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Commit 3e4555ab authored by Mark Brown's avatar Mark Brown
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Merge remote-tracking branch 'asoc/topic/intel' into asoc-next

parents 7c310f16 987da3fe
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Showing with 801 additions and 1712 deletions
include/sound/soc-acpi.h
View file @ 3e4555ab
......@@ -50,9 +50,6 @@ snd_soc_acpi_find_package_from_hid(const u8 hid[ACPI_ID_LEN],
struct snd_soc_acpi_mach *
snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines);
/* acpi check hid */
bool snd_soc_acpi_check_hid(const u8 hid[ACPI_ID_LEN]);
/**
* snd_soc_acpi_mach: ACPI-based machine descriptor. Most of the fields are
* related to the hardware, except for the firmware and topology file names.
......
include/uapi/sound/snd_sst_tokens.h
View file @ 3e4555ab
......@@ -222,6 +222,17 @@
* %SKL_TKN_MM_U32_NUM_IN_FMT: Number of input formats
* %SKL_TKN_MM_U32_NUM_OUT_FMT: Number of output formats
*
* %SKL_TKN_U32_ASTATE_IDX: Table Index for the A-State entry to be filled
* with kcps and clock source
*
* %SKL_TKN_U32_ASTATE_COUNT: Number of valid entries in A-State table
*
* %SKL_TKN_U32_ASTATE_KCPS: Specifies the core load threshold (in kilo
* cycles per second) below which DSP is clocked
* from source specified by clock source.
*
* %SKL_TKN_U32_ASTATE_CLK_SRC: Clock source for A-State entry
*
* module_id and loadable flags dont have tokens as these values will be
* read from the DSP FW manifest
*
......@@ -309,7 +320,11 @@ enum SKL_TKNS {
SKL_TKN_MM_U32_NUM_IN_FMT,
SKL_TKN_MM_U32_NUM_OUT_FMT,
SKL_TKN_MAX = SKL_TKN_MM_U32_NUM_OUT_FMT,
SKL_TKN_U32_ASTATE_IDX,
SKL_TKN_U32_ASTATE_COUNT,
SKL_TKN_U32_ASTATE_KCPS,
SKL_TKN_U32_ASTATE_CLK_SRC,
SKL_TKN_MAX = SKL_TKN_U32_ASTATE_CLK_SRC,
};
#endif
sound/soc/codecs/Kconfig
View file @ 3e4555ab
......@@ -133,7 +133,6 @@ config SND_SOC_ALL_CODECS
select SND_SOC_SGTL5000 if I2C
select SND_SOC_SI476X if MFD_SI476X_CORE
select SND_SOC_SIRF_AUDIO_CODEC
select SND_SOC_SN95031 if INTEL_SCU_IPC
select SND_SOC_SPDIF
select SND_SOC_SSM2518 if I2C
select SND_SOC_SSM2602_SPI if SPI_MASTER
......@@ -818,9 +817,6 @@ config SND_SOC_SIRF_AUDIO_CODEC
tristate "SiRF SoC internal audio codec"
select REGMAP_MMIO
config SND_SOC_SN95031
tristate
config SND_SOC_SPDIF
tristate "S/PDIF CODEC"
......
sound/soc/codecs/Makefile
View file @ 3e4555ab
......@@ -140,7 +140,6 @@ snd-soc-sigmadsp-i2c-objs := sigmadsp-i2c.o
snd-soc-sigmadsp-regmap-objs := sigmadsp-regmap.o
snd-soc-si476x-objs := si476x.o
snd-soc-sirf-audio-codec-objs := sirf-audio-codec.o
snd-soc-sn95031-objs := sn95031.o
snd-soc-spdif-tx-objs := spdif_transmitter.o
snd-soc-spdif-rx-objs := spdif_receiver.o
snd-soc-ssm2518-objs := ssm2518.o
......
sound/soc/codecs/sn95031.c deleted 100644 → 0
View file @ 7c310f16
/*
* sn95031.c - TI sn95031 Codec driver
*
* Copyright (C) 2010 Intel Corp
* Author: Vinod Koul <vinod.koul@intel.com>
* Author: Harsha Priya <priya.harsha@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.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
*
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*
*
*/
#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt
#include <linux/platform_device.h>
#include <linux/delay.h>
#include <linux/slab.h>
#include <linux/module.h>
#include <asm/intel_scu_ipc.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 "sn95031.h"
#define SN95031_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100)
#define SN95031_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE)
/* adc helper functions */
/* enables mic bias voltage */
static void sn95031_enable_mic_bias(struct snd_soc_codec *codec)
{
snd_soc_write(codec, SN95031_VAUD, BIT(2)|BIT(1)|BIT(0));
snd_soc_update_bits(codec, SN95031_MICBIAS, BIT(2), BIT(2));
}
/* Enable/Disable the ADC depending on the argument */
static void configure_adc(struct snd_soc_codec *sn95031_codec, int val)
{
int value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1);
if (val) {
/* Enable and start the ADC */
value |= (SN95031_ADC_ENBL | SN95031_ADC_START);
value &= (~SN95031_ADC_NO_LOOP);
} else {
/* Just stop the ADC */
value &= (~SN95031_ADC_START);
}
snd_soc_write(sn95031_codec, SN95031_ADC1CNTL1, value);
}
/*
* finds an empty channel for conversion
* If the ADC is not enabled then start using 0th channel
* itself. Otherwise find an empty channel by looking for a
* channel in which the stopbit is set to 1. returns the index
* of the first free channel if succeeds or an error code.
*
* Context: can sleep
*
*/
static int find_free_channel(struct snd_soc_codec *sn95031_codec)
{
int i, value;
/* check whether ADC is enabled */
value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1);
if ((value & SN95031_ADC_ENBL) == 0)
return 0;
/* ADC is already enabled; Looking for an empty channel */
for (i = 0; i < SN95031_ADC_CHANLS_MAX; i++) {
value = snd_soc_read(sn95031_codec,
SN95031_ADC_CHNL_START_ADDR + i);
if (value & SN95031_STOPBIT_MASK)
break;
}
return (i == SN95031_ADC_CHANLS_MAX) ? (-EINVAL) : i;
}
/* Initialize the ADC for reading micbias values. Can sleep. */
static int sn95031_initialize_adc(struct snd_soc_codec *sn95031_codec)
{
int base_addr, chnl_addr;
int value;
int channel_index;
/* Index of the first channel in which the stop bit is set */
channel_index = find_free_channel(sn95031_codec);
if (channel_index < 0) {
pr_err("No free ADC channels");
return channel_index;
}
base_addr = SN95031_ADC_CHNL_START_ADDR + channel_index;
if (!(channel_index == 0 || channel_index == SN95031_ADC_LOOP_MAX)) {
/* Reset stop bit for channels other than 0 and 12 */
value = snd_soc_read(sn95031_codec, base_addr);
/* Set the stop bit to zero */
snd_soc_write(sn95031_codec, base_addr, value & 0xEF);
/* Index of the first free channel */
base_addr++;
channel_index++;
}
/* Since this is the last channel, set the stop bit
to 1 by ORing the DIE_SENSOR_CODE with 0x10 */
snd_soc_write(sn95031_codec, base_addr,
SN95031_AUDIO_DETECT_CODE | 0x10);
chnl_addr = SN95031_ADC_DATA_START_ADDR + 2 * channel_index;
pr_debug("mid_initialize : %x", chnl_addr);
configure_adc(sn95031_codec, 1);
return chnl_addr;
}
/* reads the ADC registers and gets the mic bias value in mV. */
static unsigned int sn95031_get_mic_bias(struct snd_soc_codec *codec)
{
u16 adc_adr = sn95031_initialize_adc(codec);
u16 adc_val1, adc_val2;
unsigned int mic_bias;
sn95031_enable_mic_bias(codec);
/* Enable the sound card for conversion before reading */
snd_soc_write(codec, SN95031_ADC1CNTL3, 0x05);
/* Re-toggle the RRDATARD bit */
snd_soc_write(codec, SN95031_ADC1CNTL3, 0x04);
/* Read the higher bits of data */
msleep(1000);
adc_val1 = snd_soc_read(codec, adc_adr);
adc_adr++;
adc_val2 = snd_soc_read(codec, adc_adr);
/* Adding lower two bits to the higher bits */
mic_bias = (adc_val1 << 2) + (adc_val2 & 3);
mic_bias = (mic_bias * SN95031_ADC_ONE_LSB_MULTIPLIER) / 1000;
pr_debug("mic bias = %dmV\n", mic_bias);
return mic_bias;
}
/*end - adc helper functions */
static int sn95031_read(void *ctx, unsigned int reg, unsigned int *val)
{
u8 value = 0;
int ret;
ret = intel_scu_ipc_ioread8(reg, &value);
if (ret == 0)
*val = value;
return ret;
}
static int sn95031_write(void *ctx, unsigned int reg, unsigned int value)
{
return intel_scu_ipc_iowrite8(reg, value);
}
static const struct regmap_config sn95031_regmap = {
.reg_read = sn95031_read,
.reg_write = sn95031_write,
};
static int sn95031_set_vaud_bias(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
switch (level) {
case SND_SOC_BIAS_ON:
break;
case SND_SOC_BIAS_PREPARE:
if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_STANDBY) {
pr_debug("vaud_bias powering up pll\n");
/* power up the pll */
snd_soc_write(codec, SN95031_AUDPLLCTRL, BIT(5));
/* enable pcm 2 */
snd_soc_update_bits(codec, SN95031_PCM2C2,
BIT(0), BIT(0));
}
break;
case SND_SOC_BIAS_STANDBY:
switch (snd_soc_codec_get_bias_level(codec)) {
case SND_SOC_BIAS_OFF:
pr_debug("vaud_bias power up rail\n");
/* power up the rail */
snd_soc_write(codec, SN95031_VAUD,
BIT(2)|BIT(1)|BIT(0));
msleep(1);
break;
case SND_SOC_BIAS_PREPARE:
/* turn off pcm */
pr_debug("vaud_bias power dn pcm\n");
snd_soc_update_bits(codec, SN95031_PCM2C2, BIT(0), 0);
snd_soc_write(codec, SN95031_AUDPLLCTRL, 0);
break;
default:
break;
}
break;
case SND_SOC_BIAS_OFF:
pr_debug("vaud_bias _OFF doing rail shutdown\n");
snd_soc_write(codec, SN95031_VAUD, BIT(3));
break;
}
return 0;
}
static int sn95031_vhs_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
if (SND_SOC_DAPM_EVENT_ON(event)) {
pr_debug("VHS SND_SOC_DAPM_EVENT_ON doing rail startup now\n");
/* power up the rail */
snd_soc_write(codec, SN95031_VHSP, 0x3D);
snd_soc_write(codec, SN95031_VHSN, 0x3F);
msleep(1);
} else if (SND_SOC_DAPM_EVENT_OFF(event)) {
pr_debug("VHS SND_SOC_DAPM_EVENT_OFF doing rail shutdown\n");
snd_soc_write(codec, SN95031_VHSP, 0xC4);
snd_soc_write(codec, SN95031_VHSN, 0x04);
}
return 0;
}
static int sn95031_vihf_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
if (SND_SOC_DAPM_EVENT_ON(event)) {
pr_debug("VIHF SND_SOC_DAPM_EVENT_ON doing rail startup now\n");
/* power up the rail */
snd_soc_write(codec, SN95031_VIHF, 0x27);
msleep(1);
} else if (SND_SOC_DAPM_EVENT_OFF(event)) {
pr_debug("VIHF SND_SOC_DAPM_EVENT_OFF doing rail shutdown\n");
snd_soc_write(codec, SN95031_VIHF, 0x24);
}
return 0;
}
static int sn95031_dmic12_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
unsigned int ldo = 0, clk_dir = 0, data_dir = 0;
if (SND_SOC_DAPM_EVENT_ON(event)) {
ldo = BIT(5)|BIT(4);
clk_dir = BIT(0);
data_dir = BIT(7);
}
/* program DMIC LDO, clock and set clock */
snd_soc_update_bits(codec, SN95031_MICBIAS, BIT(5)|BIT(4), ldo);
snd_soc_update_bits(codec, SN95031_DMICBUF0123, BIT(0), clk_dir);
snd_soc_update_bits(codec, SN95031_DMICBUF0123, BIT(7), data_dir);
return 0;
}
static int sn95031_dmic34_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
unsigned int ldo = 0, clk_dir = 0, data_dir = 0;
if (SND_SOC_DAPM_EVENT_ON(event)) {
ldo = BIT(5)|BIT(4);
clk_dir = BIT(2);
data_dir = BIT(1);
}
/* program DMIC LDO, clock and set clock */
snd_soc_update_bits(codec, SN95031_MICBIAS, BIT(5)|BIT(4), ldo);
snd_soc_update_bits(codec, SN95031_DMICBUF0123, BIT(2), clk_dir);
snd_soc_update_bits(codec, SN95031_DMICBUF45, BIT(1), data_dir);
return 0;
}
static int sn95031_dmic56_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
unsigned int ldo = 0;
if (SND_SOC_DAPM_EVENT_ON(event))
ldo = BIT(7)|BIT(6);
/* program DMIC LDO */
snd_soc_update_bits(codec, SN95031_MICBIAS, BIT(7)|BIT(6), ldo);
return 0;
}
/* mux controls */
static const char *sn95031_mic_texts[] = { "AMIC", "LineIn" };
static SOC_ENUM_SINGLE_DECL(sn95031_micl_enum,
SN95031_ADCCONFIG, 1, sn95031_mic_texts);
static const struct snd_kcontrol_new sn95031_micl_mux_control =
SOC_DAPM_ENUM("Route", sn95031_micl_enum);
static SOC_ENUM_SINGLE_DECL(sn95031_micr_enum,
SN95031_ADCCONFIG, 3, sn95031_mic_texts);
static const struct snd_kcontrol_new sn95031_micr_mux_control =
SOC_DAPM_ENUM("Route", sn95031_micr_enum);
static const char *sn95031_input_texts[] = { "DMIC1", "DMIC2", "DMIC3",
"DMIC4", "DMIC5", "DMIC6",
"ADC Left", "ADC Right" };
static SOC_ENUM_SINGLE_DECL(sn95031_input1_enum,
