Required properties:
-- compatible : "maxim,max98090".
+- compatible : "maxim,max98090" or "maxim,max98091".
- reg : The I2C address of the device.
--- /dev/null
+/*
+ * platform_sst_audio.h: sst audio platform data header file
+ *
+ * Copyright (C) 2012-14 Intel Corporation
+ * Author: Jeeja KP <jeeja.kp@intel.com>
+ * Omair Mohammed Abdullah <omair.m.abdullah@intel.com>
+ * Vinod Koul ,vinod.koul@intel.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; version 2
+ * of the License.
+ */
+#ifndef _PLATFORM_SST_AUDIO_H_
+#define _PLATFORM_SST_AUDIO_H_
+
+#include <linux/sfi.h>
+
+enum sst_audio_task_id_mrfld {
+ SST_TASK_ID_NONE = 0,
+ SST_TASK_ID_SBA = 1,
+ SST_TASK_ID_MEDIA = 3,
+ SST_TASK_ID_MAX = SST_TASK_ID_MEDIA,
+};
+
+/* Device IDs for Merrifield are Pipe IDs,
+ * ref: DSP spec v0.75 */
+enum sst_audio_device_id_mrfld {
+ /* Output pipeline IDs */
+ PIPE_ID_OUT_START = 0x0,
+ PIPE_CODEC_OUT0 = 0x2,
+ PIPE_CODEC_OUT1 = 0x3,
+ PIPE_SPROT_LOOP_OUT = 0x4,
+ PIPE_MEDIA_LOOP1_OUT = 0x5,
+ PIPE_MEDIA_LOOP2_OUT = 0x6,
+ PIPE_VOIP_OUT = 0xC,
+ PIPE_PCM0_OUT = 0xD,
+ PIPE_PCM1_OUT = 0xE,
+ PIPE_PCM2_OUT = 0xF,
+ PIPE_MEDIA0_OUT = 0x12,
+ PIPE_MEDIA1_OUT = 0x13,
+/* Input Pipeline IDs */
+ PIPE_ID_IN_START = 0x80,
+ PIPE_CODEC_IN0 = 0x82,
+ PIPE_CODEC_IN1 = 0x83,
+ PIPE_SPROT_LOOP_IN = 0x84,
+ PIPE_MEDIA_LOOP1_IN = 0x85,
+ PIPE_MEDIA_LOOP2_IN = 0x86,
+ PIPE_VOIP_IN = 0x8C,
+ PIPE_PCM0_IN = 0x8D,
+ PIPE_PCM1_IN = 0x8E,
+ PIPE_MEDIA0_IN = 0x8F,
+ PIPE_MEDIA1_IN = 0x90,
+ PIPE_MEDIA2_IN = 0x91,
+ PIPE_RSVD = 0xFF,
+};
+
+/* The stream map for each platform consists of an array of the below
+ * stream map structure.
+ */
+struct sst_dev_stream_map {
+ u8 dev_num; /* device id */
+ u8 subdev_num; /* substream */
+ u8 direction;
+ u8 device_id; /* fw id */
+ u8 task_id; /* fw task */
+ u8 status;
+};
+
+struct sst_platform_data {
+ /* Intel software platform id*/
+ struct sst_dev_stream_map *pdev_strm_map;
+ unsigned int strm_map_size;
+};
+
+int add_sst_platform_device(void);
+#endif
+
--- /dev/null
+/*
+ * linux/sound/rt286.h -- Platform data for RT286
+ *
+ * Copyright 2013 Realtek Microelectronics
+ *
+ * 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 __LINUX_SND_RT286_H
+#define __LINUX_SND_RT286_H
+
+struct rt286_platform_data {
+ bool cbj_en; /*combo jack enable*/
+ bool gpio2_en; /*GPIO2 enable*/
+};
+
+#endif
select SND_SOC_PCM3008
select SND_SOC_PCM512x_I2C if I2C
select SND_SOC_PCM512x_SPI if SPI_MASTER
+ select SND_SOC_RT286 if I2C
select SND_SOC_RT5631 if I2C
select SND_SOC_RT5640 if I2C
select SND_SOC_RT5645 if I2C
default m if SND_SOC_RT5645=m
default m if SND_SOC_RT5651=m
+config SND_SOC_RT286
+ tristate
+
config SND_SOC_RT5631
tristate
snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o
snd-soc-pcm512x-spi-objs := pcm512x-spi.o
snd-soc-rl6231-objs := rl6231.o
+snd-soc-rt286-objs := rt286.o
snd-soc-rt5631-objs := rt5631.o
snd-soc-rt5640-objs := rt5640.o
snd-soc-rt5645-objs := rt5645.o
obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o
obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o
obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o
+obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o
obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o
obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o
obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o
#include <sound/max98090.h>
#include "max98090.h"
-#define DEBUG
-#define EXTMIC_METHOD
-#define EXTMIC_METHOD_TEST
-
/* Allows for sparsely populated register maps */
static struct reg_default max98090_reg[] = {
{ 0x00, 0x00 }, /* 00 Software Reset */
else
val = (val & M98090_MIC_PA2EN_MASK) >> M98090_MIC_PA2EN_SHIFT;
-
if (val >= 1) {
if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) {
max98090->pa1en = val - 1; /* Update for volatile */
SOC_DAPM_ENUM("MIXHPRSEL Mux", mixhprsel_mux_enum);
static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = {
-
SND_SOC_DAPM_INPUT("MIC1"),
SND_SOC_DAPM_INPUT("MIC2"),
SND_SOC_DAPM_INPUT("DMICL"),
};
static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = {
-
SND_SOC_DAPM_INPUT("DMIC3"),
SND_SOC_DAPM_INPUT("DMIC4"),
};
static const struct snd_soc_dapm_route max98090_dapm_routes[] = {
-
{"MIC1 Input", NULL, "MIC1"},
{"MIC2 Input", NULL, "MIC2"},
{"SPKR", NULL, "SPK Right Out"},
{"RCVL", NULL, "RCV Left Out"},
{"RCVR", NULL, "RCV Right Out"},
-
};
static const struct snd_soc_dapm_route max98091_dapm_routes[] = {
-
/* DMIC inputs */
{"DMIC3", NULL, "DMIC3_ENA"},
{"DMIC4", NULL, "DMIC4_ENA"},
{"DMIC3", NULL, "AHPF"},
{"DMIC4", NULL, "AHPF"},
-
};
static int max98090_add_widgets(struct snd_soc_codec *codec)
snd_soc_dapm_add_routes(dapm, max98091_dapm_routes,
ARRAY_SIZE(max98091_dapm_routes));
-
}
return 0;
}
};
-static void max98090_handle_pdata(struct snd_soc_codec *codec)
-{
- struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec);
- struct max98090_pdata *pdata = max98090->pdata;
-
- if (!pdata) {
- dev_err(codec->dev, "No platform data\n");
- return;
- }
-
-}
-
static int max98090_probe(struct snd_soc_codec *codec)
{
struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec);
struct max98090_cdata *cdata;
+ enum max98090_type devtype;
int ret = 0;
dev_dbg(codec->dev, "max98090_probe\n");
}
if ((ret >= M98090_REVA) && (ret <= M98090_REVA + 0x0f)) {
- max98090->devtype = MAX98090;
+ devtype = MAX98090;
dev_info(codec->dev, "MAX98090 REVID=0x%02x\n", ret);
} else if ((ret >= M98091_REVA) && (ret <= M98091_REVA + 0x0f)) {
- max98090->devtype = MAX98091;
+ devtype = MAX98091;
dev_info(codec->dev, "MAX98091 REVID=0x%02x\n", ret);
} else {
- max98090->devtype = MAX98090;
+ devtype = MAX98090;
dev_err(codec->dev, "Unrecognized revision 0x%02x\n", ret);
}
+ if (max98090->devtype != devtype) {
+ dev_warn(codec->dev, "Mismatch in DT specified CODEC type.\n");
+ max98090->devtype = devtype;
+ }
+
max98090->jack_state = M98090_JACK_STATE_NO_HEADSET;
INIT_DELAYED_WORK(&max98090->jack_work, max98090_jack_work);
snd_soc_update_bits(codec, M98090_REG_MIC_BIAS_VOLTAGE,
M98090_MBVSEL_MASK, M98090_MBVSEL_2V8);
- max98090_handle_pdata(codec);
-
max98090_add_widgets(codec);
err_access:
}
#endif
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
static int max98090_resume(struct device *dev)
{
struct max98090_priv *max98090 = dev_get_drvdata(dev);
static const struct i2c_device_id max98090_i2c_id[] = {
{ "max98090", MAX98090 },
+ { "max98091", MAX98091 },
{ }
};
MODULE_DEVICE_TABLE(i2c, max98090_i2c_id);
static const struct of_device_id max98090_of_match[] = {
{ .compatible = "maxim,max98090", },
+ { .compatible = "maxim,max98091", },
{ }
};
MODULE_DEVICE_TABLE(of, max98090_of_match);
ret = of_property_read_u32(np, "adc-port", &priv->adc_ssi_port);
if (ret)
- return ret;
+ goto out;
ret = of_property_read_u32(np, "dac-port", &priv->dac_ssi_port);
if (ret)
- return ret;
+ goto out;
}
dev_set_drvdata(&pdev->dev, priv);
ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783,
mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async));
+out:
+ of_node_put(np);
return ret;
}
--- /dev/null
+/*
+ * rt286.c -- RT286 ALSA SoC audio codec driver
+ *
+ * Copyright 2013 Realtek Semiconductor Corp.
+ * Author: Bard Liao <bardliao@realtek.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.
+ */
+
+#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/spi/spi.h>
+#include <linux/acpi.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 <sound/jack.h>
+#include <linux/workqueue.h>
+#include <sound/rt286.h>
+#include <sound/hda_verbs.h>
+
+#include "rt286.h"
+
+#define RT286_VENDOR_ID 0x10ec0286
+
+struct rt286_priv {
+ struct regmap *regmap;
+ struct snd_soc_codec *codec;
+ struct rt286_platform_data pdata;
+ struct i2c_client *i2c;
+ struct snd_soc_jack *jack;
+ struct delayed_work jack_detect_work;
+ int sys_clk;
+ struct reg_default *index_cache;
+};
+
+static struct reg_default rt286_index_def[] = {
+ { 0x01, 0xaaaa },
+ { 0x02, 0x8aaa },
+ { 0x03, 0x0002 },
+ { 0x04, 0xaf01 },
+ { 0x08, 0x000d },
+ { 0x09, 0xd810 },
+ { 0x0a, 0x0060 },
+ { 0x0b, 0x0000 },
+ { 0x0d, 0x2800 },
+ { 0x0f, 0x0000 },
+ { 0x19, 0x0a17 },
+ { 0x20, 0x0020 },
+ { 0x33, 0x0208 },
+ { 0x49, 0x0004 },
+ { 0x4f, 0x50e9 },
+ { 0x50, 0x2c00 },
+ { 0x63, 0x2902 },
+ { 0x67, 0x1111 },
+ { 0x68, 0x1016 },
+ { 0x69, 0x273f },
+};
+#define INDEX_CACHE_SIZE ARRAY_SIZE(rt286_index_def)
+
+static const struct reg_default rt286_reg[] = {
+ { 0x00170500, 0x00000400 },
+ { 0x00220000, 0x00000031 },
+ { 0x00239000, 0x0000007f },
+ { 0x0023a000, 0x0000007f },
+ { 0x00270500, 0x00000400 },
+ { 0x00370500, 0x00000400 },
+ { 0x00870500, 0x00000400 },
+ { 0x00920000, 0x00000031 },
+ { 0x00935000, 0x000000c3 },
+ { 0x00936000, 0x000000c3 },
+ { 0x00970500, 0x00000400 },
+ { 0x00b37000, 0x00000097 },
+ { 0x00b37200, 0x00000097 },
+ { 0x00b37300, 0x00000097 },
+ { 0x00c37000, 0x00000000 },
+ { 0x00c37100, 0x00000080 },
+ { 0x01270500, 0x00000400 },
+ { 0x01370500, 0x00000400 },
+ { 0x01371f00, 0x411111f0 },
+ { 0x01439000, 0x00000080 },
+ { 0x0143a000, 0x00000080 },
+ { 0x01470700, 0x00000000 },
+ { 0x01470500, 0x00000400 },
+ { 0x01470c00, 0x00000000 },
+ { 0x01470100, 0x00000000 },
+ { 0x01837000, 0x00000000 },
+ { 0x01870500, 0x00000400 },
+ { 0x02050000, 0x00000000 },
+ { 0x02139000, 0x00000080 },
+ { 0x0213a000, 0x00000080 },
+ { 0x02170100, 0x00000000 },
+ { 0x02170500, 0x00000400 },
+ { 0x02170700, 0x00000000 },
+ { 0x02270100, 0x00000000 },
+ { 0x02370100, 0x00000000 },
+ { 0x02040000, 0x00004002 },
+ { 0x01870700, 0x00000020 },
+ { 0x00830000, 0x000000c3 },
+ { 0x00930000, 0x000000c3 },
+ { 0x01270700, 0x00000000 },
+};
+
+static bool rt286_volatile_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case 0 ... 0xff:
+ case RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID):
+ case RT286_GET_HP_SENSE:
+ case RT286_GET_MIC1_SENSE:
+ case RT286_PROC_COEF:
+ return true;
+ default:
+ return false;
+ }
+
+
+}
+
+static bool rt286_readable_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case 0 ... 0xff:
+ case RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID):
+ case RT286_GET_HP_SENSE:
+ case RT286_GET_MIC1_SENSE:
+ case RT286_SET_AUDIO_POWER:
+ case RT286_SET_HPO_POWER:
+ case RT286_SET_SPK_POWER:
+ case RT286_SET_DMIC1_POWER:
+ case RT286_SPK_MUX:
+ case RT286_HPO_MUX:
+ case RT286_ADC0_MUX:
+ case RT286_ADC1_MUX:
+ case RT286_SET_MIC1:
+ case RT286_SET_PIN_HPO:
+ case RT286_SET_PIN_SPK:
+ case RT286_SET_PIN_DMIC1:
+ case RT286_SPK_EAPD:
+ case RT286_SET_AMP_GAIN_HPO:
+ case RT286_SET_DMIC2_DEFAULT:
+ case RT286_DACL_GAIN:
+ case RT286_DACR_GAIN:
+ case RT286_ADCL_GAIN:
+ case RT286_ADCR_GAIN:
+ case RT286_MIC_GAIN:
+ case RT286_SPOL_GAIN:
+ case RT286_SPOR_GAIN:
+ case RT286_HPOL_GAIN:
+ case RT286_HPOR_GAIN:
+ case RT286_F_DAC_SWITCH:
+ case RT286_F_RECMIX_SWITCH:
+ case RT286_REC_MIC_SWITCH:
+ case RT286_REC_I2S_SWITCH:
+ case RT286_REC_LINE_SWITCH:
+ case RT286_REC_BEEP_SWITCH:
+ case RT286_DAC_FORMAT:
+ case RT286_ADC_FORMAT:
+ case RT286_COEF_INDEX:
+ case RT286_PROC_COEF:
+ case RT286_SET_AMP_GAIN_ADC_IN1:
+ case RT286_SET_AMP_GAIN_ADC_IN2:
+ case RT286_SET_POWER(RT286_DAC_OUT1):
+ case RT286_SET_POWER(RT286_DAC_OUT2):
+ case RT286_SET_POWER(RT286_ADC_IN1):
+ case RT286_SET_POWER(RT286_ADC_IN2):
+ case RT286_SET_POWER(RT286_DMIC2):
+ case RT286_SET_POWER(RT286_MIC1):
+ return true;
+ default:
+ return false;
+ }
+}
+
+static int rt286_hw_write(void *context, unsigned int reg, unsigned int value)
+{
+ struct i2c_client *client = context;
+ struct rt286_priv *rt286 = i2c_get_clientdata(client);
+ u8 data[4];
+ int ret, i;
+
+ /*handle index registers*/
+ if (reg <= 0xff) {
+ rt286_hw_write(client, RT286_COEF_INDEX, reg);
+ reg = RT286_PROC_COEF;
+ for (i = 0; i < INDEX_CACHE_SIZE; i++) {
+ if (reg == rt286->index_cache[i].reg) {
+ rt286->index_cache[i].def = value;
+ break;
+ }
+
+ }
+ }
+
+ data[0] = (reg >> 24) & 0xff;
+ data[1] = (reg >> 16) & 0xff;
+ /*
+ * 4 bit VID: reg should be 0
+ * 12 bit VID: value should be 0
+ * So we use an OR operator to handle it rather than use if condition.
