_attack_rate_enum), +SOC_ENUM("Limiter2 Attack Rate (dB/ms)", sta32x_limiter2_attack_rate_enum), +SOC_ENUM("Limiter1 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum), +SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum), + +/* depending on mode, the attack/release thresholds have + * two different enum definitions; provide both + */ +SOC_SINGLE_TLV("Limiter1 Attack Threshold (AC Mode)", STA32X_L1ATRT, STA32X_LxA_SHIFT, + 16, 0, sta32x_limiter_ac_attack_tlv), +SOC_SINGLE_TLV("Limiter2 Attack Threshold (AC Mode)", STA32X_L2ATRT, STA32X_LxA_SHIFT, + 16, 0, sta32x_limiter_ac_attack_tlv), +SOC_SINGLE_TLV("Limiter1 Release Threshold (AC Mode)", STA32X_L1ATRT, STA32X_LxR_SHIFT, + 16, 0, sta32x_limiter_ac_release_tlv), +SOC_SINGLE_TLV("Limiter2 Release Threshold (AC Mode)", STA32X_L2ATRT, STA32X_LxR_SHIFT, + 16, 0, sta32x_limiter_ac_release_tlv), +SOC_SINGLE_TLV("Limiter1 Attack Threshold (DRC Mode)", STA32X_L1ATRT, STA32X_LxA_SHIFT, + 16, 0, sta32x_limiter_drc_attack_tlv), +SOC_SINGLE_TLV("Limiter2 Attack Threshold (DRC Mode)", STA32X_L2ATRT, STA32X_LxA_SHIFT, + 16, 0, sta32x_limiter_drc_attack_tlv), +SOC_SINGLE_TLV("Limiter1 Release Threshold (DRC Mode)", STA32X_L1ATRT, STA32X_LxR_SHIFT, + 16, 0, sta32x_limiter_drc_release_tlv), +SOC_SINGLE_TLV("Limiter2 Release Threshold (DRC Mode)", STA32X_L2ATRT, STA32X_LxR_SHIFT, + 16, 0, sta32x_limiter_drc_release_tlv), +}; + +static const struct snd_soc_dapm_widget sta32x_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_OUTPUT("LEFT"), +SND_SOC_DAPM_OUTPUT("RIGHT"), +SND_SOC_DAPM_OUTPUT("SUB"), +}; + +static const struct snd_soc_dapm_route sta32x_dapm_routes[] = { + { "LEFT", NULL, "DAC" }, + { "RIGHT", NULL, "DAC" }, + { "SUB", NULL, "DAC" }, +}; + +/* MCLK interpolation ratio per fs */ +static struct { + int fs; + int ir; +} interpolation_ratios[] = { + { 32000, 0 }, + { 44100, 0 }, + { 48000, 0 }, + { 88200, 1 }, + { 96000, 1 }, + { 176400, 2 }, + { 192000, 2 }, +}; + +/* MCLK to fs clock ratios */ +static struct { + int ratio; + int mcs; +} mclk_ratios[3][7] = { + { { 768, 0 }, { 512, 1 }, { 384, 2 }, { 256, 3 }, + { 128, 4 }, { 576, 5 }, { 0, 0 } }, + { { 384, 2 }, { 256, 3 }, { 192, 4 }, { 128, 5 }, {64, 0 }, { 0, 0 } }, + { { 384, 2 }, { 256, 3 }, { 192, 4 }, { 128, 5 }, {64, 0 }, { 0, 0 } }, +}; + + +/** + * sta32x_set_dai_sysclk - configure MCLK + * @codec_dai: the codec DAI + * @clk_id: the clock ID (ignored) + * @freq: the MCLK input frequency + * @dir: the clock direction (ignored) + * + * The value of MCLK is used to determine which sample rates are supported + * by the STA32X, based on the mclk_ratios table. + * + * This function must be called by the machine driver's 'startup' function, + * otherwise the list of supported sample rates will not be available in + * time for ALSA. + * + * For setups with variable MCLKs, pass 0 as 'freq' argument. This will cause + * theoretically possible sample rates to be enabled. Call it again with a + * proper value set one the external clock is set (most probably you would do + * that from a machine's driver 'hw_param' hook. + */ +static int sta32x_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 sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + int i, j, ir, fs; + unsigned int rates = 0; + unsigned int rate_min = -1; + unsigned int rate_max = 0; + + pr_debug("mclk=%u\n", freq); + sta32x->mclk = freq; + + if (sta32x->mclk) { + for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) { + ir = interpolation_ratios[i].ir; + fs = interpolation_ratios[i].fs; + for (j = 0; mclk_ratios[ir][j].ratio; j++) { + if (mclk_ratios[ir][j].ratio * fs == freq) { + rates |= snd_pcm_rate_to_rate_bit(fs); + if (fs < rate_min) + rate_min = fs; + if (fs > rate_max) + rate_max = fs; + } + } + } + /* FIXME: soc should support a rate list */ + rates &= ~SNDRV_PCM_RATE_KNOT; + + if (!rates) { + dev_err(codec->dev, "could not find a valid sample rate\n"); + return -EINVAL; + } + } else { + /* enable all possible rates */ + rates = STA32X_RATES; + rate_min = 32000; + rate_max = 192000; + } + + codec_dai->driver->playback.rates = rates; + codec_dai->driver->playback.rate_min = rate_min; + codec_dai->driver->playback.rate_max = rate_max; + return 0; +} + +/** + * sta32x_set_dai_fmt - configure the codec for the selected audio format + * @codec_dai: the codec DAI + * @fmt: a SND_SOC_DAIFMT_x value indicating the data format + * + * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the + * codec accordingly. + */ +static int sta32x_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + u8 confb = snd_soc_read(codec, STA32X_CONFB); + + pr_debug("\n"); + confb &= ~(STA32X_CONFB_C1IM | STA32X_CONFB_C2IM); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + sta32x->format = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + confb |= STA32X_CONFB_C2IM; + break; + case SND_SOC_DAIFMT_NB_IF: + confb |= STA32X_CONFB_C1IM; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, STA32X_CONFB, confb); + return 0; +} + +/** + * sta32x_hw_params - program the STA32X with the given hardware parameters. + * @substream: the audio stream + * @params: the hardware parameters to set + * @dai: the SOC DAI (ignored) + * + * This function programs the hardware with the values provided. + * Specifically, the sample rate and the data format. + */ +static int sta32x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + unsigned int rate; + int i, mcs = -1, ir = -1; + u8 confa, confb; + + rate = params_rate(params); + pr_debug("rate: %u\n", rate); + for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) + if (interpolation_ratios[i].fs == rate) + ir = interpolation_ratios[i].ir; + if (ir < 0) + return -EINVAL; + for (i = 0; mclk_ratios[ir][i].ratio; i++) + if (mclk_ratios[ir][i].ratio * rate == sta32x->mclk) + mcs = mclk_ratios[ir][i].mcs; + if (mcs < 0) + return -EINVAL; + + confa = snd_soc_read(codec, STA32X_CONFA); + confa &= ~(STA32X_CONFA_MCS_MASK | STA32X_CONFA_IR_MASK); + confa |= (ir << STA32X_CONFA_IR_SHIFT) | (mcs << STA32X_CONFA_MCS_SHIFT); + + confb = snd_soc_read(codec, STA32X_CONFB); + confb &= ~(STA32X_CONFB_SAI_MASK | STA32X_CONFB_SAIFB); + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S24_BE: + case SNDRV_PCM_FORMAT_S24_3LE: + case SNDRV_PCM_FORMAT_S24_3BE: + pr_debug("24bit\n"); + /* fall through */ + case SNDRV_PCM_FORMAT_S32_LE: + case SNDRV_PCM_FORMAT_S32_BE: + pr_debug("24bit or 32bit\n"); + switch (sta32x->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x0; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x1; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0x2; + break; + } + + break; + case SNDRV_PCM_FORMAT_S20_3LE: + case SNDRV_PCM_FORMAT_S20_3BE: + pr_debug("20bit\n"); + switch (sta32x->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x4; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x5; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0x6; + break; + } + + break; + case SNDRV_PCM_FORMAT_S18_3LE: + case SNDRV_PCM_FORMAT_S18_3BE: + pr_debug("18bit\n"); + switch (sta32x->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x8; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x9; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0xa; + break; + } + + break; + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S16_BE: + pr_debug("16bit\n"); + switch (sta32x->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x0; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0xd; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0xe; + break; + } + + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, STA32X_CONFA, confa); + snd_soc_write(codec, STA32X_CONFB, confb); + return 0; +} + +/** + * sta32x_set_bias_level - DAPM callback + * @codec: the codec device + * @level: DAPM power level + * + * This is called by ALSA to put the codec into low power mode + * or to wake it up. If the codec is powered off completely + * all registers must be restored after power on. + */ +static int sta32x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int ret; + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + + pr_debug("level = %d\n", level); + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* Full power on */ + snd_soc_update_bits(codec, STA32X_CONFF, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), + sta32x->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", ret); + return ret; + } + + snd_soc_cache_sync(codec); + } + + /* Power up to mute */ + /* FIXME */ + snd_soc_update_bits(codec, STA32X_CONFF, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD); + + break; + + case SND_SOC_BIAS_OFF: + /* The chip runs through the power down sequence for us. */ + snd_soc_update_bits(codec, STA32X_CONFF, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD, + STA32X_CONFF_PWDN); + msleep(300); + + regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), + sta32x->supplies); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static struct snd_soc_dai_ops sta32x_dai_ops = { + .hw_params = sta32x_hw_params, + .set_sysclk = sta32x_set_dai_sysclk, + .set_fmt = sta32x_set_dai_fmt, +}; + +static struct snd_soc_dai_driver sta32x_dai = { + .name = "STA32X", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = STA32X_RATES, + .formats = STA32X_FORMATS, + }, + .ops = &sta32x_dai_ops, +}; + +#ifdef CONFIG_PM +static int sta32x_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int sta32x_resume(struct snd_soc_codec *codec) +{ + sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} +#else +#define sta32x_suspend NULL +#define sta32x_resume NULL +#endif + +static int sta32x_probe(struct snd_soc_codec *codec) +{ + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + int i, ret = 0; + + sta32x->codec = codec; + + /* regulators */ + for (i = 0; i < ARRAY_SIZE(sta32x->supplies); i++) + sta32x->supplies[i].supply = sta32x_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sta32x->supplies), + sta32x->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), + sta32x->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_get; + } + + /* Tell ASoC what kind of I/O to use to read the registers. ASoC will + * then do the I2C transactions itself. + */ + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret); + return ret; + } + + /* read reg reset values into cache */ + for (i = 0; i < STA32X_REGISTER_COUNT; i++) + snd_soc_cache_write(codec, i, sta32x_regs[i]); + + /* FIXME enable thermal warning adjustment and recovery */ + snd_soc_update_bits(codec, STA32X_CONFA, + STA32X_CONFA_TWAB | STA32X_CONFA_TWRB, 0); + + /* FIXME select 2.1 mode */ + snd_soc_update_bits(codec, STA32X_CONFF, + STA32X_CONFF_OCFG_MASK, + 1 << STA32X_CONFF_OCFG_SHIFT); + + /* FIXME channel to output mapping */ + snd_soc_update_bits(codec, STA32X_C1CFG, + STA32X_CxCFG_OM_MASK, + 0 << STA32X_CxCFG_OM_SHIFT); + snd_soc_update_bits(codec, STA32X_C2CFG, + STA32X_CxCFG_OM_MASK, + 1 << STA32X_CxCFG_OM_SHIFT); + snd_soc_update_bits(codec, STA32X_C3CFG, + STA32X_CxCFG_OM_MASK, + 