SN95031_AUDIOMUX12, 0, sn95031_input_texts);
static const struct snd_kcontrol_new sn95031_input1_mux_control =
SOC_DAPM_ENUM("Route", sn95031_input1_enum);
static SOC_ENUM_SINGLE_DECL(sn95031_input2_enum,
SN95031_AUDIOMUX12, 4, sn95031_input_texts);
static const struct snd_kcontrol_new sn95031_input2_mux_control =
SOC_DAPM_ENUM("Route", sn95031_input2_enum);
static SOC_ENUM_SINGLE_DECL(sn95031_input3_enum,
SN95031_AUDIOMUX34, 0, sn95031_input_texts);
static const struct snd_kcontrol_new sn95031_input3_mux_control =
SOC_DAPM_ENUM("Route", sn95031_input3_enum);
static SOC_ENUM_SINGLE_DECL(sn95031_input4_enum,
SN95031_AUDIOMUX34, 4, sn95031_input_texts);
static const struct snd_kcontrol_new sn95031_input4_mux_control =
SOC_DAPM_ENUM("Route", sn95031_input4_enum);
/* capture path controls */
static const char *sn95031_micmode_text[] = {"Single Ended", "Differential"};
/* 0dB to 30dB in 10dB steps */
static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 10, 0);
static SOC_ENUM_SINGLE_DECL(sn95031_micmode1_enum,
SN95031_MICAMP1, 1, sn95031_micmode_text);
static SOC_ENUM_SINGLE_DECL(sn95031_micmode2_enum,
SN95031_MICAMP2, 1, sn95031_micmode_text);
static const char *sn95031_dmic_cfg_text[] = {"GPO", "DMIC"};
static SOC_ENUM_SINGLE_DECL(sn95031_dmic12_cfg_enum,
SN95031_DMICMUX, 0, sn95031_dmic_cfg_text);
static SOC_ENUM_SINGLE_DECL(sn95031_dmic34_cfg_enum,
SN95031_DMICMUX, 1, sn95031_dmic_cfg_text);
static SOC_ENUM_SINGLE_DECL(sn95031_dmic56_cfg_enum,
SN95031_DMICMUX, 2, sn95031_dmic_cfg_text);
static const struct snd_kcontrol_new sn95031_snd_controls[] = {
SOC_ENUM("Mic1Mode Capture Route", sn95031_micmode1_enum),
SOC_ENUM("Mic2Mode Capture Route", sn95031_micmode2_enum),
SOC_ENUM("DMIC12 Capture Route", sn95031_dmic12_cfg_enum),
SOC_ENUM("DMIC34 Capture Route", sn95031_dmic34_cfg_enum),
SOC_ENUM("DMIC56 Capture Route", sn95031_dmic56_cfg_enum),
SOC_SINGLE_TLV("Mic1 Capture Volume", SN95031_MICAMP1,
2, 4, 0, mic_tlv),
SOC_SINGLE_TLV("Mic2 Capture Volume", SN95031_MICAMP2,
2, 4, 0, mic_tlv),
};
/* DAPM widgets */
static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = {
/* all end points mic, hs etc */
SND_SOC_DAPM_OUTPUT("HPOUTL"),
SND_SOC_DAPM_OUTPUT("HPOUTR"),
SND_SOC_DAPM_OUTPUT("EPOUT"),
SND_SOC_DAPM_OUTPUT("IHFOUTL"),
SND_SOC_DAPM_OUTPUT("IHFOUTR"),
SND_SOC_DAPM_OUTPUT("LINEOUTL"),
SND_SOC_DAPM_OUTPUT("LINEOUTR"),
SND_SOC_DAPM_OUTPUT("VIB1OUT"),
SND_SOC_DAPM_OUTPUT("VIB2OUT"),
SND_SOC_DAPM_INPUT("AMIC1"), /* headset mic */
SND_SOC_DAPM_INPUT("AMIC2"),
SND_SOC_DAPM_INPUT("DMIC1"),
SND_SOC_DAPM_INPUT("DMIC2"),
SND_SOC_DAPM_INPUT("DMIC3"),
SND_SOC_DAPM_INPUT("DMIC4"),
SND_SOC_DAPM_INPUT("DMIC5"),
SND_SOC_DAPM_INPUT("DMIC6"),
SND_SOC_DAPM_INPUT("LINEINL"),
SND_SOC_DAPM_INPUT("LINEINR"),
SND_SOC_DAPM_MICBIAS("AMIC1Bias", SN95031_MICBIAS, 2, 0),
SND_SOC_DAPM_MICBIAS("AMIC2Bias", SN95031_MICBIAS, 3, 0),
SND_SOC_DAPM_MICBIAS("DMIC12Bias", SN95031_DMICMUX, 3, 0),
SND_SOC_DAPM_MICBIAS("DMIC34Bias", SN95031_DMICMUX, 4, 0),
SND_SOC_DAPM_MICBIAS("DMIC56Bias", SN95031_DMICMUX, 5, 0),
SND_SOC_DAPM_SUPPLY("DMIC12supply", SN95031_DMICLK, 0, 0,
sn95031_dmic12_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_SUPPLY("DMIC34supply", SN95031_DMICLK, 1, 0,
sn95031_dmic34_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_SUPPLY("DMIC56supply", SN95031_DMICLK, 2, 0,
sn95031_dmic56_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_AIF_OUT("PCM_Out", "Capture", 0,
SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_SUPPLY("Headset Rail", SND_SOC_NOPM, 0, 0,
sn95031_vhs_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_SUPPLY("Speaker Rail", SND_SOC_NOPM, 0, 0,
sn95031_vihf_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
/* playback path driver enables */
SND_SOC_DAPM_PGA("Headset Left Playback",
SN95031_DRIVEREN, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("Headset Right Playback",
SN95031_DRIVEREN, 1, 0, NULL, 0),
SND_SOC_DAPM_PGA("Speaker Left Playback",
SN95031_DRIVEREN, 2, 0, NULL, 0),
SND_SOC_DAPM_PGA("Speaker Right Playback",
SN95031_DRIVEREN, 3, 0, NULL, 0),
SND_SOC_DAPM_PGA("Vibra1 Playback",
SN95031_DRIVEREN, 4, 0, NULL, 0),
SND_SOC_DAPM_PGA("Vibra2 Playback",
SN95031_DRIVEREN, 5, 0, NULL, 0),
SND_SOC_DAPM_PGA("Earpiece Playback",
SN95031_DRIVEREN, 6, 0, NULL, 0),
SND_SOC_DAPM_PGA("Lineout Left Playback",
SN95031_LOCTL, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("Lineout Right Playback",
SN95031_LOCTL, 4, 0, NULL, 0),
/* playback path filter enable */
SND_SOC_DAPM_PGA("Headset Left Filter",
SN95031_HSEPRXCTRL, 4, 0, NULL, 0),
SND_SOC_DAPM_PGA("Headset Right Filter",
SN95031_HSEPRXCTRL, 5, 0, NULL, 0),
SND_SOC_DAPM_PGA("Speaker Left Filter",
SN95031_IHFRXCTRL, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("Speaker Right Filter",
SN95031_IHFRXCTRL, 1, 0, NULL, 0),
/* DACs */
SND_SOC_DAPM_DAC("HSDAC Left", "Headset",
SN95031_DACCONFIG, 0, 0),
SND_SOC_DAPM_DAC("HSDAC Right", "Headset",
SN95031_DACCONFIG, 1, 0),
SND_SOC_DAPM_DAC("IHFDAC Left", "Speaker",
SN95031_DACCONFIG, 2, 0),
SND_SOC_DAPM_DAC("IHFDAC Right", "Speaker",
SN95031_DACCONFIG, 3, 0),
SND_SOC_DAPM_DAC("Vibra1 DAC", "Vibra1",
SN95031_VIB1C5, 1, 0),
SND_SOC_DAPM_DAC("Vibra2 DAC", "Vibra2",
SN95031_VIB2C5, 1, 0),
/* capture widgets */
SND_SOC_DAPM_PGA("LineIn Enable Left", SN95031_MICAMP1,
7, 0, NULL, 0),
SND_SOC_DAPM_PGA("LineIn Enable Right", SN95031_MICAMP2,
7, 0, NULL, 0),
SND_SOC_DAPM_PGA("MIC1 Enable", SN95031_MICAMP1, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("MIC2 Enable", SN95031_MICAMP2, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("TX1 Enable", SN95031_AUDIOTXEN, 2, 0, NULL, 0),
SND_SOC_DAPM_PGA("TX2 Enable", SN95031_AUDIOTXEN, 3, 0, NULL, 0),
SND_SOC_DAPM_PGA("TX3 Enable", SN95031_AUDIOTXEN, 4, 0, NULL, 0),
SND_SOC_DAPM_PGA("TX4 Enable", SN95031_AUDIOTXEN, 5, 0, NULL, 0),
/* ADC have null stream as they will be turned ON by TX path */
SND_SOC_DAPM_ADC("ADC Left", NULL,
SN95031_ADCCONFIG, 0, 0),
SND_SOC_DAPM_ADC("ADC Right", NULL,
SN95031_ADCCONFIG, 2, 0),
SND_SOC_DAPM_MUX("Mic_InputL Capture Route",
SND_SOC_NOPM, 0, 0, &sn95031_micl_mux_control),
SND_SOC_DAPM_MUX("Mic_InputR Capture Route",
SND_SOC_NOPM, 0, 0, &sn95031_micr_mux_control),
SND_SOC_DAPM_MUX("Txpath1 Capture Route",
SND_SOC_NOPM, 0, 0, &sn95031_input1_mux_control),
SND_SOC_DAPM_MUX("Txpath2 Capture Route",
SND_SOC_NOPM, 0, 0, &sn95031_input2_mux_control),
SND_SOC_DAPM_MUX("Txpath3 Capture Route",
SND_SOC_NOPM, 0, 0, &sn95031_input3_mux_control),
SND_SOC_DAPM_MUX("Txpath4 Capture Route",
SND_SOC_NOPM, 0, 0, &sn95031_input4_mux_control),
};
static const struct snd_soc_dapm_route sn95031_audio_map[] = {
/* headset and earpiece map */
{ "HPOUTL", NULL, "Headset Rail"},
{ "HPOUTR", NULL, "Headset Rail"},
{ "HPOUTL", NULL, "Headset Left Playback" },
{ "HPOUTR", NULL, "Headset Right Playback" },
{ "EPOUT", NULL, "Earpiece Playback" },
{ "Headset Left Playback", NULL, "Headset Left Filter"},
{ "Headset Right Playback", NULL, "Headset Right Filter"},
{ "Earpiece Playback", NULL, "Headset Left Filter"},
{ "Headset Left Filter", NULL, "HSDAC Left"},
{ "Headset Right Filter", NULL, "HSDAC Right"},
/* speaker map */
{ "IHFOUTL", NULL, "Speaker Rail"},
{ "IHFOUTR", NULL, "Speaker Rail"},
{ "IHFOUTL", NULL, "Speaker Left Playback"},
{ "IHFOUTR", NULL, "Speaker Right Playback"},
{ "Speaker Left Playback", NULL, "Speaker Left Filter"},
{ "Speaker Right Playback", NULL, "Speaker Right Filter"},
{ "Speaker Left Filter", NULL, "IHFDAC Left"},
{ "Speaker Right Filter", NULL, "IHFDAC Right"},
/* vibra map */
{ "VIB1OUT", NULL, "Vibra1 Playback"},
{ "Vibra1 Playback", NULL, "Vibra1 DAC"},
{ "VIB2OUT", NULL, "Vibra2 Playback"},
{ "Vibra2 Playback", NULL, "Vibra2 DAC"},
/* lineout */
{ "LINEOUTL", NULL, "Lineout Left Playback"},
{ "LINEOUTR", NULL, "Lineout Right Playback"},
{ "Lineout Left Playback", NULL, "Headset Left Filter"},
{ "Lineout Left Playback", NULL, "Speaker Left Filter"},
{ "Lineout Left Playback", NULL, "Vibra1 DAC"},
{ "Lineout Right Playback", NULL, "Headset Right Filter"},
{ "Lineout Right Playback", NULL, "Speaker Right Filter"},
{ "Lineout Right Playback", NULL, "Vibra2 DAC"},
/* Headset (AMIC1) mic */
{ "AMIC1Bias", NULL, "AMIC1"},
{ "MIC1 Enable", NULL, "AMIC1Bias"},
{ "Mic_InputL Capture Route", "AMIC", "MIC1 Enable"},
/* AMIC2 */
{ "AMIC2Bias", NULL, "AMIC2"},
{ "MIC2 Enable", NULL, "AMIC2Bias"},
{ "Mic_InputR Capture Route", "AMIC", "MIC2 Enable"},
/* Linein */
{ "LineIn Enable Left", NULL, "LINEINL"},
{ "LineIn Enable Right", NULL, "LINEINR"},
{ "Mic_InputL Capture Route", "LineIn", "LineIn Enable Left"},
{ "Mic_InputR Capture Route", "LineIn", "LineIn Enable Right"},
/* ADC connection */
{ "ADC Left", NULL, "Mic_InputL Capture Route"},
{ "ADC Right", NULL, "Mic_InputR Capture Route"},
/*DMIC connections */
{ "DMIC1", NULL, "DMIC12supply"},
{ "DMIC2", NULL, "DMIC12supply"},
{ "DMIC3", NULL, "DMIC34supply"},
{ "DMIC4", NULL, "DMIC34supply"},
{ "DMIC5", NULL, "DMIC56supply"},
{ "DMIC6", NULL, "DMIC56supply"},
{ "DMIC12Bias", NULL, "DMIC1"},
{ "DMIC12Bias", NULL, "DMIC2"},
{ "DMIC34Bias", NULL, "DMIC3"},
{ "DMIC34Bias", NULL, "DMIC4"},
{ "DMIC56Bias", NULL, "DMIC5"},
{ "DMIC56Bias", NULL, "DMIC6"},
/*TX path inputs*/
{ "Txpath1 Capture Route", "ADC Left", "ADC Left"},
{ "Txpath2 Capture Route", "ADC Left", "ADC Left"},
{ "Txpath3 Capture Route", "ADC Left", "ADC Left"},
{ "Txpath4 Capture Route", "ADC Left", "ADC Left"},
{ "Txpath1 Capture Route", "ADC Right", "ADC Right"},
{ "Txpath2 Capture Route", "ADC Right", "ADC Right"},
{ "Txpath3 Capture Route", "ADC Right", "ADC Right"},
{ "Txpath4 Capture Route", "ADC Right", "ADC Right"},
{ "Txpath1 Capture Route", "DMIC1", "DMIC1"},
{ "Txpath2 Capture Route", "DMIC1", "DMIC1"},
{ "Txpath3 Capture Route", "DMIC1", "DMIC1"},
{ "Txpath4 Capture Route", "DMIC1", "DMIC1"},
{ "Txpath1 Capture Route", "DMIC2", "DMIC2"},
{ "Txpath2 Capture Route", "DMIC2", "DMIC2"},
{ "Txpath3 Capture Route", "DMIC2", "DMIC2"},
{ "Txpath4 Capture Route", "DMIC2", "DMIC2"},
{ "Txpath1 Capture Route", "DMIC3", "DMIC3"},
{ "Txpath2 Capture Route", "DMIC3", "DMIC3"},
{ "Txpath3 Capture Route", "DMIC3", "DMIC3"},
{ "Txpath4 Capture Route", "DMIC3", "DMIC3"},
{ "Txpath1 Capture Route", "DMIC4", "DMIC4"},
{ "Txpath2 Capture Route", "DMIC4", "DMIC4"},
{ "Txpath3 Capture Route", "DMIC4", "DMIC4"},
{ "Txpath4 Capture Route", "DMIC4", "DMIC4"},
{ "Txpath1 Capture Route", "DMIC5", "DMIC5"},
{ "Txpath2 Capture Route", "DMIC5", "DMIC5"},
{ "Txpath3 Capture Route", "DMIC5", "DMIC5"},
{ "Txpath4 Capture Route", "DMIC5", "DMIC5"},
{ "Txpath1 Capture Route", "DMIC6", "DMIC6"},
{ "Txpath2 Capture Route", "DMIC6", "DMIC6"},
{ "Txpath3 Capture Route", "DMIC6", "DMIC6"},
{ "Txpath4 Capture Route", "DMIC6", "DMIC6"},
/* tx path */
{ "TX1 Enable", NULL, "Txpath1 Capture Route"},
{ "TX2 Enable", NULL, "Txpath2 Capture Route"},
{ "TX3 Enable", NULL, "Txpath3 Capture Route"},
{ "TX4 Enable", NULL, "Txpath4 Capture Route"},
{ "PCM_Out", NULL, "TX1 Enable"},
{ "PCM_Out", NULL, "TX2 Enable"},
{ "PCM_Out", NULL, "TX3 Enable"},
{ "PCM_Out", NULL, "TX4 Enable"},
};
/* speaker and headset mutes, for audio pops and clicks */
static int sn95031_pcm_hs_mute(struct snd_soc_dai *dai, int mute)
{
snd_soc_update_bits(dai->codec,
SN95031_HSLVOLCTRL, BIT(7), (!mute << 7));