+ */
+ data[2] = ((reg >> 8) & 0xff) | ((value >> 8) & 0xff);
+ data[3] = value & 0xff;
+
+ ret = i2c_master_send(client, data, 4);
+
+ if (ret == 4)
+ return 0;
+ else
+ pr_err("ret=%d\n", ret);
+ if (ret < 0)
+ return ret;
+ else
+ return -EIO;
+}
+
+static int rt286_hw_read(void *context, unsigned int reg, unsigned int *value)
+{
+ struct i2c_client *client = context;
+ struct i2c_msg xfer[2];
+ int ret;
+ __be32 be_reg;
+ unsigned int index, vid, buf = 0x0;
+
+ /*handle index registers*/
+ if (reg <= 0xff) {
+ rt286_hw_write(client, RT286_COEF_INDEX, reg);
+ reg = RT286_PROC_COEF;
+ }
+
+ reg = reg | 0x80000;
+ vid = (reg >> 8) & 0xfff;
+
+ if (AC_VERB_GET_AMP_GAIN_MUTE == (vid & 0xf00)) {
+ index = (reg >> 8) & 0xf;
+ reg = (reg & ~0xf0f) | index;
+ }
+ be_reg = cpu_to_be32(reg);
+
+ /* Write register */
+ xfer[0].addr = client->addr;
+ xfer[0].flags = 0;
+ xfer[0].len = 4;
+ xfer[0].buf = (u8 *)&be_reg;
+
+ /* Read data */
+ xfer[1].addr = client->addr;
+ xfer[1].flags = I2C_M_RD;
+ xfer[1].len = 4;
+ xfer[1].buf = (u8 *)&buf;
+
+ ret = i2c_transfer(client->adapter, xfer, 2);
+ if (ret < 0)
+ return ret;
+ else if (ret != 2)
+ return -EIO;
+
+ *value = be32_to_cpu(buf);
+
+ return 0;
+}
+
+static void rt286_index_sync(struct snd_soc_codec *codec)
+{
+ struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+ int i;
+
+ for (i = 0; i < INDEX_CACHE_SIZE; i++) {
+ snd_soc_write(codec, rt286->index_cache[i].reg,
+ rt286->index_cache[i].def);
+ }
+}
+
+static int rt286_support_power_controls[] = {
+ RT286_DAC_OUT1,
+ RT286_DAC_OUT2,
+ RT286_ADC_IN1,
+ RT286_ADC_IN2,
+ RT286_MIC1,
+ RT286_DMIC1,
+ RT286_DMIC2,
+ RT286_SPK_OUT,
+ RT286_HP_OUT,
+};
+#define RT286_POWER_REG_LEN ARRAY_SIZE(rt286_support_power_controls)
+
+static int rt286_jack_detect(struct snd_soc_codec *codec, bool *hp, bool *mic)
+{
+ struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+ unsigned int val, buf;
+ int i;
+
+ *hp = false;
+ *mic = false;
+
+ if (rt286->pdata.cbj_en) {
+ buf = snd_soc_read(codec, RT286_GET_HP_SENSE);
+ *hp = buf & 0x80000000;
+ if (*hp) {
+ /* power on HV,VERF */
+ snd_soc_update_bits(codec,
+ RT286_POWER_CTRL1, 0x1001, 0x0);
+ /* power LDO1 */
+ snd_soc_update_bits(codec,
+ RT286_POWER_CTRL2, 0x4, 0x4);
+ snd_soc_write(codec, RT286_SET_MIC1, 0x24);
+ val = snd_soc_read(codec, RT286_CBJ_CTRL2);
+
+ msleep(200);
+ i = 40;
+ while (((val & 0x0800) == 0) && (i > 0)) {
+ val = snd_soc_read(codec,
+ RT286_CBJ_CTRL2);
+ i--;
+ msleep(20);
+ }
+
+ if (0x0400 == (val & 0x0700)) {
+ *mic = false;
+
+ snd_soc_write(codec,
+ RT286_SET_MIC1, 0x20);
+ /* power off HV,VERF */
+ snd_soc_update_bits(codec,
+ RT286_POWER_CTRL1, 0x1001, 0x1001);
+ snd_soc_update_bits(codec,
+ RT286_A_BIAS_CTRL3, 0xc000, 0x0000);
+ snd_soc_update_bits(codec,
+ RT286_CBJ_CTRL1, 0x0030, 0x0000);
+ snd_soc_update_bits(codec,
+ RT286_A_BIAS_CTRL2, 0xc000, 0x0000);
+ } else if ((0x0200 == (val & 0x0700)) ||
+ (0x0100 == (val & 0x0700))) {
+ *mic = true;
+ snd_soc_update_bits(codec,
+ RT286_A_BIAS_CTRL3, 0xc000, 0x8000);
+ snd_soc_update_bits(codec,
+ RT286_CBJ_CTRL1, 0x0030, 0x0020);
+ snd_soc_update_bits(codec,
+ RT286_A_BIAS_CTRL2, 0xc000, 0x8000);
+ } else {
+ *mic = false;
+ }
+
+ snd_soc_update_bits(codec,
+ RT286_MISC_CTRL1,
+ 0x0060, 0x0000);
+ } else {
+ snd_soc_update_bits(codec,
+ RT286_MISC_CTRL1,
+ 0x0060, 0x0020);
+ snd_soc_update_bits(codec,
+ RT286_A_BIAS_CTRL3,
+ 0xc000, 0x8000);
+ snd_soc_update_bits(codec,
+ RT286_CBJ_CTRL1,
+ 0x0030, 0x0020);
+ snd_soc_update_bits(codec,
+ RT286_A_BIAS_CTRL2,
+ 0xc000, 0x8000);
+
+ *mic = false;
+ }
+ } else {
+ buf = snd_soc_read(codec, RT286_GET_HP_SENSE);
+ *hp = buf & 0x80000000;
+ buf = snd_soc_read(codec, RT286_GET_MIC1_SENSE);
+ *mic = buf & 0x80000000;
+ }
+
+ return 0;
+}
+
+static void rt286_jack_detect_work(struct work_struct *work)
+{
+ struct rt286_priv *rt286 =
+ container_of(work, struct rt286_priv, jack_detect_work.work);
+ int status = 0;
+ bool hp = false;
+ bool mic = false;
+
+ rt286_jack_detect(rt286->codec, &hp, &mic);
+
+ if (hp == true)
+ status |= SND_JACK_HEADPHONE;
+
+ if (mic == true)
+ status |= SND_JACK_MICROPHONE;
+
+ snd_soc_jack_report(rt286->jack, status,
+ SND_JACK_MICROPHONE | SND_JACK_HEADPHONE);
+}
+
+int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack)
+{
+ struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+
+ rt286->jack = jack;
+
+ /* Send an initial empty report */
+ snd_soc_jack_report(rt286->jack, 0,
+ SND_JACK_MICROPHONE | SND_JACK_HEADPHONE);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(rt286_mic_detect);
+
+static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0);
+static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0);
+
+static const struct snd_kcontrol_new rt286_snd_controls[] = {
+ SOC_DOUBLE_R_TLV("DAC0 Playback Volume", RT286_DACL_GAIN,
+ RT286_DACR_GAIN, 0, 0x7f, 0, out_vol_tlv),
+ SOC_DOUBLE_R_TLV("ADC0 Capture Volume", RT286_ADCL_GAIN,
+ RT286_ADCR_GAIN, 0, 0x7f, 0, out_vol_tlv),
+ SOC_SINGLE_TLV("AMIC Volume", RT286_MIC_GAIN,
+ 0, 0x3, 0, mic_vol_tlv),
+ SOC_DOUBLE_R("Speaker Playback Switch", RT286_SPOL_GAIN,
+ RT286_SPOR_GAIN, RT286_MUTE_SFT, 1, 1),
+};
+
+/* Digital Mixer */
+static const struct snd_kcontrol_new rt286_front_mix[] = {
+ SOC_DAPM_SINGLE("DAC Switch", RT286_F_DAC_SWITCH,
+ RT286_MUTE_SFT, 1, 1),
+ SOC_DAPM_SINGLE("RECMIX Switch", RT286_F_RECMIX_SWITCH,
+ RT286_MUTE_SFT, 1, 1),
+};
+
+/* Analog Input Mixer */
+static const struct snd_kcontrol_new rt286_rec_mix[] = {
+ SOC_DAPM_SINGLE("Mic1 Switch", RT286_REC_MIC_SWITCH,
+ RT286_MUTE_SFT, 1, 1),
+ SOC_DAPM_SINGLE("I2S Switch", RT286_REC_I2S_SWITCH,
+ RT286_MUTE_SFT, 1, 1),
+ SOC_DAPM_SINGLE("Line1 Switch", RT286_REC_LINE_SWITCH,
+ RT286_MUTE_SFT, 1, 1),
+ SOC_DAPM_SINGLE("Beep Switch", RT286_REC_BEEP_SWITCH,
+ RT286_MUTE_SFT, 1, 1),
+};
+
+static const struct snd_kcontrol_new spo_enable_control =
+ SOC_DAPM_SINGLE("Switch", RT286_SET_PIN_SPK,
+ RT286_SET_PIN_SFT, 1, 0);
+
+static const struct snd_kcontrol_new hpol_enable_control =
+ SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT286_HPOL_GAIN,
+ RT286_MUTE_SFT, 1, 1);
+
+static const struct snd_kcontrol_new hpor_enable_control =
+ SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT286_HPOR_GAIN,
+ RT286_MUTE_SFT, 1, 1);
+
+/* ADC0 source */
+static const char * const rt286_adc_src[] = {
+ "Mic", "RECMIX", "Dmic"
+};
+
+static const int rt286_adc_values[] = {
+ 0, 4, 5,
+};
+
+static SOC_VALUE_ENUM_SINGLE_DECL(
+ rt286_adc0_enum, RT286_ADC0_MUX, RT286_ADC_SEL_SFT,
+ RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values);
+
+static const struct snd_kcontrol_new rt286_adc0_mux =
+ SOC_DAPM_ENUM("ADC 0 source", rt286_adc0_enum);
+
+static SOC_VALUE_ENUM_SINGLE_DECL(
+ rt286_adc1_enum, RT286_ADC1_MUX, RT286_ADC_SEL_SFT,
+ RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values);
+
+static const struct snd_kcontrol_new rt286_adc1_mux =
+ SOC_DAPM_ENUM("ADC 1 source", rt286_adc1_enum);
+
+static const char * const rt286_dac_src[] = {
+ "Front", "Surround"
+};
+/* HP-OUT source */
+static SOC_ENUM_SINGLE_DECL(rt286_hpo_enum, RT286_HPO_MUX,
+ 0, rt286_dac_src);
+
+static const struct snd_kcontrol_new rt286_hpo_mux =
+SOC_DAPM_ENUM("HPO source", rt286_hpo_enum);
+
+/* SPK-OUT source */
+static SOC_ENUM_SINGLE_DECL(rt286_spo_enum, RT286_SPK_MUX,
+ 0, rt286_dac_src);
+
+static const struct snd_kcontrol_new rt286_spo_mux =
+SOC_DAPM_ENUM("SPO source", rt286_spo_enum);
+
+static int rt286_spk_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ snd_soc_write(codec,
+ RT286_SPK_EAPD, RT286_SET_EAPD_HIGH);
+ break;
+ case SND_SOC_DAPM_PRE_PMD:
+ snd_soc_write(codec,
+ RT286_SPK_EAPD, RT286_SET_EAPD_LOW);
+ break;
+
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
+static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ snd_soc_write(codec, RT286_SET_PIN_DMIC1, 0x20);
+ break;
+ case SND_SOC_DAPM_PRE_PMD:
+ snd_soc_write(codec, RT286_SET_PIN_DMIC1, 0);
+ break;
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
+static int rt286_adc_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ unsigned int nid;
+
+ nid = (w->reg >> 20) & 0xff;
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ snd_soc_update_bits(codec,
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0),
+ 0x7080, 0x7000);
+ break;
+ case SND_SOC_DAPM_PRE_PMD:
+ snd_soc_update_bits(codec,
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0),
+ 0x7080, 0x7080);
+ break;
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = {
+ /* Input Lines */
+ SND_SOC_DAPM_INPUT("DMIC1 Pin"),
+ SND_SOC_DAPM_INPUT("DMIC2 Pin"),
+ SND_SOC_DAPM_INPUT("MIC1"),
+ SND_SOC_DAPM_INPUT("LINE1"),
+ SND_SOC_DAPM_INPUT("Beep"),
+
+ /* DMIC */
+ SND_SOC_DAPM_PGA_E("DMIC1", RT286_SET_POWER(RT286_DMIC1), 0, 1,
+ NULL, 0, rt286_set_dmic1_event,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PGA("DMIC2", RT286_SET_POWER(RT286_DMIC2), 0, 1,
+ NULL, 0),
+ SND_SOC_DAPM_SUPPLY("DMIC Receiver", SND_SOC_NOPM,
+ 0, 0, NULL, 0),
+
+ /* REC Mixer */
+ SND_SOC_DAPM_MIXER("RECMIX", SND_SOC_NOPM, 0, 0,
+ rt286_rec_mix, ARRAY_SIZE(rt286_rec_mix)),
+
+ /* ADCs */
+ SND_SOC_DAPM_ADC("ADC 0", NULL, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0),
+
+ /* ADC Mux */
+ SND_SOC_DAPM_MUX_E("ADC 0 Mux", RT286_SET_POWER(RT286_ADC_IN1), 0, 1,
+ &rt286_adc0_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD |
+ SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_MUX_E("ADC 1 Mux", RT286_SET_POWER(RT286_ADC_IN2), 0, 1,
+ &rt286_adc1_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD |
+ SND_SOC_DAPM_POST_PMU),
+
+ /* Audio Interface */
+ SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0),
+
+ /* Output Side */
+ /* DACs */
+ SND_SOC_DAPM_DAC("DAC 0", NULL, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("DAC 1", NULL, SND_SOC_NOPM, 0, 0),
+
+ /* Output Mux */
+ SND_SOC_DAPM_MUX("SPK Mux", SND_SOC_NOPM, 0, 0, &rt286_spo_mux),
+ SND_SOC_DAPM_MUX("HPO Mux", SND_SOC_NOPM, 0, 0, &rt286_hpo_mux),
+
+ SND_SOC_DAPM_SUPPLY("HP Power", RT286_SET_PIN_HPO,
+ RT286_SET_PIN_SFT, 0, NULL, 0),
+
+ /* Output Mixer */
+ SND_SOC_DAPM_MIXER("Front", RT286_SET_POWER(RT286_DAC_OUT1), 0, 1,
+ rt286_front_mix, ARRAY_SIZE(rt286_front_mix)),
+ SND_SOC_DAPM_PGA("Surround", RT286_SET_POWER(RT286_DAC_OUT2), 0, 1,
+ NULL, 0),
+
+ /* Output Pga */
+ SND_SOC_DAPM_SWITCH_E("SPO", SND_SOC_NOPM, 0, 0,
+ &spo_enable_control, rt286_spk_event,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_SWITCH("HPO L", SND_SOC_NOPM, 0, 0,
+ &hpol_enable_control),
+ SND_SOC_DAPM_SWITCH("HPO R", SND_SOC_NOPM, 0, 0,
+ &hpor_enable_control),
+
+ /* Output Lines */
+ SND_SOC_DAPM_OUTPUT("SPOL"),
+ SND_SOC_DAPM_OUTPUT("SPOR"),
+ SND_SOC_DAPM_OUTPUT("HPO Pin"),
+ SND_SOC_DAPM_OUTPUT("SPDIF"),
+};
+
+static const struct snd_soc_dapm_route rt286_dapm_routes[] = {
+ {"DMIC1", NULL, "DMIC1 Pin"},
+ {"DMIC2", NULL, "DMIC2 Pin"},
+ {"DMIC1", NULL, "DMIC Receiver"},
+ {"DMIC2", NULL, "DMIC Receiver"},
+
+ {"RECMIX", "Beep Switch", "Beep"},
+ {"RECMIX", "Line1 Switch", "LINE1"},
+ {"RECMIX", "Mic1 Switch", "MIC1"},
+
+ {"ADC 0 Mux", "Dmic", "DMIC1"},
+ {"ADC 0 Mux", "RECMIX", "RECMIX"},
+ {"ADC 0 Mux", "Mic", "MIC1"},
+ {"ADC 1 Mux", "Dmic", "DMIC2"},
+ {"ADC 1 Mux", "RECMIX", "RECMIX"},
+ {"ADC 1 Mux", "Mic", "MIC1"},
+
+ {"ADC 0", NULL, "ADC 0 Mux"},
+ {"ADC 1", NULL, "ADC 1 Mux"},
+
+ {"AIF1TX", NULL, "ADC 0"},
+ {"AIF2TX", NULL, "ADC 1"},
+
+ {"DAC 0", NULL, "AIF1RX"},
+ {"DAC 1", NULL, "AIF2RX"},
+
+ {"Front", "DAC Switch", "DAC 0"},
+ {"Front", "RECMIX Switch", "RECMIX"},
+
+ {"Surround", NULL, "DAC 1"},
+
+ {"SPK Mux", "Front", "Front"},
+ {"SPK Mux", "Surround", "Surround"},
+
+ {"HPO Mux", "Front", "Front"},
+ {"HPO Mux", "Surround", "Surround"},
+
+ {"SPO", "Switch", "SPK Mux"},
+ {"HPO L", "Switch", "HPO Mux"},
+ {"HPO R", "Switch", "HPO Mux"},
+ {"HPO L", NULL, "HP Power"},
+ {"HPO R", NULL, "HP Power"},
+
+ {"SPOL", NULL, "SPO"},
+ {"SPOR", NULL, "SPO"},
+ {"HPO Pin", NULL, "HPO L"},
+ {"HPO Pin", NULL, "HPO R"},
+};
+
+static int rt286_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+ unsigned int val = 0;
+ int d_len_code;
+
+ switch (params_rate(params)) {
+ /* bit 14 0:48K 1:44.1K */
+ case 44100:
+ val |= 0x4000;
+ break;
+ case 48000:
+ break;
+ default:
+ dev_err(codec->dev, "Unsupported sample rate %d\n",
+ params_rate(params));
+ return -EINVAL;
+ }
+ switch (rt286->sys_clk) {
+ case 12288000:
+ case 24576000:
+ if (params_rate(params) != 48000) {
+ dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n",
+ params_rate(params), rt286->sys_clk);
+ return -EINVAL;
+ }
+ break;
+ case 11289600:
+ case 22579200:
+ if (params_rate(params) != 44100) {
+ dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n",
+ params_rate(params), rt286->sys_clk);
+ return -EINVAL;
+ }
+ break;
+ }
+
+ if (params_channels(params) <= 16) {
+ /* bit 3:0 Number of Channel */
+ val |= (params_channels(params) - 1);
+ } else {
+ dev_err(codec->dev, "Unsupported channels %d\n",
+ params_channels(params));
+ return -EINVAL;
+ }
+
+ d_len_code = 0;
+ switch (params_width(params)) {
+ /* bit 6:4 Bits per Sample */
+ case 16:
+ d_len_code = 0;
+ val |= (0x1 << 4);
+ break;
+ case 32:
+ d_len_code = 2;
+ val |= (0x4 << 4);
+ break;
+ case 20:
+ d_len_code = 1;
+ val |= (0x2 << 4);
+ break;
+ case 24:
+ d_len_code = 2;
+ val |= (0x3 << 4);
+ break;
+ case 8:
+ d_len_code = 3;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL1, 0x0018, d_len_code << 3);
+ dev_dbg(codec->dev, "format val = 0x%x\n", val);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val);
+ else
+ snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val);
+
+ return 0;
+}
+
+static int rt286_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL1, 0x800, 0x800);
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL1, 0x800, 0x0);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL1, 0x300, 0x0);
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL1, 0x300, 0x1 << 8);
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL1, 0x300, 0x2 << 8);
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL1, 0x300, 0x3 << 8);
+ break;
+ default:
+ return -EINVAL;
+ }
+ /* bit 15 Stream Type 0:PCM 1:Non-PCM */
+ snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x8000, 0);
+ snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x8000, 0);
+
+ return 0;
+}
+
+static int rt286_set_dai_sysclk(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+
+ dev_dbg(codec->dev, "%s freq=%d\n", __func__, freq);
+
+ if (RT286_SCLK_S_MCLK == clk_id) {
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL2, 0x0100, 0x0);
+ snd_soc_update_bits(codec,
+ RT286_PLL_CTRL1, 0x20, 0x20);
+ } else {
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL2, 0x0100, 0x0100);
+ snd_soc_update_bits(codec,
+ RT286_PLL_CTRL, 0x4, 0x4);
+ snd_soc_update_bits(codec,
+ RT286_PLL_CTRL1, 0x20, 0x0);
+ }
+
+ switch (freq) {
+ case 19200000:
+ if (RT286_SCLK_S_MCLK == clk_id) {
+ dev_err(codec->dev, "Should not use MCLK\n");
+ return -EINVAL;
+ }
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL2, 0x40, 0x40);
+ break;
+ case 24000000:
+ if (RT286_SCLK_S_MCLK == clk_id) {
+ dev_err(codec->dev, "Should not use MCLK\n");
+ return -EINVAL;
+ }
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL2, 0x40, 0x0);
+ break;
+ case 12288000:
+ case 11289600:
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL2, 0x8, 0x0);
+ snd_soc_update_bits(codec,
+ RT286_CLK_DIV, 0xfc1e, 0x0004);
+ break;
+ case 24576000:
+ case 22579200:
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL2, 0x8, 0x8);
+ snd_soc_update_bits(codec,
+ RT286_CLK_DIV, 0xfc1e, 0x5406);
+ break;
+ default:
+ dev_err(codec->dev, "Unsupported system clock\n");
+ return -EINVAL;
+ }
+
+ rt286->sys_clk = freq;
+
+ return 0;
+}
+
+static int rt286_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio)
+{
+ struct snd_soc_codec *codec = dai->codec;
+
+ dev_dbg(codec->dev, "%s ratio=%d\n", __func__, ratio);
+ if (50 == ratio)
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL1, 0x1000, 0x1000);
+ else
+ snd_soc_update_bits(codec,
+ RT286_I2S_CTRL1, 0x1000, 0x0);
+
+
+ return 0;
+}
+
+static int rt286_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ switch (level) {
+ case SND_SOC_BIAS_PREPARE:
+ if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) {
+ snd_soc_write(codec,
+ RT286_SET_AUDIO_POWER, AC_PWRST_D0);
+ snd_soc_update_bits(codec,
+ RT286_DC_GAIN, 0x200, 0x200);
+ }
+ break;
+
+ case SND_SOC_BIAS_ON:
+ mdelay(10);
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ snd_soc_write(codec,
+ RT286_SET_AUDIO_POWER, AC_PWRST_D3);
+ snd_soc_update_bits(codec,
+ RT286_DC_GAIN, 0x200, 0x0);
+ break;
+
+ default:
+ break;
+ }
+ codec->dapm.bias_level = level;
+
+ return 0;
+}
+
+static irqreturn_t rt286_irq(int irq, void *data)
+{
+ struct rt286_priv *rt286 = data;
+ bool hp = false;
+ bool mic = false;
+ int status = 0;
+
+ rt286_jack_detect(rt286->codec, &hp, &mic);
+
+ /* Clear IRQ */
+ snd_soc_update_bits(rt286->codec,
+ RT286_IRQ_CTRL, 0x1, 0x1);
+
+ if (hp == true)
+ status |= SND_JACK_HEADPHONE;
+
+ if (mic == true)
+ status |= SND_JACK_MICROPHONE;
+
+ snd_soc_jack_report(rt286->jack, status,
+ SND_JACK_MICROPHONE | SND_JACK_HEADPHONE);
+
+ pm_wakeup_event(&rt286->i2c->dev, 300);
+
+ return IRQ_HANDLED;
+}
+
+static int rt286_probe(struct snd_soc_codec *codec)
+{
+ struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+
+ codec->dapm.bias_level = SND_SOC_BIAS_OFF;
+ rt286->codec = codec;
+
+ return 0;
+}
+
+static int rt286_remove(struct snd_soc_codec *codec)
+{
+ struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+
+ cancel_delayed_work_sync(&rt286->jack_detect_work);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int rt286_suspend(struct snd_soc_codec *codec)
+{
+ struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+
+ regcache_cache_only(rt286->regmap, true);
+ regcache_mark_dirty(rt286->regmap);
+
+ return 0;
+}
+
+static int rt286_resume(struct snd_soc_codec *codec)
+{
+ struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+
+ regcache_cache_only(rt286->regmap, false);
+ rt286_index_sync(codec);
+ regcache_sync(rt286->regmap);
+
+ return 0;
+}
+#else
+#define rt286_suspend NULL
+#define rt286_resume NULL
+#endif
+
+#define RT286_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+#define RT286_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8)
+
+static const struct snd_soc_dai_ops rt286_aif_dai_ops = {
+ .hw_params = rt286_hw_params,
+ .set_fmt = rt286_set_dai_fmt,
+ .set_sysclk = rt286_set_dai_sysclk,
+ .set_bclk_ratio = rt286_set_bclk_ratio,
+};
+
+static struct snd_soc_dai_driver rt286_dai[] = {
+ {
+ .name = "rt286-aif1",
+ .id = RT286_AIF1,
+ .playback = {
+ .stream_name = "AIF1 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = RT286_STEREO_RATES,
+ .formats = RT286_FORMATS,
+ },
+ .capture = {
+ .stream_name = "AIF1 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = RT286_STEREO_RATES,
+ .formats = RT286_FORMATS,
+ },
+ .ops = &rt286_aif_dai_ops,
+ .symmetric_rates = 1,
+ },
+ {
+ .name = "rt286-aif2",
+ .id = RT286_AIF2,
+ .playback = {
+ .stream_name = "AIF2 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = RT286_STEREO_RATES,
+ .formats = RT286_FORMATS,
+ },
+ .capture = {
+ .stream_name = "AIF2 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = RT286_STEREO_RATES,
+ .formats = RT286_FORMATS,
+ },
+ .ops = &rt286_aif_dai_ops,
+ .symmetric_rates = 1,
+ },
+
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_rt286 = {
+ .probe = rt286_probe,
+ .remove = rt286_remove,
+ .suspend = rt286_suspend,
+ .resume = rt286_resume,
+ .set_bias_level = rt286_set_bias_level,
+ .idle_bias_off = true,
+ .controls = rt286_snd_controls,
+ .num_controls = ARRAY_SIZE(rt286_snd_controls),
+ .dapm_widgets = rt286_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(rt286_dapm_widgets),
+ .dapm_routes = rt286_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(rt286_dapm_routes),
+};
+
+static const struct regmap_config rt286_regmap = {
+ .reg_bits = 32,
+ .val_bits = 32,
+ .max_register = 0x02370100,
+ .volatile_reg = rt286_volatile_register,
+ .readable_reg = rt286_readable_register,
+ .reg_write = rt286_hw_write,
+ .reg_read = rt286_hw_read,
+ .cache_type = REGCACHE_RBTREE,
+ .reg_defaults = rt286_reg,
+ .num_reg_defaults = ARRAY_SIZE(rt286_reg),
+};
+
+static const struct i2c_device_id rt286_i2c_id[] = {
+ {"rt286", 0},
+ {}
+};
+MODULE_DEVICE_TABLE(i2c, rt286_i2c_id);
+
+static const struct acpi_device_id rt286_acpi_match[] = {
+ { "INT343A", 0 },
+ {},
+};
+MODULE_DEVICE_TABLE(acpi, rt286_acpi_match);
+
+static int rt286_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev);
+ struct rt286_priv *rt286;
+ int i, ret;
+
+ rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286),
+ GFP_KERNEL);
+ if (NULL == rt286)
+ return -ENOMEM;
+
+ rt286->regmap = devm_regmap_init(&i2c->dev, NULL, i2c, &rt286_regmap);
+ if (IS_ERR(rt286->regmap)) {
+ ret = PTR_ERR(rt286->regmap);
+ dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
+ ret);
+ return ret;
+ }
+
+ regmap_read(rt286->regmap,
+ RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &ret);
+ if (ret != RT286_VENDOR_ID) {
+ dev_err(&i2c->dev,
+ "Device with ID register %x is not rt286\n", ret);
+ return -ENODEV;
+ }
+
+ rt286->index_cache = rt286_index_def;
+ rt286->i2c = i2c;
+ i2c_set_clientdata(i2c, rt286);
+
+ if (pdata)
+ rt286->pdata = *pdata;
+
+ regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3);
+
+ for (i = 0; i < RT286_POWER_REG_LEN; i++)
+ regmap_write(rt286->regmap,
+ RT286_SET_POWER(rt286_support_power_controls[i]),
+ AC_PWRST_D1);
+
+ if (!rt286->pdata.cbj_en) {
+ regmap_write(rt286->regmap, RT286_CBJ_CTRL2, 0x0000);
+ regmap_write(rt286->regmap, RT286_MIC1_DET_CTRL, 0x0816);
+ regmap_write(rt286->regmap, RT286_MISC_CTRL1, 0x0000);
+ regmap_update_bits(rt286->regmap,
+ RT286_CBJ_CTRL1, 0xf000, 0xb000);
+ } else {
+ regmap_update_bits(rt286->regmap,
+ RT286_CBJ_CTRL1, 0xf000, 0x5000);
+ }
+
+ mdelay(10);
+
+ if (!rt286->pdata.gpio2_en)
+ regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT, 0x4000);
+ else
+ regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT, 0);
+
+ mdelay(10);
+
+ /*Power down LDO2*/
+ regmap_update_bits(rt286->regmap, RT286_POWER_CTRL2, 0x8, 0x0);
+
+ /*Set depop parameter*/
+ regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL2, 0x403a, 0x401a);
+ regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737);
+ regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f);
+
+ if (rt286->i2c->irq) {
+ regmap_update_bits(rt286->regmap,
+ RT286_IRQ_CTRL, 0x2, 0x2);
+
+ INIT_DELAYED_WORK(&rt286->jack_detect_work,
+ rt286_jack_detect_work);
+ schedule_delayed_work(&rt286->jack_detect_work,
+ msecs_to_jiffies(1250));
+
+ ret = request_threaded_irq(rt286->i2c->irq, NULL, rt286_irq,
+ IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt286", rt286);
+ if (ret != 0) {
+ dev_err(&i2c->dev,
+ "Failed to reguest IRQ: %d\n", ret);
+ return ret;
+ }
+ }
+
+ ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt286,
+ rt286_dai, ARRAY_SIZE(rt286_dai));
+
+ return ret;
+}
+
+static int rt286_i2c_remove(struct i2c_client *i2c)
+{
+ struct rt286_priv *rt286 = i2c_get_clientdata(i2c);
+
+ if (i2c->irq)
+ free_irq(i2c->irq, rt286);
+ snd_soc_unregister_codec(&i2c->dev);
+
+ return 0;
+}
+
+
+static struct i2c_driver rt286_i2c_driver = {
+ .driver = {
+ .name = "rt286",
+ .owner = THIS_MODULE,
+ .acpi_match_table = ACPI_PTR(rt286_acpi_match),
+ },
+ .probe = rt286_i2c_probe,
+ .remove = rt286_i2c_remove,
+ .id_table = rt286_i2c_id,
+};
+
+module_i2c_driver(rt286_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC RT286 driver");
+MODULE_AUTHOR("Bard Liao <bardliao@realtek.com>");
+MODULE_LICENSE("GPL");
--- /dev/null
+/*
+ * rt286.h -- RT286 ALSA SoC audio driver
+ *
+ * Copyright 2011 Realtek Microelectronics
+ * Author: Johnny Hsu <johnnyhsu@realtek.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 __RT286_H__
+#define __RT286_H__
+
+#define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D)
+
+#define RT286_AUDIO_FUNCTION_GROUP 0x01
+#define RT286_DAC_OUT1 0x02
+#define RT286_DAC_OUT2 0x03
+#define RT286_ADC_IN1 0x09
+#define RT286_ADC_IN2 0x08
+#define RT286_MIXER_IN 0x0b
+#define RT286_MIXER_OUT1 0x0c
+#define RT286_MIXER_OUT2 0x0d
+#define RT286_DMIC1 0x12
+#define RT286_DMIC2 0x13
+#define RT286_SPK_OUT 0x14
+#define RT286_MIC1 0x18
+#define RT286_LINE1 0x1a
+#define RT286_BEEP 0x1d
+#define RT286_SPDIF 0x1e
+#define RT286_VENDOR_REGISTERS 0x20
+#define RT286_HP_OUT 0x21
+#define RT286_MIXER_IN1 0x22
+#define RT286_MIXER_IN2 0x23
+
+#define RT286_SET_PIN_SFT 6
+#define RT286_SET_PIN_ENABLE 0x40
+#define RT286_SET_PIN_DISABLE 0
+#define RT286_SET_EAPD_HIGH 0x2
+#define RT286_SET_EAPD_LOW 0
+
+#define RT286_MUTE_SFT 7
+
+/* Verb commands */
+#define RT286_GET_PARAM(NID, PARAM) VERB_CMD(AC_VERB_PARAMETERS, NID, PARAM)
+#define RT286_SET_POWER(NID) VERB_CMD(AC_VERB_SET_POWER_STATE, NID, 0)
+#define RT286_SET_AUDIO_POWER RT286_SET_POWER(RT286_AUDIO_FUNCTION_GROUP)
+#define RT286_SET_HPO_POWER RT286_SET_POWER(RT286_HP_OUT)
+#define RT286_SET_SPK_POWER RT286_SET_POWER(RT286_SPK_OUT)
+#define RT286_SET_DMIC1_POWER RT286_SET_POWER(RT286_DMIC1)
+#define RT286_SPK_MUX\
+ VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_SPK_OUT, 0)