2 << STA32X_CxCFG_OM_SHIFT); + + sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Bias level configuration will have done an extra enable */ + regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + return 0; + +err_get: + regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); +err: + return ret; +} + +static int sta32x_remove(struct snd_soc_codec *codec) +{ + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + + regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + return 0; +} + +static const struct snd_soc_codec_driver sta32x_codec = { + .probe = sta32x_probe, + .remove = sta32x_remove, + .suspend = sta32x_suspend, + .resume = sta32x_resume, + .reg_cache_size = STA32X_REGISTER_COUNT, + .reg_word_size = sizeof(u8), + .set_bias_level = sta32x_set_bias_level, + .controls = sta32x_snd_controls, + .num_controls = ARRAY_SIZE(sta32x_snd_controls), + .dapm_widgets = sta32x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sta32x_dapm_widgets), + .dapm_routes = sta32x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(sta32x_dapm_routes), +}; + +static __devinit int sta32x_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct sta32x_priv *sta32x; + int ret; + + sta32x = kzalloc(sizeof(struct sta32x_priv), GFP_KERNEL); + if (!sta32x) + return -ENOMEM; + + i2c_set_clientdata(i2c, sta32x); + + ret = snd_soc_register_codec(&i2c->dev, &sta32x_codec, &sta32x_dai, 1); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret); + return ret; + } + + return 0; +} + +static __devexit int sta32x_i2c_remove(struct i2c_client *client) +{ + struct sta32x_priv *sta32x = i2c_get_clientdata(client); + struct snd_soc_codec *codec = sta32x->codec; + + if (codec) + sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); + + regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + if (codec) { + snd_soc_unregister_codec(&client->dev); + snd_soc_codec_set_drvdata(codec, NULL); + } + + kfree(sta32x); + return 0; +} + +static const struct i2c_device_id sta32x_i2c_id[] = { + { "sta326", 0 }, + { "sta328", 0 }, + { "sta329", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, sta32x_i2c_id); + +static struct i2c_driver sta32x_i2c_driver = { + .driver = { + .name = "sta32x", + .owner = THIS_MODULE, + }, + .probe = sta32x_i2c_probe, + .remove = __devexit_p(sta32x_i2c_remove), + .id_table = sta32x_i2c_id, +}; + +static int __init sta32x_init(void) +{ + return i2c_add_driver(&sta32x_i2c_driver); +} +module_init(sta32x_init); + +static void __exit sta32x_exit(void) +{ + i2c_del_driver(&sta32x_i2c_driver); +} +module_exit(sta32x_exit); + +MODULE_DESCRIPTION("ASoC STA32X driver"); +MODULE_AUTHOR("Johannes Stezenbach <js@sig21.net>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sta32x.h b/sound/soc/codecs/sta32x.h new file mode 100644 index 0000000..b97ee5a --- /dev/null +++ b/sound/soc/codecs/sta32x.h @@ -0,0 +1,210 @@ +/* + * Codec driver for ST STA32x 2.1-channel high-efficiency digital audio system + * + * Copyright: 2011 Raumfeld GmbH + * Author: Johannes Stezenbach <js@sig21.net> + * + * based on code from: + * Wolfson Microelectronics PLC. + * Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#ifndef _ASOC_STA_32X_H +#define _ASOC_STA_32X_H + +/* STA326 register addresses */ + +#define STA32X_REGISTER_COUNT 0x2d + +#define STA32X_CONFA 0x00 +#define STA32X_CONFB 0x01 +#define STA32X_CONFC 0x02 +#define STA32X_CONFD 0x03 +#define STA32X_CONFE 0x04 +#define STA32X_CONFF 0x05 +#define STA32X_MMUTE 0x06 +#define STA32X_MVOL 0x07 +#define STA32X_C1VOL 0x08 +#define STA32X_C2VOL 0x09 +#define STA32X_C3VOL 0x0a +#define STA32X_AUTO1 0x0b +#define STA32X_AUTO2 0x0c +#define STA32X_AUTO3 0x0d +#define STA32X_C1CFG 0x0e +#define STA32X_C2CFG 0x0f +#define STA32X_C3CFG 0x10 +#define STA32X_TONE 0x11 +#define STA32X_L1AR 0x12 +#define STA32X_L1ATRT 0x13 +#define STA32X_L2AR 0x14 +#define STA32X_L2ATRT 0x15 +#define STA32X_CFADDR2 0x16 +#define STA32X_B1CF1 0x17 +#define STA32X_B1CF2 0x18 +#define STA32X_B1CF3 0x19 +#define STA32X_B2CF1 0x1a +#define STA32X_B2CF2 0x1b +#define STA32X_B2CF3 0x1c +#define STA32X_A1CF1 0x1d +#define STA32X_A1CF2 0x1e +#define STA32X_A1CF3 0x1f +#define STA32X_A2CF1 0x20 +#define STA32X_A2CF2 0x21 +#define STA32X_A2CF3 0x22 +#define STA32X_B0CF1 0x23 +#define STA32X_B0CF2 0x24 +#define STA32X_B0CF3 0x25 +#define STA32X_CFUD 0x26 +#define STA32X_MPCC1 0x27 +#define STA32X_MPCC2 0x28 +/* Reserved 0x29 */ +/* Reserved 0x2a */ +#define STA32X_Reserved 0x2a +#define STA32X_FDRC1 0x2b +#define STA32X_FDRC2 0x2c +/* Reserved 0x2d */ + + +/* STA326 register field definitions */ + +/* 0x00 CONFA */ +#define STA32X_CONFA_MCS_MASK 0x03 +#define STA32X_CONFA_MCS_SHIFT 0 +#define STA32X_CONFA_IR_MASK 0x18 +#define STA32X_CONFA_IR_SHIFT 3 +#define STA32X_CONFA_TWRB 0x20 +#define STA32X_CONFA_TWAB 0x40 +#define STA32X_CONFA_FDRB 0x80 + +/* 0x01 CONFB */ +#define STA32X_CONFB_SAI_MASK 0x0f +#define STA32X_CONFB_SAI_SHIFT 0 +#define STA32X_CONFB_SAIFB 0x10 +#define STA32X_CONFB_DSCKE 0x20 +#define STA32X_CONFB_C1IM 0x40 +#define STA32X_CONFB_C2IM 0x80 + +/* 0x02 CONFC */ +#define STA32X_CONFC_OM_MASK 0x03 +#define STA32X_CONFC_OM_SHIFT 0 +#define STA32X_CONFC_CSZ_MASK 0x7c +#define STA32X_CONFC_CSZ_SHIFT 2 + +/* 0x03 CONFD */ +#define STA32X_CONFD_HPB 0x01 +#define STA32X_CONFD_HPB_SHIFT 0 +#define STA32X_CONFD_DEMP 0x02 +#define STA32X_CONFD_DEMP_SHIFT 1 +#define STA32X_CONFD_DSPB 0x04 +#define STA32X_CONFD_DSPB_SHIFT 2 +#define STA32X_CONFD_PSL 0x08 +#define