snd_soc_update_bits(dai->codec,
SN95031_HSRVOLCTRL, BIT(7), (!mute << 7));
return 0;
}
static int sn95031_pcm_spkr_mute(struct snd_soc_dai *dai, int mute)
{
snd_soc_update_bits(dai->codec,
SN95031_IHFLVOLCTRL, BIT(7), (!mute << 7));
snd_soc_update_bits(dai->codec,
SN95031_IHFRVOLCTRL, BIT(7), (!mute << 7));
return 0;
}
static int sn95031_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
{
unsigned int format, rate;
switch (params_width(params)) {
case 16:
format = BIT(4)|BIT(5);
break;
case 24:
format = 0;
break;
default:
return -EINVAL;
}
snd_soc_update_bits(dai->codec, SN95031_PCM2C2,
BIT(4)|BIT(5), format);
switch (params_rate(params)) {
case 48000:
pr_debug("RATE_48000\n");
rate = 0;
break;
case 44100:
pr_debug("RATE_44100\n");
rate = BIT(7);
break;
default:
pr_err("ERR rate %d\n", params_rate(params));
return -EINVAL;
}
snd_soc_update_bits(dai->codec, SN95031_PCM1C1, BIT(7), rate);
return 0;
}
/* Codec DAI section */
static const struct snd_soc_dai_ops sn95031_headset_dai_ops = {
.digital_mute = sn95031_pcm_hs_mute,
.hw_params = sn95031_pcm_hw_params,
};
static const struct snd_soc_dai_ops sn95031_speaker_dai_ops = {
.digital_mute = sn95031_pcm_spkr_mute,
.hw_params = sn95031_pcm_hw_params,
};
static const struct snd_soc_dai_ops sn95031_vib1_dai_ops = {
.hw_params = sn95031_pcm_hw_params,
};
static const struct snd_soc_dai_ops sn95031_vib2_dai_ops = {
.hw_params = sn95031_pcm_hw_params,
};
static struct snd_soc_dai_driver sn95031_dais[] = {
{
.name = "SN95031 Headset",
.playback = {
.stream_name = "Headset",
.channels_min = 2,
.channels_max = 2,
.rates = SN95031_RATES,
.formats = SN95031_FORMATS,
},
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 5,
.rates = SN95031_RATES,
.formats = SN95031_FORMATS,
},
.ops = &sn95031_headset_dai_ops,
},
{ .name = "SN95031 Speaker",
.playback = {
.stream_name = "Speaker",
.channels_min = 2,
.channels_max = 2,
.rates = SN95031_RATES,
.formats = SN95031_FORMATS,
},
.ops = &sn95031_speaker_dai_ops,
},
{ .name = "SN95031 Vibra1",
.playback = {
.stream_name = "Vibra1",
.channels_min = 1,
.channels_max = 1,
.rates = SN95031_RATES,
.formats = SN95031_FORMATS,
},
.ops = &sn95031_vib1_dai_ops,
},
{ .name = "SN95031 Vibra2",
.playback = {
.stream_name = "Vibra2",
.channels_min = 1,
.channels_max = 1,
.rates = SN95031_RATES,
.formats = SN95031_FORMATS,
},
.ops = &sn95031_vib2_dai_ops,
},
};
static inline void sn95031_disable_jack_btn(struct snd_soc_codec *codec)
{
snd_soc_write(codec, SN95031_BTNCTRL2, 0x00);
}
static inline void sn95031_enable_jack_btn(struct snd_soc_codec *codec)
{
snd_soc_write(codec, SN95031_BTNCTRL1, 0x77);
snd_soc_write(codec, SN95031_BTNCTRL2, 0x01);
}
static int sn95031_get_headset_state(struct snd_soc_codec *codec,
struct snd_soc_jack *mfld_jack)
{
int micbias = sn95031_get_mic_bias(codec);
int jack_type = snd_soc_jack_get_type(mfld_jack, micbias);
pr_debug("jack type detected = %d\n", jack_type);
if (jack_type == SND_JACK_HEADSET)
sn95031_enable_jack_btn(codec);
return jack_type;
}
void sn95031_jack_detection(struct snd_soc_codec *codec,
struct mfld_jack_data *jack_data)
{
unsigned int status;
unsigned int mask = SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_HEADSET;
pr_debug("interrupt id read in sram = 0x%x\n", jack_data->intr_id);
if (jack_data->intr_id & 0x1) {
pr_debug("short_push detected\n");
status = SND_JACK_HEADSET | SND_JACK_BTN_0;
} else if (jack_data->intr_id & 0x2) {
pr_debug("long_push detected\n");
status = SND_JACK_HEADSET | SND_JACK_BTN_1;
} else if (jack_data->intr_id & 0x4) {
pr_debug("headset or headphones inserted\n");
status = sn95031_get_headset_state(codec, jack_data->mfld_jack);
} else if (jack_data->intr_id & 0x8) {
pr_debug("headset or headphones removed\n");
status = 0;
sn95031_disable_jack_btn(codec);
} else {
pr_err("unidentified interrupt\n");
return;
}
snd_soc_jack_report(jack_data->mfld_jack, status, mask);
/*button pressed and released so we send explicit button release */
if ((status & SND_JACK_BTN_0) | (status & SND_JACK_BTN_1))
snd_soc_jack_report(jack_data->mfld_jack,
SND_JACK_HEADSET, mask);
}
EXPORT_SYMBOL_GPL(sn95031_jack_detection);
/* codec registration */
static int sn95031_codec_probe(struct snd_soc_codec *codec)
{
pr_debug("codec_probe called\n");
/* PCM interface config
* This sets the pcm rx slot conguration to max 6 slots
* for max 4 dais (2 stereo and 2 mono)
*/
snd_soc_write(codec, SN95031_PCM2RXSLOT01, 0x10);
snd_soc_write(codec, SN95031_PCM2RXSLOT23, 0x32);
snd_soc_write(codec, SN95031_PCM2RXSLOT45, 0x54);
snd_soc_write(codec, SN95031_PCM2TXSLOT01, 0x10);
snd_soc_write(codec, SN95031_PCM2TXSLOT23, 0x32);
/* pcm port setting
* This sets the pcm port to slave and clock at 19.2Mhz which
* can support 6slots, sampling rate set per stream in hw-params
*/
snd_soc_write(codec, SN95031_PCM1C1, 0x00);
snd_soc_write(codec, SN95031_PCM2C1, 0x01);
snd_soc_write(codec, SN95031_PCM2C2, 0x0A);
snd_soc_write(codec, SN95031_HSMIXER, BIT(0)|BIT(4));
/* vendor vibra workround, the vibras are muted by
* custom register so unmute them
*/
snd_soc_write(codec, SN95031_SSR5, 0x80);
snd_soc_write(codec, SN95031_SSR6, 0x80);
snd_soc_write(codec, SN95031_VIB1C5, 0x00);
snd_soc_write(codec, SN95031_VIB2C5, 0x00);
/* configure vibras for pcm port */
snd_soc_write(codec, SN95031_VIB1C3, 0x00);
snd_soc_write(codec, SN95031_VIB2C3, 0x00);
/* soft mute ramp time */
snd_soc_write(codec, SN95031_SOFTMUTE, 0x3);
/* fix the initial volume at 1dB,
* default in +9dB,
* 1dB give optimal swing on DAC, amps
*/
snd_soc_write(codec, SN95031_HSLVOLCTRL, 0x08);
snd_soc_write(codec, SN95031_HSRVOLCTRL, 0x08);
snd_soc_write(codec, SN95031_IHFLVOLCTRL, 0x08);
snd_soc_write(codec, SN95031_IHFRVOLCTRL, 0x08);
/* dac mode and lineout workaround */
snd_soc_write(codec, SN95031_SSR2, 0x10);
snd_soc_write(codec, SN95031_SSR3, 0x40);
return 0;
}
static const struct snd_soc_codec_driver sn95031_codec = {
.probe = sn95031_codec_probe,
.set_bias_level = sn95031_set_vaud_bias,
.idle_bias_off = true,
.component_driver = {
.controls = sn95031_snd_controls,
.num_controls = ARRAY_SIZE(sn95031_snd_controls),
.dapm_widgets = sn95031_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(sn95031_dapm_widgets),
.dapm_routes = sn95031_audio_map,
.num_dapm_routes = ARRAY_SIZE(sn95031_audio_map),
},
};
static int sn95031_device_probe(struct platform_device *pdev)
{
struct regmap *regmap;
pr_debug("codec device probe called for %s\n", dev_name(&pdev->dev));
regmap = devm_regmap_init(&pdev->dev, NULL, NULL, &sn95031_regmap);
if (IS_ERR(regmap))
return PTR_ERR(regmap);
return snd_soc_register_codec(&pdev->dev, &sn95031_codec,
sn95031_dais, ARRAY_SIZE(sn95031_dais));
}
static int sn95031_device_remove(struct platform_device *pdev)
{
pr_debug("codec device remove called\n");
snd_soc_unregister_codec(&pdev->dev);
return 0;
}
static struct platform_driver sn95031_codec_driver = {
.driver = {
.name = "sn95031",
},
.probe = sn95031_device_probe,
.remove = sn95031_device_remove,
};
module_platform_driver(sn95031_codec_driver);
MODULE_DESCRIPTION("ASoC TI SN95031 codec driver");
MODULE_AUTHOR("Vinod Koul <vinod.koul@intel.com>");
MODULE_AUTHOR("Harsha Priya <priya.harsha@intel.com>");
MODULE_LICENSE("GPL v2");
MODULE_ALIAS("platform:sn95031");
sound/soc/codecs/sn95031.h deleted 100644 → 0
View file @ 7c310f16
/*
* sn95031.h - TI sn95031 Codec driver
*
* Copyright (C) 2010 Intel Corp
* Author: Vinod Koul <vinod.koul@intel.com>
* Author: Harsha Priya <priya.harsha@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.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
*
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*
*
*/
#ifndef _SN95031_H
#define _SN95031_H
/*register map*/
#define SN95031_VAUD 0xDB
#define SN95031_VHSP 0xDC
#define SN95031_VHSN 0xDD
#define SN95031_VIHF 0xC9
#define SN95031_AUDPLLCTRL 0x240
#define SN95031_DMICBUF0123 0x241
#define SN95031_DMICBUF45 0x242
#define SN95031_DMICGPO 0x244
#define SN95031_DMICMUX 0x245
#define SN95031_DMICLK 0x246
#define SN95031_MICBIAS 0x247
#define SN95031_ADCCONFIG 0x248
#define SN95031_MICAMP1 0x249
#define SN95031_MICAMP2 0x24A
#define SN95031_NOISEMUX 0x24B
#define SN95031_AUDIOMUX12 0x24C
#define SN95031_AUDIOMUX34 0x24D
#define SN95031_AUDIOSINC 0x24E
#define SN95031_AUDIOTXEN 0x24F
#define SN95031_HSEPRXCTRL 0x250
#define SN95031_IHFRXCTRL 0x251
#define SN95031_HSMIXER 0x256
#define SN95031_DACCONFIG 0x257
#define SN95031_SOFTMUTE 0x258
#define SN95031_HSLVOLCTRL 0x259
#define SN95031_HSRVOLCTRL 0x25A
#define SN95031_IHFLVOLCTRL 0x25B
#define SN95031_IHFRVOLCTRL 0x25C
#define SN95031_DRIVEREN 0x25D
#define SN95031_LOCTL 0x25E
#define SN95031_VIB1C1 0x25F
#define SN95031_VIB1C2 0x260
#define SN95031_VIB1C3 0x261
#define SN95031_VIB1SPIPCM1 0x262
#define SN95031_VIB1SPIPCM2 0x263
#define SN95031_VIB1C5 0x264
#define SN95031_VIB2C1 0x265
#define SN95031_VIB2C2 0x266
#define SN95031_VIB2C3 0x267
#define SN95031_VIB2SPIPCM1 0x268
#define SN95031_VIB2SPIPCM2 0x269
#define SN95031_VIB2C5 0x26A
#define SN95031_BTNCTRL1 0x26B
#define SN95031_BTNCTRL2 0x26C
#define SN95031_PCM1TXSLOT01 0x26D
#define SN95031_PCM1TXSLOT23 0x26E
#define SN95031_PCM1TXSLOT45 0x26F
#define SN95031_PCM1RXSLOT0_3 0x270
#define SN95031_PCM1RXSLOT45 0x271
#define SN95031_PCM2TXSLOT01 0x272
#define SN95031_PCM2TXSLOT23 0x273
#define SN95031_PCM2TXSLOT45 0x274
#define SN95031_PCM2RXSLOT01 0x275
#define SN95031_PCM2RXSLOT23 0x276
#define SN95031_PCM2RXSLOT45 0x277
#define SN95031_PCM1C1 0x278
#define SN95031_PCM1C2 0x279
#define SN95031_PCM1C3 0x27A
#define SN95031_PCM2C1 0x27B
#define SN95031_PCM2C2 0x27C
/*end codec register defn*/
/*vendor defn these are not part of avp*/
#define SN95031_SSR2 0x381
#define SN95031_SSR3 0x382
#define SN95031_SSR5 0x384
#define SN95031_SSR6 0x385
/* ADC registers */
#define SN95031_ADC1CNTL1 0x1C0
#define SN95031_ADC_ENBL 0x10
#define SN95031_ADC_START 0x08
#define SN95031_ADC1CNTL3 0x1C2
#define SN95031_ADCTHERM_ENBL 0x04
#define SN95031_ADCRRDATA_ENBL 0x05
#define SN95031_STOPBIT_MASK 16
#define SN95031_ADCTHERM_MASK 4
#define SN95031_ADC_CHANLS_MAX 15 /* Number of ADC channels */
#define SN95031_ADC_LOOP_MAX (SN95031_ADC_CHANLS_MAX - 1)
#define SN95031_ADC_NO_LOOP 0x07
#define SN95031_AUDIO_GPIO_CTRL 0x070
/* ADC channel code values */
#define SN95031_AUDIO_DETECT_CODE 0x06
/* ADC base addresses */
#define SN95031_ADC_CHNL_START_ADDR 0x1C5 /* increments by 1 */
#define SN95031_ADC_DATA_START_ADDR 0x1D4 /* increments by 2 */
/* multipier to convert to mV */
#define SN95031_ADC_ONE_LSB_MULTIPLIER 2346
struct mfld_jack_data {
int intr_id;
int micbias_vol;
struct snd_soc_jack *mfld_jack;
};
extern void sn95031_jack_detection(struct snd_soc_codec *codec,
struct mfld_jack_data *jack_data);
#endif
sound/soc/intel/Kconfig
View file @ 3e4555ab
config SND_SOC_INTEL_SST_TOPLEVEL
bool "Intel ASoC SST drivers"
default y
depends on X86 || COMPILE_TEST
select SND_SOC_INTEL_MACH
help
Intel ASoC SST Platform Drivers. If you have a Intel machine that
has an audio controller with a DSP and I2S or DMIC port, then
enable this option by saying Y
Note that the answer to this question doesn't directly affect the
kernel: saying N will just cause the configurator to skip all
the questions about Intel SST drivers.
