[alsa-devel] [PATCH v5 1/7] ASoC: sirf: Add SiRF internal audio codec driver
RongJun Ying
rjying at gmail.com
Wed Mar 5 03:36:21 CET 2014
From: Rongjun Ying <rongjun.ying at csr.com>
SiRF internal audio codec is integrated in SiRF atlas6 and prima2 SoC.
Features include:
1. Stereo DAC and ADC with 16-bit resolution amd 48KHz sample rate
2. Support headphone and/or speaker output
3. Integrate headphone and speaker output amp
4. Support LINE and MIC input
5. Support single ended and differential input mode
---
-v5:
1. Drop all inlines.
2. Reordering the Kconfig and Makefile
3. Remove the sirf_audio_codec_reg_bits struct, use the new controls instead it.
4. Add some SND_SOC_DAPM_OUT_DRV instead of HP and SPK enable driver
5. Add audio codec clock supply instead of adc event callback
6. Fixed playback and capture can't concurrent work bug.
.../devicetree/bindings/sound/sirf-audio-codec.txt | 17 +
sound/soc/codecs/Kconfig | 6 +-
sound/soc/codecs/Makefile | 2 +
sound/soc/codecs/sirf-audio-codec.c | 534 ++++++++++++++++++++
sound/soc/codecs/sirf-audio-codec.h | 75 +++
5 files changed, 633 insertions(+), 1 deletions(-)
create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-codec.txt
create mode 100644 sound/soc/codecs/sirf-audio-codec.c
create mode 100644 sound/soc/codecs/sirf-audio-codec.h
diff --git a/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt b/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt
new file mode 100644
index 0000000..062f5ec
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt
@@ -0,0 +1,17 @@
+SiRF internal audio CODEC
+
+Required properties:
+
+ - compatible : "sirf,atlas6-audio-codec" or "sirf,prima2-audio-codec"
+
+ - reg : the register address of the device.
+
+ - clocks: the clock of SiRF internal audio codec
+
+Example:
+
+audiocodec: audiocodec at b0040000 {
+ compatible = "sirf,atlas6-audio-codec";
+ reg = <0xb0040000 0x10000>;
+ clocks = <&clks 27>;
+};
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index f2383eb..41a886d 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -71,6 +71,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_RT5640 if I2C
select SND_SOC_SGTL5000 if I2C
select SND_SOC_SI476X if MFD_SI476X_CORE
+ select SND_SOC_SIRF_AUDIO_CODEC
select SND_SOC_SN95031 if INTEL_SCU_IPC
select SND_SOC_SPDIF
select SND_SOC_SSM2518 if I2C
@@ -383,6 +384,10 @@ config SND_SOC_SIGMADSP
tristate
select CRC32
+config SND_SOC_SIRF_AUDIO_CODEC
+ tristate "SiRF SoC internal audio codec"
+ select REGMAP_MMIO
+
config SND_SOC_SN95031
tristate
@@ -633,5 +638,4 @@ config SND_SOC_ML26124
config SND_SOC_TPA6130A2
tristate "Texas Instruments TPA6130A2 headphone amplifier"
depends on I2C
-
endmenu
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 6af7a55..6bea5ba 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -134,6 +134,7 @@ snd-soc-wm9705-objs := wm9705.o
snd-soc-wm9712-objs := wm9712.o
snd-soc-wm9713-objs := wm9713.o
snd-soc-wm-hubs-objs := wm_hubs.o
+snd-soc-sirf-audio-codec-objs := sirf-audio-codec.o
# Amp
snd-soc-max9877-objs := max9877.o
@@ -274,6 +275,7 @@ obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o
obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o
obj-$(CONFIG_SND_SOC_WM_ADSP) += snd-soc-wm-adsp.o
obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o
+obj-$(CONFIG_SND_SOC_SIRF_AUDIO_CODEC) += snd-soc-sirf-audio-codec.o
# Amp
obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o
diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c
new file mode 100644
index 0000000..20e3ca9
--- /dev/null
+++ b/sound/soc/codecs/sirf-audio-codec.c
@@ -0,0 +1,534 @@
+/*
+ * SiRF audio codec driver
+ *
+ * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company.
