iv/linux
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
337d348b83
linux/sound/soc/codecs/mc13783.c
Alexander A. Klimov 5856d8bd30
ASoC: Replace HTTP links with HTTPS ones 
Rationale:
Reduces attack surface on kernel devs opening the links for MITM
as HTTPS traffic is much harder to manipulate.

Deterministic algorithm:
For each file:
  If not .svg:
    For each line:
      If doesn't contain `\bxmlns\b`:
        For each link, `\bhttp://[^# \t\r\n]*(?:\w|/)`:
	  If neither `\bgnu\.org/license`, nor `\bmozilla\.org/MPL\b`:
            If both the HTTP and HTTPS versions
            return 200 OK and serve the same content:
              Replace HTTP with HTTPS.

Signed-off-by: Alexander A. Klimov <grandmaster@al2klimov.de>
Acked-by: Rob Herring <robh@kernel.org>
Link: https://lore.kernel.org/r/20200719153822.59788-1-grandmaster@al2klimov.de
Signed-off-by: Mark Brown <broonie@kernel.org>
2020-07-22 14:31:55 +01:00

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, https://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