if SND_SOC_INTEL_SST_TOPLEVEL
config SND_SST_IPC
tristate
# This option controls the IPC core for HiFi2 platforms
config SND_SST_IPC_PCI
tristate
select SND_SST_IPC
# This option controls the PCI-based IPC for HiFi2 platforms
# (Medfield, Merrifield).
config SND_SST_IPC_ACPI
tristate
select SND_SST_IPC
select SND_SOC_INTEL_SST
select IOSF_MBI
# This option controls the ACPI-based IPC for HiFi2 platforms
# (Baytrail, Cherrytrail)
config SND_SOC_INTEL_COMMON
config SND_SOC_INTEL_SST_ACPI
tristate
# This option controls ACPI-based probing on
# Haswell/Broadwell/Baytrail legacy and will be set
# when these platforms are enabled
config SND_SOC_INTEL_SST
tristate
select SND_SOC_INTEL_SST_ACPI if ACPI
config SND_SOC_INTEL_SST_FIRMWARE
tristate
select DW_DMAC_CORE
config SND_SOC_INTEL_SST_ACPI
tristate
config SND_SOC_ACPI_INTEL_MATCH
tristate
select SND_SOC_ACPI if ACPI
config SND_SOC_INTEL_SST_TOPLEVEL
tristate "Intel ASoC SST drivers"
depends on X86 || COMPILE_TEST
select SND_SOC_INTEL_MACH
select SND_SOC_INTEL_COMMON
help
Intel ASoC Audio Drivers. If you have a Intel machine that
has audio controller with a DSP and I2S or DMIC port, then
enable this option by saying Y or M
If unsure select "N".
# This option controls firmware download on
# Haswell/Broadwell/Baytrail legacy and will be set
# when these platforms are enabled
config SND_SOC_INTEL_HASWELL
tristate "Intel ASoC SST driver for Haswell/Broadwell"
depends on SND_SOC_INTEL_SST_TOPLEVEL && SND_DMA_SGBUF
depends on DMADEVICES
tristate "Haswell/Broadwell Platforms"
depends on SND_DMA_SGBUF
depends on DMADEVICES && ACPI
select SND_SOC_INTEL_SST
select SND_SOC_INTEL_SST_ACPI
select SND_SOC_INTEL_SST_FIRMWARE
select SND_SOC_ACPI_INTEL_MATCH
help
If you have a Intel Haswell or Broadwell platform connected to
an I2S codec, then enable this option by saying Y or m. This is
typically used for Chromebooks. This is a recommended option.
config SND_SOC_INTEL_BAYTRAIL
tristate "Intel ASoC SST driver for Baytrail (legacy)"
depends on SND_SOC_INTEL_SST_TOPLEVEL
depends on DMADEVICES
tristate "Baytrail (legacy) Platforms"
depends on DMADEVICES && ACPI
select SND_SOC_INTEL_SST
select SND_SOC_INTEL_SST_ACPI
select SND_SOC_INTEL_SST_FIRMWARE
select SND_SOC_ACPI_INTEL_MATCH
help
If you have a Intel Baytrail platform connected to an I2S codec,
then enable this option by saying Y or m. This was typically used
for Baytrail Chromebooks but this option is now deprecated and is
not recommended, use SND_SST_ATOM_HIFI2_PLATFORM instead.
config SND_SST_ATOM_HIFI2_PLATFORM_PCI
tristate "PCI HiFi2 (Medfield, Merrifield) Platforms"
depends on X86 && PCI
select SND_SST_IPC_PCI
select SND_SOC_COMPRESS
select SND_SOC_INTEL_COMMON
help
If you have a Intel Medfield or Merrifield/Edison platform, then
enable this option by saying Y or m. Distros will typically not
enable this option: Medfield devices are not available to
developers and while Merrifield/Edison can run a mainline kernel with
limited functionality it will require a firmware file which
is not in the standard firmware tree
config SND_SST_ATOM_HIFI2_PLATFORM
tristate "Intel ASoC SST driver for HiFi2 platforms (*field, *trail)"
depends on SND_SOC_INTEL_SST_TOPLEVEL && X86
tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms"
depends on X86 && ACPI
select SND_SST_IPC_ACPI
select SND_SOC_COMPRESS
select SND_SOC_ACPI_INTEL_MATCH
select IOSF_MBI
help
If you have a Intel Baytrail or Cherrytrail platform with an I2S
codec, then enable this option by saying Y or m. This is a
recommended option
config SND_SOC_INTEL_SKYLAKE
tristate "Intel ASoC SST driver for SKL/BXT/KBL/GLK/CNL"
depends on SND_SOC_INTEL_SST_TOPLEVEL && PCI && ACPI
tristate "SKL/BXT/KBL/GLK/CNL... Platforms"
depends on PCI && ACPI
select SND_HDA_EXT_CORE
select SND_HDA_DSP_LOADER
select SND_SOC_TOPOLOGY
select SND_SOC_INTEL_SST
select SND_SOC_ACPI_INTEL_MATCH
help
If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/
GeminiLake or CannonLake platform with the DSP enabled in the BIOS
then enable this option by saying Y or m.
config SND_SOC_ACPI_INTEL_MATCH
tristate
select SND_SOC_ACPI if ACPI
# this option controls the compilation of ACPI matching tables and
# helpers and is not meant to be selected by the user.
endif ## SND_SOC_INTEL_SST_TOPLEVEL
# ASoC codec drivers
source "sound/soc/intel/boards/Kconfig"
sound/soc/intel/Makefile
View file @ 3e4555ab
# SPDX-License-Identifier: GPL-2.0
# Core support
obj-$(CONFIG_SND_SOC_INTEL_COMMON) += common/
obj-$(CONFIG_SND_SOC) += common/
# Platform Support
obj-$(CONFIG_SND_SOC_INTEL_HASWELL) += haswell/
......
sound/soc/intel/atom/sst/sst_acpi.c
View file @ 3e4555ab
......@@ -236,6 +236,9 @@ static int sst_platform_get_resources(struct intel_sst_drv *ctx)
/* Find the IRQ */
ctx->irq_num = platform_get_irq(pdev,
ctx->pdata->res_info->acpi_ipc_irq_index);
if (ctx->irq_num <= 0)
return ctx->irq_num < 0 ? ctx->irq_num : -EIO;
return 0;
}
......
sound/soc/intel/atom/sst/sst_stream.c
View file @ 3e4555ab
......@@ -220,10 +220,10 @@ int sst_send_byte_stream_mrfld(struct intel_sst_drv *sst_drv_ctx,
sst_free_block(sst_drv_ctx, block);
out:
test_and_clear_bit(pvt_id, &sst_drv_ctx->pvt_id);
return 0;
return ret;
}
/*
/**
* sst_pause_stream - Send msg for a pausing stream
* @str_id: stream ID
*
......@@ -261,7 +261,7 @@ int sst_pause_stream(struct intel_sst_drv *sst_drv_ctx, int str_id)
}
} else {
retval = -EBADRQC;
dev_dbg(sst_drv_ctx->dev, "SST DBG:BADRQC for stream\n ");
dev_dbg(sst_drv_ctx->dev, "SST DBG:BADRQC for stream\n");
}
return retval;
......@@ -284,7 +284,7 @@ int sst_resume_stream(struct intel_sst_drv *sst_drv_ctx, int str_id)
if (!str_info)
return -EINVAL;
if (str_info->status == STREAM_RUNNING)
return 0;
return 0;
if (str_info->status == STREAM_PAUSED) {
retval = sst_prepare_and_post_msg(sst_drv_ctx, str_info->task_id,
IPC_CMD, IPC_IA_RESUME_STREAM_MRFLD,
......
sound/soc/intel/boards/Kconfig
View file @ 3e4555ab
config SND_SOC_INTEL_MACH
tristate "Intel Audio machine drivers"
menuconfig SND_SOC_INTEL_MACH
bool "Intel Machine drivers"
depends on SND_SOC_INTEL_SST_TOPLEVEL
select SND_SOC_ACPI_INTEL_MATCH if ACPI
help
Intel ASoC Machine Drivers. If you have a Intel machine that
has an audio controller with a DSP and I2S or DMIC port, then
enable this option by saying Y
Note that the answer to this question doesn't directly affect the
kernel: saying N will just cause the configurator to skip all
the questions about Intel ASoC machine drivers.
if SND_SOC_INTEL_MACH
config SND_MFLD_MACHINE
tristate "SOC Machine Audio driver for Intel Medfield MID platform"
depends on INTEL_SCU_IPC
select SND_SOC_SN95031
depends on SND_SST_ATOM_HIFI2_PLATFORM
select SND_SST_IPC_PCI
help
This adds support for ASoC machine driver for Intel(R) MID Medfield platform
used as alsa device in audio substem in Intel(R) MID devices
Say Y if you have such a device.
If unsure select "N".
if SND_SOC_INTEL_HASWELL
config SND_SOC_INTEL_HASWELL_MACH
tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint"
tristate "Haswell Lynxpoint"
depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM
depends on SND_SOC_INTEL_HASWELL
select SND_SOC_RT5640
help
This adds support for the Lynxpoint Audio DSP on Intel(R) Haswell
Ultrabook platforms.
Say Y if you have such a device.
Ultrabook platforms. This is a recommended option.
Say Y or m if you have such a device.
If unsure select "N".
config SND_SOC_INTEL_BDW_RT5677_MACH
tristate "ASoC Audio driver for Intel Broadwell with RT5677 codec"
depends on X86_INTEL_LPSS && GPIOLIB && I2C
depends on SND_SOC_INTEL_HASWELL
tristate "Broadwell with RT5677 codec"
depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM && GPIOLIB
select SND_SOC_RT5677
help
This adds support for Intel Broadwell platform based boards with
the RT5677 audio codec.
the RT5677 audio codec. This is a recommended option.
Say Y or m if you have such a device.
If unsure select "N".
config SND_SOC_INTEL_BROADWELL_MACH
tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint"
tristate "Broadwell Wildcatpoint"
depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM
depends on SND_SOC_INTEL_HASWELL
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.
Say Y or m if you have such a device. This is a recommended option.
If unsure select "N".
endif ## SND_SOC_INTEL_HASWELL
if SND_SOC_INTEL_BAYTRAIL
config SND_SOC_INTEL_BYT_MAX98090_MACH
tristate "ASoC Audio driver for Intel Baytrail with MAX98090 codec"
tristate "Baytrail with MAX98090 codec"
depends on X86_INTEL_LPSS && I2C
depends on SND_SST_IPC_ACPI = n
depends on SND_SOC_INTEL_BAYTRAIL
select SND_SOC_MAX98090
help
This adds audio driver for Intel Baytrail platform based boards
with the MAX98090 audio codec.
with the MAX98090 audio codec. This driver is deprecated, use
SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH instead for better
functionality.
config SND_SOC_INTEL_BYT_RT5640_MACH
tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec"
tristate "Baytrail with RT5640 codec"
depends on X86_INTEL_LPSS && I2C
depends on SND_SST_IPC_ACPI = n
depends on SND_SOC_INTEL_BAYTRAIL
select SND_SOC_RT5640
help
This adds audio driver for Intel Baytrail platform based boards
with the RT5640 audio codec. This driver is deprecated, use
SND_SOC_INTEL_BYTCR_RT5640_MACH instead for better functionality.
endif ## SND_SOC_INTEL_BAYTRAIL
if SND_SST_ATOM_HIFI2_PLATFORM
config SND_SOC_INTEL_BYTCR_RT5640_MACH
tristate "ASoC Audio driver for Intel Baytrail and Baytrail-CR with RT5640 codec"
depends on X86 && I2C && ACPI
tristate "Baytrail and Baytrail-CR with RT5640 codec"
depends on X86_INTEL_LPSS && I2C && ACPI
select SND_SOC_ACPI
select SND_SOC_RT5640
depends on SND_SST_ATOM_HIFI2_PLATFORM
select SND_SST_IPC_ACPI
help
This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR
platforms with RT5640 audio codec.
Say Y if you have such a device.
If unsure select "N".
This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR
platforms with RT5640 audio codec.
Say Y or m if you have such a device. This is a recommended option.
If unsure select "N".
config SND_SOC_INTEL_BYTCR_RT5651_MACH
tristate "ASoC Audio driver for Intel Baytrail and Baytrail-CR with RT5651 codec"
depends on X86 && I2C && ACPI
tristate "Baytrail and Baytrail-CR with RT5651 codec"
depends on X86_INTEL_LPSS && I2C && ACPI
select SND_SOC_ACPI
select SND_SOC_RT5651
depends on SND_SST_ATOM_HIFI2_PLATFORM
select SND_SST_IPC_ACPI
help
This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR
platforms with RT5651 audio codec.
Say Y if you have such a device.
If unsure select "N".
This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR
platforms with RT5651 audio codec.
Say Y or m if you have such a device. This is a recommended option.
If unsure select "N".
config SND_SOC_INTEL_CHT_BSW_RT5672_MACH
tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec"
tristate "Cherrytrail & Braswell with RT5672 codec"
depends on X86_INTEL_LPSS && I2C && ACPI
select SND_SOC_RT5670
depends on SND_SST_ATOM_HIFI2_PLATFORM
select SND_SST_IPC_ACPI
select SND_SOC_ACPI
select SND_SOC_RT5670
help
This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell
platforms with RT5672 audio codec.
Say Y if you have such a device.