+ *
+ * Licensed under GPLv2 or later.
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/pm_runtime.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/io.h>
+#include <linux/regmap.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "sirf-audio-codec.h"
+
+struct sirf_audio_codec {
+ struct clk *clk;
+ struct regmap *regmap;
+ u32 reg_ctrl0, reg_ctrl1;
+};
+
+static const char * const input_mode_mux[] = {"Single-ended",
+ "Differential"};
+
+static const struct soc_enum input_mode_mux_enum =
+ SOC_ENUM_SINGLE(AUDIO_IC_CODEC_CTRL1, 4, 2, input_mode_mux);
+
+static const struct snd_kcontrol_new sirf_audio_codec_input_mode_control =
+ SOC_DAPM_ENUM("Route", input_mode_mux_enum);
+
+static const DECLARE_TLV_DB_SCALE(playback_vol_tlv, -12400, 100, 0);
+static const DECLARE_TLV_DB_SCALE(capture_vol_tlv_prima2, 500, 100, 0);
+static const DECLARE_TLV_DB_RANGE(capture_vol_tlv_atlas6,
+ 0, 7, TLV_DB_SCALE_ITEM(-100, 100, 0),
+ 0x22, 0x3F, TLV_DB_SCALE_ITEM(700, 100, 0),
+);
+
+static struct snd_kcontrol_new volume_controls_atlas6[] = {
+ SOC_DOUBLE_TLV("Playback Volume", AUDIO_IC_CODEC_CTRL0, 21, 14,
+ 0x7F, 0, playback_vol_tlv),
+ SOC_DOUBLE_TLV("Capture Volume", AUDIO_IC_CODEC_CTRL1, 16, 10,
+ 0x3F, 0, capture_vol_tlv_atlas6),
+};
+
+static struct snd_kcontrol_new volume_controls_prima2[] = {
+ SOC_DOUBLE_TLV("Speaker Volume", AUDIO_IC_CODEC_CTRL0, 21, 14,
+ 0x7F, 0, playback_vol_tlv),
+ SOC_DOUBLE_TLV("Capture Volume", AUDIO_IC_CODEC_CTRL1, 15, 10,
+ 0x1F, 0, capture_vol_tlv_prima2),
+};
+
+static struct snd_kcontrol_new left_input_path_controls[] = {
+ SOC_DAPM_SINGLE("Line Left Switch", AUDIO_IC_CODEC_CTRL1, 6, 1, 0),
+ SOC_DAPM_SINGLE("Mic Left Switch", AUDIO_IC_CODEC_CTRL1, 3, 1, 0),
+};
+
+static struct snd_kcontrol_new right_input_path_controls[] = {
+ SOC_DAPM_SINGLE("Line Right Switch", AUDIO_IC_CODEC_CTRL1, 5, 1, 0),
+ SOC_DAPM_SINGLE("Mic Right Switch", AUDIO_IC_CODEC_CTRL1, 2, 1, 0),
+};
+
+static struct snd_kcontrol_new left_dac_to_hp_left_amp_switch_control =
+ SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 9, 1, 0);
+
+static struct snd_kcontrol_new left_dac_to_hp_right_amp_switch_control =
+ SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 8, 1, 0);
+
+static struct snd_kcontrol_new right_dac_to_hp_left_amp_switch_control =
+ SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 7, 1, 0);
+
+static struct snd_kcontrol_new right_dac_to_hp_right_amp_switch_control =
+ SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 6, 1, 0);
+
+static struct snd_kcontrol_new left_dac_to_speaker_lineout_switch_control =
+ SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 11, 1, 0);
+
+static struct snd_kcontrol_new right_dac_to_speaker_lineout_switch_control =
+ SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 10, 1, 0);
+
+/* After enable adc, Delay 200ms to avoid pop noise */
+static int adc_enable_delay_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ msleep(200);
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static void enable_and_reset_codec(struct regmap *regmap,
+ u32 codec_enable_bits, u32 codec_reset_bits)
+{
+ regmap_update_bits(regmap, AUDIO_IC_CODEC_CTRL1,
+ codec_enable_bits | codec_reset_bits,
+ codec_enable_bits | ~codec_reset_bits);
+ msleep(20);
+ regmap_update_bits(regmap, AUDIO_IC_CODEC_CTRL1,
+ codec_reset_bits, codec_reset_bits);
+}
+
