From: Jarkko Nikula <jarkko.nikula@xxxxxxxxxx>
This codec driver template represents an I2C controlled multichannel audio
codec that has many typical ASoC codec driver features like volume controls,
mixer stages, mux selection, output power control, in-codec audio routings,
codec bias management and DAI link configuration.
Updates from Stefan Roese, 2014-04-28:
Port the HA DSP codec driver to Linux v3.15-rc. This includes
support for DT based probing. No platform-data code is needed
any more, DT nodes are sufficient.
Signed-off-by: Jarkko Nikula <jarkko.nikula@xxxxxxxxxx>
Signed-off-by: Stefan Roese <sr@xxxxxxx>
Cc: Thorsten Eisbein <thorsten.eisbein@xxxxxxxxxxxxxxxxx>
---
sound/soc/codecs/Kconfig | 4 +
sound/soc/codecs/Makefile | 2 +
sound/soc/codecs/ha-dsp.c | 419 ++++++++++++++++++++++++++++++++++++++++++++++
sound/soc/codecs/ha-dsp.h | 50 ++++++
4 files changed, 475 insertions(+)
create mode 100644 sound/soc/codecs/ha-dsp.c
create mode 100644 sound/soc/codecs/ha-dsp.h
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index f0e8401..f357988 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -51,6 +51,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_DA732X if I2C
select SND_SOC_DA9055 if I2C
select SND_SOC_BT_SCO
+ select SND_SOC_HA_DSP if I2C
select SND_SOC_ISABELLE if I2C
select SND_SOC_JZ4740_CODEC
select SND_SOC_LM4857 if I2C
@@ -343,6 +344,9 @@ config SND_SOC_BT_SCO
config SND_SOC_DMIC
tristate
+config SND_SOC_HA_DSP
+ tristate
+
config SND_SOC_HDMI_CODEC
tristate "HDMI stub CODEC"
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 3c4d275..f296bec 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -39,6 +39,7 @@ snd-soc-da732x-objs := da732x.o
snd-soc-da9055-objs := da9055.o
snd-soc-bt-sco-objs := bt-sco.o
snd-soc-dmic-objs := dmic.o
+snd-soc-ha-dsp-objs := ha-dsp.o
snd-soc-isabelle-objs := isabelle.o
snd-soc-jz4740-codec-objs := jz4740.o
snd-soc-l3-objs := l3.o
@@ -190,6 +191,7 @@ obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o
obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o
obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o
obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o
+obj-$(CONFIG_SND_SOC_HA_DSP) += snd-soc-ha-dsp.o
obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o
obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o
obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o
diff --git a/sound/soc/codecs/ha-dsp.c b/sound/soc/codecs/ha-dsp.c
new file mode 100644
index 0000000..7cf24dc
--- /dev/null
+++ b/sound/soc/codecs/ha-dsp.c
@@ -0,0 +1,419 @@
+/*
+ * ha-dsp.c -- HA DSP ALSA SoC Audio driver
+ *
+ * Copyright 2011 Head acoustics GmbH
+ *
+ * Author: Jarkko Nikula <jarkko.nikula@xxxxxxxxxx>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/regmap.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/tlv.h>
+#include <sound/initval.h>
+
+#include "ha-dsp.h"
+
+/* Reset default register values for soc-cache */
+static const struct reg_default ha_dsp_reg_defaults[] = {
+ { 0x00, 0x00 },
+ { 0x01, 0x55 },
+ { 0x02, 0x55 },