Say Y or m if you have such a device. This is a recommended option.
If unsure select "N".
config SND_SOC_INTEL_CHT_BSW_RT5645_MACH
tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5645/5650 codec"
tristate "Cherrytrail & Braswell with RT5645/5650 codec"
depends on X86_INTEL_LPSS && I2C && ACPI
select SND_SOC_ACPI
select SND_SOC_RT5645
depends on SND_SST_ATOM_HIFI2_PLATFORM
select SND_SST_IPC_ACPI
help
This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell
platforms with RT5645/5650 audio codec.
Say Y or m if you have such a device. This is a recommended option.
If unsure select "N".
config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH
tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with MAX98090 & TI codec"
tristate "Cherrytrail & Braswell with MAX98090 & TI codec"
depends on X86_INTEL_LPSS && I2C && ACPI
select SND_SOC_MAX98090
select SND_SOC_TS3A227E
depends on SND_SST_ATOM_HIFI2_PLATFORM
select SND_SST_IPC_ACPI
help
This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell
platforms with MAX98090 audio codec it also can support TI jack chip as aux device.
Say Y or m if you have such a device. This is a recommended option.
If unsure select "N".
config SND_SOC_INTEL_BYT_CHT_DA7213_MACH
tristate "ASoC Audio driver for Intel Baytrail & Cherrytrail with DA7212/7213 codec"
tristate "Baytrail & Cherrytrail with DA7212/7213 codec"
depends on X86_INTEL_LPSS && I2C && ACPI
select SND_SOC_ACPI
select SND_SOC_DA7213
depends on SND_SST_ATOM_HIFI2_PLATFORM
select SND_SST_IPC_ACPI
help
This adds support for ASoC machine driver for Intel(R) Baytrail & CherryTrail
platforms with DA7212/7213 audio codec.
Say Y or m if you have such a device. This is a recommended option.
If unsure select "N".
config SND_SOC_INTEL_BYT_CHT_ES8316_MACH
tristate "ASoC Audio driver for Intel Baytrail & Cherrytrail with ES8316 codec"
tristate "Baytrail & Cherrytrail with ES8316 codec"
depends on X86_INTEL_LPSS && I2C && ACPI
select SND_SOC_ES8316
depends on SND_SST_ATOM_HIFI2_PLATFORM
select SND_SST_IPC_ACPI
help
This adds support for ASoC machine driver for Intel(R) Baytrail &
Cherrytrail platforms with ES8316 audio codec.
Say Y or m if you have such a device. This is a recommended option.
If unsure select "N".
config SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH
tristate "ASoC Audio driver for Intel Baytrail & Cherrytrail platform with no codec (MinnowBoard MAX, Up)"
tristate "Baytrail & Cherrytrail platform with no codec (MinnowBoard MAX, Up)"
depends on X86_INTEL_LPSS && I2C && ACPI
depends on SND_SST_ATOM_HIFI2_PLATFORM
select SND_SST_IPC_ACPI
help
This adds support for ASoC machine driver for the MinnowBoard Max or
Up boards and provides access to I2S signals on the Low-Speed
connector
connector. This is not a recommended option outside of these cases.
It is not intended to be enabled by distros by default.
Say Y or m if you have such a device.
If unsure select "N".
endif ## SND_SST_ATOM_HIFI2_PLATFORM
if SND_SOC_INTEL_SKYLAKE
config SND_SOC_INTEL_SKL_RT286_MACH
tristate "ASoC Audio driver for SKL with RT286 I2S mode"
depends on X86 && ACPI && I2C
depends on SND_SOC_INTEL_SKYLAKE
tristate "SKL with RT286 I2S mode"
depends on MFD_INTEL_LPSS && I2C && ACPI
select SND_SOC_RT286
select SND_SOC_DMIC
select SND_SOC_HDAC_HDMI
help
This adds support for ASoC machine driver for Skylake platforms
with RT286 I2S audio codec.
Say Y if you have such a device.
Say Y or m if you have such a device.
If unsure select "N".
config SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH
tristate "ASoC Audio driver for SKL with NAU88L25 and SSM4567 in I2S Mode"
depends on X86_INTEL_LPSS && I2C
depends on SND_SOC_INTEL_SKYLAKE
tristate "SKL with NAU88L25 and SSM4567 in I2S Mode"
depends on MFD_INTEL_LPSS && I2C && ACPI
select SND_SOC_NAU8825
select SND_SOC_SSM4567
select SND_SOC_DMIC
......@@ -185,13 +184,12 @@ config SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH
help
This adds support for ASoC Onboard Codec I2S machine driver. This will
create an alsa sound card for NAU88L25 + SSM4567.
Say Y if you have such a device.
Say Y or m if you have such a device. This is a recommended option.
If unsure select "N".
config SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH
tristate "ASoC Audio driver for SKL with NAU88L25 and MAX98357A in I2S Mode"
depends on X86_INTEL_LPSS && I2C
depends on SND_SOC_INTEL_SKYLAKE
tristate "SKL with NAU88L25 and MAX98357A in I2S Mode"
depends on MFD_INTEL_LPSS && I2C && ACPI
select SND_SOC_NAU8825
select SND_SOC_MAX98357A
select SND_SOC_DMIC
......@@ -199,13 +197,12 @@ config SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH
help
This adds support for ASoC Onboard Codec I2S machine driver. This will
create an alsa sound card for NAU88L25 + MAX98357A.
Say Y if you have such a device.
Say Y or m if you have such a device. This is a recommended option.
If unsure select "N".
config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH
tristate "ASoC Audio driver for Broxton with DA7219 and MAX98357A in I2S Mode"
depends on X86 && ACPI && I2C
depends on SND_SOC_INTEL_SKYLAKE
tristate "Broxton with DA7219 and MAX98357A in I2S Mode"
depends on MFD_INTEL_LPSS && I2C && ACPI
select SND_SOC_DA7219
select SND_SOC_MAX98357A
select SND_SOC_DMIC
......@@ -214,13 +211,12 @@ config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH
help
This adds support for ASoC machine driver for Broxton-P platforms
with DA7219 + MAX98357A I2S audio codec.
Say Y if you have such a device.
Say Y or m if you have such a device. This is a recommended option.
If unsure select "N".
config SND_SOC_INTEL_BXT_RT298_MACH
tristate "ASoC Audio driver for Broxton with RT298 I2S mode"
depends on X86 && ACPI && I2C
depends on SND_SOC_INTEL_SKYLAKE
tristate "Broxton with RT298 I2S mode"
depends on MFD_INTEL_LPSS && I2C && ACPI
select SND_SOC_RT298
select SND_SOC_DMIC
select SND_SOC_HDAC_HDMI
......@@ -228,14 +224,12 @@ config SND_SOC_INTEL_BXT_RT298_MACH
help
This adds support for ASoC machine driver for Broxton platforms
with RT286 I2S audio codec.
Say Y if you have such a device.
Say Y or m if you have such a device. This is a recommended option.
If unsure select "N".
config SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH
tristate "ASoC Audio driver for KBL with RT5663 and MAX98927 in I2S Mode"
depends on X86_INTEL_LPSS && I2C
select SND_SOC_INTEL_SST
depends on SND_SOC_INTEL_SKYLAKE
tristate "KBL with RT5663 and MAX98927 in I2S Mode"
depends on MFD_INTEL_LPSS && I2C && ACPI
select SND_SOC_RT5663
select SND_SOC_MAX98927
select SND_SOC_DMIC
......@@ -243,14 +237,13 @@ config SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH
help
This adds support for ASoC Onboard Codec I2S machine driver. This will
create an alsa sound card for RT5663 + MAX98927.
Say Y if you have such a device.
Say Y or m if you have such a device. This is a recommended option.
If unsure select "N".
config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH
tristate "ASoC Audio driver for KBL with RT5663, RT5514 and MAX98927 in I2S Mode"
depends on X86_INTEL_LPSS && I2C && SPI
select SND_SOC_INTEL_SST
depends on SND_SOC_INTEL_SKYLAKE
tristate "KBL with RT5663, RT5514 and MAX98927 in I2S Mode"
depends on MFD_INTEL_LPSS && I2C && ACPI
depends on SPI
select SND_SOC_RT5663
select SND_SOC_RT5514
select SND_SOC_RT5514_SPI
......@@ -259,7 +252,8 @@ config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH
help
This adds support for ASoC Onboard Codec I2S machine driver. This will
create an alsa sound card for RT5663 + RT5514 + MAX98927.
Say Y if you have such a device.
Say Y or m if you have such a device. This is a recommended option.
If unsure select "N".
endif ## SND_SOC_INTEL_SKYLAKE
endif
endif ## SND_SOC_INTEL_MACH
sound/soc/intel/boards/bytcr_rt5651.c
View file @ 3e4555ab
......@@ -38,6 +38,8 @@ enum {
BYT_RT5651_DMIC_MAP,
BYT_RT5651_IN1_MAP,
BYT_RT5651_IN2_MAP,
BYT_RT5651_IN1_IN2_MAP,
BYT_RT5651_IN3_MAP,
};
#define BYT_RT5651_MAP(quirk) ((quirk) & GENMASK(7, 0))
......@@ -62,6 +64,8 @@ static void log_quirks(struct device *dev)
dev_info(dev, "quirk IN1_MAP enabled");
if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN2_MAP)
dev_info(dev, "quirk IN2_MAP enabled");
if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN3_MAP)
dev_info(dev, "quirk IN3_MAP enabled");
if (byt_rt5651_quirk & BYT_RT5651_DMIC_EN)
dev_info(dev, "quirk DMIC enabled");
if (byt_rt5651_quirk & BYT_RT5651_MCLK_EN)
......@@ -127,6 +131,7 @@ static const struct snd_soc_dapm_widget byt_rt5651_widgets[] = {
SND_SOC_DAPM_MIC("Headset Mic", NULL),
SND_SOC_DAPM_MIC("Internal Mic", NULL),
SND_SOC_DAPM_SPK("Speaker", NULL),
SND_SOC_DAPM_LINE("Line In", NULL),
SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
platform_clock_control, SND_SOC_DAPM_PRE_PMU |
SND_SOC_DAPM_POST_PMD),
......@@ -138,6 +143,7 @@ static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = {
{"Headset Mic", NULL, "Platform Clock"},
{"Internal Mic", NULL, "Platform Clock"},
{"Speaker", NULL, "Platform Clock"},
{"Line In", NULL, "Platform Clock"},
{"AIF1 Playback", NULL, "ssp2 Tx"},
{"ssp2 Tx", NULL, "codec_out0"},
......@@ -151,6 +157,9 @@ static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = {
{"Headphone", NULL, "HPOR"},
{"Speaker", NULL, "LOUTL"},
{"Speaker", NULL, "LOUTR"},
{"IN2P", NULL, "Line In"},
{"IN2N", NULL, "Line In"},
};
static const struct snd_soc_dapm_route byt_rt5651_intmic_dmic_map[] = {
......@@ -171,11 +180,25 @@ static const struct snd_soc_dapm_route byt_rt5651_intmic_in2_map[] = {
{"IN2P", NULL, "Internal Mic"},
};
static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_in2_map[] = {
{"Internal Mic", NULL, "micbias1"},
{"IN1P", NULL, "Internal Mic"},
{"IN2P", NULL, "Internal Mic"},
{"IN3P", NULL, "Headset Mic"},
};
static const struct snd_soc_dapm_route byt_rt5651_intmic_in3_map[] = {
{"Internal Mic", NULL, "micbias1"},
{"IN3P", NULL, "Headset Mic"},
{"IN1P", NULL, "Internal Mic"},
};
static const struct snd_kcontrol_new byt_rt5651_controls[] = {
SOC_DAPM_PIN_SWITCH("Headphone"),
SOC_DAPM_PIN_SWITCH("Headset Mic"),
SOC_DAPM_PIN_SWITCH("Internal Mic"),
SOC_DAPM_PIN_SWITCH("Speaker"),
SOC_DAPM_PIN_SWITCH("Line In"),
};
static struct snd_soc_jack_pin bytcr_jack_pins[] = {
......@@ -247,8 +270,16 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = {
DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"),
DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max B3 PLATFORM"),
},
.driver_data = (void *)(BYT_RT5651_DMIC_MAP |
BYT_RT5651_DMIC_EN),
.driver_data = (void *)(BYT_RT5651_IN3_MAP),
},
{
.callback = byt_rt5651_quirk_cb,
.matches = {
DMI_MATCH(DMI_SYS_VENDOR, "ADI"),
DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Turbot"),
},
.driver_data = (void *)(BYT_RT5651_MCLK_EN |
BYT_RT5651_IN3_MAP),
},
{
.callback = byt_rt5651_quirk_cb,
......@@ -256,7 +287,8 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = {
DMI_MATCH(DMI_SYS_VENDOR, "KIANO"),
DMI_MATCH(DMI_PRODUCT_NAME, "KIANO SlimNote 14.2"),
},
.driver_data = (void *)(BYT_RT5651_IN2_MAP),
.driver_data = (void *)(BYT_RT5651_MCLK_EN |
BYT_RT5651_IN1_IN2_MAP),
},
{}
};
......@@ -281,6 +313,14 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime)
custom_map = byt_rt5651_intmic_in2_map;
num_routes = ARRAY_SIZE(byt_rt5651_intmic_in2_map);
break;
case BYT_RT5651_IN1_IN2_MAP:
custom_map = byt_rt5651_intmic_in1_in2_map;
num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_in2_map);
break;
case BYT_RT5651_IN3_MAP:
custom_map = byt_rt5651_intmic_in3_map;
num_routes = ARRAY_SIZE(byt_rt5651_intmic_in3_map);
break;
default:
custom_map = byt_rt5651_intmic_dmic_map;
num_routes = ARRAY_SIZE(byt_rt5651_intmic_dmic_map);
......
sound/soc/intel/boards/haswell.c
View file @ 3e4555ab
......@@ -76,7 +76,7 @@ static int haswell_rt5640_hw_params(struct snd_pcm_substream *substream,
}
/* set correct codec filter for DAI format and clock config */
snd_soc_update_bits(rtd->codec, 0x83, 0xffff, 0x8000);
snd_soc_component_update_bits(codec_dai->component, 0x83, 0xffff, 0x8000);
return ret;
}
......
sound/soc/intel/boards/kbl_rt5663_max98927.c
View file @ 3e4555ab
......@@ -225,7 +225,7 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd)
}
jack = &ctx->kabylake_headset;
snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA);
snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND);
snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP);
snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN);
......
sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
View file @ 3e4555ab
......@@ -195,7 +195,7 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd)
}
jack = &ctx->kabylake_headset;
snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA);
snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND);
snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP);
snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN);
......