+#define RT286_HPO_MUX\
+ VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_HP_OUT, 0)
+#define RT286_ADC0_MUX\
+ VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_MIXER_IN1, 0)
+#define RT286_ADC1_MUX\
+ VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_MIXER_IN2, 0)
+#define RT286_SET_MIC1\
+ VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_MIC1, 0)
+#define RT286_SET_PIN_HPO\
+ VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_HP_OUT, 0)
+#define RT286_SET_PIN_SPK\
+ VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_SPK_OUT, 0)
+#define RT286_SET_PIN_DMIC1\
+ VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_DMIC1, 0)
+#define RT286_SPK_EAPD\
+ VERB_CMD(AC_VERB_SET_EAPD_BTLENABLE, RT286_SPK_OUT, 0)
+#define RT286_SET_AMP_GAIN_HPO\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0)
+#define RT286_SET_AMP_GAIN_ADC_IN1\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0)
+#define RT286_SET_AMP_GAIN_ADC_IN2\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN2, 0)
+#define RT286_GET_HP_SENSE\
+ VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_HP_OUT, 0)
+#define RT286_GET_MIC1_SENSE\
+ VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_MIC1, 0)
+#define RT286_SET_DMIC2_DEFAULT\
+ VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT286_DMIC2, 0)
+#define RT286_DACL_GAIN\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_DAC_OUT1, 0xa000)
+#define RT286_DACR_GAIN\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_DAC_OUT1, 0x9000)
+#define RT286_ADCL_GAIN\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0x6000)
+#define RT286_ADCR_GAIN\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0x5000)
+#define RT286_MIC_GAIN\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIC1, 0x7000)
+#define RT286_SPOL_GAIN\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_SPK_OUT, 0xa000)
+#define RT286_SPOR_GAIN\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_SPK_OUT, 0x9000)
+#define RT286_HPOL_GAIN\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0xa000)
+#define RT286_HPOR_GAIN\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0x9000)
+#define RT286_F_DAC_SWITCH\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_OUT1, 0x7000)
+#define RT286_F_RECMIX_SWITCH\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_OUT1, 0x7100)
+#define RT286_REC_MIC_SWITCH\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7000)
+#define RT286_REC_I2S_SWITCH\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7100)
+#define RT286_REC_LINE_SWITCH\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7200)
+#define RT286_REC_BEEP_SWITCH\
+ VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7300)
+#define RT286_DAC_FORMAT\
+ VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT286_DAC_OUT1, 0)
+#define RT286_ADC_FORMAT\
+ VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT286_ADC_IN1, 0)
+#define RT286_COEF_INDEX\
+ VERB_CMD(AC_VERB_SET_COEF_INDEX, RT286_VENDOR_REGISTERS, 0)
+#define RT286_PROC_COEF\
+ VERB_CMD(AC_VERB_SET_PROC_COEF, RT286_VENDOR_REGISTERS, 0)
+
+/* Index registers */
+#define RT286_A_BIAS_CTRL1 0x01
+#define RT286_A_BIAS_CTRL2 0x02
+#define RT286_POWER_CTRL1 0x03
+#define RT286_A_BIAS_CTRL3 0x04
+#define RT286_POWER_CTRL2 0x08
+#define RT286_I2S_CTRL1 0x09
+#define RT286_I2S_CTRL2 0x0a
+#define RT286_CLK_DIV 0x0b
+#define RT286_DC_GAIN 0x0d
+#define RT286_POWER_CTRL3 0x0f
+#define RT286_MIC1_DET_CTRL 0x19
+#define RT286_MISC_CTRL1 0x20
+#define RT286_IRQ_CTRL 0x33
+#define RT286_PLL_CTRL1 0x49
+#define RT286_CBJ_CTRL1 0x4f
+#define RT286_CBJ_CTRL2 0x50
+#define RT286_PLL_CTRL 0x63
+#define RT286_DEPOP_CTRL1 0x66
+#define RT286_DEPOP_CTRL2 0x67
+#define RT286_DEPOP_CTRL3 0x68
+#define RT286_DEPOP_CTRL4 0x69
+
+/* SPDIF (0x06) */
+#define RT286_SPDIF_SEL_SFT 0
+#define RT286_SPDIF_SEL_PCM0 0
+#define RT286_SPDIF_SEL_PCM1 1
+#define RT286_SPDIF_SEL_SPOUT 2
+#define RT286_SPDIF_SEL_PP 3
+
+/* RECMIX (0x0b) */
+#define RT286_M_REC_BEEP_SFT 0
+#define RT286_M_REC_LINE1_SFT 1
+#define RT286_M_REC_MIC1_SFT 2
+#define RT286_M_REC_I2S_SFT 3
+
+/* Front (0x0c) */
+#define RT286_M_FRONT_DAC_SFT 0
+#define RT286_M_FRONT_REC_SFT 1
+
+/* SPK-OUT (0x14) */
+#define RT286_M_SPK_MUX_SFT 14
+#define RT286_SPK_SEL_MASK 0x1
+#define RT286_SPK_SEL_SFT 0
+#define RT286_SPK_SEL_F 0
+#define RT286_SPK_SEL_S 1
+
+/* HP-OUT (0x21) */
+#define RT286_M_HP_MUX_SFT 14
+#define RT286_HP_SEL_MASK 0x1
+#define RT286_HP_SEL_SFT 0
+#define RT286_HP_SEL_F 0
+#define RT286_HP_SEL_S 1
+
+/* ADC (0x22) (0x23) */
+#define RT286_ADC_SEL_MASK 0x7
+#define RT286_ADC_SEL_SFT 0
+#define RT286_ADC_SEL_SURR 0
+#define RT286_ADC_SEL_FRONT 1
+#define RT286_ADC_SEL_DMIC 2
+#define RT286_ADC_SEL_BEEP 4
+#define RT286_ADC_SEL_LINE1 5
+#define RT286_ADC_SEL_I2S 6
+#define RT286_ADC_SEL_MIC1 7
+
+#define RT286_SCLK_S_MCLK 0
+#define RT286_SCLK_S_PLL 1
+
+enum {
+ RT286_AIF1,
+ RT286_AIF2,
+ RT286_AIFS,
+};
+
+int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack);
+
+#endif /* __RT286_H__ */
+
help
This adds audio driver for Intel Baytrail platform based boards
with the MAX98090 audio codec.
+
+config SND_SOC_INTEL_BROADWELL_MACH
+ tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint"
+ depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC
+ select SND_SOC_INTEL_HASWELL
+ select SND_COMPRESS_OFFLOAD
+ select SND_SOC_RT286
+ help
+ This adds support for the Wilcatpoint Audio DSP on Intel(R) Broadwell
+ Ultrabook platforms.
+ Say Y if you have such a device
+ If unsure select "N".
snd-soc-sst-haswell-objs := haswell.o
snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o
snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o
+snd-soc-sst-broadwell-objs := broadwell.o
obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o
obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o
obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o
+obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o
--- /dev/null
+/*
+ * Intel Broadwell Wildcatpoint SST Audio
+ *
+ * Copyright (C) 2013, Intel Corporation. All rights reserved.
+ *
+ * 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.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+
+#include "sst-dsp.h"
+#include "sst-haswell-ipc.h"
+
+#include "../codecs/rt286.h"
+
+static const struct snd_soc_dapm_widget broadwell_widgets[] = {
+ SND_SOC_DAPM_HP("Headphones", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+ SND_SOC_DAPM_MIC("DMIC1", NULL),
+ SND_SOC_DAPM_MIC("DMIC2", NULL),
+ SND_SOC_DAPM_LINE("Line Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route broadwell_rt286_map[] = {
+
+ /* speaker */
+ {"Speaker", NULL, "SPOR"},
+ {"Speaker", NULL, "SPOL"},
+
+ /* HP jack connectors - unknown if we have jack deteck */
+ {"Headphones", NULL, "HPO Pin"},
+
+ /* other jacks */
+ {"MIC1", NULL, "Mic Jack"},
+ {"LINE1", NULL, "Line Jack"},
+
+ /* digital mics */
+ {"DMIC1 Pin", NULL, "DMIC1"},
+ {"DMIC2 Pin", NULL, "DMIC2"},
+
+ /* CODEC BE connections */
+ {"SSP0 CODEC IN", NULL, "AIF1 Capture"},
+ {"AIF1 Playback", NULL, "SSP0 CODEC OUT"},
+};
+
+static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ /* The ADSP will covert the FE rate to 48k, stereo */
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = 2;
+
+ /* set SSP0 to 16 bit */
+ snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT -
+ SNDRV_PCM_HW_PARAM_FIRST_MASK],
+ SNDRV_PCM_FORMAT_S16_LE);
+ return 0;
+}
+
+static int broadwell_rt286_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000,
+ SND_SOC_CLOCK_IN);
+
+ if (ret < 0) {
+ dev_err(rtd->dev, "can't set codec sysclk configuration\n");
+ return ret;
+ }
+
+ return ret;
+}
+
+static struct snd_soc_ops broadwell_rt286_ops = {
+ .hw_params = broadwell_rt286_hw_params,
+};
+
+static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev);
+ struct sst_hsw *broadwell = pdata->dsp;
+ int ret;
+
+ /* Set ADSP SSP port settings */
+ ret = sst_hsw_device_set_config(broadwell, SST_HSW_DEVICE_SSP_0,
+ SST_HSW_DEVICE_MCLK_FREQ_24_MHZ,
+ SST_HSW_DEVICE_CLOCK_MASTER, 9);
+ if (ret < 0) {
+ dev_err(rtd->dev, "error: failed to set device config\n");
+ return ret;
+ }
+
+ /* always connected - check HP for jack detect */
+ snd_soc_dapm_enable_pin(dapm, "Headphones");
+ snd_soc_dapm_enable_pin(dapm, "Speaker");
+ snd_soc_dapm_enable_pin(dapm, "Mic Jack");
+ snd_soc_dapm_enable_pin(dapm, "Line Jack");
+ snd_soc_dapm_enable_pin(dapm, "DMIC1");
+ snd_soc_dapm_enable_pin(dapm, "DMIC2");
+
+ return 0;
+}
+
+/* broadwell digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link broadwell_rt286_dais[] = {
+ /* Front End DAI links */
+ {
+ .name = "System PCM",
+ .stream_name = "System Playback",
+ .cpu_dai_name = "System Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .init = broadwell_rtd_init,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ },
+ {
+ .name = "Offload0",
+ .stream_name = "Offload0 Playback",
+ .cpu_dai_name = "Offload0 Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ },
+ {
+ .name = "Offload1",
+ .stream_name = "Offload1 Playback",
+ .cpu_dai_name = "Offload1 Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ },
+ {
+ .name = "Loopback PCM",
+ .stream_name = "Loopback",
+ .cpu_dai_name = "Loopback Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 0,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_capture = 1,
+ },
+ {
+ .name = "Capture PCM",
+ .stream_name = "Capture",
+ .cpu_dai_name = "Capture Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_capture = 1,
+ },
+
+ /* Back End DAI links */
+ {
+ /* SSP0 - Codec */
+ .name = "Codec",
+ .be_id = 0,
+ .cpu_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "snd-soc-dummy",
+ .no_pcm = 1,
+ .codec_name = "i2c-INT343A:00",
+ .codec_dai_name = "rt286-aif1",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1,
+ .be_hw_params_fixup = broadwell_ssp0_fixup,
+ .ops = &broadwell_rt286_ops,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ },
+};
+
+/* broadwell audio machine driver for WPT + RT286S */
+static struct snd_soc_card broadwell_rt286 = {
+ .name = "broadwell-rt286",
+ .owner = THIS_MODULE,
+ .dai_link = broadwell_rt286_dais,
+ .num_links = ARRAY_SIZE(broadwell_rt286_dais),
+ .dapm_widgets = broadwell_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(broadwell_widgets),
+ .dapm_routes = broadwell_rt286_map,
+ .num_dapm_routes = ARRAY_SIZE(broadwell_rt286_map),
+ .fully_routed = true,
+};
+
+static int broadwell_audio_probe(struct platform_device *pdev)
+{
+ broadwell_rt286.dev = &pdev->dev;
+
+ return snd_soc_register_card(&broadwell_rt286);
+}
+
+static int broadwell_audio_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_card(&broadwell_rt286);
+ return 0;
+}
+
+static struct platform_driver broadwell_audio = {
+ .probe = broadwell_audio_probe,
+ .remove = broadwell_audio_remove,
+ .driver = {
+ .name = "broadwell-audio",
+ .owner = THIS_MODULE,
+ },
+};
+
+module_platform_driver(broadwell_audio)
+
+/* Module information */
+MODULE_AUTHOR("Liam Girdwood, Xingchao Wang");
+MODULE_DESCRIPTION("Intel SST Audio for WPT/Broadwell");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:broadwell-audio");
.pin = "Headset Mic",
.mask = SND_JACK_MICROPHONE,
},
- {
- .pin = "Ext Spk",
- .mask = SND_JACK_LINEOUT,
- },
- {
- .pin = "Int Mic",
- .mask = SND_JACK_LINEIN,
- },
};
static struct snd_soc_jack_gpio hs_jack_gpios[] = {
};
static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = {
+ {"Headset Mic", NULL, "MICBIAS1"},
{"IN2P", NULL, "Headset Mic"},
{"IN2N", NULL, "Headset Mic"},
{"DMIC1", NULL, "Internal Mic"},
--- /dev/null
+/*
+ * Copyright (C) 2013-14 Intel Corp
+ * Author: Ramesh Babu <ramesh.babu.koul@intel.com>
+ * Omair M Abdullah <omair.m.abdullah@intel.com>
+ * Samreen Nilofer <samreen.nilofer@intel.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; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ */
+
+#ifndef __SST_CONTROLS_V2_H__
+#define __SST_CONTROLS_V2_H__
+
+enum {
+ MERR_DPCM_AUDIO = 0,
+ MERR_DPCM_COMPR,
+};
+
+
+#endif
u32 channel_peak[8];
} __packed;
+struct sst_byt_fw_version {
+ u8 build;
+ u8 minor;
+ u8 major;
+ u8 type;
+} __packed;
+
+struct sst_byt_fw_build_info {
+ u8 date[16];
+ u8 time[16];
+} __packed;
+
+struct sst_byt_fw_init {
+ struct sst_byt_fw_version fw_version;
+ struct sst_byt_fw_build_info build_info;
+ u16 result;
+ u8 module_id;
+ u8 debug_info;
+} __packed;
+
/* driver internal IPC message structure */
struct ipc_message {
struct list_head list;
{
struct sst_byt *byt;
struct sst_fw *byt_sst_fw;
+ struct sst_byt_fw_init init;
int err;
dev_dbg(dev, "initialising Byt DSP IPC\n");
goto boot_err;
}
+ /* show firmware information */
