lisa/android_kernel_xiaomi_sm8350
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
20c8ccb197
android_kernel_xiaomi_sm8350/sound/soc/codecs/mc13783.c
Thomas Gleixner 1621633323 treewide: Replace GPLv2 boilerplate/reference with SPDX - rule 1 
Based on 2 normalized pattern(s):

  this program is free software you can redistribute it and or modify
  it under the terms of the gnu general public license as published by
  the free software foundation either version 2 of the license or at
  your option any later version 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
  51 franklin street fifth floor boston ma 02110 1301 usa

  this program is free software you can redistribute it and or modify
  it under the terms of the gnu general public license as published by
  the free software foundation either version 2 of the license or at
  your option [no]_[pad]_[ctrl] any later version 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 51 franklin street fifth floor boston ma
  02110 1301 usa

extracted by the scancode license scanner the SPDX license identifier

  GPL-2.0-or-later

has been chosen to replace the boilerplate/reference in 176 file(s).

Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Reviewed-by: Jilayne Lovejoy <opensource@jilayne.com>
Reviewed-by: Steve Winslow <swinslow@gmail.com>
Reviewed-by: Allison Randal <allison@lohutok.net>
Reviewed-by: Kate Stewart <kstewart@linuxfoundation.org>
Cc: linux-spdx@vger.kernel.org
Link: https://lkml.kernel.org/r/20190519154040.652910950@linutronix.de
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2019-05-21 11:28:39 +02:00

798 lines
22 KiB
C
Raw Blame History

// SPDX-License-Identifier: GPL-2.0-or-later
/*
* Copyright 2008 Juergen Beisert, kernel@pengutronix.de
* Copyright 2009 Sascha Hauer, s.hauer@pengutronix.de
* Copyright 2012 Philippe Retornaz, philippe.retornaz@epfl.ch
*
* Initial development of this code was funded by
* Phytec Messtechnik GmbH, http://www.phytec.de
*/
#include <linux/module.h>
#include <linux/device.h>
#include <linux/of.h>
#include <linux/mfd/mc13xxx.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/control.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/initval.h>
#include <sound/soc-dapm.h>
#include <linux/regmap.h>
#include "mc13783.h"
#define AUDIO_RX0_ALSPEN (1 << 5)
#define AUDIO_RX0_ALSPSEL (1 << 7)
#define AUDIO_RX0_ADDCDC (1 << 21)
#define AUDIO_RX0_ADDSTDC (1 << 22)
#define AUDIO_RX0_ADDRXIN (1 << 23)
#define AUDIO_RX1_PGARXEN (1 << 0);
#define AUDIO_RX1_PGASTEN (1 << 5)
#define AUDIO_RX1_ARXINEN (1 << 10)
#define AUDIO_TX_AMC1REN (1 << 5)
#define AUDIO_TX_AMC1LEN (1 << 7)
#define AUDIO_TX_AMC2EN (1 << 9)
#define AUDIO_TX_ATXINEN (1 << 11)
#define AUDIO_TX_RXINREC (1 << 13)
#define SSI_NETWORK_CDCTXRXSLOT(x) (((x) & 0x3) << 2)
#define SSI_NETWORK_CDCTXSECSLOT(x) (((x) & 0x3) << 4)
#define SSI_NETWORK_CDCRXSECSLOT(x) (((x) & 0x3) << 6)