+static int atlas6_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+#define ATLAS6_CODEC_ENABLE_BITS (1 << 29)
+#define ATLAS6_CODEC_RESET_BITS (1 << 28)
+ struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(w->codec->dev);
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ enable_and_reset_codec(sirf_audio_codec->regmap,
+ ATLAS6_CODEC_ENABLE_BITS, ATLAS6_CODEC_RESET_BITS);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ regmap_update_bits(sirf_audio_codec->regmap,
+ AUDIO_IC_CODEC_CTRL1, ATLAS6_CODEC_ENABLE_BITS,
+ ~ATLAS6_CODEC_ENABLE_BITS);
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static int prima2_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+#define PRIMA2_CODEC_ENABLE_BITS (1 << 27)
+#define PRIMA2_CODEC_RESET_BITS (1 << 26)
+ struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(w->codec->dev);
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ enable_and_reset_codec(sirf_audio_codec->regmap,
+ PRIMA2_CODEC_ENABLE_BITS, PRIMA2_CODEC_RESET_BITS);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ regmap_update_bits(sirf_audio_codec->regmap,
+ AUDIO_IC_CODEC_CTRL1, PRIMA2_CODEC_ENABLE_BITS,
+ ~PRIMA2_CODEC_ENABLE_BITS);
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget atlas6_output_driver_dapm_widgets[] = {
+ SND_SOC_DAPM_OUT_DRV("HP Left Driver", AUDIO_IC_CODEC_CTRL1,
+ 25, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("HP Right Driver", AUDIO_IC_CODEC_CTRL1,
+ 26, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("Speaker Driver", AUDIO_IC_CODEC_CTRL1,
+ 27, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_widget prima2_output_driver_dapm_widgets[] = {
+ SND_SOC_DAPM_OUT_DRV("HP Left Driver", AUDIO_IC_CODEC_CTRL1,
+ 23, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("HP Right Driver", AUDIO_IC_CODEC_CTRL1,
+ 24, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("Speaker Driver", AUDIO_IC_CODEC_CTRL1,
+ 25, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_widget atlas6_codec_clock_dapm_widget =
+ SND_SOC_DAPM_SUPPLY("codecclk", SND_SOC_NOPM, 0, 0,
+ atlas6_codec_enable_and_reset_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD);
+
+static const struct snd_soc_dapm_widget prima2_codec_clock_dapm_widget =
+ SND_SOC_DAPM_SUPPLY("codecclk", SND_SOC_NOPM, 0, 0,
+ prima2_codec_enable_and_reset_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD);
+
+static const struct snd_soc_dapm_widget sirf_audio_codec_dapm_widgets[] = {
+ SND_SOC_DAPM_DAC("DAC left", NULL, AUDIO_IC_CODEC_CTRL0, 1, 0),
+ SND_SOC_DAPM_DAC("DAC right", NULL, AUDIO_IC_CODEC_CTRL0, 0, 0),
+ SND_SOC_DAPM_SWITCH("Left dac to hp left amp", SND_SOC_NOPM, 0, 0,
+ &left_dac_to_hp_left_amp_switch_control),
+ SND_SOC_DAPM_SWITCH("Left dac to hp right amp", SND_SOC_NOPM, 0, 0,
+ &left_dac_to_hp_right_amp_switch_control),
+ SND_SOC_DAPM_SWITCH("Right dac to hp left amp", SND_SOC_NOPM, 0, 0,
+ &right_dac_to_hp_left_amp_switch_control),
+ SND_SOC_DAPM_SWITCH("Right dac to hp right amp", SND_SOC_NOPM, 0, 0,
+ &right_dac_to_hp_right_amp_switch_control),
+ SND_SOC_DAPM_OUT_DRV("HP amp left driver", AUDIO_IC_CODEC_CTRL0, 3, 0,
+ NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("HP amp right driver", AUDIO_IC_CODEC_CTRL0, 3, 0,
+ NULL, 0),
+
+ SND_SOC_DAPM_SWITCH("Left dac to speaker lineout", SND_SOC_NOPM, 0, 0,
+ &left_dac_to_speaker_lineout_switch_control),
