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.
This driver is based on an early version provided by Jarkko Nikula.
Signed-off-by: Jarkko Nikula jarkko.nikula@bitmer.com Signed-off-by: Stefan Roese sr@denx.de Cc: Thorsten Eisbein thorsten.eisbein@head-acoustics.de Cc: Lars-Peter Clausen lars@metafoo.de Cc: Mark Brown broonie@kernel.org --- v2: - Added/changed copyright line - Changed authorship (as suggested by Jarkko the original author) - Added Thorsten as maintainer - Remove ha_dsp_hw_params() and ha_dsp_set_dai_fmt() as its not used (only needed for CODEC as clock master which is currently not suported). - Removed some unneeded include files - "const char *const foo" used instead of "const char *foo" - SOC_MIXER_ARRAY() helper macro used - Removed ha_dsp_set_bias_level() and use default implementation - Use codec->dev instead of codec->dev->parent in dev_get_regmap() - Added CODEC reset to probe - Remove "ret" in ha_dsp_i2c_probe()
sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ha-dsp.c | 333 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ha-dsp.h | 50 +++++++ 4 files changed, 389 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..5a3c7ef --- /dev/null +++ b/sound/soc/codecs/ha-dsp.c @@ -0,0 +1,333 @@ +/* + * ha-dsp.c -- HA DSP ALSA SoC Audio driver + * + * Copyright 2011-2014 HEAD acoustics GmbH + * + * Authors: + * Jarkko Nikula jarkko.nikula@bitmer.com + * Stefan Roese sr@denx.de + * Thorsten Eisbein thorsten.eisbein@head-acoustics.de + * + * 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. + * + * Maintainer: Thorsten Eisbein thorsten.eisbein@head-acoustics.de + */ + +#include <linux/module.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.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 *const 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 *const 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), + + SOC_MIXER_ARRAY("OUT1 Mixer", SND_SOC_NOPM, 0, 0, + ha_dsp_out1_mixer_controls), + SOC_MIXER_ARRAY("OUT2 Mixer", SND_SOC_NOPM, 0, 0, + ha_dsp_out2_mixer_controls), + SOC_MIXER_ARRAY("OUT3 Mixer", SND_SOC_NOPM, 0, 0, + ha_dsp_out3_mixer_controls), + SOC_MIXER_ARRAY("OUT4 Mixer", SND_SOC_NOPM, 0, 0, + ha_dsp_out4_mixer_controls), + SOC_MIXER_ARRAY("OUT5 Mixer", SND_SOC_NOPM, 0, 0, + ha_dsp_out5_mixer_controls), + SOC_MIXER_ARRAY("OUT6 Mixer", SND_SOC_NOPM, 0, 0, + ha_dsp_out6_mixer_controls), + SOC_MIXER_ARRAY("OUT7 Mixer", SND_SOC_NOPM, 0, 0, + ha_dsp_out7_mixer_controls), + SOC_MIXER_ARRAY("OUT8 Mixer", SND_SOC_NOPM, 0, 0, + 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 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, + }, +}; + +static int ha_dsp_probe(struct snd_soc_codec *codec) +{ + int ret; + + codec->control_data = dev_get_regmap(codec->dev, 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; + } + + snd_soc_write(codec, HA_DSP_CTRL, HA_DSP_SW_RESET); + + 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, + + .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; + + regmap = devm_regmap_init_i2c(client, &ha_dsp_regmap); + if (IS_ERR(regmap)) { + dev_err(&client->dev, "Failed to create regmap: %ld\n", + PTR_ERR(regmap)); + return PTR_ERR(regmap); + } + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_ha_dsp, + &ha_dsp_dai, 1); +} + +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@bitmer.com"); +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..6622f8a --- /dev/null +++ b/sound/soc/codecs/ha-dsp.h @@ -0,0 +1,50 @@ +/* + * ha-dsp.h -- HA DSP ALSA SoC Audio driver + * + * Copyright 2011-2014 HEAD acoustics GmbH + * + * Author: Jarkko Nikula jhnikula@gmail.com + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + */ + +#ifndef __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