#define SSI_NETWORK_CDCRXSECGAIN(x) (((x) & 0x3) << 8)
#define SSI_NETWORK_CDCSUMGAIN(x) (1 << 10)
#define SSI_NETWORK_CDCFSDLY(x) (1 << 11)
#define SSI_NETWORK_DAC_SLOTS_8 (1 << 12)
#define SSI_NETWORK_DAC_SLOTS_4 (2 << 12)
#define SSI_NETWORK_DAC_SLOTS_2 (3 << 12)
#define SSI_NETWORK_DAC_SLOT_MASK (3 << 12)
#define SSI_NETWORK_DAC_RXSLOT_0_1 (0 << 14)
#define SSI_NETWORK_DAC_RXSLOT_2_3 (1 << 14)
#define SSI_NETWORK_DAC_RXSLOT_4_5 (2 << 14)
#define SSI_NETWORK_DAC_RXSLOT_6_7 (3 << 14)
#define SSI_NETWORK_DAC_RXSLOT_MASK (3 << 14)
#define SSI_NETWORK_STDCRXSECSLOT(x) (((x) & 0x3) << 16)
#define SSI_NETWORK_STDCRXSECGAIN(x) (((x) & 0x3) << 18)
#define SSI_NETWORK_STDCSUMGAIN (1 << 20)
/*
* MC13783_AUDIO_CODEC and MC13783_AUDIO_DAC mostly share the same
* register layout
*/
#define AUDIO_SSI_SEL (1 << 0)
#define AUDIO_CLK_SEL (1 << 1)
#define AUDIO_CSM (1 << 2)
#define AUDIO_BCL_INV (1 << 3)
#define AUDIO_CFS_INV (1 << 4)
#define AUDIO_CFS(x) (((x) & 0x3) << 5)
#define AUDIO_CLK(x) (((x) & 0x7) << 7)
#define AUDIO_C_EN (1 << 11)
#define AUDIO_C_CLK_EN (1 << 12)
#define AUDIO_C_RESET (1 << 15)
#define AUDIO_CODEC_CDCFS8K16K (1 << 10)
#define AUDIO_DAC_CFS_DLY_B (1 << 10)
struct mc13783_priv {
struct mc13xxx *mc13xxx;
struct regmap *regmap;
enum mc13783_ssi_port adc_ssi_port;
enum mc13783_ssi_port dac_ssi_port;
};
/* Mapping between sample rates and register value */
static unsigned int mc13783_rates[] = {
8000, 11025, 12000, 16000,
22050, 24000, 32000, 44100,
48000, 64000, 96000
};
static int mc13783_pcm_hw_params_dac(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_component *component = dai->component;
unsigned int rate = params_rate(params);
int i;
for (i = 0; i < ARRAY_SIZE(mc13783_rates); i++) {
if (rate == mc13783_rates[i]) {
snd_soc_component_update_bits(component, MC13783_AUDIO_DAC,
0xf << 17, i << 17);
return 0;
}
}
return -EINVAL;
}
static int mc13783_pcm_hw_params_codec(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_component *component = dai->component;
unsigned int rate = params_rate(params);
unsigned int val;
switch (rate) {
case 8000:
val = 0;
break;
case 16000:
val = AUDIO_CODEC_CDCFS8K16K;
break;
default:
return -EINVAL;
}
snd_soc_component_update_bits(component, MC13783_AUDIO_CODEC, AUDIO_CODEC_CDCFS8K16K,
val);
return 0;
}
static int mc13783_pcm_hw_params_sync(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
return mc13783_pcm_hw_params_dac(substream, params, dai);
else
return mc13783_pcm_hw_params_codec(substream, params, dai);
}
static int mc13783_set_fmt(struct snd_soc_dai *dai, unsigned int fmt,
unsigned int reg)
{
struct snd_soc_component *component = dai->component;
unsigned int val = 0;
unsigned int mask = AUDIO_CFS(3) | AUDIO_BCL_INV | AUDIO_CFS_INV |
AUDIO_CSM | AUDIO_C_CLK_EN | AUDIO_C_RESET;
/* DAI mode */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
val |= AUDIO_CFS(2);
break;
case SND_SOC_DAIFMT_DSP_A:
val |= AUDIO_CFS(1);
break;
default:
return -EINVAL;
}
/* DAI clock inversion */
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
val |= AUDIO_BCL_INV;
break;
case SND_SOC_DAIFMT_NB_IF:
val |= AUDIO_BCL_INV | AUDIO_CFS_INV;
break;
case SND_SOC_DAIFMT_IB_NF:
break;
case SND_SOC_DAIFMT_IB_IF:
val |= AUDIO_CFS_INV;
break;
}
/* DAI clock master masks */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
val |= AUDIO_C_CLK_EN;
break;
case SND_SOC_DAIFMT_CBS_CFS:
val |= AUDIO_CSM;
break;
case SND_SOC_DAIFMT_CBM_CFS:
case SND_SOC_DAIFMT_CBS_CFM:
return -EINVAL;
}
val |= AUDIO_C_RESET;
snd_soc_component_update_bits(component, reg, mask, val);
return 0;
}
static int mc13783_set_fmt_async(struct snd_soc_dai *dai, unsigned int fmt)
{
if (dai->id == MC13783_ID_STEREO_DAC)
return mc13783_set_fmt(dai, fmt, MC13783_AUDIO_DAC);
else
return mc13783_set_fmt(dai, fmt, MC13783_AUDIO_CODEC);
}
static int mc13783_set_fmt_sync(struct snd_soc_dai *dai, unsigned int fmt)
{
int ret;
ret = mc13783_set_fmt(dai, fmt, MC13783_AUDIO_DAC);
if (ret)
return ret;
/*
* In synchronous mode force the voice codec into slave mode
* so that the clock / framesync from the stereo DAC is used
*/
fmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
fmt |= SND_SOC_DAIFMT_CBS_CFS;
ret = mc13783_set_fmt(dai, fmt, MC13783_AUDIO_CODEC);
return ret;
}
static int mc13783_sysclk[] = {
13000000,
15360000,
16800000,
-1,
26000000,
-1, /* 12000000, invalid for voice codec */
-1, /* 3686400, invalid for voice codec */
33600000,
};
static int mc13783_set_sysclk(struct snd_soc_dai *dai,
int clk_id, unsigned int freq, int dir,
unsigned int reg)
{
struct snd_soc_component *component = dai->component;
int clk;
unsigned int val = 0;
unsigned int mask = AUDIO_CLK(0x7) | AUDIO_CLK_SEL;
for (clk = 0; clk < ARRAY_SIZE(mc13783_sysclk); clk++) {
if (mc13783_sysclk[clk] < 0)
continue;
if (mc13783_sysclk[clk] == freq)
break;
}
if (clk == ARRAY_SIZE(mc13783_sysclk))
return -EINVAL;
if (clk_id == MC13783_CLK_CLIB)
val |= AUDIO_CLK_SEL;
val |= AUDIO_CLK(clk);
snd_soc_component_update_bits(component, reg, mask, val);
return 0;
}
static int mc13783_set_sysclk_dac(struct snd_soc_dai *dai,
int clk_id, unsigned int freq, int dir)
{
return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_DAC);
}
static int mc13783_set_sysclk_codec(struct snd_soc_dai *dai,
int clk_id, unsigned int freq, int dir)
{
return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_CODEC);
}
static int mc13783_set_sysclk_sync(struct snd_soc_dai *dai,
int clk_id, unsigned int freq, int dir)
{
int ret;
ret = mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_DAC);
if (ret)
return ret;
return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_CODEC);
}
static int mc13783_set_tdm_slot_dac(struct snd_soc_dai *dai,
unsigned int tx_mask, unsigned int rx_mask, int slots,
int slot_width)
{
struct snd_soc_component *component = dai->component;
unsigned int val = 0;
unsigned int mask = SSI_NETWORK_DAC_SLOT_MASK |