+ SND_SOC_DAPM_SWITCH("Right dac to speaker lineout", SND_SOC_NOPM, 0, 0,
+ &right_dac_to_speaker_lineout_switch_control),
+ SND_SOC_DAPM_OUT_DRV("Speaker amp driver", AUDIO_IC_CODEC_CTRL0, 4, 0,
+ NULL, 0),
+
+ SND_SOC_DAPM_OUTPUT("HPOUTL"),
+ SND_SOC_DAPM_OUTPUT("HPOUTR"),
+ SND_SOC_DAPM_OUTPUT("SPKOUT"),
+
+ SND_SOC_DAPM_ADC_E("ADC left", NULL, AUDIO_IC_CODEC_CTRL1, 8, 0,
+ adc_enable_delay_event, SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_ADC_E("ADC right", NULL, AUDIO_IC_CODEC_CTRL1, 7, 0,
+ adc_enable_delay_event, SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_MIXER("Left PGA mixer", AUDIO_IC_CODEC_CTRL1, 1, 0,
+ &left_input_path_controls[0],
+ ARRAY_SIZE(left_input_path_controls)),
+ SND_SOC_DAPM_MIXER("Right PGA mixer", AUDIO_IC_CODEC_CTRL1, 0, 0,
+ &right_input_path_controls[0],
+ ARRAY_SIZE(right_input_path_controls)),
+
+ SND_SOC_DAPM_MUX("Mic input mode mux", SND_SOC_NOPM, 0, 0,
+ &sirf_audio_codec_input_mode_control),
+ SND_SOC_DAPM_MICBIAS("Mic Bias", AUDIO_IC_CODEC_PWR, 3, 0),
+ SND_SOC_DAPM_INPUT("MICIN1"),
+ SND_SOC_DAPM_INPUT("MICIN2"),
+ SND_SOC_DAPM_INPUT("LINEIN1"),
+ SND_SOC_DAPM_INPUT("LINEIN2"),
+
+ SND_SOC_DAPM_SUPPLY("HSL Phase Opposite", AUDIO_IC_CODEC_CTRL0,
+ 30, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_route sirf_audio_codec_map[] = {
+ {"SPKOUT", NULL, "Speaker Driver"},
+ {"Speaker Driver", NULL, "Speaker amp driver"},
+ {"Speaker amp driver", NULL, "Left dac to speaker lineout"},
+ {"Speaker amp driver", NULL, "Right dac to speaker lineout"},
+ {"Left dac to speaker lineout", "Switch", "DAC left"},
+ {"Right dac to speaker lineout", "Switch", "DAC right"},
+ {"HPOUTL", NULL, "HP Left Driver"},
+ {"HPOUTR", NULL, "HP Right Driver"},
+ {"HP Left Driver", NULL, "HP amp left driver"},
+ {"HP Right Driver", NULL, "HP amp right driver"},
+ {"HP amp left driver", NULL, "Right dac to hp left amp"},
+ {"HP amp right driver", NULL , "Right dac to hp right amp"},
+ {"HP amp left driver", NULL, "Left dac to hp left amp"},
+ {"HP amp right driver", NULL , "Right dac to hp right amp"},
+ {"Right dac to hp left amp", "Switch", "DAC left"},
+ {"Right dac to hp right amp", "Switch", "DAC right"},
+ {"Left dac to hp left amp", "Switch", "DAC left"},
+ {"Left dac to hp right amp", "Switch", "DAC right"},
+ {"DAC left", NULL, "codecclk"},
+ {"DAC right", NULL, "codecclk"},
+ {"DAC left", NULL, "Playback"},
+ {"DAC right", NULL, "Playback"},
+ {"DAC left", NULL, "HSL Phase Opposite"},
+ {"DAC right", NULL, "HSL Phase Opposite"},
+
+ {"Capture", NULL, "ADC left"},
+ {"Capture", NULL, "ADC right"},
+ {"ADC left", NULL, "codecclk"},
+ {"ADC right", NULL, "codecclk"},
+ {"ADC left", NULL, "Left PGA mixer"},
+ {"ADC right", NULL, "Right PGA mixer"},
+ {"Left PGA mixer", "Line Left Switch", "LINEIN2"},
+ {"Right PGA mixer", "Line Right Switch", "LINEIN1"},
+ {"Left PGA mixer", "Mic Left Switch", "MICIN2"},
+ {"Right PGA mixer", "Mic Right Switch", "Mic input mode mux"},
+ {"Mic input mode mux", "Single-ended", "MICIN1"},
+ {"Mic input mode mux", "Differential", "MICIN1"},
+};
+
+static int sirf_audio_codec_trigger(struct snd_pcm_substream *substream,
+ int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u32 val = 0;
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
+ /*
+ * This is a workaround, When stop playback,
+ * need disable HP amp, avoid the current noise.