sound/soc/intel/boards/mfld_machine.c deleted 100644 → 0
View file @ 7c310f16
/*
* mfld_machine.c - ASoc Machine driver for Intel Medfield MID platform
*
* Copyright (C) 2010 Intel Corp
* Author: Vinod Koul <vinod.koul@intel.com>
* Author: Harsha Priya <priya.harsha@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.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
*
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*/
#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt
#include <linux/init.h>
#include <linux/device.h>
#include <linux/slab.h>
#include <linux/io.h>
#include <linux/module.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/jack.h>
#include "../codecs/sn95031.h"
#define MID_MONO 1
#define MID_STEREO 2
#define MID_MAX_CAP 5
#define MFLD_JACK_INSERT 0x04
enum soc_mic_bias_zones {
MFLD_MV_START = 0,
/* mic bias volutage range for Headphones*/
MFLD_MV_HP = 400,
/* mic bias volutage range for American Headset*/
MFLD_MV_AM_HS = 650,
/* mic bias volutage range for Headset*/
MFLD_MV_HS = 2000,
MFLD_MV_UNDEFINED,
};
static unsigned int hs_switch;
static unsigned int lo_dac;
static struct snd_soc_codec *mfld_codec;
struct mfld_mc_private {
void __iomem *int_base;
u8 interrupt_status;
};
struct snd_soc_jack mfld_jack;
/*Headset jack detection DAPM pins */
static struct snd_soc_jack_pin mfld_jack_pins[] = {
{
.pin = "Headphones",
.mask = SND_JACK_HEADPHONE,
},
{
.pin = "AMIC1",
.mask = SND_JACK_MICROPHONE,
},
};
/* jack detection voltage zones */
static struct snd_soc_jack_zone mfld_zones[] = {
{MFLD_MV_START, MFLD_MV_AM_HS, SND_JACK_HEADPHONE},
{MFLD_MV_AM_HS, MFLD_MV_HS, SND_JACK_HEADSET},
};
/* sound card controls */
static const char * const headset_switch_text[] = {"Earpiece", "Headset"};
static const char * const lo_text[] = {"Vibra", "Headset", "IHF", "None"};
static const struct soc_enum headset_enum =
SOC_ENUM_SINGLE_EXT(2, headset_switch_text);
static const struct soc_enum lo_enum =
SOC_ENUM_SINGLE_EXT(4, lo_text);
static int headset_get_switch(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.enumerated.item[0] = hs_switch;
return 0;
}
static int headset_set_switch(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
struct snd_soc_dapm_context *dapm = &card->dapm;
if (ucontrol->value.enumerated.item[0] == hs_switch)
return 0;
snd_soc_dapm_mutex_lock(dapm);
if (ucontrol->value.enumerated.item[0]) {
pr_debug("hs_set HS path\n");
snd_soc_dapm_enable_pin_unlocked(dapm, "Headphones");
snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT");
} else {
pr_debug("hs_set EP path\n");
snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones");
snd_soc_dapm_enable_pin_unlocked(dapm, "EPOUT");
}
snd_soc_dapm_sync_unlocked(dapm);
snd_soc_dapm_mutex_unlock(dapm);
hs_switch = ucontrol->value.enumerated.item[0];
return 0;
}
static void lo_enable_out_pins(struct snd_soc_dapm_context *dapm)
{
snd_soc_dapm_enable_pin_unlocked(dapm, "IHFOUTL");
snd_soc_dapm_enable_pin_unlocked(dapm, "IHFOUTR");
snd_soc_dapm_enable_pin_unlocked(dapm, "LINEOUTL");
snd_soc_dapm_enable_pin_unlocked(dapm, "LINEOUTR");
snd_soc_dapm_enable_pin_unlocked(dapm, "VIB1OUT");
snd_soc_dapm_enable_pin_unlocked(dapm, "VIB2OUT");
if (hs_switch) {
snd_soc_dapm_enable_pin_unlocked(dapm, "Headphones");
snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT");
} else {
snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones");
snd_soc_dapm_enable_pin_unlocked(dapm, "EPOUT");
}
}
static int lo_get_switch(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.enumerated.item[0] = lo_dac;
return 0;
}
static int lo_set_switch(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
struct snd_soc_dapm_context *dapm = &card->dapm;
if (ucontrol->value.enumerated.item[0] == lo_dac)
return 0;
snd_soc_dapm_mutex_lock(dapm);
/* we dont want to work with last state of lineout so just enable all
* pins and then disable pins not required
*/
lo_enable_out_pins(dapm);
switch (ucontrol->value.enumerated.item[0]) {
case 0:
pr_debug("set vibra path\n");
snd_soc_dapm_disable_pin_unlocked(dapm, "VIB1OUT");
snd_soc_dapm_disable_pin_unlocked(dapm, "VIB2OUT");
snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0);
break;
case 1:
pr_debug("set hs path\n");
snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones");
snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT");
snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x22);
break;
case 2:
pr_debug("set spkr path\n");
snd_soc_dapm_disable_pin_unlocked(dapm, "IHFOUTL");
snd_soc_dapm_disable_pin_unlocked(dapm, "IHFOUTR");
snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x44);
break;
case 3:
pr_debug("set null path\n");
snd_soc_dapm_disable_pin_unlocked(dapm, "LINEOUTL");
snd_soc_dapm_disable_pin_unlocked(dapm, "LINEOUTR");
snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x66);
break;
}
snd_soc_dapm_sync_unlocked(dapm);
snd_soc_dapm_mutex_unlock(dapm);
lo_dac = ucontrol->value.enumerated.item[0];
return 0;
}
static const struct snd_kcontrol_new mfld_snd_controls[] = {
SOC_ENUM_EXT("Playback Switch", headset_enum,
headset_get_switch, headset_set_switch),
SOC_ENUM_EXT("Lineout Mux", lo_enum,
lo_get_switch, lo_set_switch),
};
static const struct snd_soc_dapm_widget mfld_widgets[] = {
SND_SOC_DAPM_HP("Headphones", NULL),
SND_SOC_DAPM_MIC("Mic", NULL),
};
static const struct snd_soc_dapm_route mfld_map[] = {
{"Headphones", NULL, "HPOUTR"},
{"Headphones", NULL, "HPOUTL"},
{"Mic", NULL, "AMIC1"},
};
static void mfld_jack_check(unsigned int intr_status)
{
struct mfld_jack_data jack_data;
if (!mfld_codec)
return;
jack_data.mfld_jack = &mfld_jack;
jack_data.intr_id = intr_status;
sn95031_jack_detection(mfld_codec, &jack_data);
/* TODO: add american headset detection post gpiolib support */
}
static int mfld_init(struct snd_soc_pcm_runtime *runtime)
{
struct snd_soc_dapm_context *dapm = &runtime->card->dapm;
int ret_val;
/* default is earpiece pin, userspace sets it explcitly */
snd_soc_dapm_disable_pin(dapm, "Headphones");
/* default is lineout NC, userspace sets it explcitly */
snd_soc_dapm_disable_pin(dapm, "LINEOUTL");
snd_soc_dapm_disable_pin(dapm, "LINEOUTR");
lo_dac = 3;
hs_switch = 0;
/* we dont use linein in this so set to NC */
snd_soc_dapm_disable_pin(dapm, "LINEINL");
snd_soc_dapm_disable_pin(dapm, "LINEINR");
/* Headset and button jack detection */
ret_val = snd_soc_card_jack_new(runtime->card,
"Intel(R) MID Audio Jack", SND_JACK_HEADSET |
SND_JACK_BTN_0 | SND_JACK_BTN_1, &mfld_jack,
mfld_jack_pins, ARRAY_SIZE(mfld_jack_pins));
if (ret_val) {
pr_err("jack creation failed\n");
return ret_val;
}
ret_val = snd_soc_jack_add_zones(&mfld_jack,
ARRAY_SIZE(mfld_zones), mfld_zones);
if (ret_val) {
pr_err("adding jack zones failed\n");
return ret_val;
}
mfld_codec = runtime->codec;
/* we want to check if anything is inserted at boot,
* so send a fake event to codec and it will read adc
* to find if anything is there or not */
mfld_jack_check(MFLD_JACK_INSERT);
return ret_val;
}
static struct snd_soc_dai_link mfld_msic_dailink[] = {
{
.name = "Medfield Headset",
.stream_name = "Headset",
.cpu_dai_name = "Headset-cpu-dai",
.codec_dai_name = "SN95031 Headset",
.codec_name = "sn95031",
.platform_name = "sst-platform",
.init = mfld_init,
},
{
.name = "Medfield Speaker",
.stream_name = "Speaker",
.cpu_dai_name = "Speaker-cpu-dai",
.codec_dai_name = "SN95031 Speaker",
.codec_name = "sn95031",
.platform_name = "sst-platform",
.init = NULL,
},
{
.name = "Medfield Vibra",
.stream_name = "Vibra1",
.cpu_dai_name = "Vibra1-cpu-dai",
.codec_dai_name = "SN95031 Vibra1",
.codec_name = "sn95031",
.platform_name = "sst-platform",
.init = NULL,
},
{
.name = "Medfield Haptics",
.stream_name = "Vibra2",
.cpu_dai_name = "Vibra2-cpu-dai",
.codec_dai_name = "SN95031 Vibra2",
.codec_name = "sn95031",
.platform_name = "sst-platform",
.init = NULL,
},
{
.name = "Medfield Compress",
.stream_name = "Speaker",
.cpu_dai_name = "Compress-cpu-dai",
.codec_dai_name = "SN95031 Speaker",
.codec_name = "sn95031",
.platform_name = "sst-platform",
.init = NULL,
},
};
/* SoC card */
static struct snd_soc_card snd_soc_card_mfld = {
.name = "medfield_audio",
.owner = THIS_MODULE,
.dai_link = mfld_msic_dailink,
.num_links = ARRAY_SIZE(mfld_msic_dailink),
.controls = mfld_snd_controls,
.num_controls = ARRAY_SIZE(mfld_snd_controls),
.dapm_widgets = mfld_widgets,
.num_dapm_widgets = ARRAY_SIZE(mfld_widgets),
.dapm_routes = mfld_map,
.num_dapm_routes = ARRAY_SIZE(mfld_map),
};
static irqreturn_t snd_mfld_jack_intr_handler(int irq, void *dev)
{
struct mfld_mc_private *mc_private = (struct mfld_mc_private *) dev;
memcpy_fromio(&mc_private->interrupt_status,
((void *)(mc_private->int_base)),
sizeof(u8));
return IRQ_WAKE_THREAD;
}
static irqreturn_t snd_mfld_jack_detection(int irq, void *data)
{
struct mfld_mc_private *mc_drv_ctx = (struct mfld_mc_private *) data;
mfld_jack_check(mc_drv_ctx->interrupt_status);
return IRQ_HANDLED;
}
static int snd_mfld_mc_probe(struct platform_device *pdev)
{
int ret_val = 0, irq;
struct mfld_mc_private *mc_drv_ctx;
struct resource *irq_mem;
pr_debug("snd_mfld_mc_probe called\n");
/* retrive the irq number */
irq = platform_get_irq(pdev, 0);
/* audio interrupt base of SRAM location where
* interrupts are stored by System FW */
mc_drv_ctx = devm_kzalloc(&pdev->dev, sizeof(*mc_drv_ctx), GFP_ATOMIC);
if (!mc_drv_ctx)
return -ENOMEM;
irq_mem = platform_get_resource_byname(
pdev, IORESOURCE_MEM, "IRQ_BASE");
if (!irq_mem) {
pr_err("no mem resource given\n");
return -ENODEV;
}
mc_drv_ctx->int_base = devm_ioremap_nocache(&pdev->dev, irq_mem->start,
resource_size(irq_mem));
if (!mc_drv_ctx->int_base) {
pr_err("Mapping of cache failed\n");
return -ENOMEM;
}
/* register for interrupt */
ret_val = devm_request_threaded_irq(&pdev->dev, irq,
snd_mfld_jack_intr_handler,
snd_mfld_jack_detection,
IRQF_SHARED, pdev->dev.driver->name, mc_drv_ctx);
if (ret_val) {
pr_err("cannot register IRQ\n");
return ret_val;
}
/* register the soc card */
snd_soc_card_mfld.dev = &pdev->dev;
ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_mfld);
if (ret_val) {
pr_debug("snd_soc_register_card failed %d\n", ret_val);
return ret_val;
}
platform_set_drvdata(pdev, mc_drv_ctx);
pr_debug("successfully exited probe\n");
return 0;
}
static struct platform_driver snd_mfld_mc_driver = {
.driver = {
.name = "msic_audio",
},
.probe = snd_mfld_mc_probe,
};
module_platform_driver(snd_mfld_mc_driver);
MODULE_DESCRIPTION("ASoC Intel(R) MID Machine driver");
MODULE_AUTHOR("Vinod Koul <vinod.koul@intel.com>");
MODULE_AUTHOR("Harsha Priya <priya.harsha@intel.com>");
MODULE_LICENSE("GPL v2");
MODULE_ALIAS("platform:msic-audio");
sound/soc/intel/common/sst-dsp.c
View file @ 3e4555ab
......@@ -269,7 +269,7 @@ int sst_dsp_register_poll(struct sst_dsp *ctx, u32 offset, u32 mask,
*/
timeout = jiffies + msecs_to_jiffies(time);
while (((sst_dsp_shim_read_unlocked(ctx, offset) & mask) != target)
while ((((reg = sst_dsp_shim_read_unlocked(ctx, offset)) & mask) != target)
&& time_before(jiffies, timeout)) {
k++;
if (k > 10)
......@@ -278,8 +278,6 @@ int sst_dsp_register_poll(struct sst_dsp *ctx, u32 offset, u32 mask,
usleep_range(s, 2*s);
}
reg = sst_dsp_shim_read_unlocked(ctx, offset);
if ((reg & mask) == target) {
dev_dbg(ctx->dev, "FW Poll Status: reg=%#x %s successful\n",
reg, operation);
......
sound/soc/intel/skylake/bxt-sst.c
View file @ 3e4555ab
......@@ -595,7 +595,7 @@ int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
INIT_DELAYED_WORK(&skl->d0i3.work, bxt_set_dsp_D0i3);
skl->d0i3.state = SKL_DSP_D0I3_NONE;
return 0;
return skl_dsp_acquire_irq(sst);
}
EXPORT_SYMBOL_GPL(bxt_sst_dsp_init);
......
sound/soc/intel/skylake/cnl-sst.c
View file @ 3e4555ab
......@@ -458,7 +458,7 @@ int cnl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
cnl->boot_complete = false;
init_waitqueue_head(&cnl->boot_wait);
return 0;
return skl_dsp_acquire_irq(sst);
}
EXPORT_SYMBOL_GPL(cnl_sst_dsp_init);
......
sound/soc/intel/skylake/skl-i2s.h 0 → 100644
View file @ 3e4555ab
/*
* skl-i2s.h - i2s blob mapping
*
* Copyright (C) 2017 Intel Corp
* Author: Subhransu S. Prusty < subhransu.s.prusty@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 __SOUND_SOC_SKL_I2S_H
#define __SOUND_SOC_SKL_I2S_H
#define SKL_I2S_MAX_TIME_SLOTS 8
#define SKL_MCLK_DIV_CLK_SRC_MASK GENMASK(17, 16)
#define SKL_MNDSS_DIV_CLK_SRC_MASK GENMASK(21, 20)
#define SKL_SHIFT(x) (ffs(x) - 1)
#define SKL_MCLK_DIV_RATIO_MASK GENMASK(11, 0)
struct skl_i2s_config {
u32 ssc0;
u32 ssc1;
u32 sscto;
u32 sspsp;
u32 sstsa;
u32 ssrsa;
u32 ssc2;
u32 sspsp2;
u32 ssc3;
u32 ssioc;
} __packed;
struct skl_i2s_config_mclk {
u32 mdivctrl;
u32 mdivr;
};
/**
* struct skl_i2s_config_blob_legacy - Structure defines I2S Gateway
* configuration legacy blob
*
* @gtw_attr: Gateway attribute for the I2S Gateway
* @tdm_ts_group: TDM slot mapping against channels in the Gateway.