SSI_NETWORK_DAC_RXSLOT_MASK;
switch (slots) {
case 2:
val |= SSI_NETWORK_DAC_SLOTS_2;
break;
case 4:
val |= SSI_NETWORK_DAC_SLOTS_4;
break;
case 8:
val |= SSI_NETWORK_DAC_SLOTS_8;
break;
default:
return -EINVAL;
}
switch (rx_mask) {
case 0x03:
val |= SSI_NETWORK_DAC_RXSLOT_0_1;
break;
case 0x0c:
val |= SSI_NETWORK_DAC_RXSLOT_2_3;
break;
case 0x30:
val |= SSI_NETWORK_DAC_RXSLOT_4_5;
break;
case 0xc0:
val |= SSI_NETWORK_DAC_RXSLOT_6_7;
break;
default:
return -EINVAL;
}
snd_soc_component_update_bits(component, MC13783_SSI_NETWORK, mask, val);
return 0;
}
static int mc13783_set_tdm_slot_codec(struct snd_soc_dai *dai,
unsigned int tx_mask, unsigned int rx_mask, int slots,
int slot_width)
{
struct snd_soc_component *component = dai->component;
unsigned int val = 0;
unsigned int mask = 0x3f;
if (slots != 4)
return -EINVAL;
if (tx_mask != 0x3)
return -EINVAL;
val |= (0x00 << 2); /* primary timeslot RX/TX(?) is 0 */
val |= (0x01 << 4); /* secondary timeslot TX is 1 */
snd_soc_component_update_bits(component, MC13783_SSI_NETWORK, mask, val);
return 0;
}
static int mc13783_set_tdm_slot_sync(struct snd_soc_dai *dai,
unsigned int tx_mask, unsigned int rx_mask, int slots,
int slot_width)
{
int ret;
ret = mc13783_set_tdm_slot_dac(dai, tx_mask, rx_mask, slots,
slot_width);
if (ret)
return ret;
ret = mc13783_set_tdm_slot_codec(dai, tx_mask, rx_mask, slots,
slot_width);
return ret;
}
static const struct snd_kcontrol_new mc1l_amp_ctl =
SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 7, 1, 0);
static const struct snd_kcontrol_new mc1r_amp_ctl =
SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 5, 1, 0);
static const struct snd_kcontrol_new mc2_amp_ctl =
SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 9, 1, 0);
static const struct snd_kcontrol_new atx_amp_ctl =
SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 11, 1, 0);
/* Virtual mux. The chip does the input selection automatically
* as soon as we enable one input. */
static const char * const adcl_enum_text[] = {
"MC1L", "RXINL",
};
static SOC_ENUM_SINGLE_VIRT_DECL(adcl_enum, adcl_enum_text);
static const struct snd_kcontrol_new left_input_mux =
SOC_DAPM_ENUM("Route", adcl_enum);
static const char * const adcr_enum_text[] = {
"MC1R", "MC2", "RXINR", "TXIN",
};
static SOC_ENUM_SINGLE_VIRT_DECL(adcr_enum, adcr_enum_text);
static const struct snd_kcontrol_new right_input_mux =
SOC_DAPM_ENUM("Route", adcr_enum);
static const struct snd_kcontrol_new samp_ctl =
SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 3, 1, 0);
static const char * const speaker_amp_source_text[] = {
"CODEC", "Right"
};
static SOC_ENUM_SINGLE_DECL(speaker_amp_source, MC13783_AUDIO_RX0, 4,
speaker_amp_source_text);
static const struct snd_kcontrol_new speaker_amp_source_mux =
SOC_DAPM_ENUM("Speaker Amp Source MUX", speaker_amp_source);
static const char * const headset_amp_source_text[] = {
"CODEC", "Mixer"
};