+ */
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ break;
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ val = IC_HSLEN | IC_HSREN;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, AUDIO_IC_CODEC_CTRL0,
+ IC_HSLEN | IC_HSREN, val);
+ return 0;
+}
+
+struct snd_soc_dai_ops sirf_audio_codec_dai_ops = {
+ .trigger = sirf_audio_codec_trigger,
+};
+
+struct snd_soc_dai_driver sirf_audio_codec_dai = {
+ .name = "sirf-audio-codec",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .ops = &sirf_audio_codec_dai_ops,
+};
+
+static int sirf_audio_codec_probe(struct snd_soc_codec *codec)
+{
+ int ret;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ struct sirf_audio_codec *sirf_audio_codec = snd_soc_codec_get_drvdata(codec);
+
+ pm_runtime_enable(codec->dev);
+ codec->control_data = sirf_audio_codec->regmap;
+
+ ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ return ret;
+ }
+
+ if (of_device_is_compatible(codec->dev->of_node, "sirf,prima2-audio-codec")) {
+ snd_soc_dapm_new_controls(dapm,
+ prima2_output_driver_dapm_widgets,
+ ARRAY_SIZE(prima2_output_driver_dapm_widgets));
+ snd_soc_dapm_new_controls(dapm,
+ &prima2_codec_clock_dapm_widget, 1);
+ return snd_soc_add_codec_controls(codec,
+ volume_controls_prima2,
+ ARRAY_SIZE(volume_controls_prima2));
+ }
+ if (of_device_is_compatible(codec->dev->of_node, "sirf,atlas6-audio-codec")) {
+ snd_soc_dapm_new_controls(dapm,
+ atlas6_output_driver_dapm_widgets,
+ ARRAY_SIZE(atlas6_output_driver_dapm_widgets));
+ snd_soc_dapm_new_controls(dapm,
+ &atlas6_codec_clock_dapm_widget, 1);
+ return snd_soc_add_codec_controls(codec,
+ volume_controls_atlas6,
+ ARRAY_SIZE(volume_controls_atlas6));
+ }
+
+ return -EINVAL;
+}
+
+static int sirf_audio_codec_remove(struct snd_soc_codec *codec)
+{
+ pm_runtime_disable(codec->dev);
+ return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_device_sirf_audio_codec = {
+ .probe = sirf_audio_codec_probe,
+ .remove = sirf_audio_codec_remove,
+ .dapm_widgets = sirf_audio_codec_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(sirf_audio_codec_dapm_widgets),
+ .dapm_routes = sirf_audio_codec_map,
+ .num_dapm_routes = ARRAY_SIZE(sirf_audio_codec_map),
+ .idle_bias_off = true,
+};
+
+static const struct of_device_id sirf_audio_codec_of_match[] = {
+ { .compatible = "sirf,prima2-audio-codec" },
+ { .compatible = "sirf,atlas6-audio-codec" },
+ {}
+};
+MODULE_DEVICE_TABLE(of, sirf_audio_codec_of_match);
+
+static const struct regmap_config sirf_audio_codec_regmap_config = {
+ .reg_bits = 32,
+ .reg_stride = 4,
+ .val_bits = 32,
+ .max_register = AUDIO_IC_CODEC_CTRL3,
+ .cache_type = REGCACHE_NONE,
+};
+
+static int sirf_audio_codec_driver_probe(struct platform_device *pdev)
+{
+ int ret;
+ struct sirf_audio_codec *sirf_audio_codec;
+ void __iomem *base;
+ struct resource *mem_res;
+ const struct of_device_id *match;
+
+ match = of_match_node(sirf_audio_codec_of_match, pdev->dev.of_node);
+
+ sirf_audio_codec = devm_kzalloc(&pdev->dev,
+ sizeof(struct sirf_audio_codec), GFP_KERNEL);
+ if (!sirf_audio_codec)
+ return -ENOMEM;
+
+ platform_set_drvdata(pdev, sirf_audio_codec);
+
+ mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ base = devm_ioremap_resource(&pdev->dev, mem_res);
+ if (base == NULL)
+ return -ENOMEM;
+