+ sst_dsp_inbox_read(byt->dsp, &init, sizeof(init));
+ dev_info(byt->dev, "FW version: %02x.%02x.%02x.%02x\n",
+ init.fw_version.major, init.fw_version.minor,
+ init.fw_version.build, init.fw_version.type);
+ dev_info(byt->dev, "Build type: %x\n", init.fw_version.type);
+ dev_info(byt->dev, "Build date: %s %s\n",
+ init.build_info.date, init.build_info.time);
+
pdata->dsp = byt;
byt->fw = byt_sst_fw;
void sst_dsp_dump(struct sst_dsp *sst)
{
- sst->ops->dump(sst);
+ if (sst->ops->dump)
+ sst->ops->dump(sst);
}
EXPORT_SYMBOL_GPL(sst_dsp_dump);
void sst_dsp_reset(struct sst_dsp *sst)
{
- sst->ops->reset(sst);
+ if (sst->ops->reset)
+ sst->ops->reset(sst);
}
EXPORT_SYMBOL_GPL(sst_dsp_reset);
int sst_dsp_boot(struct sst_dsp *sst)
{
- sst->ops->boot(sst);
+ if (sst->ops->boot)
+ sst->ops->boot(sst);
+
return 0;
}
EXPORT_SYMBOL_GPL(sst_dsp_boot);
#define SST_CLKCTL 0x78
#define SST_CSR2 0x80
#define SST_LTRC 0xE0
-#define SST_HDMC 0xE8
+#define SST_HMDC 0xE8
+
+#define SST_SHIM_BEGIN SST_CSR
+#define SST_SHIM_END SST_HDMC
+
#define SST_DBGO 0xF0
#define SST_SHIM_SIZE 0x100
#define SST_CSR_S0IOCS (0x1 << 21)
#define SST_CSR_S1IOCS (0x1 << 23)
#define SST_CSR_LPCS (0x1 << 31)
+#define SST_CSR_24MHZ_LPCS (SST_CSR_SBCS0 | SST_CSR_SBCS1 | SST_CSR_LPCS)
+#define SST_CSR_24MHZ_NO_LPCS (SST_CSR_SBCS0 | SST_CSR_SBCS1)
#define SST_BYT_CSR_RST (0x1 << 0)
#define SST_BYT_CSR_VECTOR_SEL (0x1 << 1)
#define SST_BYT_CSR_STALL (0x1 << 2)
#define SST_IMRX_DONE (0x1 << 0)
#define SST_BYT_IMRX_REQUEST (0x1 << 1)
+/* IMRD / IMD */
+#define SST_IMRD_DONE (0x1 << 0)
+#define SST_IMRD_BUSY (0x1 << 1)
+#define SST_IMRD_SSP0 (0x1 << 16)
+#define SST_IMRD_DMAC0 (0x1 << 21)
+#define SST_IMRD_DMAC1 (0x1 << 22)
+#define SST_IMRD_DMAC (SST_IMRD_DMAC0 | SST_IMRD_DMAC1)
+
/* IPCX / IPCC */
#define SST_IPCX_DONE (0x1 << 30)
#define SST_IPCX_BUSY (0x1 << 31)
/* LTRC */
#define SST_LTRC_VAL(x) (x << 0)
-/* HDMC */
-#define SST_HDMC_HDDA0(x) (x << 0)
-#define SST_HDMC_HDDA1(x) (x << 7)
+/* HMDC */
+#define SST_HMDC_HDDA0(x) (x << 0)
+#define SST_HMDC_HDDA1(x) (x << 7)
+#define SST_HMDC_HDDA_E0_CH0 1
+#define SST_HMDC_HDDA_E0_CH1 2
+#define SST_HMDC_HDDA_E0_CH2 4
+#define SST_HMDC_HDDA_E0_CH3 8
+#define SST_HMDC_HDDA_E1_CH0 SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH0)
+#define SST_HMDC_HDDA_E1_CH1 SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH1)
+#define SST_HMDC_HDDA_E1_CH2 SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH2)
+#define SST_HMDC_HDDA_E1_CH3 SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH3)
+#define SST_HMDC_HDDA_E0_ALLCH (SST_HMDC_HDDA_E0_CH0 | SST_HMDC_HDDA_E0_CH1 | \
+ SST_HMDC_HDDA_E0_CH2 | SST_HMDC_HDDA_E0_CH3)
+#define SST_HMDC_HDDA_E1_ALLCH (SST_HMDC_HDDA_E1_CH0 | SST_HMDC_HDDA_E1_CH1 | \
+ SST_HMDC_HDDA_E1_CH2 | SST_HMDC_HDDA_E1_CH3)
/* SST Vendor Defined Registers and bits */
#define SST_VDRTCTL3 0xaC
/* VDRTCTL0 */
+#define SST_VDRTCL0_APLLSE_MASK 1
#define SST_VDRTCL0_DSRAMPGE_SHIFT 16
#define SST_VDRTCL0_DSRAMPGE_MASK (0xffff << SST_VDRTCL0_DSRAMPGE_SHIFT)
#define SST_VDRTCL0_ISRAMPGE_SHIFT 6
#define SST_VDRTCL0_ISRAMPGE_MASK (0x3ff << SST_VDRTCL0_ISRAMPGE_SHIFT)
+/* PMCS */
+#define SST_PMCS 0x84
+#define SST_PMCS_PS_MASK 0x3
+
struct sst_dsp;
/*
#include <linux/firmware.h>
#include <linux/pm_runtime.h>
-#include <linux/acpi.h>
-#include <acpi/acpi_bus.h>
-
#include "sst-dsp.h"
#include "sst-dsp-priv.h"
#include "sst-haswell-ipc.h"
SST_CSR2_SDFD_SSP1);
/* enable DMA engine 0,1 all channels to access host memory */
- sst_dsp_shim_update_bits_unlocked(sst, SST_HDMC,
- SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff),
- SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff));
+ sst_dsp_shim_update_bits_unlocked(sst, SST_HMDC,
+ SST_HMDC_HDDA1(0xff) | SST_HMDC_HDDA0(0xff),
+ SST_HMDC_HDDA1(0xff) | SST_HMDC_HDDA0(0xff));
/* disable all clock gating */
writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL2);
/* wild cat point ADSP mem regions */
static const struct sst_adsp_memregion wpt_region[] = {
- {0x00000, 0x40000, 8, SST_MEM_DRAM}, /* D-SRAM0 - 8 * 32kB */
- {0x40000, 0x80000, 8, SST_MEM_DRAM}, /* D-SRAM1 - 8 * 32kB */
- {0x80000, 0xA0000, 4, SST_MEM_DRAM}, /* D-SRAM2 - 4 * 32kB */
+ {0x00000, 0xA0000, 20, SST_MEM_DRAM}, /* D-SRAM0,D-SRAM1,D-SRAM2 - 20 * 32kB */
{0xA0000, 0xF0000, 10, SST_MEM_IRAM}, /* I-SRAM - 10 * 32kB */
};
return 0;
}
+struct sst_sram_shift {
+ u32 dev_id; /* SST Device IDs */
+ u32 iram_shift;
+ u32 dram_shift;
+};
+
+static const struct sst_sram_shift sram_shift[] = {
+ {SST_DEV_ID_LYNX_POINT, 6, 16}, /* lp */
+ {SST_DEV_ID_WILDCAT_POINT, 2, 12}, /* wpt */
+};
static u32 hsw_block_get_bit(struct sst_mem_block *block)
{
- u32 bit = 0, shift = 0;
+ u32 bit = 0, shift = 0, index;
+ struct sst_dsp *sst = block->dsp;
- switch (block->type) {
- case SST_MEM_DRAM:
- shift = 16;
- break;
- case SST_MEM_IRAM:
- shift = 6;
- break;
- default:
- return 0;
+ for (index = 0; index < ARRAY_SIZE(sram_shift); index++) {
+ if (sram_shift[index].dev_id == sst->id)
+ break;
}
+ if (index < ARRAY_SIZE(sram_shift)) {
+ switch (block->type) {
+ case SST_MEM_DRAM:
+ shift = sram_shift[index].dram_shift;
+ break;
+ case SST_MEM_IRAM:
+ shift = sram_shift[index].iram_shift;
+ break;
+ default:
+ shift = 0;
+ }
+ } else
+ shift = 0;
+
bit = 1 << (block->index + shift);
return bit;
}
}
- /* set default power gating mask */
- writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL0);
+ /* set default power gating control, enable power gating control for all blocks. that is,
+ can't be accessed, please enable each block before accessing. */
+ writel(0xffffffff, sst->addr.pci_cfg + SST_VDRTCTL0);
return 0;
}
u32 inbox_size;
u32 outbox_size;
u32 fw_info_size;
- u8 fw_info[1];
+ u8 fw_info[IPC_MAX_MAILBOX_BYTES - 5 * sizeof(u32)];
} __attribute__((packed));
struct ipc_message {
return;
}
- /* if the DSP is busy we will TX messages after IRQ */
+ /* if the DSP is busy, we will TX messages after IRQ.
+ * also postpone if we are in the middle of procesing completion irq*/
ipcx = sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCX);
- if (ipcx & SST_IPCX_BUSY) {
+ if (ipcx & (SST_IPCX_BUSY | SST_IPCX_DONE)) {
spin_unlock_irqrestore(&hsw->dsp->spinlock, flags);
return;
}
ipc_shim_dbg(hsw, "message timeout");
trace_ipc_error("error message timeout for", msg->header);
+ list_del(&msg->list);
ret = -ETIMEDOUT;
} else {
{
struct sst_hsw_ipc_fw_ready fw_ready;
u32 offset;
+ u8 fw_info[IPC_MAX_MAILBOX_BYTES - 5 * sizeof(u32)];
+ char *tmp[5], *pinfo;
+ int i = 0;
offset = (header & 0x1FFFFFFF) << 3;
fw_ready.inbox_offset, fw_ready.inbox_size);
dev_dbg(hsw->dev, " mailbox downstream 0x%x - size 0x%x\n",
fw_ready.outbox_offset, fw_ready.outbox_size);
+ if (fw_ready.fw_info_size < sizeof(fw_ready.fw_info)) {
+ fw_ready.fw_info[fw_ready.fw_info_size] = 0;
+ dev_dbg(hsw->dev, " Firmware info: %s \n", fw_ready.fw_info);
+
+ /* log the FW version info got from the mailbox here. */
+ memcpy(fw_info, fw_ready.fw_info, fw_ready.fw_info_size);
+ pinfo = &fw_info[0];
+ for (i = 0; i < sizeof(tmp) / sizeof(char *); i++)
+ tmp[i] = strsep(&pinfo, " ");
+ dev_info(hsw->dev, "FW loaded, mailbox readback FW info: type %s, - "
+ "version: %s.%s, build %s, source commit id: %s\n",
+ tmp[0], tmp[1], tmp[2], tmp[3], tmp[4]);
+ }
}
static void hsw_notification_work(struct work_struct *work)
switch (stream_msg) {
case IPC_STR_STAGE_MESSAGE:
case IPC_STR_NOTIFICATION:
+ break;
case IPC_STR_RESET:
+ trace_ipc_notification("stream reset", stream->reply.stream_hw_id);
break;
case IPC_STR_PAUSE:
stream->running = false;
}
/* update any stream states */
- hsw_stream_update(hsw, msg);
+ if (msg_get_global_type(header) == IPC_GLB_STREAM_MESSAGE)
+ hsw_stream_update(hsw, msg);
/* wake up and return the error if we have waiters on this message ? */
list_del(&msg->list);
enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx)
{
u32 header, state_;
- int ret;
+ int ret, item;
header = IPC_GLB_TYPE(IPC_GLB_ENTER_DX_STATE);
state_ = state;
return ret;
}
+ for (item = 0; item < dx->entries_no; item++) {
+ dev_dbg(hsw->dev,
+ "Item[%d] offset[%x] - size[%x] - source[%x]\n",
+ item, dx->mem_info[item].offset,
+ dx->mem_info[item].size,
+ dx->mem_info[item].source);
+ }
dev_dbg(hsw->dev, "ipc: got %d entry numbers for state %d\n",
dx->entries_no, state);
/* get the FW version */
sst_hsw_fw_get_version(hsw, &version);
- dev_info(hsw->dev, "FW loaded: type %d - version: %d.%d build %d\n",
- version.type, version.major, version.minor, version.build);
/* get the globalmixer */
ret = sst_hsw_mixer_get_info(hsw);
/*
* sst_mfld_dsp.h - Intel SST Driver for audio engine
*
- * Copyright (C) 2008-12 Intel Corporation
+ * Copyright (C) 2008-14 Intel Corporation
* Authors: Vinod Koul <vinod.koul@linux.intel.com>
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*/
+#define SST_MAX_BIN_BYTES 1024
+
+#define MAX_DBG_RW_BYTES 80
+#define MAX_NUM_SCATTER_BUFFERS 8
+#define MAX_LOOP_BACK_DWORDS 8
+/* IPC base address and mailbox, timestamp offsets */
+#define SST_MAILBOX_SIZE 0x0400
+#define SST_MAILBOX_SEND 0x0000
+#define SST_TIME_STAMP 0x1800
+#define SST_TIME_STAMP_MRFLD 0x800
+#define SST_RESERVED_OFFSET 0x1A00
+#define SST_SCU_LPE_MAILBOX 0x1000
+#define SST_LPE_SCU_MAILBOX 0x1400
+#define SST_SCU_LPE_LOG_BUF (SST_SCU_LPE_MAILBOX+16)
+#define PROCESS_MSG 0x80
+
+/* Message ID's for IPC messages */
+/* Bits B7: SST or IA/SC ; B6-B4: Msg Category; B3-B0: Msg Type */
+
+/* I2L Firmware/Codec Download msgs */
+#define IPC_IA_PREP_LIB_DNLD 0x01
+#define IPC_IA_LIB_DNLD_CMPLT 0x02
+#define IPC_IA_GET_FW_VERSION 0x04
+#define IPC_IA_GET_FW_BUILD_INF 0x05
+#define IPC_IA_GET_FW_INFO 0x06
+#define IPC_IA_GET_FW_CTXT 0x07
+#define IPC_IA_SET_FW_CTXT 0x08
+#define IPC_IA_PREPARE_SHUTDOWN 0x31
+/* I2L Codec Config/control msgs */
+#define IPC_PREP_D3 0x10
+#define IPC_IA_SET_CODEC_PARAMS 0x10
+#define IPC_IA_GET_CODEC_PARAMS 0x11
+#define IPC_IA_SET_PPP_PARAMS 0x12
+#define IPC_IA_GET_PPP_PARAMS 0x13
+#define IPC_SST_PERIOD_ELAPSED_MRFLD 0xA
+#define IPC_IA_ALG_PARAMS 0x1A
+#define IPC_IA_TUNING_PARAMS 0x1B
+#define IPC_IA_SET_RUNTIME_PARAMS 0x1C
+#define IPC_IA_SET_PARAMS 0x1
+#define IPC_IA_GET_PARAMS 0x2
+
+#define IPC_EFFECTS_CREATE 0xE
+#define IPC_EFFECTS_DESTROY 0xF
+
+/* I2L Stream config/control msgs */
+#define IPC_IA_ALLOC_STREAM_MRFLD 0x2
+#define IPC_IA_ALLOC_STREAM 0x20 /* Allocate a stream ID */
+#define IPC_IA_FREE_STREAM_MRFLD 0x03
+#define IPC_IA_FREE_STREAM 0x21 /* Free the stream ID */
+#define IPC_IA_SET_STREAM_PARAMS 0x22
+#define IPC_IA_SET_STREAM_PARAMS_MRFLD 0x12
+#define IPC_IA_GET_STREAM_PARAMS 0x23
+#define IPC_IA_PAUSE_STREAM 0x24
+#define IPC_IA_PAUSE_STREAM_MRFLD 0x4
+#define IPC_IA_RESUME_STREAM 0x25
+#define IPC_IA_RESUME_STREAM_MRFLD 0x5
+#define IPC_IA_DROP_STREAM 0x26
+#define IPC_IA_DROP_STREAM_MRFLD 0x07
+#define IPC_IA_DRAIN_STREAM 0x27 /* Short msg with str_id */
+#define IPC_IA_DRAIN_STREAM_MRFLD 0x8
+#define IPC_IA_CONTROL_ROUTING 0x29
+#define IPC_IA_VTSV_UPDATE_MODULES 0x20
+#define IPC_IA_VTSV_DETECTED 0x21
+
+#define IPC_IA_START_STREAM_MRFLD 0X06
+#define IPC_IA_START_STREAM 0x30 /* Short msg with str_id */
+
+#define IPC_IA_SET_GAIN_MRFLD 0x21
+/* Debug msgs */
+#define IPC_IA_DBG_MEM_READ 0x40
+#define IPC_IA_DBG_MEM_WRITE 0x41
+#define IPC_IA_DBG_LOOP_BACK 0x42
+#define IPC_IA_DBG_LOG_ENABLE 0x45
+#define IPC_IA_DBG_SET_PROBE_PARAMS 0x47
+
+/* L2I Firmware/Codec Download msgs */
+#define IPC_IA_FW_INIT_CMPLT 0x81
+#define IPC_IA_FW_INIT_CMPLT_MRFLD 0x01
+#define IPC_IA_FW_ASYNC_ERR_MRFLD 0x11
+
+/* L2I Codec Config/control msgs */
+#define IPC_SST_FRAGMENT_ELPASED 0x90 /* Request IA more data */
+
+#define IPC_SST_BUF_UNDER_RUN 0x92 /* PB Under run and stopped */
+#define IPC_SST_BUF_OVER_RUN 0x93 /* CAP Under run and stopped */
+#define IPC_SST_DRAIN_END 0x94 /* PB Drain complete and stopped */
+#define IPC_SST_CHNGE_SSP_PARAMS 0x95 /* PB SSP parameters changed */
+#define IPC_SST_STREAM_PROCESS_FATAL_ERR 0x96/* error in processing a stream */
+#define IPC_SST_PERIOD_ELAPSED 0x97 /* period elapsed */
+
+#define IPC_SST_ERROR_EVENT 0x99 /* Buffer over run occurred */
+/* L2S messages */
+#define IPC_SC_DDR_LINK_UP 0xC0
+#define IPC_SC_DDR_LINK_DOWN 0xC1
+#define IPC_SC_SET_LPECLK_REQ 0xC2
+#define IPC_SC_SSP_BIT_BANG 0xC3
+
+/* L2I Error reporting msgs */
+#define IPC_IA_MEM_ALLOC_FAIL 0xE0
+#define IPC_IA_PROC_ERR 0xE1 /* error in processing a
+ stream can be used by playback and
+ capture modules */
+
+/* L2I Debug msgs */
+#define IPC_IA_PRINT_STRING 0xF0
+
+/* Buffer under-run */
+#define IPC_IA_BUF_UNDER_RUN_MRFLD 0x0B
+
+/* Mrfld specific defines:
+ * For asynchronous messages(INIT_CMPLT, PERIOD_ELAPSED, ASYNC_ERROR)
+ * received from FW, the format is:
+ * - IPC High: pvt_id is set to zero. Always short message.
+ * - msg_id is in lower 16-bits of IPC low payload.
+ * - pipe_id is in higher 16-bits of IPC low payload for period_elapsed.
+ * - error id is in higher 16-bits of IPC low payload for async errors.
+ */
+#define SST_ASYNC_DRV_ID 0
+
+/* Command Response or Acknowledge message to any IPC message will have
+ * same message ID and stream ID information which is sent.
+ * There is no specific Ack message ID. The data field is used as response
+ * meaning.
+ */
+enum ackData {
+ IPC_ACK_SUCCESS = 0,
+ IPC_ACK_FAILURE,
+};
+
+enum ipc_ia_msg_id {
+ IPC_CMD = 1, /*!< Task Control message ID */
+ IPC_SET_PARAMS = 2,/*!< Task Set param message ID */
+ IPC_GET_PARAMS = 3, /*!< Task Get param message ID */
+ IPC_INVALID = 0xFF, /*!<Task Get param message ID */
+};
+
enum sst_codec_types {
/* AUDIO/MUSIC CODEC Type Definitions */
SST_CODEC_TYPE_UNKNOWN = 0,
SST_STREAM_TYPE_MUSIC = 1,
};
+enum sst_error_codes {
+ /* Error code,response to msgId: Description */
+ /* Common error codes */
+ SST_SUCCESS = 0, /* Success */
+ SST_ERR_INVALID_STREAM_ID = 1,
+ SST_ERR_INVALID_MSG_ID = 2,
+ SST_ERR_INVALID_STREAM_OP = 3,
+ SST_ERR_INVALID_PARAMS = 4,
+ SST_ERR_INVALID_CODEC = 5,
+ SST_ERR_INVALID_MEDIA_TYPE = 6,
+ SST_ERR_STREAM_ERR = 7,
+
+ SST_ERR_STREAM_IN_USE = 15,
+};
+
+struct ipc_dsp_hdr {
+ u16 mod_index_id:8; /*!< DSP Command ID specific to tasks */
+ u16 pipe_id:8; /*!< instance of the module in the pipeline */
+ u16 mod_id; /*!< Pipe_id */
+ u16 cmd_id; /*!< Module ID = lpe_algo_types_t */
+ u16 length; /*!< Length of the payload only */
+} __packed;
+
+union ipc_header_high {
+ struct {
+ u32 msg_id:8; /* Message ID - Max 256 Message Types */
+ u32 task_id:4; /* Task ID associated with this comand */
+ u32 drv_id:4; /* Identifier for the driver to track*/
+ u32 rsvd1:8; /* Reserved */
+ u32 result:4; /* Reserved */
+ u32 res_rqd:1; /* Response rqd */
+ u32 large:1; /* Large Message if large = 1 */
+ u32 done:1; /* bit 30 - Done bit */
+ u32 busy:1; /* bit 31 - busy bit*/
+ } part;
+ u32 full;
+} __packed;
+/* IPC header */
+union ipc_header_mrfld {
+ struct {
+ u32 header_low_payload;
+ union ipc_header_high header_high;
+ } p;
+ u64 full;
+} __packed;
+/* CAUTION NOTE: All IPC message body must be multiple of 32 bits.*/
+
+/* IPC Header */
+union ipc_header {
+ struct {
+ u32 msg_id:8; /* Message ID - Max 256 Message Types */
+ u32 str_id:5;
+ u32 large:1; /* Large Message if large = 1 */
+ u32 reserved:2; /* Reserved for future use */
+ u32 data:14; /* Ack/Info for msg, size of msg in Mailbox */
+ u32 done:1; /* bit 30 */
+ u32 busy:1; /* bit 31 */
+ } part;
+ u32 full;
+} __packed;
+
+/* Firmware build info */
+struct sst_fw_build_info {
+ unsigned char date[16]; /* Firmware build date */
+ unsigned char time[16]; /* Firmware build time */
+} __packed;
+
+/* Firmware Version info */
+struct snd_sst_fw_version {
+ u8 build; /* build number*/
+ u8 minor; /* minor number*/
+ u8 major; /* major number*/
+ u8 type; /* build type */
+};
+
+struct ipc_header_fw_init {
+ struct snd_sst_fw_version fw_version;/* Firmware version details */
+ struct sst_fw_build_info build_info;
+ u16 result; /* Fw init result */
+ u8 module_id; /* Module ID in case of error */
+ u8 debug_info; /* Debug info from Module ID in case of fail */
+} __packed;
+
+struct snd_sst_tstamp {
+ u64 ring_buffer_counter; /* PB/CP: Bytes copied from/to DDR. */
+ u64 hardware_counter; /* PB/CP: Bytes DMAed to/from SSP. */
+ u64 frames_decoded;
+ u64 bytes_decoded;
+ u64 bytes_copied;
+ u32 sampling_frequency;
+ u32 channel_peak[8];
+} __packed;
+
+/* Stream type params struture for Alloc stream */
+struct snd_sst_str_type {
+ u8 codec_type; /* Codec type */
+ u8 str_type; /* 1 = voice 2 = music */
+ u8 operation; /* Playback or Capture */
+ u8 protected_str; /* 0=Non DRM, 1=DRM */
+ u8 time_slots;
+ u8 reserved; /* Reserved */
+ u16 result; /* Result used for acknowledgment */
+} __packed;
+
+/* Library info structure */
+struct module_info {
+ u32 lib_version;
+ u32 lib_type;/*TBD- KLOCKWORK u8 lib_type;*/
+ u32 media_type;
+ u8 lib_name[12];
+ u32 lib_caps;
+ unsigned char b_date[16]; /* Lib build date */
+ unsigned char b_time[16]; /* Lib build time */
+} __packed;
+
+/* Library slot info */
+struct lib_slot_info {
+ u8 slot_num; /* 1 or 2 */
+ u8 reserved1;
+ u16 reserved2;
+ u32 iram_size; /* slot size in IRAM */
+ u32 dram_size; /* slot size in DRAM */
+ u32 iram_offset; /* starting offset of slot in IRAM */
+ u32 dram_offset; /* starting offset of slot in DRAM */
+} __packed;
+
+struct snd_ppp_mixer_params {
+ __u32 type; /*Type of the parameter */
+ __u32 size;
+ __u32 input_stream_bitmap; /*Input stream Bit Map*/
+} __packed;
+
+struct snd_sst_lib_download {
+ struct module_info lib_info; /* library info type, capabilities etc */
+ struct lib_slot_info slot_info; /* slot info to be downloaded */
+ u32 mod_entry_pt;
+};
+
+struct snd_sst_lib_download_info {
+ struct snd_sst_lib_download dload_lib;
+ u16 result; /* Result used for acknowledgment */
+ u8 pvt_id; /* Private ID */
+ u8 reserved; /* for alignment */
+};
struct snd_pcm_params {
u8 num_chan; /* 1=Mono, 2=Stereo */
u8 pcm_wd_sz; /* 16/24 - bit*/
- u32 reserved; /* Bitrate in bits per second */
- u32 sfreq; /* Sampling rate in Hz */
- u8 use_offload_path;
+ u8 use_offload_path; /* 0-PCM using period elpased & ALSA interfaces
+ 1-PCM stream via compressed interface */
u8 reserved2;
- u16 reserved3;
+ u32 sfreq; /* Sampling rate in Hz */
u8 channel_map[8];
} __packed;
struct snd_wma_params {
u8 num_chan; /* 1=Mono, 2=Stereo */
u8 pcm_wd_sz; /* 16/24 - bit*/
+ u16 reserved1;
u32 brate; /* Use the hard coded value. */
u32 sfreq; /* Sampling freq eg. 8000, 441000, 48000 */
u32 channel_mask; /* Channel Mask */
};
struct snd_sst_alloc_params_ext {
- struct sst_address_info ring_buf_info[8];
- u8 sg_count;
- u8 reserved;
- u16 reserved2;
- u32 frag_size; /*Number of samples after which period elapsed
+ __u16 sg_count;
+ __u16 reserved;
+ __u32 frag_size; /*Number of samples after which period elapsed
message is sent valid only if path = 0*/
-} __packed;
+ struct sst_address_info ring_buf_info[8];
+};
struct snd_sst_stream_params {
union snd_sst_codec_params uc;
} __packed;
struct snd_sst_params {
+ u32 result;
u32 stream_id;
u8 codec;
u8 ops;
u8 stream_type;
u8 device_type;
+ u8 task;
struct snd_sst_stream_params sparams;
struct snd_sst_alloc_params_ext aparams;
};
+struct snd_sst_alloc_mrfld {
+ u16 codec_type;
+ u8 operation;
+ u8 sg_count;
+ struct sst_address_info ring_buf_info[8];
+ u32 frag_size;
+ u32 ts;
+ struct snd_sst_stream_params codec_params;
+} __packed;
+
+/* Alloc stream params structure */
+struct snd_sst_alloc_params {
+ struct snd_sst_str_type str_type;
+ struct snd_sst_stream_params stream_params;
+ struct snd_sst_alloc_params_ext alloc_params;
+} __packed;
+
+/* Alloc stream response message */
+struct snd_sst_alloc_response {
+ struct snd_sst_str_type str_type; /* Stream type for allocation */
+ struct snd_sst_lib_download lib_dnld; /* Valid only for codec dnld */
+};
+
+/* Drop response */
+struct snd_sst_drop_response {
+ u32 result;
+ u32 bytes;
+};
+
+struct snd_sst_async_msg {
+ u32 msg_id; /* Async msg id */
+ u32 payload[0];
+};
+
+struct snd_sst_async_err_msg {
+ u32 fw_resp; /* Firmware Result */
+ u32 lib_resp; /*Library result */
+} __packed;
+
+struct snd_sst_vol {
+ u32 stream_id;
+ s32 volume;
+ u32 ramp_duration;
+ u32 ramp_type; /* Ramp type, default=0 */
+};
+
+/* Gain library parameters for mrfld
+ * based on DSP command spec v0.82
+ */
+struct snd_sst_gain_v2 {
+ u16 gain_cell_num; /* num of gain cells to modify*/
+ u8 cell_nbr_idx; /* instance index*/
+ u8 cell_path_idx; /* pipe-id */
+ u16 module_id; /*module id */
+ u16 left_cell_gain; /* left gain value in dB*/
+ u16 right_cell_gain; /* right gain value in dB*/
+ u16 gain_time_const; /* gain time constant*/
+} __packed;
+
+struct snd_sst_mute {
+ u32 stream_id;
+ u32 mute;
+};
+
+struct snd_sst_runtime_params {
+ u8 type;
+ u8 str_id;
+ u8 size;
+ u8 rsvd;
+ void *addr;
+} __packed;
+
+enum stream_param_type {
+ SST_SET_TIME_SLOT = 0,
+ SST_SET_CHANNEL_INFO = 1,
+ OTHERS = 2, /*reserved for future params*/
+};
+
+/* CSV Voice call routing structure */
+struct snd_sst_control_routing {
+ u8 control; /* 0=start, 1=Stop */
+ u8 reserved[3]; /* Reserved- for 32 bit alignment */
+};
+
+struct ipc_post {
+ struct list_head node;
+ union ipc_header header; /* driver specific */
+ bool is_large;
+ bool is_process_reply;
+ union ipc_header_mrfld mrfld_header;
+ char *mailbox_data;
+};
+
+struct snd_sst_ctxt_params {
+ u32 address; /* Physical Address in DDR where the context is stored */
+ u32 size; /* size of the context */
+};
+
+struct snd_sst_lpe_log_params {
+ u8 dbg_type;
+ u8 module_id;
+ u8 log_level;
+ u8 reserved;
+} __packed;
+
+enum snd_sst_bytes_type {
+ SND_SST_BYTES_SET = 0x1,
+ SND_SST_BYTES_GET = 0x2,
+};
+
+struct snd_sst_bytes_v2 {
+ u8 type;
+ u8 ipc_msg;
+ u8 block;
+ u8 task_id;
+ u8 pipe_id;
+ u8 rsvd;
+ u16 len;
+ char bytes[0];
+};
+
+#define MAX_VTSV_FILES 2
+struct snd_sst_vtsv_info {
+ struct sst_address_info vfiles[MAX_VTSV_FILES];
+} __packed;
+
#endif /* __SST_MFLD_DSP_H__ */
int retval;
struct snd_sst_params str_params;
struct sst_compress_cb cb;
+ struct snd_soc_pcm_runtime *rtd = cstream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct sst_data *ctx = snd_soc_platform_get_drvdata(platform);
stream = cstream->runtime->private_data;
/* construct fw structure for this*/
memset(&str_params, 0, sizeof(str_params));
- str_params.ops = STREAM_OPS_PLAYBACK;
- str_params.stream_type = SST_STREAM_TYPE_MUSIC;
- str_params.device_type = SND_SST_DEVICE_COMPRESS;
+ /* fill the device type and stream id to pass to SST driver */
+ retval = sst_fill_stream_params(cstream, ctx, &str_params, true);
+ pr_debug("compr_set_params: fill stream params ret_val = 0x%x\n", retval);
+ if (retval < 0)
+ return retval;
switch (params->codec.id) {
case SND_AUDIOCODEC_MP3: {
/*
* sst_mfld_platform.c - Intel MID Platform driver
*
- * Copyright (C) 2010-2013 Intel Corp
+ * Copyright (C) 2010-2014 Intel Corp
* Author: Vinod Koul <vinod.koul@intel.com>
* Author: Harsha Priya <priya.harsha@intel.com>
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/compress_driver.h>
+#include <asm/platform_sst_audio.h>
#include "sst-mfld-platform.h"
+#include "sst-atom-controls.h"
struct sst_device *sst;
static DEFINE_MUTEX(sst_lock);
.fifo_size = SST_FIFO_SIZE,
};
+static struct sst_dev_stream_map dpcm_strm_map[] = {
+ {0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF}, /* Reserved, not in use */
+ {MERR_DPCM_AUDIO, 0, SNDRV_PCM_STREAM_PLAYBACK, PIPE_MEDIA1_IN, SST_TASK_ID_MEDIA, 0},
+ {MERR_DPCM_COMPR, 0, SNDRV_PCM_STREAM_PLAYBACK, PIPE_MEDIA0_IN, SST_TASK_ID_MEDIA, 0},
+ {MERR_DPCM_AUDIO, 0, SNDRV_PCM_STREAM_CAPTURE, PIPE_PCM1_OUT, SST_TASK_ID_MEDIA, 0},
+};
+
/* MFLD - MSIC */
static struct snd_soc_dai_driver sst_platform_dai[] = {
{
return state;
}
+static void sst_fill_alloc_params(struct snd_pcm_substream *substream,
+ struct snd_sst_alloc_params_ext *alloc_param)
+{
+ unsigned int channels;
+ snd_pcm_uframes_t period_size;
+ ssize_t periodbytes;
+ ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream);
+ u32 buffer_addr = virt_to_phys(substream->dma_buffer.area);
+
+ channels = substream->runtime->channels;
+ period_size = substream->runtime->period_size;
+ periodbytes = samples_to_bytes(substream->runtime, period_size);
+ alloc_param->ring_buf_info[0].addr = buffer_addr;
+ alloc_param->ring_buf_info[0].size = buffer_bytes;
+ alloc_param->sg_count = 1;
+ alloc_param->reserved = 0;
+ alloc_param->frag_size = periodbytes * channels;
+
+}
static void sst_fill_pcm_params(struct snd_pcm_substream *substream,
- struct sst_pcm_params *param)
+ struct snd_sst_stream_params *param)
{
+ param->uc.pcm_params.num_chan = (u8) substream->runtime->channels;
+ param->uc.pcm_params.pcm_wd_sz = substream->runtime->sample_bits;
+ param->uc.pcm_params.sfreq = substream->runtime->rate;
+
+ /* PCM stream via ALSA interface */
+ param->uc.pcm_params.use_offload_path = 0;
+ param->uc.pcm_params.reserved2 = 0;
+ memset(param->uc.pcm_params.channel_map, 0, sizeof(u8));
- param->num_chan = (u8) substream->runtime->channels;
- param->pcm_wd_sz = substream->runtime->sample_bits;
- param->reserved = 0;
- param->sfreq = substream->runtime->rate;
- param->ring_buffer_size = snd_pcm_lib_buffer_bytes(substream);
- param->period_count = substream->runtime->period_size;