STA32X_CONFD_PSL_SHIFT 3 +#define STA32X_CONFD_BQL 0x10 +#define STA32X_CONFD_BQL_SHIFT 4 +#define STA32X_CONFD_DRC 0x20 +#define STA32X_CONFD_DRC_SHIFT 5 +#define STA32X_CONFD_ZDE 0x40 +#define STA32X_CONFD_ZDE_SHIFT 6 +#define STA32X_CONFD_MME 0x80 +#define STA32X_CONFD_MME_SHIFT 7 + +/* 0x04 CONFE */ +#define STA32X_CONFE_MPCV 0x01 +#define STA32X_CONFE_MPCV_SHIFT 0 +#define STA32X_CONFE_MPC 0x02 +#define STA32X_CONFE_MPC_SHIFT 1 +#define STA32X_CONFE_AME 0x08 +#define STA32X_CONFE_AME_SHIFT 3 +#define STA32X_CONFE_PWMS 0x10 +#define STA32X_CONFE_PWMS_SHIFT 4 +#define STA32X_CONFE_ZCE 0x40 +#define STA32X_CONFE_ZCE_SHIFT 6 +#define STA32X_CONFE_SVE 0x80 +#define STA32X_CONFE_SVE_SHIFT 7 + +/* 0x05 CONFF */ +#define STA32X_CONFF_OCFG_MASK 0x03 +#define STA32X_CONFF_OCFG_SHIFT 0 +#define STA32X_CONFF_IDE 0x04 +#define STA32X_CONFF_IDE_SHIFT 3 +#define STA32X_CONFF_BCLE 0x08 +#define STA32X_CONFF_ECLE 0x20 +#define STA32X_CONFF_PWDN 0x40 +#define STA32X_CONFF_EAPD 0x80 + +/* 0x06 MMUTE */ +#define STA32X_MMUTE_MMUTE 0x01 + +/* 0x0b AUTO1 */ +#define STA32X_AUTO1_AMEQ_MASK 0x03 +#define STA32X_AUTO1_AMEQ_SHIFT 0 +#define STA32X_AUTO1_AMV_MASK 0xc0 +#define STA32X_AUTO1_AMV_SHIFT 2 +#define STA32X_AUTO1_AMGC_MASK 0x30 +#define STA32X_AUTO1_AMGC_SHIFT 4 +#define STA32X_AUTO1_AMPS 0x80 + +/* 0x0c AUTO2 */ +#define STA32X_AUTO2_AMAME 0x01 +#define STA32X_AUTO2_AMAM_MASK 0x0e +#define STA32X_AUTO2_AMAM_SHIFT 1 +#define STA32X_AUTO2_XO_MASK 0xf0 +#define STA32X_AUTO2_XO_SHIFT 4 + +/* 0x0d AUTO3 */ +#define STA32X_AUTO3_PEQ_MASK 0x1f +#define STA32X_AUTO3_PEQ_SHIFT 0 + +/* 0x0e 0x0f 0x10 CxCFG */ +#define STA32X_CxCFG_TCB 0x01 /* only C1 and C2 */ +#define STA32X_CxCFG_TCB_SHIFT 0 +#define STA32X_CxCFG_EQBP 0x02 /* only C1 and C2 */ +#define STA32X_CxCFG_EQBP_SHIFT 1 +#define STA32X_CxCFG_VBP 0x03 +#define STA32X_CxCFG_VBP_SHIFT 2 +#define STA32X_CxCFG_BO 0x04 +#define STA32X_CxCFG_LS_MASK 0x30 +#define STA32X_CxCFG_LS_SHIFT 4 +#define STA32X_CxCFG_OM_MASK 0xc0 +#define STA32X_CxCFG_OM_SHIFT 6 + +/* 0x11 TONE */ +#define STA32X_TONE_BTC_SHIFT 0 +#define STA32X_TONE_TTC_SHIFT 4 + +/* 0x12 0x13 0x14 0x15 limiter attack/release */ +#define STA32X_LxA_SHIFT 0 +#define STA32X_LxR_SHIFT 4 + +/* 0x26 CFUD */ +#define STA32X_CFUD_W1 0x01 +#define STA32X_CFUD_WA 0x02 +#define STA32X_CFUD_R1 0x04 +#define STA32X_CFUD_RA 0x08 + + +/* biquad filter coefficient table offsets */ +#define STA32X_C1_BQ_BASE 0 +#define STA32X_C2_BQ_BASE 20 +#define STA32X_CH_BQ_NUM 4 +#define STA32X_BQ_NUM_COEF 5 +#define STA32X_XO_HP_BQ_BASE 40 +#define STA32X_XO_LP_BQ_BASE 45 +#define STA32X_C1_PRESCALE 50 +#define STA32X_C2_PRESCALE 51 +#define STA32X_C1_POSTSCALE 52 +#define STA32X_C2_POSTSCALE 53 +#define STA32X_C3_POSTSCALE 54 +#define STA32X_TW_POSTSCALE 55 +#define STA32X_C1_MIX1 56 +#define STA32X_C1_MIX2 57 +#define STA32X_C2_MIX1 58 +#define STA32X_C2_MIX2 59 +#define STA32X_C3_MIX1 60 +#define STA32X_C3_MIX2 61 + +#endif /* _ASOC_STA_32X_H */ -- 1.7.5.4 ^ permalink raw reply related [flat|nested] 38+ messages in thread
* [PATCH 2/2] ALSA: ASoC: add WM8782 ADC Codec Driver 2011-06-22 12:59 ` [PATCH 1/2] " Daniel Mack @ 2011-06-22 12:59 ` Daniel Mack 2011-06-22 13:26 ` [PATCH 1/2] ALSA: ASoC: add STA32X codec driver Mark Brown 1 sibling, 0 replies; 38+ messages in thread From: Daniel Mack @ 2011-06-22 12:59 UTC (permalink / raw) To: alsa-devel; +Cc: broonie, Daniel Mack From: Johannes Stezenbach <js@sig21.net> Signed-off-by: Johannes Stezenbach <js@sig21.net> [zonque@gmail.com: transform to new ASoC structure] Signed-off-by: Daniel Mack <zonque@gmail.com> --- sound/soc/codecs/Kconfig | 4 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8782.c | 80 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 86 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/wm8782.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index dd075f2..64bc0d2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -72,6 +72,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8770 if SPI_MASTER select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI + select SND_SOC_WM8782 select SND_SOC_WM8804 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8900 if I2C select SND_SOC_WM8903 if I2C @@ -303,6 +304,9 @@ config SND_SOC_WM8770 config SND_SOC_WM8776 tristate +config SND_SOC_WM8782 + tristate + config SND_SOC_WM8804 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 2ad1310..0ef7cd1 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -56,6 +56,7 @@ snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o snd-soc-wm8770-objs := wm8770.o snd-soc-wm8776-objs := wm8776.o +snd-soc-wm8782-objs := wm8782.o snd-soc-wm8804-objs := wm8804.o snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o @@ -149,6 +150,7 @@ obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o obj-$(CONFIG_SND_SOC_WM8770) += snd-soc-wm8770.o obj-$(CONFIG_SND_SOC_WM8776) += snd-soc-wm8776.o +obj-$(CONFIG_SND_SOC_WM8782) += snd-soc-wm8782.o obj-$(CONFIG_SND_SOC_WM8804) += snd-soc-wm8804.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c new file mode 100644 index 0000000..a2a09f8 --- /dev/null +++ b/sound/soc/codecs/wm8782.c @@ -0,0 +1,80 @@ +/* + * sound/soc/codecs/wm8782.c + * simple, strap-pin configured 24bit 2ch ADC + * + * Copyright: 2011 Raumfeld GmbH + * Author: Johannes Stezenbach <js@sig21.net> + * + * based on ad73311.c + * Copyright: Analog Device Inc. + * Author: Cliff Cai <cliff.cai@analog.