+ { 0x03, 0x00 },
+ { 0x04, 0x00 },
+ { 0x05, 0x00 },
+ { 0x06, 0x00 },
+ { 0x07, 0x00 },
+ { 0x08, 0x02 },
+ { 0x09, 0x02 },
+ { 0x0a, 0x02 },
+ { 0x0b, 0x02 },
+ { 0x0c, 0x02 },
+ { 0x0d, 0x02 },
+ { 0x0e, 0x02 },
+ { 0x0f, 0x02 },
+};
+
+/* DSP mode selection */
+static const char *ha_dsp_mode_texts[] = {"Mode 1", "Mode 2"};
+static SOC_ENUM_SINGLE_DECL(ha_dsp_mode_enum, HA_DSP_CTRL, 0,
+ ha_dsp_mode_texts);
+
+/* Monitor output mux selection */
+static const char *ha_dsp_monitor_texts[] = {"Off", "ADC", "DAC"};
+static SOC_ENUM_SINGLE_DECL(ha_dsp_monitor_enum, HA_DSP_CTRL, 1,
+ ha_dsp_monitor_texts);
+
+static const struct snd_kcontrol_new ha_dsp_monitor_control =
+ SOC_DAPM_ENUM("Route", ha_dsp_monitor_enum);
+
+/* Output mixers */
+static const struct snd_kcontrol_new ha_dsp_out1_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DAC Switch", HA_DSP_OUT1_CTRL, 1, 1, 0),
+ SOC_DAPM_SINGLE("IN Bypass Switch", HA_DSP_OUT1_CTRL, 2, 1, 0),
+};
+static const struct snd_kcontrol_new ha_dsp_out2_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DAC Switch", HA_DSP_OUT2_CTRL, 1, 1, 0),
+ SOC_DAPM_SINGLE("IN Bypass Switch", HA_DSP_OUT2_CTRL, 2, 1, 0),
+};
+static const struct snd_kcontrol_new ha_dsp_out3_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DAC Switch", HA_DSP_OUT3_CTRL, 1, 1, 0),
+ SOC_DAPM_SINGLE("IN Bypass Switch", HA_DSP_OUT3_CTRL, 2, 1, 0),
+};
+static const struct snd_kcontrol_new ha_dsp_out4_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DAC Switch", HA_DSP_OUT4_CTRL, 1, 1, 0),
+ SOC_DAPM_SINGLE("IN Bypass Switch", HA_DSP_OUT4_CTRL, 2, 1, 0),
+};
+static const struct snd_kcontrol_new ha_dsp_out5_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DAC Switch", HA_DSP_OUT5_CTRL, 1, 1, 0),
+ SOC_DAPM_SINGLE("IN Bypass Switch", HA_DSP_OUT5_CTRL, 2, 1, 0),
+};
+static const struct snd_kcontrol_new ha_dsp_out6_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DAC Switch", HA_DSP_OUT6_CTRL, 1, 1, 0),
+ SOC_DAPM_SINGLE("IN Bypass Switch", HA_DSP_OUT6_CTRL, 2, 1, 0),
+};
+static const struct snd_kcontrol_new ha_dsp_out7_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DAC Switch", HA_DSP_OUT7_CTRL, 1, 1, 0),
+ SOC_DAPM_SINGLE("IN Bypass Switch", HA_DSP_OUT1_CTRL, 2, 1, 0),
+};
+static const struct snd_kcontrol_new ha_dsp_out8_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DAC Switch", HA_DSP_OUT8_CTRL, 1, 1, 0),
+ SOC_DAPM_SINGLE("IN Bypass Switch", HA_DSP_OUT8_CTRL, 2, 1, 0),
+};
+
+static const struct snd_kcontrol_new ha_dsp_snd_controls[] = {
+ SOC_SINGLE("ADC Capture Volume",
+ HA_DSP_ADC_VOL, 0, 0x7f, 0),
+ SOC_SINGLE("ADC Capture Switch",
+ HA_DSP_ADC_VOL, 7, 0x01, 1),
+
+ SOC_SINGLE("PCM Playback Volume",
+ HA_DSP_DAC_VOL, 0, 0x7f, 0),
+ SOC_SINGLE("PCM Playback Switch",
+ HA_DSP_DAC_VOL, 7, 0x01, 1),
+
+ SOC_ENUM("DSP Mode", ha_dsp_mode_enum),
+};
+
+static const struct snd_soc_dapm_widget ha_dsp_widgets[] = {
+ SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0),
+
+ SND_SOC_DAPM_MIXER("OUT1 Mixer", SND_SOC_NOPM, 0, 0,
+ &ha_dsp_out1_mixer_controls[0],