- param->ring_buffer_addr = virt_to_phys(substream->dma_buffer.area);
- pr_debug("period_cnt = %d\n", param->period_count);
- pr_debug("sfreq= %d, wd_sz = %d\n", param->sfreq, param->pcm_wd_sz);
}
-static int sst_platform_alloc_stream(struct snd_pcm_substream *substream)
+static int sst_get_stream_mapping(int dev, int sdev, int dir,
+ struct sst_dev_stream_map *map, int size)
+{
+ int i;
+
+ if (map == NULL)
+ return -EINVAL;
+
+
+ /* index 0 is not used in stream map */
+ for (i = 1; i < size; i++) {
+ if ((map[i].dev_num == dev) && (map[i].direction == dir))
+ return i;
+ }
+ return 0;
+}
+
+int sst_fill_stream_params(void *substream,
+ const struct sst_data *ctx, struct snd_sst_params *str_params, bool is_compress)
+{
+ int map_size;
+ int index;
+ struct sst_dev_stream_map *map;
+ struct snd_pcm_substream *pstream = NULL;
+ struct snd_compr_stream *cstream = NULL;
+
+ map = ctx->pdata->pdev_strm_map;
+ map_size = ctx->pdata->strm_map_size;
+
+ if (is_compress == true)
+ cstream = (struct snd_compr_stream *)substream;
+ else
+ pstream = (struct snd_pcm_substream *)substream;
+
+ str_params->stream_type = SST_STREAM_TYPE_MUSIC;
+
+ /* For pcm streams */
+ if (pstream) {
+ index = sst_get_stream_mapping(pstream->pcm->device,
+ pstream->number, pstream->stream,
+ map, map_size);
+ if (index <= 0)
+ return -EINVAL;
+
+ str_params->stream_id = index;
+ str_params->device_type = map[index].device_id;
+ str_params->task = map[index].task_id;
+
+ str_params->ops = (u8)pstream->stream;
+ }
+
+ if (cstream) {
+ index = sst_get_stream_mapping(cstream->device->device,
+ 0, cstream->direction,
+ map, map_size);
+ if (index <= 0)
+ return -EINVAL;
+ str_params->stream_id = index;
+ str_params->device_type = map[index].device_id;
+ str_params->task = map[index].task_id;
+
+ str_params->ops = (u8)cstream->direction;
+ }
+ return 0;
+}
+
+static int sst_platform_alloc_stream(struct snd_pcm_substream *substream,
+ struct snd_soc_platform *platform)
{
struct sst_runtime_stream *stream =
substream->runtime->private_data;
- struct sst_pcm_params param = {0};
- struct sst_stream_params str_params = {0};
- int ret_val;
+ struct snd_sst_stream_params param = {{{0,},},};
+ struct snd_sst_params str_params = {0};
+ struct snd_sst_alloc_params_ext alloc_params = {0};
+ int ret_val = 0;
+ struct sst_data *ctx = snd_soc_platform_get_drvdata(platform);
/* set codec params and inform SST driver the same */
sst_fill_pcm_params(substream, ¶m);
+ sst_fill_alloc_params(substream, &alloc_params);
substream->runtime->dma_area = substream->dma_buffer.area;
str_params.sparams = param;
- str_params.codec = param.codec;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- str_params.ops = STREAM_OPS_PLAYBACK;
- str_params.device_type = substream->pcm->device + 1;
- pr_debug("Playbck stream,Device %d\n",
- substream->pcm->device);
- } else {
- str_params.ops = STREAM_OPS_CAPTURE;
- str_params.device_type = SND_SST_DEVICE_CAPTURE;
- pr_debug("Capture stream,Device %d\n",
- substream->pcm->device);
- }
- ret_val = stream->ops->open(&str_params);
- pr_debug("SST_SND_PLAY/CAPTURE ret_val = %x\n", ret_val);
+ str_params.aparams = alloc_params;
+ str_params.codec = SST_CODEC_TYPE_PCM;
+
+ /* fill the device type and stream id to pass to SST driver */
+ ret_val = sst_fill_stream_params(substream, ctx, &str_params, false);
if (ret_val < 0)
return ret_val;
- stream->stream_info.str_id = ret_val;
- pr_debug("str id : %d\n", stream->stream_info.str_id);
+ stream->stream_info.str_id = str_params.stream_id;
+
+ ret_val = stream->ops->open(&str_params);
+ if (ret_val <= 0)
+ return ret_val;
+
+
return ret_val;
}
-static void sst_period_elapsed(void *mad_substream)
+static void sst_period_elapsed(void *arg)
{
- struct snd_pcm_substream *substream = mad_substream;
+ struct snd_pcm_substream *substream = arg;
struct sst_runtime_stream *stream;
int status;
pr_debug("setting buffer ptr param\n");
sst_set_stream_status(stream, SST_PLATFORM_INIT);
stream->stream_info.period_elapsed = sst_period_elapsed;
- stream->stream_info.mad_substream = substream;
+ stream->stream_info.arg = substream;
stream->stream_info.buffer_ptr = 0;
stream->stream_info.sfreq = substream->runtime->rate;
ret_val = stream->ops->device_control(
}
/* end -- helper functions */
-static int sst_platform_open(struct snd_pcm_substream *substream)
+static int sst_media_open(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
+ int ret_val = 0;
struct snd_pcm_runtime *runtime = substream->runtime;
struct sst_runtime_stream *stream;
- int ret_val;
-
- pr_debug("sst_platform_open called\n");
-
- snd_soc_set_runtime_hwparams(substream, &sst_platform_pcm_hw);
- ret_val = snd_pcm_hw_constraint_integer(runtime,
- SNDRV_PCM_HW_PARAM_PERIODS);
- if (ret_val < 0)
- return ret_val;
stream = kzalloc(sizeof(*stream), GFP_KERNEL);
if (!stream)
/* get the sst ops */
mutex_lock(&sst_lock);
- if (!sst) {
+ if (!sst ||
+ !try_module_get(sst->dev->driver->owner)) {
pr_err("no device available to run\n");
- mutex_unlock(&sst_lock);
- kfree(stream);
- return -ENODEV;
- }
- if (!try_module_get(sst->dev->driver->owner)) {
- mutex_unlock(&sst_lock);
- kfree(stream);
- return -ENODEV;
+ ret_val = -ENODEV;
+ goto out_ops;
}
stream->ops = sst->ops;
mutex_unlock(&sst_lock);
stream->stream_info.str_id = 0;
- sst_set_stream_status(stream, SST_PLATFORM_INIT);
- stream->stream_info.mad_substream = substream;
+
+ stream->stream_info.arg = substream;
/* allocate memory for SST API set */
runtime->private_data = stream;
- return 0;
+ /* Make sure, that the period size is always even */
+ snd_pcm_hw_constraint_step(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIODS, 2);
+
+ return snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+out_ops:
+ kfree(stream);
+ mutex_unlock(&sst_lock);
+ return ret_val;
}
-static int sst_platform_close(struct snd_pcm_substream *substream)
+static void sst_media_close(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct sst_runtime_stream *stream;
int ret_val = 0, str_id;
- pr_debug("sst_platform_close called\n");
stream = substream->runtime->private_data;
str_id = stream->stream_info.str_id;
if (str_id)
ret_val = stream->ops->close(str_id);
module_put(sst->dev->driver->owner);
kfree(stream);
- return ret_val;
}
-static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream)
+static inline unsigned int get_current_pipe_id(struct snd_soc_platform *platform,
+ struct snd_pcm_substream *substream)
+{
+ struct sst_data *sst = snd_soc_platform_get_drvdata(platform);
+ struct sst_dev_stream_map *map = sst->pdata->pdev_strm_map;
+ struct sst_runtime_stream *stream =
+ substream->runtime->private_data;
+ u32 str_id = stream->stream_info.str_id;
+ unsigned int pipe_id;
+ pipe_id = map[str_id].device_id;
+
+ pr_debug("%s: got pipe_id = %#x for str_id = %d\n",
+ __func__, pipe_id, str_id);
+ return pipe_id;
+}
+
+static int sst_media_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct sst_runtime_stream *stream;
int ret_val = 0, str_id;
- pr_debug("sst_platform_pcm_prepare called\n");
stream = substream->runtime->private_data;
str_id = stream->stream_info.str_id;
if (stream->stream_info.str_id) {
return ret_val;
}
- ret_val = sst_platform_alloc_stream(substream);
- if (ret_val < 0)
+ ret_val = sst_platform_alloc_stream(substream, dai->platform);
+ if (ret_val <= 0)
return ret_val;
snprintf(substream->pcm->id, sizeof(substream->pcm->id),
"%d", stream->stream_info.str_id);
return ret_val;
}
+static int sst_media_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+ memset(substream->runtime->dma_area, 0, params_buffer_bytes(params));
+ return 0;
+}
+
+static int sst_media_hw_free(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static struct snd_soc_dai_ops sst_media_dai_ops = {
+ .startup = sst_media_open,
+ .shutdown = sst_media_close,
+ .prepare = sst_media_prepare,
+ .hw_params = sst_media_hw_params,
+ .hw_free = sst_media_hw_free,
+};
+
+static int sst_platform_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime;
+
+ if (substream->pcm->internal)
+ return 0;
+
+ runtime = substream->runtime;
+ runtime->hw = sst_platform_pcm_hw;
+ return 0;
+}
+
static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream,
int cmd)
{
pr_debug("sst: Trigger Start\n");
str_cmd = SST_SND_START;
status = SST_PLATFORM_RUNNING;
- stream->stream_info.mad_substream = substream;
+ stream->stream_info.arg = substream;
break;
case SNDRV_PCM_TRIGGER_STOP:
pr_debug("sst: in stop\n");
pr_err("sst: error code = %d\n", ret_val);
return ret_val;
}
- return stream->stream_info.buffer_ptr;
-}
-
-static int sst_platform_pcm_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
- memset(substream->runtime->dma_area, 0, params_buffer_bytes(params));
-
- return 0;
-}
-
-static int sst_platform_pcm_hw_free(struct snd_pcm_substream *substream)
-{
- return snd_pcm_lib_free_pages(substream);
+ substream->runtime->delay = str_info->pcm_delay;
+ return str_info->buffer_ptr;
}
static struct snd_pcm_ops sst_platform_ops = {
.open = sst_platform_open,
- .close = sst_platform_close,
.ioctl = snd_pcm_lib_ioctl,
- .prepare = sst_platform_pcm_prepare,
.trigger = sst_platform_pcm_trigger,
.pointer = sst_platform_pcm_pointer,
- .hw_params = sst_platform_pcm_hw_params,
- .hw_free = sst_platform_pcm_hw_free,
};
static void sst_pcm_free(struct snd_pcm *pcm)
static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_soc_dai *dai = rtd->cpu_dai;
struct snd_pcm *pcm = rtd->pcm;
int retval = 0;
- pr_debug("sst_pcm_new called\n");
- if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream ||
- pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+ if (dai->driver->playback.channels_min ||
+ dai->driver->capture.channels_min) {
retval = snd_pcm_lib_preallocate_pages_for_all(pcm,
SNDRV_DMA_TYPE_CONTINUOUS,
- snd_dma_continuous_data(GFP_KERNEL),
+ snd_dma_continuous_data(GFP_DMA),
SST_MIN_BUFFER, SST_MAX_BUFFER);
if (retval) {
pr_err("dma buffer allocationf fail\n");
static int sst_platform_probe(struct platform_device *pdev)
{
+ struct sst_data *drv;
int ret;
+ struct sst_platform_data *pdata;
+
+ drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL);
+ if (drv == NULL) {
+ pr_err("kzalloc failed\n");
+ return -ENOMEM;
+ }
+
+ pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL);
+ if (pdata == NULL) {
+ pr_err("kzalloc failed for pdata\n");
+ return -ENOMEM;
+ }
+
+ pdata->pdev_strm_map = dpcm_strm_map;
+ pdata->strm_map_size = ARRAY_SIZE(dpcm_strm_map);
+ drv->pdata = pdata;
+ mutex_init(&drv->lock);
+ dev_set_drvdata(&pdev->dev, drv);
- pr_debug("sst_platform_probe called\n");
- sst = NULL;
ret = snd_soc_register_platform(&pdev->dev, &sst_soc_platform_drv);
if (ret) {
pr_err("registering soc platform failed\n");
struct pcm_stream_info {
int str_id;
- void *mad_substream;
- void (*period_elapsed) (void *mad_substream);
+ void *arg;
+ void (*period_elapsed) (void *arg);
unsigned long long buffer_ptr;
+ unsigned long long pcm_delay;
int sfreq;
};
SST_SND_BUFFER_POINTER = 0x05,
SST_SND_STREAM_INIT = 0x06,
SST_SND_START = 0x07,
- SST_MAX_CONTROLS = 0x07,
+ SST_SET_BYTE_STREAM = 0x100A,
+ SST_GET_BYTE_STREAM = 0x100B,
+ SST_MAX_CONTROLS = SST_GET_BYTE_STREAM,
};
enum sst_stream_ops {
};
struct sst_ops {
- int (*open) (struct sst_stream_params *str_param);
+ int (*open) (struct snd_sst_params *str_param);
int (*device_control) (int cmd, void *arg);
+ int (*set_generic_params)(enum sst_controls cmd, void *arg);
int (*close) (unsigned int str_id);
};
char *name;
struct device *dev;
struct sst_ops *ops;
+ struct platform_device *pdev;
struct compress_sst_ops *compr_ops;
};
+struct sst_data;
void sst_set_stream_status(struct sst_runtime_stream *stream, int state);
+int sst_fill_stream_params(void *substream, const struct sst_data *ctx,
+ struct snd_sst_params *str_params, bool is_compress);
+
+struct sst_algo_int_control_v2 {
+ struct soc_mixer_control mc;
+ u16 module_id; /* module identifieer */
+ u16 pipe_id; /* location info: pipe_id + instance_id */
+ u16 instance_id;
+ unsigned int value; /* Value received is stored here */
+};
+struct sst_data {
+ struct platform_device *pdev;
+ struct sst_platform_data *pdata;
+ struct mutex lock;
+};
int sst_register_dsp(struct sst_device *sst);
int sst_unregister_dsp(struct sst_device *sst);
#endif
config SND_KIRKWOOD_SOC
tristate "SoC Audio for the Marvell Kirkwood and Dove chips"
- depends on ARCH_KIRKWOOD || ARCH_DOVE || ARCH_MVEBU || MACH_KIRKWOOD || COMPILE_TEST
+ depends on ARCH_DOVE || ARCH_MVEBU || COMPILE_TEST
help
Say Y or M if you want to add support for codecs attached to
the Kirkwood I2S interface. You will also need to select the
Say Y if you want to add support for SoC audio on
the Armada 370 Development Board.