static SOC_ENUM_SINGLE_DECL(headset_amp_source, MC13783_AUDIO_RX0, 11,
headset_amp_source_text);
static const struct snd_kcontrol_new headset_amp_source_mux =
SOC_DAPM_ENUM("Headset Amp Source MUX", headset_amp_source);
static const struct snd_kcontrol_new cdcout_ctl =
SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 18, 1, 0);
static const struct snd_kcontrol_new adc_bypass_ctl =
SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_CODEC, 16, 1, 0);
static const struct snd_kcontrol_new lamp_ctl =
SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 5, 1, 0);
static const struct snd_kcontrol_new hlamp_ctl =
SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 10, 1, 0);
static const struct snd_kcontrol_new hramp_ctl =
SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 9, 1, 0);
static const struct snd_kcontrol_new llamp_ctl =
SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 16, 1, 0);
static const struct snd_kcontrol_new lramp_ctl =
SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 15, 1, 0);
static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = {
/* Input */
SND_SOC_DAPM_INPUT("MC1LIN"),
SND_SOC_DAPM_INPUT("MC1RIN"),
SND_SOC_DAPM_INPUT("MC2IN"),
SND_SOC_DAPM_INPUT("RXINR"),
SND_SOC_DAPM_INPUT("RXINL"),
SND_SOC_DAPM_INPUT("TXIN"),
SND_SOC_DAPM_SUPPLY("MC1 Bias", MC13783_AUDIO_TX, 0, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("MC2 Bias", MC13783_AUDIO_TX, 1, 0, NULL, 0),
SND_SOC_DAPM_SWITCH("MC1L Amp", MC13783_AUDIO_TX, 7, 0, &mc1l_amp_ctl),
SND_SOC_DAPM_SWITCH("MC1R Amp", MC13783_AUDIO_TX, 5, 0, &mc1r_amp_ctl),
SND_SOC_DAPM_SWITCH("MC2 Amp", MC13783_AUDIO_TX, 9, 0, &mc2_amp_ctl),
SND_SOC_DAPM_SWITCH("TXIN Amp", MC13783_AUDIO_TX, 11, 0, &atx_amp_ctl),
SND_SOC_DAPM_MUX("PGA Left Input Mux", SND_SOC_NOPM, 0, 0,
&left_input_mux),
SND_SOC_DAPM_MUX("PGA Right Input Mux", SND_SOC_NOPM, 0, 0,
&right_input_mux),
SND_SOC_DAPM_MUX("Speaker Amp Source MUX", SND_SOC_NOPM, 0, 0,
&speaker_amp_source_mux),
SND_SOC_DAPM_MUX("Headset Amp Source MUX", SND_SOC_NOPM, 0, 0,
&headset_amp_source_mux),
SND_SOC_DAPM_PGA("PGA Left Input", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("PGA Right Input", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_ADC("ADC", "Capture", MC13783_AUDIO_CODEC, 11, 0),
SND_SOC_DAPM_SUPPLY("ADC_Reset", MC13783_AUDIO_CODEC, 15, 0, NULL, 0),
SND_SOC_DAPM_PGA("Voice CODEC PGA", MC13783_AUDIO_RX1, 0, 0, NULL, 0),
SND_SOC_DAPM_SWITCH("Voice CODEC Bypass", MC13783_AUDIO_CODEC, 16, 0,
&adc_bypass_ctl),
/* Output */
SND_SOC_DAPM_SUPPLY("DAC_E", MC13783_AUDIO_DAC, 11, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("DAC_Reset", MC13783_AUDIO_DAC, 15, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("RXOUTL"),
SND_SOC_DAPM_OUTPUT("RXOUTR"),
SND_SOC_DAPM_OUTPUT("HSL"),
SND_SOC_DAPM_OUTPUT("HSR"),
SND_SOC_DAPM_OUTPUT("LSPL"),
SND_SOC_DAPM_OUTPUT("LSP"),
SND_SOC_DAPM_OUTPUT("SP"),
SND_SOC_DAPM_OUTPUT("CDCOUT"),
