提交 f516e368 编写于 作者: R Rongjun Ying 提交者: Mark Brown

ASoC: sirf: Add SiRF internal audio codec driver

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
Signed-off-by: NRongjun Ying <rongjun.ying@csr.com>
--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                           |    5 +
 sound/soc/codecs/Makefile                          |    1 +
 sound/soc/codecs/sirf-audio-codec.c                |  533 ++++++++++++++++++++
 sound/soc/codecs/sirf-audio-codec.h                |   75 +++
 5 files changed, 631 insertions(+), 0 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
Signed-off-by: NMark Brown <broonie@linaro.org>
上级 38dbfb59
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@b0040000 {
compatible = "sirf,atlas6-audio-codec";
reg = <0xb0040000 0x10000>;
clocks = <&clks 27>;
};
......@@ -63,6 +63,7 @@ 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
......@@ -330,6 +331,10 @@ 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
......
......@@ -53,6 +53,7 @@ snd-soc-alc5623-objs := alc5623.o
snd-soc-alc5632-objs := alc5632.o
snd-soc-sigmadsp-objs := sigmadsp.o
snd-soc-si476x-objs := si476x.o
snd-soc-sirf-audio-codec-objs := sirf-audio-codec.o
snd-soc-sn95031-objs := sn95031.o
snd-soc-spdif-tx-objs := spdif_transmitter.o
snd-soc-spdif-rx-objs := spdif_receiver.o
......
/*
* 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)
{
int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
struct snd_soc_codec *codec = dai->codec;
u32 val = 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:
if (playback)
val = IC_HSLEN | IC_HSREN;
break;
default:
return -EINVAL;
}
if (playback)
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@csr.com>");
MODULE_LICENSE("GPL v2");
/*
* 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*/
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