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/soc.h> + +static struct snd_soc_dai_driver wm8782_dai = { + .name = "wm8782", + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + /* For configurations with FSAMPEN=0 */ + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_LE, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_wm8782; + +static __devinit int wm8782_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, + &soc_codec_dev_wm8782, &wm8782_dai, 1); +} + +static int __devexit wm8782_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver wm8782_codec_driver = { + .driver = { + .name = "wm8782", + .owner = THIS_MODULE, + }, + .probe = wm8782_probe, + .remove = wm8782_remove, +}; + +static int __init wm8782_init(void) +{ + return platform_driver_register(&wm8782_codec_driver); +} +module_init(wm8782_init); + +static void __exit wm8782_exit(void) +{ + platform_driver_unregister(&wm8782_codec_driver); +} +module_exit(wm8782_exit); + +MODULE_DESCRIPTION("ASoC WM8782 driver"); +MODULE_AUTHOR("Johannes Stezenbach <js@sig21.net>"); +MODULE_LICENSE("GPL"); -- 1.7.5.4 ^ permalink raw reply related [flat|nested] 38+ messages in thread
* Re: [PATCH 1/2] ALSA: ASoC: add STA32X codec driver 2011-06-22 12:59 ` [PATCH 1/2] " Daniel Mack 2011-06-22 12:59 ` [PATCH 2/2] ALSA: ASoC: add WM8782 ADC Codec Driver Daniel Mack @ 2011-06-22 13:26 ` Mark Brown 2011-06-22 13:34 ` Daniel Mack 1 sibling, 1 reply; 38+ messages in thread From: Mark Brown @ 2011-06-22 13:26 UTC (permalink / raw) To: Daniel Mack; +Cc: alsa-devel On Wed, Jun 22, 2011 at 02:59:24PM +0200, Daniel Mack wrote: > From: Johannes Stezenbach <js@sig21.net> > > Signed-off-by: Johannes Stezenbach <js@sig21.net> > [zonque@gmail.com: transform to new ASoC structure] > Signed-off-by: Daniel Mack <zonque@gmail.com> Have you changed these paches in some way so that Liam's ack no longer applies? Please also don't post new versions of patches as replies in the middle of old serieses, it's much harder to find the current version and the threading can get exceptionally deep. ^ permalink raw reply [flat|nested] 38+ messages in thread
* Re: [PATCH 1/2] ALSA: ASoC: add STA32X codec driver 2011-06-22 13:26 ` [PATCH 1/2] ALSA: ASoC: add STA32X codec driver Mark Brown @ 2011-06-22 13:34 ` Daniel Mack 2011-06-22 13:59 ` Mark Brown 2011-06-22 14:09 ` Paul Menzel 0 siblings, 2 replies; 38+ messages in thread From: Daniel Mack @ 2011-06-22 13:34 UTC (permalink / raw) To: Mark Brown; +Cc: alsa-devel On Wed, Jun 22, 2011 at 3:26 PM, Mark Brown <broonie@opensource.wolfsonmicro.com> wrote: > On Wed, Jun 22, 2011 at 02:59:24PM +0200, Daniel Mack wrote: >> From: Johannes Stezenbach <js@sig21.net> >> >> Signed-off-by: Johannes Stezenbach <js@sig21.net> >> [zonque@gmail.com: transform to new ASoC structure] >> Signed-off-by: Daniel Mack <zonque@gmail.com> > > Have you changed these paches in some way so that Liam's ack no longer > applies? No, I just added my S-o-b. > Please also don't post new versions of patches as replies in the middle > of old serieses, it's much harder to find the current version and the > threading can get exceptionally deep. Ok, I thought it's actually easier to have them as follow-ups to your comments. But I don't care - next time I'll start a new thread for each round :) Daniel ^ permalink raw reply [flat|nested] 38+ messages in thread
* Re: [PATCH 1/2] ALSA: ASoC: add STA32X codec driver 2011-06-22 13:34 ` Daniel Mack @ 2011-06-22 13:59 ` Mark Brown 2011-06-22 14:09 ` Paul Menzel 1 sibling, 0 replies; 38+ messages in thread From: Mark Brown @ 2011-06-22 13:59 UTC (permalink / raw) To: Daniel Mack; +Cc: alsa-devel On Wed, Jun 22, 2011 at 03:34:15PM +0200, Daniel Mack wrote: > On Wed, Jun 22, 2011 at 3:26 PM, Mark Brown > > Have you changed these paches in some way so that Liam's ack no longer > > applies? > No, I just added my S-o-b. OK, as ever if people have acked code please keep the acks when reposting. Anyway, applied now. ^ permalink raw reply [flat|nested] 38+ messages in thread
* Re: [PATCH 1/2] ALSA: ASoC: add STA32X codec driver 2011-06-22 13:34 ` Daniel Mack 2011-06-22 13:59 ` Mark Brown @ 2011-06-22 14:09 ` Paul Menzel 1 sibling, 0 replies; 38+ messages in thread From: Paul Menzel @ 2011-06-22 14:09 UTC (permalink / raw) To: alsa-devel [-- Attachment #1.1: Type: text/plain, Size: 681 bytes --] Am Mittwoch, den 22.06.2011, 15:34 +0200 schrieb Daniel Mack: > On Wed, Jun 22, 2011 at 3:26 PM, Mark Brown wrote: […] > > Please also don't post new versions of patches as replies in the middle > > of old serieses, it's much harder to find the current version and the > > threading can get exceptionally deep. > > Ok, I thought it's actually easier to have them as follow-ups to your comments. > But I don't care - next time I'll start a new thread for each round :) To easier keep an overview, you could also add a version suffix like »[PATCH v2]« or »[PATCH] [RESEND]« to your patch iterations and add a small comment what changed. Thanks, Paul [-- Attachment #1.2: This is a digitally signed message part --] [-- Type: application/pgp-signature, Size: 198 bytes --] [-- Attachment #2: Type: text/plain, Size: 0 bytes --] ^ permalink raw reply [flat|nested] 38+ messages in thread