+ sirf_audio_codec->regmap = devm_regmap_init_mmio(&pdev->dev, base,
+ &sirf_audio_codec_regmap_config);
+ if (IS_ERR(sirf_audio_codec->regmap))
+ return PTR_ERR(sirf_audio_codec->regmap);
+
+ sirf_audio_codec->clk = devm_clk_get(&pdev->dev, NULL);
+ if (IS_ERR(sirf_audio_codec->clk)) {
+ dev_err(&pdev->dev, "Get clock failed.\n");
+ return PTR_ERR(sirf_audio_codec->clk);
+ }
+
+ ret = clk_prepare_enable(sirf_audio_codec->clk);
+ if (ret) {
+ dev_err(&pdev->dev, "Enable clock failed.\n");
+ return ret;
+ }
+
+ ret = snd_soc_register_codec(&(pdev->dev),
+ &soc_codec_device_sirf_audio_codec,
+ &sirf_audio_codec_dai, 1);
+ if (ret) {
+ dev_err(&pdev->dev, "Register Audio Codec dai failed.\n");
+ goto err_clk_put;
+ }
+
+ /*
+ * Always open charge pump, if not, when the charge pump closed the
+ * adc will not stable
+ */
+ regmap_update_bits(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL0,
+ IC_CPFREQ, IC_CPFREQ);
+
+ if (of_device_is_compatible(pdev->dev.of_node, "sirf,atlas6-audio-codec"))
+ regmap_update_bits(sirf_audio_codec->regmap,
+ AUDIO_IC_CODEC_CTRL0, IC_CPEN, IC_CPEN);
+ return 0;
+
+err_clk_put:
+ clk_disable_unprepare(sirf_audio_codec->clk);
+ return ret;
+}
+
+static int sirf_audio_codec_driver_remove(struct platform_device *pdev)
+{
+ struct sirf_audio_codec *sirf_audio_codec = platform_get_drvdata(pdev);
+
+ clk_disable_unprepare(sirf_audio_codec->clk);
+ snd_soc_unregister_codec(&(pdev->dev));
+
+ return 0;
+}
+
+#ifdef CONFIG_PM_SLEEP
+static int sirf_audio_codec_suspend(struct device *dev)
+{
+ struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(dev);
+
+ regmap_read(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL0,
+ &sirf_audio_codec->reg_ctrl0);
+ regmap_read(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL1,
+ &sirf_audio_codec->reg_ctrl1);
+ clk_disable_unprepare(sirf_audio_codec->clk);
+
+ return 0;
+}
+
+static int sirf_audio_codec_resume(struct device *dev)
+{
+ struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(dev);
+ int ret;
+
+ ret = clk_prepare_enable(sirf_audio_codec->clk);
+ if (ret)
+ return ret;
+
+ regmap_write(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL0,
+ sirf_audio_codec->reg_ctrl0);
+ regmap_write(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL1,
+ sirf_audio_codec->reg_ctrl1);
+
+ return 0;
+}
+#endif
+
+static const struct dev_pm_ops sirf_audio_codec_pm_ops = {
+ SET_SYSTEM_SLEEP_PM_OPS(sirf_audio_codec_suspend, sirf_audio_codec_resume)
+};
+
+static struct platform_driver sirf_audio_codec_driver = {
+ .driver = {
+ .name = "sirf-audio-codec",
+ .owner = THIS_MODULE,
+ .of_match_table = sirf_audio_codec_of_match,
+ .pm = &sirf_audio_codec_pm_ops,
+ },
+ .probe = sirf_audio_codec_driver_probe,
+ .remove = sirf_audio_codec_driver_remove,
+};
+
+module_platform_driver(sirf_audio_codec_driver);
+
+MODULE_DESCRIPTION("SiRF audio codec driver");
+MODULE_AUTHOR("RongJun Ying <Rongjun.Ying at csr.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/sirf-audio-codec.h b/sound/soc/codecs/sirf-audio-codec.h
new file mode 100644
index 0000000..d4c187b
--- /dev/null
+++ b/sound/soc/codecs/sirf-audio-codec.h
@@ -0,0 +1,75 @@
+/*
+ * SiRF inner codec controllers define
+ *
+ * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company.