SND_SOC_DAPM_SWITCH("CDCOUT Switch", MC13783_AUDIO_RX0, 18, 0,
&cdcout_ctl),
SND_SOC_DAPM_SWITCH("Speaker Amp Switch", MC13783_AUDIO_RX0, 3, 0,
&samp_ctl),
SND_SOC_DAPM_SWITCH("Loudspeaker Amp", SND_SOC_NOPM, 0, 0, &lamp_ctl),
SND_SOC_DAPM_SWITCH("Headset Amp Left", MC13783_AUDIO_RX0, 10, 0,
&hlamp_ctl),
SND_SOC_DAPM_SWITCH("Headset Amp Right", MC13783_AUDIO_RX0, 9, 0,
&hramp_ctl),
SND_SOC_DAPM_SWITCH("Line out Amp Left", MC13783_AUDIO_RX0, 16, 0,
&llamp_ctl),
SND_SOC_DAPM_SWITCH("Line out Amp Right", MC13783_AUDIO_RX0, 15, 0,
&lramp_ctl),
SND_SOC_DAPM_DAC("DAC", "Playback", MC13783_AUDIO_RX0, 22, 0),
SND_SOC_DAPM_PGA("DAC PGA", MC13783_AUDIO_RX1, 5, 0, NULL, 0),
};
static struct snd_soc_dapm_route mc13783_routes[] = {
/* Input */
{ "MC1L Amp", NULL, "MC1LIN"},
{ "MC1R Amp", NULL, "MC1RIN" },
{ "MC2 Amp", NULL, "MC2IN" },
{ "TXIN Amp", NULL, "TXIN"},
{ "PGA Left Input Mux", "MC1L", "MC1L Amp" },
{ "PGA Left Input Mux", "RXINL", "RXINL"},
{ "PGA Right Input Mux", "MC1R", "MC1R Amp" },
{ "PGA Right Input Mux", "MC2", "MC2 Amp"},
{ "PGA Right Input Mux", "TXIN", "TXIN Amp"},
{ "PGA Right Input Mux", "RXINR", "RXINR"},
{ "PGA Left Input", NULL, "PGA Left Input Mux"},
{ "PGA Right Input", NULL, "PGA Right Input Mux"},
{ "ADC", NULL, "PGA Left Input"},
{ "ADC", NULL, "PGA Right Input"},
{ "ADC", NULL, "ADC_Reset"},
{ "Voice CODEC PGA", "Voice CODEC Bypass", "ADC" },
{ "Speaker Amp Source MUX", "CODEC", "Voice CODEC PGA"},
{ "Speaker Amp Source MUX", "Right", "DAC PGA"},
{ "Headset Amp Source MUX", "CODEC", "Voice CODEC PGA"},
{ "Headset Amp Source MUX", "Mixer", "DAC PGA"},
/* Output */
{ "HSL", NULL, "Headset Amp Left" },
{ "HSR", NULL, "Headset Amp Right"},
{ "RXOUTL", NULL, "Line out Amp Left"},
{ "RXOUTR", NULL, "Line out Amp Right"},
{ "SP", "Speaker Amp Switch", "Speaker Amp Source MUX"},
{ "LSP", "Loudspeaker Amp", "Speaker Amp Source MUX"},
{ "HSL", "Headset Amp Left", "Headset Amp Source MUX"},
{ "HSR", "Headset Amp Right", "Headset Amp Source MUX"},
{ "Line out Amp Left", NULL, "DAC PGA"},
{ "Line out Amp Right", NULL, "DAC PGA"},
{ "DAC PGA", NULL, "DAC"},
{ "DAC", NULL, "DAC_E"},
{ "CDCOUT", "CDCOUT Switch", "Voice CODEC PGA"},
};
static const char * const mc13783_3d_mixer[] = {"Stereo", "Phase Mix",
"Mono", "Mono Mix"};
static SOC_ENUM_SINGLE_DECL(mc13783_enum_3d_mixer,
MC13783_AUDIO_RX1, 16,
mc13783_3d_mixer);
static struct snd_kcontrol_new mc13783_control_list[] = {
SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0),
SOC_SINGLE("PCM Playback Volume", MC13783_AUDIO_RX1, 6, 15, 0),
SOC_SINGLE("PCM Playback Switch", MC13783_AUDIO_RX1, 5, 1, 0),
SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0),
SOC_ENUM("3D Control", mc13783_enum_3d_mixer),
SOC_SINGLE("CDCOUT Switch", MC13783_AUDIO_RX0, 18, 1, 0),
SOC_SINGLE("Earpiece Amp Switch", MC13783_AUDIO_RX0, 3, 1, 0),
SOC_DOUBLE("Headset Amp Switch", MC13783_AUDIO_RX0, 10, 9, 1, 0),
SOC_DOUBLE("Line out Amp Switch", MC13783_AUDIO_RX0, 16, 15, 1, 0),