* Re: [PATCH] ALSA: ASoC: add STA32X codec driver 2011-06-17 9:45 ` Mark Brown 2011-06-17 9:54 ` Daniel Mack @ 2011-06-20 6:07 ` Daniel Mack 2011-06-22 11:06 ` Daniel Mack 1 sibling, 1 reply; 38+ messages in thread From: Daniel Mack @ 2011-06-20 6:07 UTC (permalink / raw) To: Mark Brown; +Cc: alsa-devel, lrg On Fri, Jun 17, 2011 at 11:45 AM, Mark Brown <broonie@opensource.wolfsonmicro.com> wrote: > On Thu, Jun 16, 2011 at 02:07:23PM +0200, Daniel Mack wrote: > > Looks very good, a few small nits, if Liam's OK I guess we could fix > these incrementally: > >> +#define DEBUG > > This shouldn't be in the submission. > >> +SOC_SINGLE_TLV("Master Volume", STA32X_MVOL, 0, 0xff, 1, mvol_tlv), >> +SOC_SINGLE("Master Playback Switch", STA32X_MMUTE, 0, 1, 1), >> +SOC_SINGLE("Ch1 Playback Switch", STA32X_MMUTE, 1, 1, 1), >> +SOC_SINGLE("Ch2 Playback Switch", STA32X_MMUTE, 2, 1, 1), >> +SOC_SINGLE("Ch3 Playback Switch", STA32X_MMUTE, 3, 1, 1), >> +SOC_SINGLE_TLV("Ch1 Volume", STA32X_C1VOL, 0, 0xff, 1, chvol_tlv), >> +SOC_SINGLE_TLV("Ch2 Volume", STA32X_C2VOL, 0, 0xff, 1, chvol_tlv), >> +SOC_SINGLE_TLV("Ch3 Volume", STA32X_C3VOL, 0, 0xff, 1, chvol_tlv), > > The "Playback" should be consistently either present or not present in > both volume and mute controls so UIs can match them up easily. > I've sent a fixed version of this patch just after you mentioned these issues. Hope it's all good now. Daniel ^ permalink raw reply [flat|nested] 38+ messages in thread
* Re: [PATCH] ALSA: ASoC: add STA32X codec driver 2011-06-20 6:07 ` [PATCH] " Daniel Mack @ 2011-06-22 11:06 ` Daniel Mack 2011-06-22 11:12 ` Mark Brown 0 siblings, 1 reply; 38+ messages in thread From: Daniel Mack @ 2011-06-22 11:06 UTC (permalink / raw) To: Mark Brown; +Cc: alsa-devel, lrg I really hate to bother you, but can we merge this before it gets lost? Daniel On Mon, Jun 20, 2011 at 8:07 AM, Daniel Mack <zonque@gmail.com> wrote: > On Fri, Jun 17, 2011 at 11:45 AM, Mark Brown > <broonie@opensource.wolfsonmicro.com> wrote: >> On Thu, Jun 16, 2011 at 02:07:23PM +0200, Daniel Mack wrote: >> >> Looks very good, a few small nits, if Liam's OK I guess we could fix >> these incrementally: >> >>> +#define DEBUG >> >> This shouldn't be in the submission. >> >>> +SOC_SINGLE_TLV("Master Volume", STA32X_MVOL, 0, 0xff, 1, mvol_tlv), >>> +SOC_SINGLE("Master Playback Switch", STA32X_MMUTE, 0, 1, 1), >>> +SOC_SINGLE("Ch1 Playback Switch", STA32X_MMUTE, 1, 1, 1), >>> +SOC_SINGLE("Ch2 Playback Switch", STA32X_MMUTE, 2, 1, 1), >>> +SOC_SINGLE("Ch3 Playback Switch", STA32X_MMUTE, 3, 1, 1), >>> +SOC_SINGLE_TLV("Ch1 Volume", STA32X_C1VOL, 0, 0xff, 1, chvol_tlv), >>> +SOC_SINGLE_TLV("Ch2 Volume", STA32X_C2VOL, 0, 0xff, 1, chvol_tlv), >>> +SOC_SINGLE_TLV("Ch3 Volume", STA32X_C3VOL, 0, 0xff, 1, chvol_tlv), >> >> The "Playback" should be consistently either present or not present in >> both volume and mute controls so UIs can match them up easily. >> > > I've sent a fixed version of this patch just after you mentioned these > issues. Hope it's all good now. > > Daniel > ^ permalink raw reply [flat|nested] 38+ messages in thread
* Re: [PATCH] ALSA: ASoC: add STA32X codec driver 2011-06-22 11:06 ` Daniel Mack @ 2011-06-22 11:12 ` Mark Brown 2011-06-22 12:10 ` Liam Girdwood 0 siblings, 1 reply; 38+ messages in thread From: Mark Brown @ 2011-06-22 11:12 UTC (permalink / raw) To: Daniel Mack; +Cc: alsa-devel, lrg On Wed, Jun 22, 2011 at 01:06:33PM +0200, Daniel Mack wrote: > I really hate to bother you, but can we merge this before it gets lost? I sent Liam a reminder about reviewing it the other day. ^ permalink raw reply [flat|nested] 38+ messages in thread
* Re: [PATCH] ALSA: ASoC: add STA32X codec driver 2011-06-22 11:12 ` Mark Brown @ 2011-06-22 12:10 ` Liam Girdwood 0 siblings, 0 replies; 38+ messages in thread From: Liam Girdwood @ 2011-06-22 12:10 UTC (permalink / raw) To: Mark Brown; +Cc: alsa-devel, lrg, Daniel Mack On 22/06/11 12:12, Mark Brown wrote: > On Wed, Jun 22, 2011 at 01:06:33PM +0200, Daniel Mack wrote: >> I really hate to bother you, but can we merge this before it gets lost? > > I sent Liam a reminder about reviewing it the other day. > _______________________________________________ Sorry, been on some training courses this week. Looks fine to me. Acked-by: Liam Girdwood <lrg@ti.com> ^ permalink raw reply [flat|nested] 38+ messages in thread
* [PATCH 2/2] ALSA: ASoC: add WM8782 ADC Codec Driver 2011-06-14 19:27 ASoC: Patches for an STA32X and WM8782 Daniel Mack 2011-06-14 19:27 ` [PATCH 1/2] ALSA: ASoC: add STA32X codec driver Daniel Mack @ 2011-06-14 19:27 ` Daniel Mack 2011-06-15 15:17 ` Mark Brown 2011-06-15 10:00 ` ASoC: Patches for an STA32X and WM8782 Johannes Stezenbach 2 siblings, 1 reply; 38+ messages in thread From: Daniel Mack @ 2011-06-14 19:27 UTC (permalink / raw) To: alsa-devel; +Cc: broonie, lrg From: Johannes Stezenbach <js@sig21.net> Signed-off-by: Johannes Stezenbach <js@sig21.net> [zonque@gmail.com: transform to new ASoC structure] --- sound/soc/codecs/Kconfig | 4 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8782.c | 77 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 83 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/wm8782.