[alsa-devel] [PATCH 1/2] ALSA: ASoC: add STA32X codec driver
Daniel Mack
zonque at gmail.com
Tue Jun 14 21:27:05 CEST 2011
From: Johannes Stezenbach <js at sig21.net>
Signed-off-by: Johannes Stezenbach <js at sig21.net>
[zonque at 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 at sig21.net>
+ *
+ * based on code from:
+ * Wolfson Microelectronics PLC.
+ * Mark Brown <broonie at opensource.wolfsonmicro.com>
+ * Freescale Semiconductor, Inc.
+ * Timur Tabi <timur at 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 at 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 at sig21.net>
+ *
+ * based on code from:
+ * Wolfson Microelectronics PLC.
+ * Mark Brown <broonie at 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
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