* @i2s_cfg: I2S HW registers
* @mclk: MCLK clock source and divider values
*/
struct skl_i2s_config_blob_legacy {
u32 gtw_attr;
u32 tdm_ts_group[SKL_I2S_MAX_TIME_SLOTS];
struct skl_i2s_config i2s_cfg;
struct skl_i2s_config_mclk mclk;
};
#endif /* __SOUND_SOC_SKL_I2S_H */
sound/soc/intel/skylake/skl-messages.c
View file @ 3e4555ab
......@@ -55,6 +55,19 @@ static int skl_free_dma_buf(struct device *dev, struct snd_dma_buffer *dmab)
return 0;
}
#define SKL_ASTATE_PARAM_ID 4
void skl_dsp_set_astate_cfg(struct skl_sst *ctx, u32 cnt, void *data)
{
struct skl_ipc_large_config_msg msg = {0};
msg.large_param_id = SKL_ASTATE_PARAM_ID;
msg.param_data_size = (cnt * sizeof(struct skl_astate_param) +
sizeof(cnt));
skl_ipc_set_large_config(&ctx->ipc, &msg, data);
}
#define NOTIFICATION_PARAM_ID 3
#define NOTIFICATION_MASK 0xf
......@@ -404,11 +417,20 @@ int skl_resume_dsp(struct skl *skl)
if (skl->skl_sst->is_first_boot == true)
return 0;
/* disable dynamic clock gating during fw and lib download */
ctx->enable_miscbdcge(ctx->dev, false);
ret = skl_dsp_wake(ctx->dsp);
ctx->enable_miscbdcge(ctx->dev, true);
if (ret < 0)
return ret;
skl_dsp_enable_notification(skl->skl_sst, false);
if (skl->cfg.astate_cfg != NULL) {
skl_dsp_set_astate_cfg(skl->skl_sst, skl->cfg.astate_cfg->count,
skl->cfg.astate_cfg);
}
return ret;
}
......
sound/soc/intel/skylake/skl-nhlt.c
View file @ 3e4555ab
......@@ -19,6 +19,7 @@
*/
#include <linux/pci.h>
#include "skl.h"
#include "skl-i2s.h"
#define NHLT_ACPI_HEADER_SIG "NHLT"
......@@ -277,3 +278,157 @@ void skl_nhlt_remove_sysfs(struct skl *skl)
sysfs_remove_file(&dev->kobj, &dev_attr_platform_id.attr);
}
/*
* Queries NHLT for all the fmt configuration for a particular endpoint and
* stores all possible rates supported in a rate table for the corresponding
* sclk/sclkfs.
*/
static void skl_get_ssp_clks(struct skl *skl, struct skl_ssp_clk *ssp_clks,
struct nhlt_fmt *fmt, u8 id)
{
struct skl_i2s_config_blob_legacy *i2s_config;
struct skl_clk_parent_src *parent;
struct skl_ssp_clk *sclk, *sclkfs;
struct nhlt_fmt_cfg *fmt_cfg;
struct wav_fmt_ext *wav_fmt;
unsigned long rate = 0;
bool present = false;
int rate_index = 0;
u16 channels, bps;
u8 clk_src;
int i, j;
u32 fs;
sclk = &ssp_clks[SKL_SCLK_OFS];
sclkfs = &ssp_clks[SKL_SCLKFS_OFS];
if (fmt->fmt_count == 0)
return;
for (i = 0; i < fmt->fmt_count; i++) {
fmt_cfg = &fmt->fmt_config[i];
wav_fmt = &fmt_cfg->fmt_ext;
channels = wav_fmt->fmt.channels;
bps = wav_fmt->fmt.bits_per_sample;
fs = wav_fmt->fmt.samples_per_sec;
/*
* In case of TDM configuration on a ssp, there can
* be more than one blob in which channel masks are
* different for each usecase for a specific rate and bps.
* But the sclk rate will be generated for the total
* number of channels used for that endpoint.
*
* So for the given fs and bps, choose blob which has
* the superset of all channels for that endpoint and
* derive the rate.
*/
for (j = i; j < fmt->fmt_count; j++) {
fmt_cfg = &fmt->fmt_config[j];
wav_fmt = &fmt_cfg->fmt_ext;
if ((fs == wav_fmt->fmt.samples_per_sec) &&
(bps == wav_fmt->fmt.bits_per_sample))
channels = max_t(u16, channels,
wav_fmt->fmt.channels);
}
rate = channels * bps * fs;
/* check if the rate is added already to the given SSP's sclk */
for (j = 0; (j < SKL_MAX_CLK_RATES) &&
(sclk[id].rate_cfg[j].rate != 0); j++) {
if (sclk[id].rate_cfg[j].rate == rate) {
present = true;
break;
}
}
/* Fill rate and parent for sclk/sclkfs */
if (!present) {
/* MCLK Divider Source Select */
i2s_config = (struct skl_i2s_config_blob_legacy *)
fmt->fmt_config[0].config.caps;
clk_src = ((i2s_config->mclk.mdivctrl)
& SKL_MNDSS_DIV_CLK_SRC_MASK) >>
SKL_SHIFT(SKL_MNDSS_DIV_CLK_SRC_MASK);
parent = skl_get_parent_clk(clk_src);
/*
* Do not copy the config data if there is no parent
* clock available for this clock source select
*/
if (!parent)
continue;
sclk[id].rate_cfg[rate_index].rate = rate;
sclk[id].rate_cfg[rate_index].config = fmt_cfg;
sclkfs[id].rate_cfg[rate_index].rate = rate;
sclkfs[id].rate_cfg[rate_index].config = fmt_cfg;
sclk[id].parent_name = parent->name;
sclkfs[id].parent_name = parent->name;
rate_index++;
}
}
}
static void skl_get_mclk(struct skl *skl, struct skl_ssp_clk *mclk,
struct nhlt_fmt *fmt, u8 id)
{
struct skl_i2s_config_blob_legacy *i2s_config;
struct nhlt_specific_cfg *fmt_cfg;
struct skl_clk_parent_src *parent;
u32 clkdiv, div_ratio;
u8 clk_src;
fmt_cfg = &fmt->fmt_config[0].config;
i2s_config = (struct skl_i2s_config_blob_legacy *)fmt_cfg->caps;
/* MCLK Divider Source Select */
clk_src = ((i2s_config->mclk.mdivctrl) & SKL_MCLK_DIV_CLK_SRC_MASK) >>
SKL_SHIFT(SKL_MCLK_DIV_CLK_SRC_MASK);
clkdiv = i2s_config->mclk.mdivr & SKL_MCLK_DIV_RATIO_MASK;
/* bypass divider */
div_ratio = 1;
if (clkdiv != SKL_MCLK_DIV_RATIO_MASK)
/* Divider is 2 + clkdiv */
div_ratio = clkdiv + 2;
/* Calculate MCLK rate from source using div value */
parent = skl_get_parent_clk(clk_src);
if (!parent)
return;
mclk[id].rate_cfg[0].rate = parent->rate/div_ratio;
mclk[id].rate_cfg[0].config = &fmt->fmt_config[0];
mclk[id].parent_name = parent->name;
}
void skl_get_clks(struct skl *skl, struct skl_ssp_clk *ssp_clks)
{
struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt;
struct nhlt_endpoint *epnt;
struct nhlt_fmt *fmt;
int i;
u8 id;
epnt = (struct nhlt_endpoint *)nhlt->desc;
for (i = 0; i < nhlt->endpoint_count; i++) {
if (epnt->linktype == NHLT_LINK_SSP) {
id = epnt->virtual_bus_id;
fmt = (struct nhlt_fmt *)(epnt->config.caps
+ epnt->config.size);
skl_get_ssp_clks(skl, ssp_clks, fmt, id);
skl_get_mclk(skl, ssp_clks, fmt, id);
}
epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length);
}
}
sound/soc/intel/skylake/skl-pcm.c
View file @ 3e4555ab
......@@ -537,7 +537,7 @@ static int skl_link_hw_params(struct snd_pcm_substream *substream,
snd_soc_dai_set_dma_data(dai, substream, (void *)link_dev);
link = snd_hdac_ext_bus_get_link(ebus, rtd->codec->component.name);
link = snd_hdac_ext_bus_get_link(ebus, codec_dai->component->name);
if (!link)
return -EINVAL;
......@@ -620,7 +620,7 @@ static int skl_link_hw_free(struct snd_pcm_substream *substream,
link_dev->link_prepared = 0;
link = snd_hdac_ext_bus_get_link(ebus, rtd->codec->component.name);
link = snd_hdac_ext_bus_get_link(ebus, rtd->codec_dai->component->name);
if (!link)
return -EINVAL;
......@@ -1343,7 +1343,11 @@ static int skl_platform_soc_probe(struct snd_soc_platform *platform)
return -EIO;
}
/* disable dynamic clock gating during fw and lib download */
skl->skl_sst->enable_miscbdcge(platform->dev, false);
ret = ops->init_fw(platform->dev, skl->skl_sst);
skl->skl_sst->enable_miscbdcge(platform->dev, true);
if (ret < 0) {
dev_err(platform->dev, "Failed to boot first fw: %d\n", ret);
return ret;
......@@ -1351,6 +1355,12 @@ static int skl_platform_soc_probe(struct snd_soc_platform *platform)
skl_populate_modules(skl);
skl->skl_sst->update_d0i3c = skl_update_d0i3c;
skl_dsp_enable_notification(skl->skl_sst, false);
if (skl->cfg.astate_cfg != NULL) {
skl_dsp_set_astate_cfg(skl->skl_sst,
skl->cfg.astate_cfg->count,
skl->cfg.astate_cfg);
}
}
pm_runtime_mark_last_busy(platform->dev);
pm_runtime_put_autosuspend(platform->dev);
......
sound/soc/intel/skylake/skl-ssp-clk.h 0 → 100644
View file @ 3e4555ab
/*
* skl-ssp-clk.h - Skylake ssp clock information and ipc structure
*
* Copyright (C) 2017 Intel Corp
* Author: Jaikrishna Nemallapudi <jaikrishnax.nemallapudi@intel.com>
* Author: Subhransu S. Prusty <subhransu.s.prusty@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 SOUND_SOC_SKL_SSP_CLK_H
#define SOUND_SOC_SKL_SSP_CLK_H
#define SKL_MAX_SSP 6
/* xtal/cardinal/pll, parent of ssp clocks and mclk */
#define SKL_MAX_CLK_SRC 3
#define SKL_MAX_SSP_CLK_TYPES 3 /* mclk, sclk, sclkfs */
#define SKL_MAX_CLK_CNT (SKL_MAX_SSP * SKL_MAX_SSP_CLK_TYPES)
/* Max number of configurations supported for each clock */
#define SKL_MAX_CLK_RATES 10
#define SKL_SCLK_OFS SKL_MAX_SSP
#define SKL_SCLKFS_OFS (SKL_SCLK_OFS + SKL_MAX_SSP)
enum skl_clk_type {
SKL_MCLK,
SKL_SCLK,
SKL_SCLK_FS,
};
enum skl_clk_src_type {
SKL_XTAL,
SKL_CARDINAL,
SKL_PLL,
};
struct skl_clk_parent_src {
u8 clk_id;
const char *name;
unsigned long rate;
const char *parent_name;
};
struct skl_clk_rate_cfg_table {
unsigned long rate;
void *config;
};
/*
* rate for mclk will be in rates[0]. For sclk and sclkfs, rates[] store
* all possible clocks ssp can generate for that platform.
*/
struct skl_ssp_clk {
const char *name;
const char *parent_name;
struct skl_clk_rate_cfg_table rate_cfg[SKL_MAX_CLK_RATES];
};
struct skl_clk_pdata {
struct skl_clk_parent_src *parent_clks;
int num_clks;
struct skl_ssp_clk *ssp_clks;
void *pvt_data;
};
#endif /* SOUND_SOC_SKL_SSP_CLK_H */
sound/soc/intel/skylake/skl-sst-dsp.c
View file @ 3e4555ab
......@@ -435,16 +435,22 @@ struct sst_dsp *skl_dsp_ctx_init(struct device *dev,
return NULL;
}
return sst;
}
int skl_dsp_acquire_irq(struct sst_dsp *sst)
{
struct sst_dsp_device *sst_dev = sst->sst_dev;
int ret;
/* Register the ISR */
ret = request_threaded_irq(sst->irq, sst->ops->irq_handler,
sst_dev->thread, IRQF_SHARED, "AudioDSP", sst);
if (ret) {
if (ret)
dev_err(sst->dev, "unable to grab threaded IRQ %d, disabling device\n",
sst->irq);
return NULL;
}
return sst;
return ret;
}
void skl_dsp_free(struct sst_dsp *dsp)
......
sound/soc/intel/skylake/skl-sst-dsp.h
View file @ 3e4555ab
......@@ -206,6 +206,7 @@ int skl_cldma_wait_interruptible(struct sst_dsp *ctx);
void skl_dsp_set_state_locked(struct sst_dsp *ctx, int state);
struct sst_dsp *skl_dsp_ctx_init(struct device *dev,
struct sst_dsp_device *sst_dev, int irq);
int skl_dsp_acquire_irq(struct sst_dsp *sst);
bool is_skl_dsp_running(struct sst_dsp *ctx);
unsigned int skl_dsp_get_enabled_cores(struct sst_dsp *ctx);
......@@ -251,6 +252,9 @@ void skl_freeup_uuid_list(struct skl_sst *ctx);
int skl_dsp_strip_extended_manifest(struct firmware *fw);
void skl_dsp_enable_notification(struct skl_sst *ctx, bool enable);
void skl_dsp_set_astate_cfg(struct skl_sst *ctx, u32 cnt, void *data);
int skl_sst_ctx_init(struct device *dev, int irq, const char *fw_name,
struct skl_dsp_loader_ops dsp_ops, struct skl_sst **dsp,
struct sst_dsp_device *skl_dev);
......