SOC_SINGLE("PCM Capture Mixin Switch", MC13783_AUDIO_RX0, 22, 1, 0),
SOC_SINGLE("Line in Capture Mixin Switch", MC13783_AUDIO_RX0, 23, 1, 0),
SOC_SINGLE("CODEC Capture Volume", MC13783_AUDIO_RX1, 1, 15, 0),
SOC_SINGLE("CODEC Capture Mixin Switch", MC13783_AUDIO_RX0, 21, 1, 0),
SOC_SINGLE("Line in Capture Volume", MC13783_AUDIO_RX1, 12, 15, 0),
SOC_SINGLE("Line in Capture Switch", MC13783_AUDIO_RX1, 10, 1, 0),
SOC_SINGLE("MC1 Capture Bias Switch", MC13783_AUDIO_TX, 0, 1, 0),
SOC_SINGLE("MC2 Capture Bias Switch", MC13783_AUDIO_TX, 1, 1, 0),
};
static int mc13783_probe(struct snd_soc_component *component)
{
struct mc13783_priv *priv = snd_soc_component_get_drvdata(component);
snd_soc_component_init_regmap(component,
dev_get_regmap(component->dev->parent, NULL));
/* these are the reset values */
mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_RX0, 0x25893);
mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_RX1, 0x00d35A);
mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_TX, 0x420000);
mc13xxx_reg_write(priv->mc13xxx, MC13783_SSI_NETWORK, 0x013060);
mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_CODEC, 0x180027);
mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_DAC, 0x0e0004);
if (priv->adc_ssi_port == MC13783_SSI1_PORT)
mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_CODEC,
AUDIO_SSI_SEL, 0);
else
mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_CODEC,
AUDIO_SSI_SEL, AUDIO_SSI_SEL);
if (priv->dac_ssi_port == MC13783_SSI1_PORT)
mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_DAC,
AUDIO_SSI_SEL, 0);
else
mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_DAC,
AUDIO_SSI_SEL, AUDIO_SSI_SEL);
return 0;
}
static void mc13783_remove(struct snd_soc_component *component)
{
struct mc13783_priv *priv = snd_soc_component_get_drvdata(component);
/* Make sure VAUDIOON is off */
mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_RX0, 0x3, 0);
}
#define MC13783_RATES_RECORD (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000)
#define MC13783_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE)
static const struct snd_soc_dai_ops mc13783_ops_dac = {
.hw_params = mc13783_pcm_hw_params_dac,
.set_fmt = mc13783_set_fmt_async,
.set_sysclk = mc13783_set_sysclk_dac,
.set_tdm_slot = mc13783_set_tdm_slot_dac,
};
static const struct snd_soc_dai_ops mc13783_ops_codec = {
.hw_params = mc13783_pcm_hw_params_codec,
.set_fmt = mc13783_set_fmt_async,
.set_sysclk = mc13783_set_sysclk_codec,
.set_tdm_slot = mc13783_set_tdm_slot_codec,
};
/*
* The mc13783 has two SSI ports, both of them can be routed either
* to the voice codec or the stereo DAC. When two different SSI ports
* are used for the voice codec and the stereo DAC we can do different
* formats and sysclock settings for playback and capture
* (mc13783-hifi-playback and mc13783-hifi-capture). Using the same port
* forces us to use symmetric rates (mc13783-hifi).