+ ARRAY_SIZE(ha_dsp_out1_mixer_controls)),
+ SND_SOC_DAPM_MIXER("OUT2 Mixer", SND_SOC_NOPM, 0, 0,
+ &ha_dsp_out2_mixer_controls[0],
+ ARRAY_SIZE(ha_dsp_out2_mixer_controls)),
+ SND_SOC_DAPM_MIXER("OUT3 Mixer", SND_SOC_NOPM, 0, 0,
+ &ha_dsp_out3_mixer_controls[0],
+ ARRAY_SIZE(ha_dsp_out3_mixer_controls)),
+ SND_SOC_DAPM_MIXER("OUT4 Mixer", SND_SOC_NOPM, 0, 0,
+ &ha_dsp_out4_mixer_controls[0],
+ ARRAY_SIZE(ha_dsp_out4_mixer_controls)),
+ SND_SOC_DAPM_MIXER("OUT5 Mixer", SND_SOC_NOPM, 0, 0,
+ &ha_dsp_out5_mixer_controls[0],
+ ARRAY_SIZE(ha_dsp_out5_mixer_controls)),
+ SND_SOC_DAPM_MIXER("OUT6 Mixer", SND_SOC_NOPM, 0, 0,
+ &ha_dsp_out6_mixer_controls[0],
+ ARRAY_SIZE(ha_dsp_out6_mixer_controls)),
+ SND_SOC_DAPM_MIXER("OUT7 Mixer", SND_SOC_NOPM, 0, 0,
+ &ha_dsp_out7_mixer_controls[0],
+ ARRAY_SIZE(ha_dsp_out7_mixer_controls)),
+ SND_SOC_DAPM_MIXER("OUT8 Mixer", SND_SOC_NOPM, 0, 0,
+ &ha_dsp_out8_mixer_controls[0],
+ ARRAY_SIZE(ha_dsp_out8_mixer_controls)),
+
+ SND_SOC_DAPM_PGA("OUT1 PGA", HA_DSP_OUT1_CTRL, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("OUT2 PGA", HA_DSP_OUT2_CTRL, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("OUT3 PGA", HA_DSP_OUT3_CTRL, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("OUT4 PGA", HA_DSP_OUT4_CTRL, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("OUT5 PGA", HA_DSP_OUT5_CTRL, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("OUT6 PGA", HA_DSP_OUT6_CTRL, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("OUT7 PGA", HA_DSP_OUT7_CTRL, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("OUT8 PGA", HA_DSP_OUT8_CTRL, 0, 0, NULL, 0),
+
+ SND_SOC_DAPM_MUX("Monitor Out Mux", SND_SOC_NOPM, 0, 0,
+ &ha_dsp_monitor_control),
+
+ /* Input pins */
+ SND_SOC_DAPM_INPUT("IN1"),
+ SND_SOC_DAPM_INPUT("IN2"),
+ SND_SOC_DAPM_INPUT("IN3"),
+ SND_SOC_DAPM_INPUT("IN4"),
+ SND_SOC_DAPM_INPUT("IN5"),
+ SND_SOC_DAPM_INPUT("IN6"),
+ SND_SOC_DAPM_INPUT("IN7"),
+ SND_SOC_DAPM_INPUT("IN8"),
+
+ /* Output pins */
+ SND_SOC_DAPM_OUTPUT("OUT1"),
+ SND_SOC_DAPM_OUTPUT("OUT2"),
+ SND_SOC_DAPM_OUTPUT("OUT3"),
+ SND_SOC_DAPM_OUTPUT("OUT4"),
+ SND_SOC_DAPM_OUTPUT("OUT5"),
+ SND_SOC_DAPM_OUTPUT("OUT6"),
+ SND_SOC_DAPM_OUTPUT("OUT7"),
+ SND_SOC_DAPM_OUTPUT("OUT8"),
+ SND_SOC_DAPM_OUTPUT("MONITOR"),
+};
+
+static const struct snd_soc_dapm_route ha_dsp_routes[] = {
+ /* Inputs to ADC */
+ {"ADC", NULL, "IN1"},
+ {"ADC", NULL, "IN2"},
+ {"ADC", NULL, "IN3"},
+ {"ADC", NULL, "IN4"},
+ {"ADC", NULL, "IN5"},
+ {"ADC", NULL, "IN6"},
+ {"ADC", NULL, "IN7"},
+ {"ADC", NULL, "IN8"},
+
+ /* DAC and input bypass paths to outputs */
+ {"OUT1 Mixer", "DAC Switch", "DAC"},
+ {"OUT1 Mixer", "IN Bypass Switch", "IN1"},
+ {"OUT1 PGA", NULL, "OUT1 Mixer"},
+ {"OUT1", NULL, "OUT1 PGA"},
+
+ {"OUT2 Mixer", "DAC Switch", "DAC"},
+ {"OUT2 Mixer", "IN Bypass Switch", "IN2"},
+ {"OUT2 PGA", NULL, "OUT2 Mixer"},
+ {"OUT2", NULL, "OUT2 PGA"},
+
+ {"OUT3 Mixer", "DAC Switch", "DAC"},
+ {"OUT3 Mixer", "IN Bypass Switch", "IN3"},
+ {"OUT3 PGA", NULL, "OUT3 Mixer"},
+ {"OUT3", NULL, "OUT3 PGA"},
+