-config SND_KIRKWOOD_SOC_OPENRD
- tristate "SoC Audio support for Kirkwood Openrd Client"
- depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE || COMPILE_TEST)
- depends on I2C
- select SND_SOC_CS42L51
- help
- Say Y if you want to add support for SoC audio on
- Openrd Client.
-
-config SND_KIRKWOOD_SOC_T5325
- tristate "SoC Audio support for HP t5325"
- depends on SND_KIRKWOOD_SOC && (MACH_T5325 || COMPILE_TEST) && I2C
- select SND_SOC_ALC5623
- help
- Say Y if you want to add support for SoC audio on
- the HP t5325 thin client.
-
obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o
-snd-soc-openrd-objs := kirkwood-openrd.o
-snd-soc-t5325-objs := kirkwood-t5325.o
snd-soc-armada-370-db-objs := armada-370-db.o
obj-$(CONFIG_SND_KIRKWOOD_SOC_ARMADA370_DB) += snd-soc-armada-370-db.o
-obj-$(CONFIG_SND_KIRKWOOD_SOC_OPENRD) += snd-soc-openrd.o
-obj-$(CONFIG_SND_KIRKWOOD_SOC_T5325) += snd-soc-t5325.o
}
static struct snd_pcm_hardware kirkwood_dma_snd_hw = {
- .info = (SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_MMAP |
- SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_PAUSE),
+ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_NO_PERIOD_WAKEUP,
.buffer_bytes_max = KIRKWOOD_SND_MAX_BUFFER_BYTES,
.period_bytes_min = KIRKWOOD_SND_MIN_PERIOD_BYTES,
.period_bytes_max = KIRKWOOD_SND_MAX_PERIOD_BYTES,
KIRKWOOD_PLAYCTL_SIZE_MASK);
priv->ctl_play |= ctl_play;
} else {
- priv->ctl_rec &= ~KIRKWOOD_RECCTL_SIZE_MASK;
+ priv->ctl_rec &= ~(KIRKWOOD_RECCTL_ENABLE_MASK |
+ KIRKWOOD_RECCTL_SIZE_MASK);
priv->ctl_rec |= ctl_rec;
}
return 0;
}
+static unsigned kirkwood_i2s_play_mute(unsigned ctl)
+{
+ if (!(ctl & KIRKWOOD_PLAYCTL_I2S_EN))
+ ctl |= KIRKWOOD_PLAYCTL_I2S_MUTE;
+ if (!(ctl & KIRKWOOD_PLAYCTL_SPDIF_EN))
+ ctl |= KIRKWOOD_PLAYCTL_SPDIF_MUTE;
+ return ctl;
+}
+
static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream,
int cmd, struct snd_soc_dai *dai)
{
+ struct snd_pcm_runtime *runtime = substream->runtime;
struct kirkwood_dma_data *priv = snd_soc_dai_get_drvdata(dai);
uint32_t ctl, value;
ctl = readl(priv->io + KIRKWOOD_PLAYCTL);
- if (ctl & KIRKWOOD_PLAYCTL_PAUSE) {
+ if ((ctl & KIRKWOOD_PLAYCTL_ENABLE_MASK) == 0) {
unsigned timeout = 5000;
/*
* The Armada510 spec says that if we enter pause mode, the
ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN; /* i2s */
else
ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN; /* spdif */
-
+ ctl = kirkwood_i2s_play_mute(ctl);
value = ctl & ~KIRKWOOD_PLAYCTL_ENABLE_MASK;
writel(value, priv->io + KIRKWOOD_PLAYCTL);
/* enable interrupts */
- value = readl(priv->io + KIRKWOOD_INT_MASK);
- value |= KIRKWOOD_INT_CAUSE_PLAY_BYTES;
- writel(value, priv->io + KIRKWOOD_INT_MASK);
+ if (!runtime->no_period_wakeup) {
+ value = readl(priv->io + KIRKWOOD_INT_MASK);
+ value |= KIRKWOOD_INT_CAUSE_PLAY_BYTES;
+ writel(value, priv->io + KIRKWOOD_INT_MASK);
+ }
/* enable playback */
writel(ctl, priv->io + KIRKWOOD_PLAYCTL);
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE |
KIRKWOOD_PLAYCTL_SPDIF_MUTE);
+ ctl = kirkwood_i2s_play_mute(ctl);
writel(ctl, priv->io + KIRKWOOD_PLAYCTL);
break;
else
ctl &= ~KIRKWOOD_RECCTL_I2S_EN; /* spdif */
- value = ctl & ~(KIRKWOOD_RECCTL_I2S_EN |
- KIRKWOOD_RECCTL_SPDIF_EN);
+ value = ctl & ~KIRKWOOD_RECCTL_ENABLE_MASK;
writel(value, priv->io + KIRKWOOD_RECCTL);
/* enable interrupts */
/* disable all records */
value = readl(priv->io + KIRKWOOD_RECCTL);
- value &= ~(KIRKWOOD_RECCTL_I2S_EN | KIRKWOOD_RECCTL_SPDIF_EN);
+ value &= ~KIRKWOOD_RECCTL_ENABLE_MASK;
writel(value, priv->io + KIRKWOOD_RECCTL);
break;
writel(value, priv->io + KIRKWOOD_PLAYCTL);
value = readl(priv->io + KIRKWOOD_RECCTL);
- value &= ~(KIRKWOOD_RECCTL_I2S_EN | KIRKWOOD_RECCTL_SPDIF_EN);
+ value &= ~KIRKWOOD_RECCTL_ENABLE_MASK;
writel(value, priv->io + KIRKWOOD_RECCTL);
return 0;
+++ /dev/null
-/*
- * kirkwood-openrd.c
- *
- * (c) 2010 Arnaud Patard <apatard@mandriva.com>
- * (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
- *
- * 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.
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/interrupt.h>
-#include <linux/platform_device.h>
-#include <linux/slab.h>
-#include <sound/soc.h>
-#include <linux/platform_data/asoc-kirkwood.h>
-#include "../codecs/cs42l51.h"
-
-static int openrd_client_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- unsigned int freq;
-
- switch (params_rate(params)) {
- default:
- case 44100:
- freq = 11289600;
- break;
- case 48000:
- freq = 12288000;
- break;
- case 96000:
- freq = 24576000;
- break;
- }
-
- return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN);
-
-}
-
-static struct snd_soc_ops openrd_client_ops = {
- .hw_params = openrd_client_hw_params,
-};
-
-
-static struct snd_soc_dai_link openrd_client_dai[] = {
-{
- .name = "CS42L51",
- .stream_name = "CS42L51 HiFi",
- .cpu_dai_name = "i2s",
- .platform_name = "mvebu-audio",
- .codec_dai_name = "cs42l51-hifi",
- .codec_name = "cs42l51-codec.0-004a",
- .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS,
- .ops = &openrd_client_ops,
-},
-};
-
-
-static struct snd_soc_card openrd_client = {
- .name = "OpenRD Client",
- .owner = THIS_MODULE,
- .dai_link = openrd_client_dai,
- .num_links = ARRAY_SIZE(openrd_client_dai),
-};
-
-static int openrd_probe(struct platform_device *pdev)
-{
- struct snd_soc_card *card = &openrd_client;
- int ret;
-
- card->dev = &pdev->dev;
-
- ret = snd_soc_register_card(card);
- if (ret)
- dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
- ret);
- return ret;
-}
-
-static int openrd_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
-static struct platform_driver openrd_driver = {
- .driver = {
- .name = "openrd-client-audio",
- .owner = THIS_MODULE,
- },
- .probe = openrd_probe,
- .remove = openrd_remove,
-};
-
-module_platform_driver(openrd_driver);
-
-/* Module information */
-MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
-MODULE_DESCRIPTION("ALSA SoC OpenRD Client");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:openrd-client-audio");
+++ /dev/null
-/*
- * kirkwood-t5325.c
- *
- * (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
- *
- * 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.
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/interrupt.h>
-#include <linux/platform_device.h>
-#include <linux/slab.h>
-#include <sound/soc.h>
-#include <linux/platform_data/asoc-kirkwood.h>
-#include "../codecs/alc5623.h"
-
-static int t5325_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- unsigned int freq;
-
- freq = params_rate(params) * 256;
-
- return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN);
-
-}
-
-static struct snd_soc_ops t5325_ops = {
- .hw_params = t5325_hw_params,
-};
-
-static const struct snd_soc_dapm_widget t5325_dapm_widgets[] = {
- SND_SOC_DAPM_HP("Headphone Jack", NULL),
- SND_SOC_DAPM_SPK("Speaker", NULL),
- SND_SOC_DAPM_MIC("Mic Jack", NULL),
-};
-
-static const struct snd_soc_dapm_route t5325_route[] = {
- { "Headphone Jack", NULL, "HPL" },
- { "Headphone Jack", NULL, "HPR" },
-
- {"Speaker", NULL, "SPKOUT"},
- {"Speaker", NULL, "SPKOUTN"},
-
- { "MIC1", NULL, "Mic Jack" },
- { "MIC2", NULL, "Mic Jack" },
-};
-
-static struct snd_soc_dai_link t5325_dai[] = {
-{
- .name = "ALC5621",
- .stream_name = "ALC5621 HiFi",
- .cpu_dai_name = "i2s",
- .platform_name = "mvebu-audio",
- .codec_dai_name = "alc5621-hifi",
- .codec_name = "alc562x-codec.0-001a",
- .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS,
- .ops = &t5325_ops,
-},
-};
-
-static struct snd_soc_card t5325 = {
- .name = "t5325",
- .owner = THIS_MODULE,
- .dai_link = t5325_dai,
- .num_links = ARRAY_SIZE(t5325_dai),
-
- .dapm_widgets = t5325_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(t5325_dapm_widgets),
- .dapm_routes = t5325_route,
- .num_dapm_routes = ARRAY_SIZE(t5325_route),
-};
-
-static int t5325_probe(struct platform_device *pdev)
-{
- struct snd_soc_card *card = &t5325;
- int ret;
-
- card->dev = &pdev->dev;
-
- ret = snd_soc_register_card(card);
- if (ret)
- dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
- ret);
- return ret;
-}
-
-static int t5325_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
-static struct platform_driver t5325_driver = {
- .driver = {
- .name = "t5325-audio",
- .owner = THIS_MODULE,
- },
- .probe = t5325_probe,
- .remove = t5325_remove,
-};
-
-module_platform_driver(t5325_driver);
-
-MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
-MODULE_DESCRIPTION("ALSA SoC t5325 audio client");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:t5325-audio");
#define KIRKWOOD_RECCTL_SIZE_24 (1<<0)
#define KIRKWOOD_RECCTL_SIZE_32 (0<<0)
+#define KIRKWOOD_RECCTL_ENABLE_MASK (KIRKWOOD_RECCTL_SPDIF_EN | \
+ KIRKWOOD_RECCTL_I2S_EN)
+
#define KIRKWOOD_REC_BUF_ADDR 0x1004
#define KIRKWOOD_REC_BUF_SIZE 0x1008
#define KIRKWOOD_REC_BYTE_COUNT 0x100C
/* Theses values come from the marvell alsa driver */
/* need to find where they come from */
-#define KIRKWOOD_SND_MIN_PERIODS 8
+#define KIRKWOOD_SND_MIN_PERIODS 2
#define KIRKWOOD_SND_MAX_PERIODS 16
-#define KIRKWOOD_SND_MIN_PERIOD_BYTES 0x800
+#define KIRKWOOD_SND_MIN_PERIOD_BYTES 256
#define KIRKWOOD_SND_MAX_PERIOD_BYTES 0x8000
#define KIRKWOOD_SND_MAX_BUFFER_BYTES (KIRKWOOD_SND_MAX_PERIOD_BYTES \
* KIRKWOOD_SND_MAX_PERIODS)
git.openpandora.org / pandora-kernel.git / commitdiff