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index dd075f2..64bc0d2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -72,6 +72,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8770 if SPI_MASTER select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI + select SND_SOC_WM8782 select SND_SOC_WM8804 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8900 if I2C select SND_SOC_WM8903 if I2C @@ -303,6 +304,9 @@ config SND_SOC_WM8770 config SND_SOC_WM8776 tristate +config SND_SOC_WM8782 + tristate + config SND_SOC_WM8804 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 2ad1310..0ef7cd1 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -56,6 +56,7 @@ snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o snd-soc-wm8770-objs := wm8770.o snd-soc-wm8776-objs := wm8776.o +snd-soc-wm8782-objs := wm8782.o snd-soc-wm8804-objs := wm8804.o snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o @@ -149,6 +150,7 @@ obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o obj-$(CONFIG_SND_SOC_WM8770) += snd-soc-wm8770.o obj-$(CONFIG_SND_SOC_WM8776) += snd-soc-wm8776.o +obj-$(CONFIG_SND_SOC_WM8782) += snd-soc-wm8782.o obj-$(CONFIG_SND_SOC_WM8804) += snd-soc-wm8804.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c new file mode 100644 index 0000000..ca2ada9 --- /dev/null +++ b/sound/soc/codecs/wm8782.c @@ -0,0 +1,77 @@ +/* + * sound/soc/codecs/wm8782.c + * simple, strap-pin configured 24bit 2ch ADC + * + * Copyright: 2011 Raumfeld GmbH + * Author: Johannes Stezenbach <js@sig21.net> + * + * based on ad73311.c + * Copyright: Analog Device Inc. + * Author: Cliff Cai <cliff.cai@analog.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/soc.h> + +static struct snd_soc_dai_driver wm8782_dai = { + .name = "wm8782", + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_wm8782; + +static __devinit int wm8782_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, + &soc_codec_dev_wm8782, &wm8782_dai, 1); +} + +static int __devexit wm8782_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver wm8782_codec_driver = { + .driver = { + .name = "wm8782-codec", + .owner = THIS_MODULE, + }, + .probe = wm8782_probe, + .remove = wm8782_remove, +}; + +static int __init wm8782_init(void) +{ + return platform_driver_register(&wm8782_codec_driver); +} +module_init(wm8782_init); + +static void __exit wm8782_exit(void) +{ + platform_driver_unregister(&wm8782_codec_driver); +} +module_exit(wm8782_exit); + +MODULE_DESCRIPTION("ASoC WM8782 driver"); +MODULE_AUTHOR("Johannes Stezenbach <js@sig21.net>"); +MODULE_LICENSE("GPL"); -- 1.7.5.1 ^ permalink raw reply related [flat|nested] 38+ messages in thread
* Re: [PATCH 2/2] ALSA: ASoC: add WM8782 ADC Codec Driver 2011-06-14 19:27 ` [PATCH 2/2] ALSA: ASoC: add WM8782 ADC Codec Driver Daniel Mack @ 2011-06-15 15:17 ` Mark Brown 2011-06-16 8:20 ` Daniel Mack 0 siblings, 1 reply; 38+ messages in thread From: Mark Brown @ 2011-06-15 15:17 UTC (permalink / raw) To: Daniel Mack; +Cc: alsa-devel, lrg On Tue, Jun 14, 2011 at 09:27:06PM +0200, Daniel Mack wrote: > + .rates = SNDRV_PCM_RATE_48000, Datasheet says 8kHz to 192kHz are supported, there's a pin strap to select 96kHz and 192kHz rates so those could be omitted with a comment at a first pass. > + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, 20 bit is also supported. > + .driver = { > + .name = "wm8782-codec", Drop the -codec - not only is it redundant but the device is an ADC only. > + .owner = THIS_MODULE, > + }, > + .probe = wm8782_probe, > + .remove = wm8782_remove, > +}; > + > +static int __init wm8782_init(void) > +{ > + return platform_driver_register(&wm8782_codec_driver); > +} > +module_init(wm8782_init); > + > +static void __exit wm8782_exit(void) > +{ > + platform_driver_unregister(&wm8782_codec_driver); > +} > +module_exit(wm8782_exit); > + > +MODULE_DESCRIPTION("ASoC WM8782 driver"); > +MODULE_AUTHOR("Johannes Stezenbach <js@sig21.net>"); > +MODULE_LICENSE("GPL"); > -- > 1.7.5.1 > ^ permalink raw reply [flat|nested] 38+ messages in thread
* [PATCH 2/2] ALSA: ASoC: add WM8782 ADC Codec Driver 2011-06-15 15:17 ` Mark Brown @ 2011-06-16 8:20 ` Daniel Mack 2011-06-22 12:12 ` Liam Girdwood 0 siblings, 1 reply; 38+ messages in thread From: Daniel Mack @ 2011-06-16 8:20 UTC (permalink / raw) To: alsa-devel; +Cc: broonie, lrg From: Johannes Stezenbach <js@sig21.net> Signed-off-by: Johannes Stezenbach <js@sig21.net> [zonque@gmail.com: transform to new ASoC structure] --- sound/soc/codecs/Kconfig | 4 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8782.c | 80 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 86 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/wm8782.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index dd075f2..64bc0d2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -72,6 +72,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8770 if SPI_MASTER select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI + select SND_SOC_WM8782 select SND_SOC_WM8804 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8900 if I2C select SND_SOC_WM8903 if I2C @@ -303,6 +304,9 @@ config SND_SOC_WM8770 config SND_SOC_WM8776 tristate +config SND_SOC_WM8782 + tristate + config SND_SOC_WM8804 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 2ad1310..0ef7cd1 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -56,6 +56,7 @@ snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o snd-soc-wm8770-objs := wm8770.o snd-soc-wm8776-objs := wm8776.o +snd-soc-wm8782-objs := wm8782.o snd-soc-wm8804-objs := wm8804.o snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o @@ -149,6 +150,7 @@ obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o obj-$(CONFIG_SND_SOC_WM8770) += snd-soc-wm8770.o obj-$(CONFIG_SND_SOC_WM8776) += snd-soc-wm8776.o +obj-$(CONFIG_SND_SOC_WM8782) += snd-soc-wm8782.o obj-$(CONFIG_SND_SOC_WM8804) += snd-soc-wm8804.