+ {"OUT4 Mixer", "DAC Switch", "DAC"},
+ {"OUT4 Mixer", "IN Bypass Switch", "IN4"},
+ {"OUT4 PGA", NULL, "OUT4 Mixer"},
+ {"OUT4", NULL, "OUT4 PGA"},
+
+ {"OUT5 Mixer", "DAC Switch", "DAC"},
+ {"OUT5 Mixer", "IN Bypass Switch", "IN5"},
+ {"OUT5 PGA", NULL, "OUT5 Mixer"},
+ {"OUT5", NULL, "OUT5 PGA"},
+
+ {"OUT6 Mixer", "DAC Switch", "DAC"},
+ {"OUT6 Mixer", "IN Bypass Switch", "IN6"},
+ {"OUT6 PGA", NULL, "OUT6 Mixer"},
+ {"OUT6", NULL, "OUT6 PGA"},
+
+ {"OUT7 Mixer", "DAC Switch", "DAC"},
+ {"OUT7 Mixer", "IN Bypass Switch", "IN7"},
+ {"OUT7 PGA", NULL, "OUT7 Mixer"},
+ {"OUT7", NULL, "OUT7 PGA"},
+
+ {"OUT8 Mixer", "DAC Switch", "DAC"},
+ {"OUT8 Mixer", "IN Bypass Switch", "IN8"},
+ {"OUT8 PGA", NULL, "OUT8 Mixer"},
+ {"OUT8", NULL, "OUT8 PGA"},
+
+ /* Monitor output */
+ {"Monitor Out Mux", "ADC", "ADC"},
+ {"Monitor Out Mux", "DAC", "DAC"},
+ {"MONITOR", NULL, "Monitor Out Mux"},
+};
+
+static int ha_dsp_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;
+
+ dev_dbg(codec->dev, "Sample format 0x%X\n", params_format(params));
+ dev_dbg(codec->dev, "Channels %d\n", params_channels(params));
+ dev_dbg(codec->dev, "Rate %d\n", params_rate(params));
+
+ return 0;
+}
+
+static int ha_dsp_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+
+ /* codec role */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ dev_dbg(codec->dev, "Codec is master\n");
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ dev_dbg(codec->dev, "Codec is slave\n");
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* DAI format */
+ dev_dbg(codec->dev, "DAI format 0x%X",
+ fmt & SND_SOC_DAIFMT_FORMAT_MASK);
+
+ /* Bit clock and frame sync polarities */
+ dev_dbg(codec->dev, "Clock polarities 0x%X\n",
+ fmt & SND_SOC_DAIFMT_INV_MASK);
+
+ return 0;
+}
+
+static int ha_dsp_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ dev_dbg(codec->dev, "Changing bias from %d to %d\n",
+ codec->dapm.bias_level, level);
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ /* Set PLL on */
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ /* Set power on, Set PLL off */
+ break;
+ case SND_SOC_BIAS_OFF:
+ /* Set power down */
+ break;
+ }
+ codec->dapm.bias_level = level;
+
+ return 0;
+}
+
+static struct snd_soc_dai_ops ha_dsp_dai_ops = {
+ .hw_params = ha_dsp_hw_params,
+ .set_fmt = ha_dsp_set_dai_fmt,
+};
+
+static struct snd_soc_dai_driver ha_dsp_dai = {
+ .name = "ha-dsp-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 16,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ /* We use only 32 Bits for Audio */
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 16,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ /* We use only 32 Bits for Audio */
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+ },
+ .ops = &ha_dsp_dai_ops,
+};
+
+static int ha_dsp_probe(struct snd_soc_codec *codec)
+{
+ int ret;
+
+ codec->control_data = dev_get_regmap(codec->dev->parent, NULL);
+ ret = snd_soc_codec_set_cache_io(codec, codec->control_data);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int ha_dsp_remove(struct snd_soc_codec *codec)