+ *
+ * Licensed under GPLv2 or later.
+ */
+
+#ifndef _SIRF_AUDIO_CODEC_H
+#define _SIRF_AUDIO_CODEC_H
+
+
+#define AUDIO_IC_CODEC_PWR (0x00E0)
+#define AUDIO_IC_CODEC_CTRL0 (0x00E4)
+#define AUDIO_IC_CODEC_CTRL1 (0x00E8)
+#define AUDIO_IC_CODEC_CTRL2 (0x00EC)
+#define AUDIO_IC_CODEC_CTRL3 (0x00F0)
+
+#define MICBIASEN (1 << 3)
+
+#define IC_RDACEN (1 << 0)
+#define IC_LDACEN (1 << 1)
+#define IC_HSREN (1 << 2)
+#define IC_HSLEN (1 << 3)
+#define IC_SPEN (1 << 4)
+#define IC_CPEN (1 << 5)
+
+#define IC_HPRSELR (1 << 6)
+#define IC_HPLSELR (1 << 7)
+#define IC_HPRSELL (1 << 8)
+#define IC_HPLSELL (1 << 9)
+#define IC_SPSELR (1 << 10)
+#define IC_SPSELL (1 << 11)
+
+#define IC_MONOR (1 << 12)
+#define IC_MONOL (1 << 13)
+
+#define IC_RXOSRSEL (1 << 28)
+#define IC_CPFREQ (1 << 29)
+#define IC_HSINVEN (1 << 30)
+
+#define IC_MICINREN (1 << 0)
+#define IC_MICINLEN (1 << 1)
+#define IC_MICIN1SEL (1 << 2)
+#define IC_MICIN2SEL (1 << 3)
+#define IC_MICDIFSEL (1 << 4)
+#define IC_LINEIN1SEL (1 << 5)
+#define IC_LINEIN2SEL (1 << 6)
+#define IC_RADCEN (1 << 7)
+#define IC_LADCEN (1 << 8)
+#define IC_ALM (1 << 9)
+
+#define IC_DIGMICEN (1 << 22)
+#define IC_DIGMICFREQ (1 << 23)
+#define IC_ADC14B_12 (1 << 24)
+#define IC_FIRDAC_HSL_EN (1 << 25)
+#define IC_FIRDAC_HSR_EN (1 << 26)
+#define IC_FIRDAC_LOUT_EN (1 << 27)
+#define IC_POR (1 << 28)
+#define IC_CODEC_CLK_EN (1 << 29)
+#define IC_HP_3DB_BOOST (1 << 30)
+
+#define IC_ADC_LEFT_GAIN_SHIFT 16
+#define IC_ADC_RIGHT_GAIN_SHIFT 10
+#define IC_ADC_GAIN_MASK 0x3F
+#define IC_MIC_MAX_GAIN 0x39
+
+#define IC_RXPGAR_MASK 0x3F
+#define IC_RXPGAR_SHIFT 14
+#define IC_RXPGAL_MASK 0x3F
+#define IC_RXPGAL_SHIFT 21
+#define IC_RXPGAR 0x7B
+#define IC_RXPGAL 0x7B
+
+#endif /*__SIRF_AUDIO_CODEC_H*/
--
1.7.5.4
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