*/
static struct snd_soc_dai_driver mc13783_dai_async[] = {
{
.name = "mc13783-hifi-playback",
.id = MC13783_ID_STEREO_DAC,
.playback = {
.stream_name = "Playback",
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_96000,
.formats = MC13783_FORMATS,
},
.ops = &mc13783_ops_dac,
}, {
.name = "mc13783-hifi-capture",
.id = MC13783_ID_STEREO_CODEC,
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
.rates = MC13783_RATES_RECORD,
.formats = MC13783_FORMATS,
},
.ops = &mc13783_ops_codec,
},
};
static const struct snd_soc_dai_ops mc13783_ops_sync = {
.hw_params = mc13783_pcm_hw_params_sync,
.set_fmt = mc13783_set_fmt_sync,
.set_sysclk = mc13783_set_sysclk_sync,
.set_tdm_slot = mc13783_set_tdm_slot_sync,
};
static struct snd_soc_dai_driver mc13783_dai_sync[] = {
{
.name = "mc13783-hifi",
.id = MC13783_ID_SYNC,
.playback = {
.stream_name = "Playback",
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_96000,
.formats = MC13783_FORMATS,
},
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
.rates = MC13783_RATES_RECORD,
.formats = MC13783_FORMATS,
},
.ops = &mc13783_ops_sync,
.symmetric_rates = 1,
}
};
static const struct snd_soc_component_driver soc_component_dev_mc13783 = {
.probe = mc13783_probe,
.remove = mc13783_remove,
.controls = mc13783_control_list,
.num_controls = ARRAY_SIZE(mc13783_control_list),
.dapm_widgets = mc13783_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(mc13783_dapm_widgets),
.dapm_routes = mc13783_routes,
.num_dapm_routes = ARRAY_SIZE(mc13783_routes),
.idle_bias_on = 1,
.use_pmdown_time = 1,
.endianness = 1,
.non_legacy_dai_naming = 1,
};
static int __init mc13783_codec_probe(struct platform_device *pdev)
{
struct mc13783_priv *priv;
struct mc13xxx_codec_platform_data *pdata = pdev->dev.platform_data;
struct device_node *np;
int ret;
priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
if (!priv)
return -ENOMEM;
if (pdata) {
priv->adc_ssi_port = pdata->adc_ssi_port;
priv->dac_ssi_port = pdata->dac_ssi_port;
} else {
np = of_get_child_by_name(pdev->dev.parent->of_node, "codec");
if (!np)
return -ENOSYS;
ret = of_property_read_u32(np, "adc-port", &priv->adc_ssi_port);
if (ret) {
of_node_put(np);
return ret;
}
ret = of_property_read_u32(np, "dac-port", &priv->dac_ssi_port);
if (ret) {
of_node_put(np);
return ret;
}
of_node_put(np);
}
dev_set_drvdata(&pdev->dev, priv);
priv->mc13xxx = dev_get_drvdata(pdev->dev.parent);
if (priv->adc_ssi_port == priv->dac_ssi_port)
ret = devm_snd_soc_register_component(&pdev->dev, &soc_component_dev_mc13783,
mc13783_dai_sync, ARRAY_SIZE(mc13783_dai_sync));
else
ret = devm_snd_soc_register_component(&pdev->dev, &soc_component_dev_mc13783,
mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async));
return ret;
}
static int mc13783_codec_remove(struct platform_device *pdev)
{
return 0;
}
static struct platform_driver mc13783_codec_driver = {
.driver = {
.name = "mc13783-codec",
},
.remove = mc13783_codec_remove,
};
module_platform_driver_probe(mc13783_codec_driver, mc13783_codec_probe);
MODULE_DESCRIPTION("ASoC MC13783 driver");
MODULE_AUTHOR("Sascha Hauer, Pengutronix <s.hauer@pengutronix.de>");
MODULE_AUTHOR("Philippe Retornaz <philippe.retornaz@epfl.ch>");
MODULE_LICENSE("GPL");
Reference in New Issue View Git Blame Copy Permalink