+{
+ snd_soc_write(codec, HA_DSP_CTRL, HA_DSP_SW_RESET);
+
+ return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_ha_dsp = {
+ .probe = ha_dsp_probe,
+ .remove = ha_dsp_remove,
+ .set_bias_level = ha_dsp_set_bias_level,
+
+ .controls = ha_dsp_snd_controls,
+ .num_controls = ARRAY_SIZE(ha_dsp_snd_controls),
+ .dapm_widgets = ha_dsp_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(ha_dsp_widgets),
+ .dapm_routes = ha_dsp_routes,
+ .num_dapm_routes = ARRAY_SIZE(ha_dsp_routes),
+};
+
+static const struct regmap_config ha_dsp_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .max_register = 0x0f,
+ .reg_defaults = ha_dsp_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(ha_dsp_reg_defaults),
+ .cache_type = REGCACHE_RBTREE,
+};
+
+static int ha_dsp_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ struct regmap *regmap;
+ int ret;
+
+ regmap = devm_regmap_init_i2c(client, &ha_dsp_regmap);
+ if (IS_ERR(regmap)) {
+ ret = PTR_ERR(regmap);
+ dev_err(&client->dev, "Failed to create regmap: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_register_codec(&client->dev, &soc_codec_dev_ha_dsp,
+ &ha_dsp_dai, 1);
+
+ return ret;
+}
+
+static int ha_dsp_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+
+ return 0;
+}
+
+/*
+ * This name/ID is neded to match the DT node for the codec
+ */
+static const struct i2c_device_id ha_dsp_i2c_id[] = {
+ { "ha-dsp-audio", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, ha_dsp_i2c_id);
+
+static struct i2c_driver ha_dsp_i2c_driver = {
+ .driver = {
+ .name = "ha-dsp-codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = ha_dsp_i2c_probe,
+ .remove = ha_dsp_i2c_remove,
+ .id_table = ha_dsp_i2c_id,
+};
+
+module_i2c_driver(ha_dsp_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC HA DSP driver");
+MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@xxxxxxxxxx>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ha-dsp.h b/sound/soc/codecs/ha-dsp.h
new file mode 100644
index 0000000..cab82f8
--- /dev/null
+++ b/sound/soc/codecs/ha-dsp.h
@@ -0,0 +1,50 @@
+/*
+ * ha-dsp.h -- HA DSP ALSA SoC Audio driver
+ *
+ * Copyright 2011 Head acoustics GmbH
+ *
+ * Author: Jarkko Nikula <jhnikula@xxxxxxxxx>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __HA_DSP_H__
+#define __HA_DSP_H__
+
+/* Registers */
+
+/*
+ * Bit 2-1: Monitor output selection: Off, ADC, DAC
+ * Bit 0: DSP Mode
+ */
+#define HA_DSP_CTRL 0x00
+
+/*
+ * Bit 7: Mute
+ * Bit 6-0: Volume
+ */
+#define HA_DSP_DAC_VOL 0x01
+#define HA_DSP_ADC_VOL 0x02
+
+/*
+ * Bit 2: INx Bypass to OUTx Switch
+ * Bit 1: DAC to OUTx switch
+ * Bit 0: Output power
+ */
+#define HA_DSP_OUT1_CTRL 0x08
+#define HA_DSP_OUT2_CTRL 0x09
+#define HA_DSP_OUT3_CTRL 0x0a
+#define HA_DSP_OUT4_CTRL 0x0b
+#define HA_DSP_OUT5_CTRL 0x0c
+#define HA_DSP_OUT6_CTRL 0x0d
+#define HA_DSP_OUT7_CTRL 0x0e
+#define HA_DSP_OUT8_CTRL 0x0f
+
+/* Register bits and values */
+
+/* HA_DSP_CTRL */
+#define HA_DSP_SW_RESET 0xff
+
+#endif
--
1.9.1
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