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c new file mode 100644 index 0000000..a2a09f8 --- /dev/null +++ b/sound/soc/codecs/wm8782.c @@ -0,0 +1,80 @@ +/* + * sound/soc/codecs/wm8782.c + * simple, strap-pin configured 24bit 2ch ADC + * + * Copyright: 2011 Raumfeld GmbH + * Author: Johannes Stezenbach <js@sig21.net> + * + * based on ad73311.c + * Copyright: Analog Device Inc. + * Author: Cliff Cai <cliff.cai@analog.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/soc.h> + +static struct snd_soc_dai_driver wm8782_dai = { + .name = "wm8782", + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + /* For configurations with FSAMPEN=0 */ + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_LE, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_wm8782; + +static __devinit int wm8782_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, + &soc_codec_dev_wm8782, &wm8782_dai, 1); +} + +static int __devexit wm8782_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver wm8782_codec_driver = { + .driver = { + .name = "wm8782", + .owner = THIS_MODULE, + }, + .probe = wm8782_probe, + .remove = wm8782_remove, +}; + +static int __init wm8782_init(void) +{ + return platform_driver_register(&wm8782_codec_driver); +} +module_init(wm8782_init); + +static void __exit wm8782_exit(void) +{ + platform_driver_unregister(&wm8782_codec_driver); +} +module_exit(wm8782_exit); + +MODULE_DESCRIPTION("ASoC WM8782 driver"); +MODULE_AUTHOR("Johannes Stezenbach <js@sig21.net>"); +MODULE_LICENSE("GPL"); -- 1.7.5.1 ^ permalink raw reply related [flat|nested] 38+ messages in thread
* Re: [PATCH 2/2] ALSA: ASoC: add WM8782 ADC Codec Driver 2011-06-16 8:20 ` Daniel Mack @ 2011-06-22 12:12 ` Liam Girdwood 0 siblings, 0 replies; 38+ messages in thread From: Liam Girdwood @ 2011-06-22 12:12 UTC (permalink / raw) To: Daniel Mack; +Cc: alsa-devel, broonie, lrg On 16/06/11 09:20, Daniel Mack wrote: > From: Johannes Stezenbach <js@sig21.net> > > Signed-off-by: Johannes Stezenbach <js@sig21.net> > [zonque@gmail.com: transform to new ASoC structure] > --- > sound/soc/codecs/Kconfig | 4 ++ > sound/soc/codecs/Makefile | 2 + > sound/soc/codecs/wm8782.c | 80 +++++++++++++++++++++++++++++++++++++++++++++ > 3 files changed, 86 insertions(+), 0 deletions(-) > create mode 100644 sound/soc/codecs/wm8782.c Acked-by: Liam Girdwood <lrg@ti.com> ^ permalink raw reply [flat|nested] 38+ messages in thread
* Re: ASoC: Patches for an STA32X and WM8782 2011-06-14 19:27 ASoC: Patches for an STA32X and WM8782 Daniel Mack 2011-06-14 19:27 ` [PATCH 1/2] ALSA: ASoC: add STA32X codec driver Daniel Mack 2011-06-14 19:27 ` [PATCH 2/2] ALSA: ASoC: add WM8782 ADC Codec Driver Daniel Mack @ 2011-06-15 10:00 ` Johannes Stezenbach 2 siblings, 0 replies; 38+ messages in thread From: Johannes Stezenbach @ 2011-06-15 10:00 UTC (permalink / raw) To: Daniel Mack; +Cc: alsa-devel, broonie, lrg Hi Daniel, On Tue, Jun 14, 2011 at 09:27:04PM +0200, Daniel Mack wrote: > I'm sending two patches to support the ST STA32x codec and the WM8782 ADC. > The code was originally written by Johannes Stezenbach for 2.6.36. > > As the ASoC internals changed quite a bit due to the multi-component > reconstruction, I took care for the code transition to make these drivers > compatible to more recent versions. If there's anything left to care for > in these drivers, please let me know. Thank you very much for doing that work! In the STA32x driver there are a few controls #if 0'd, and there are corresponding FIXMEs in sta32x_probe() where the operating mode and channel mapping is hard coded. I think the #if 0'd controls can be deleted, this was the wrong idea. The mode configuration should eventually be moved to platform data, but I thought this work is best left to a possible future second user of the driver ;-) Johannes ^ permalink raw reply [flat|nested] 38+ messages in thread
end of thread, other threads:[~2011-06-22 14:09 UTC | newest] Thread overview: 38+ messages (download: mbox.gz / follow: Atom feed) -- links below jump to the message on this page -- 2011-06-14 19:27 ASoC: Patches for an STA32X and WM8782 Daniel Mack 2011-06-14 19:27 ` [PATCH 1/2] ALSA: ASoC: add STA32X codec driver Daniel Mack 2011-06-15 15:05 ` Mark Brown 2011-06-15 20:53 ` Johannes Stezenbach 2011-06-16 8:39 ` Lars-Peter Clausen 2011-06-16 9:12 ` Johannes Stezenbach 2011-06-16 9:51 ` Daniel Mack 2011-06-16 10:13 ` Mark Brown 2011-06-16 11:03 ` Daniel Mack 2011-06-16 11:11 ` Lars-Peter Clausen 2011-06-16 11:26 ` Daniel Mack 2011-06-16 11:38 ` Lars-Peter Clausen 2011-06-16 11:47 ` Daniel Mack 2011-06-16 11:51 ` Lars-Peter Clausen 2011-06-16 11:54 ` Daniel Mack 2011-06-16 8:19 ` Daniel Mack 2011-06-16 8:49 ` Liam Girdwood 2011-06-16 8:53 ` Daniel Mack 2011-06-16 12:07 ` [PATCH] " Daniel Mack 2011-06-17 8:30 ` Daniel Mack 2011-06-17 9:45 ` Mark Brown 2011-06-17 9:54 ` Daniel Mack 2011-06-22 12:54 ` Mark Brown 2011-06-22 12:59 ` [PATCH 1/2] " Daniel Mack 2011-06-22 12:59 ` [PATCH 2/2] ALSA: ASoC: add WM8782 ADC Codec Driver Daniel Mack 2011-06-22 13:26 ` [PATCH 1/2] ALSA: ASoC: add STA32X codec driver Mark Brown 2011-06-22 13:34 ` Daniel Mack 2011-06-22 13:59 ` Mark Brown 2011-06-22 14:09 ` Paul Menzel 2011-06-20 6:07 ` [PATCH] " Daniel Mack 2011-06-22 11:06 ` Daniel Mack 2011-06-22 11:12 ` Mark Brown 2011-06-22 12:10 ` Liam Girdwood 2011-06-14 19:27 ` [PATCH 2/2] ALSA: ASoC: add WM8782 ADC Codec Driver Daniel Mack 2011-06-15 15:17 ` Mark Brown 2011-06-16 8:20 ` Daniel Mack 2011-06-22 12:12 ` Liam Girdwood 2011-06-15 10:00 ` ASoC: Patches for an STA32X and WM8782 Johannes Stezenbach
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