sound/soc/intel/skylake/skl-sst-utils.c
View file @ 3e4555ab
......@@ -178,7 +178,8 @@ static inline int skl_pvtid_128(struct uuid_module *module)
* skl_get_pvt_id: generate a private id for use as module id
*
* @ctx: driver context
* @mconfig: module configuration data
* @uuid_mod: module's uuid
* @instance_id: module's instance id
*
* This generates a 128 bit private unique id for a module TYPE so that
* module instance is unique
......@@ -208,7 +209,8 @@ EXPORT_SYMBOL_GPL(skl_get_pvt_id);
* skl_put_pvt_id: free up the private id allocated
*
* @ctx: driver context
* @mconfig: module configuration data
* @uuid_mod: module's uuid
* @pvt_id: module pvt id
*
* This frees a 128 bit private unique id previously generated
*/
......
sound/soc/intel/skylake/skl-sst.c
View file @ 3e4555ab
......@@ -569,7 +569,7 @@ int skl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
sst->fw_ops = skl_fw_ops;
return 0;
return skl_dsp_acquire_irq(sst);
}
EXPORT_SYMBOL_GPL(skl_sst_dsp_init);
......
sound/soc/intel/skylake/skl-topology.c
View file @ 3e4555ab
......@@ -3056,11 +3056,13 @@ static int skl_tplg_get_int_tkn(struct device *dev,
struct snd_soc_tplg_vendor_value_elem *tkn_elem,
struct skl *skl)
{
int tkn_count = 0, ret;
int tkn_count = 0, ret, size;
static int mod_idx, res_val_idx, intf_val_idx, dir, pin_idx;
struct skl_module_res *res = NULL;
struct skl_module_iface *fmt = NULL;
struct skl_module *mod = NULL;
static struct skl_astate_param *astate_table;
static int astate_cfg_idx, count;
int i;
if (skl->modules) {
......@@ -3093,6 +3095,46 @@ static int skl_tplg_get_int_tkn(struct device *dev,
mod_idx = tkn_elem->value;
break;
case SKL_TKN_U32_ASTATE_COUNT:
if (astate_table != NULL) {
dev_err(dev, "More than one entry for A-State count");
return -EINVAL;
}
if (tkn_elem->value > SKL_MAX_ASTATE_CFG) {
dev_err(dev, "Invalid A-State count %d\n",
tkn_elem->value);
return -EINVAL;
}
size = tkn_elem->value * sizeof(struct skl_astate_param) +
sizeof(count);
skl->cfg.astate_cfg = devm_kzalloc(dev, size, GFP_KERNEL);
if (!skl->cfg.astate_cfg)
return -ENOMEM;
astate_table = skl->cfg.astate_cfg->astate_table;
count = skl->cfg.astate_cfg->count = tkn_elem->value;
break;
case SKL_TKN_U32_ASTATE_IDX:
if (tkn_elem->value >= count) {
dev_err(dev, "Invalid A-State index %d\n",
tkn_elem->value);
return -EINVAL;
}
astate_cfg_idx = tkn_elem->value;
break;
case SKL_TKN_U32_ASTATE_KCPS:
astate_table[astate_cfg_idx].kcps = tkn_elem->value;
break;
case SKL_TKN_U32_ASTATE_CLK_SRC:
astate_table[astate_cfg_idx].clk_src = tkn_elem->value;
break;
case SKL_TKN_U8_IN_PIN_TYPE:
case SKL_TKN_U8_OUT_PIN_TYPE:
case SKL_TKN_U8_IN_QUEUE_COUNT:
......
sound/soc/intel/skylake/skl.c
View file @ 3e4555ab
......@@ -355,6 +355,7 @@ static int skl_resume(struct device *dev)
if (ebus->cmd_dma_state)
snd_hdac_bus_init_cmd_io(&ebus->bus);
ret = 0;
} else {
ret = _skl_resume(ebus);
......@@ -435,19 +436,51 @@ static int skl_free(struct hdac_ext_bus *ebus)
return 0;
}
static int skl_machine_device_register(struct skl *skl, void *driver_data)
/*
* For each ssp there are 3 clocks (mclk/sclk/sclkfs).
* e.g. for ssp0, clocks will be named as
* "ssp0_mclk", "ssp0_sclk", "ssp0_sclkfs"
* So for skl+, there are 6 ssps, so 18 clocks will be created.
*/
static struct skl_ssp_clk skl_ssp_clks[] = {
{.name = "ssp0_mclk"}, {.name = "ssp1_mclk"}, {.name = "ssp2_mclk"},
{.name = "ssp3_mclk"}, {.name = "ssp4_mclk"}, {.name = "ssp5_mclk"},
{.name = "ssp0_sclk"}, {.name = "ssp1_sclk"}, {.name = "ssp2_sclk"},
{.name = "ssp3_sclk"}, {.name = "ssp4_sclk"}, {.name = "ssp5_sclk"},
{.name = "ssp0_sclkfs"}, {.name = "ssp1_sclkfs"},
{.name = "ssp2_sclkfs"},
{.name = "ssp3_sclkfs"}, {.name = "ssp4_sclkfs"},
{.name = "ssp5_sclkfs"},
};
static int skl_find_machine(struct skl *skl, void *driver_data)
{
struct hdac_bus *bus = ebus_to_hbus(&skl->ebus);
struct platform_device *pdev;
struct snd_soc_acpi_mach *mach = driver_data;
int ret;
struct hdac_bus *bus = ebus_to_hbus(&skl->ebus);
struct skl_machine_pdata *pdata;
mach = snd_soc_acpi_find_machine(mach);
if (mach == NULL) {
dev_err(bus->dev, "No matching machine driver found\n");
return -ENODEV;
}
skl->mach = mach;
skl->fw_name = mach->fw_filename;
pdata = skl->mach->pdata;
if (mach->pdata)
skl->use_tplg_pcm = pdata->use_tplg_pcm;
return 0;
}
static int skl_machine_device_register(struct skl *skl)
{
struct hdac_bus *bus = ebus_to_hbus(&skl->ebus);
struct snd_soc_acpi_mach *mach = skl->mach;
struct platform_device *pdev;
int ret;
pdev = platform_device_alloc(mach->drv_name, -1);
if (pdev == NULL) {
......@@ -462,11 +495,8 @@ static int skl_machine_device_register(struct skl *skl, void *driver_data)
return -EIO;
}
if (mach->pdata) {
skl->use_tplg_pcm =
((struct skl_machine_pdata *)mach->pdata)->use_tplg_pcm;
if (mach->pdata)
dev_set_drvdata(&pdev->dev, mach->pdata);
}
skl->i2s_dev = pdev;
......@@ -509,6 +539,74 @@ static void skl_dmic_device_unregister(struct skl *skl)
platform_device_unregister(skl->dmic_dev);
}
static struct skl_clk_parent_src skl_clk_src[] = {
{ .clk_id = SKL_XTAL, .name = "xtal" },
{ .clk_id = SKL_CARDINAL, .name = "cardinal", .rate = 24576000 },
{ .clk_id = SKL_PLL, .name = "pll", .rate = 96000000 },
};
struct skl_clk_parent_src *skl_get_parent_clk(u8 clk_id)
{
unsigned int i;
for (i = 0; i < ARRAY_SIZE(skl_clk_src); i++) {
if (skl_clk_src[i].clk_id == clk_id)
return &skl_clk_src[i];
}
return NULL;
}
static void init_skl_xtal_rate(int pci_id)
{
switch (pci_id) {
case 0x9d70:
case 0x9d71:
skl_clk_src[0].rate = 24000000;
return;
default:
skl_clk_src[0].rate = 19200000;
return;
}
}
static int skl_clock_device_register(struct skl *skl)
{
struct platform_device_info pdevinfo = {NULL};
struct skl_clk_pdata *clk_pdata;
clk_pdata = devm_kzalloc(&skl->pci->dev, sizeof(*clk_pdata),
GFP_KERNEL);
if (!clk_pdata)
return -ENOMEM;
init_skl_xtal_rate(skl->pci->device);
clk_pdata->parent_clks = skl_clk_src;
clk_pdata->ssp_clks = skl_ssp_clks;
clk_pdata->num_clks = ARRAY_SIZE(skl_ssp_clks);
/* Query NHLT to fill the rates and parent */
skl_get_clks(skl, clk_pdata->ssp_clks);
clk_pdata->pvt_data = skl;
/* Register Platform device */
pdevinfo.parent = &skl->pci->dev;
pdevinfo.id = -1;
pdevinfo.name = "skl-ssp-clk";
pdevinfo.data = clk_pdata;
pdevinfo.size_data = sizeof(*clk_pdata);
skl->clk_dev = platform_device_register_full(&pdevinfo);
return PTR_ERR_OR_ZERO(skl->clk_dev);
}
static void skl_clock_device_unregister(struct skl *skl)
{
if (skl->clk_dev)
platform_device_unregister(skl->clk_dev);
}
/*
* Probe the given codec address
*/
......@@ -615,18 +713,30 @@ static void skl_probe_work(struct work_struct *work)
/* create codec instances */
skl_codec_create(ebus);
/* register platform dai and controls */
err = skl_platform_register(bus->dev);
if (err < 0) {
dev_err(bus->dev, "platform register failed: %d\n", err);
return;
}
if (bus->ppcap) {
err = skl_machine_device_register(skl);
if (err < 0) {
dev_err(bus->dev, "machine register failed: %d\n", err);
goto out_err;
}
}
if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) {
err = snd_hdac_display_power(bus, false);
if (err < 0) {
dev_err(bus->dev, "Cannot turn off display power on i915\n");
skl_machine_device_unregister(skl);
return;
}
}
/* register platform dai and controls */
err = skl_platform_register(bus->dev);
if (err < 0)
return;
/*
* we are done probing so decrement link counts
*/
......@@ -791,18 +901,21 @@ static int skl_probe(struct pci_dev *pci,
/* check if dsp is there */
if (bus->ppcap) {
err = skl_machine_device_register(skl,
(void *)pci_id->driver_data);
/* create device for dsp clk */
err = skl_clock_device_register(skl);
if (err < 0)
goto out_clk_free;
err = skl_find_machine(skl, (void *)pci_id->driver_data);
if (err < 0)
goto out_nhlt_free;
err = skl_init_dsp(skl);
if (err < 0) {
dev_dbg(bus->dev, "error failed to register dsp\n");
goto out_mach_free;
goto out_nhlt_free;
}
skl->skl_sst->enable_miscbdcge = skl_enable_miscbdcge;
}
if (bus->mlcap)
snd_hdac_ext_bus_get_ml_capabilities(ebus);
......@@ -820,8 +933,8 @@ static int skl_probe(struct pci_dev *pci,
out_dsp_free:
skl_free_dsp(skl);
out_mach_free:
skl_machine_device_unregister(skl);
out_clk_free:
skl_clock_device_unregister(skl);
out_nhlt_free:
skl_nhlt_free(skl->nhlt);
out_free:
......@@ -872,6 +985,7 @@ static void skl_remove(struct pci_dev *pci)
skl_free_dsp(skl);
skl_machine_device_unregister(skl);
skl_dmic_device_unregister(skl);
skl_clock_device_unregister(skl);
skl_nhlt_remove_sysfs(skl);
skl_nhlt_free(skl->nhlt);
skl_free(ebus);
......
sound/soc/intel/skylake/skl.h
View file @ 3e4555ab
......@@ -25,9 +25,12 @@
#include <sound/hdaudio_ext.h>
#include <sound/soc.h>
#include "skl-nhlt.h"
#include "skl-ssp-clk.h"
#define SKL_SUSPEND_DELAY 2000
#define SKL_MAX_ASTATE_CFG 3
#define AZX_PCIREG_PGCTL 0x44
#define AZX_PGCTL_LSRMD_MASK (1 << 4)
#define AZX_PCIREG_CGCTL 0x48
......@@ -45,6 +48,20 @@ struct skl_dsp_resource {
struct skl_debug;
struct skl_astate_param {
u32 kcps;
u32 clk_src;
};
struct skl_astate_config {
u32 count;
struct skl_astate_param astate_table[0];
};
struct skl_fw_config {
struct skl_astate_config *astate_cfg;
};
struct skl {
struct hdac_ext_bus ebus;
struct pci_dev *pci;
......@@ -52,6 +69,7 @@ struct skl {
unsigned int init_done:1; /* delayed init status */
struct platform_device *dmic_dev;
struct platform_device *i2s_dev;
struct platform_device *clk_dev;
struct snd_soc_platform *platform;
struct snd_soc_dai_driver *dais;
......@@ -75,6 +93,8 @@ struct skl {
u8 nr_modules;
struct skl_module **modules;
bool use_tplg_pcm;
struct skl_fw_config cfg;
struct snd_soc_acpi_mach *mach;
};
#define skl_to_ebus(s) (&(s)->ebus)
......@@ -125,6 +145,8 @@ const struct skl_dsp_ops *skl_get_dsp_ops(int pci_id);
void skl_update_d0i3c(struct device *dev, bool enable);
int skl_nhlt_create_sysfs(struct skl *skl);
void skl_nhlt_remove_sysfs(struct skl *skl);
void skl_get_clks(struct skl *skl, struct skl_ssp_clk *ssp_clks);
struct skl_clk_parent_src *skl_get_parent_clk(u8 clk_id);
struct skl_module_cfg;
......
sound/soc/soc-acpi.c
View file @ 3e4555ab
......@@ -49,41 +49,13 @@ const char *snd_soc_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN])
}
EXPORT_SYMBOL_GPL(snd_soc_acpi_find_name_from_hid);
static acpi_status snd_soc_acpi_mach_match(acpi_handle handle, u32 level,
void *context, void **ret)
{
unsigned long long sta;
acpi_status status;
*(bool *)context = true;
status = acpi_evaluate_integer(handle, "_STA", NULL, &sta);
if (ACPI_FAILURE(status) || !(sta & ACPI_STA_DEVICE_PRESENT))
*(bool *)context = false;
return AE_OK;
}
bool snd_soc_acpi_check_hid(const u8 hid[ACPI_ID_LEN])
{
acpi_status status;
bool found = false;
status = acpi_get_devices(hid, snd_soc_acpi_mach_match, &found, NULL);
if (ACPI_FAILURE(status))
return false;
return found;
}
EXPORT_SYMBOL_GPL(snd_soc_acpi_check_hid);
struct snd_soc_acpi_mach *
snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines)
{
struct snd_soc_acpi_mach *mach;
for (mach = machines; mach->id[0]; mach++) {
if (snd_soc_acpi_check_hid(mach->id) == true) {
if (acpi_dev_present(mach->id, NULL, -1)) {
if (mach->machine_quirk)
mach = mach->machine_quirk(mach);
return mach;
......@@ -161,7 +133,7 @@ struct snd_soc_acpi_mach *snd_soc_acpi_codec_list(void *arg)
return mach;
for (i = 0; i < codec_list->num_codecs; i++) {
if (snd_soc_acpi_check_hid(codec_list->codecs[i]) != true)
if (!acpi_dev_present(codec_list->codecs[i], NULL, -1))
return NULL;
}
......
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