android_kernel_google_msm/sound/soc/msm/msm8974.c
Duy Truong 04e554807c Update copyright to The Linux Foundation
Change-Id: Ibead64ce2e901dede2ddd1b86088b88f2350ce92
Signed-off-by: Duy Truong <dtruong@codeaurora.org>
2013-03-15 17:07:39 -07:00

1009 lines
27 KiB
C

/* Copyright (c) 2012, The Linux Foundation. All rights reserved.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 and
* only version 2 as published by the Free Software Foundation.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*/
#include <linux/clk.h>
#include <linux/delay.h>
#include <linux/gpio.h>
#include <linux/mfd/pm8xxx/pm8921.h>
#include <linux/platform_device.h>
#include <linux/gpio.h>
#include <linux/mfd/pm8xxx/pm8921.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/pcm.h>
#include <sound/jack.h>
#include <asm/mach-types.h>
#include <mach/socinfo.h>
#include <qdsp6v2/msm-pcm-routing-v2.h>
#include "../codecs/wcd9320.h"
/* 8974 machine driver */
#define PM8921_GPIO_BASE NR_GPIO_IRQS
#define PM8921_GPIO_PM_TO_SYS(pm_gpio) (pm_gpio - 1 + PM8921_GPIO_BASE)
#define MSM8974_SPK_ON 1
#define MSM8974_SPK_OFF 0
#define MSM_SLIM_0_RX_MAX_CHANNELS 2
#define MSM_SLIM_0_TX_MAX_CHANNELS 4
#define BTSCO_RATE_8KHZ 8000
#define BTSCO_RATE_16KHZ 16000
#define BOTTOM_SPK_AMP_POS 0x1
#define BOTTOM_SPK_AMP_NEG 0x2
#define TOP_SPK_AMP_POS 0x4
#define TOP_SPK_AMP_NEG 0x8
#define GPIO_AUX_PCM_DOUT 43
#define GPIO_AUX_PCM_DIN 44
#define GPIO_AUX_PCM_SYNC 45
#define GPIO_AUX_PCM_CLK 46
#define TABLA_EXT_CLK_RATE 12288000
#define TABLA_MBHC_DEF_BUTTONS 8
#define TABLA_MBHC_DEF_RLOADS 5
/* Shared channel numbers for Slimbus ports that connect APQ to MDM. */
enum {
SLIM_1_RX_1 = 145, /* BT-SCO and USB TX */
SLIM_1_TX_1 = 146, /* BT-SCO and USB RX */
SLIM_2_RX_1 = 147, /* HDMI RX */
SLIM_3_RX_1 = 148, /* In-call recording RX */
SLIM_3_RX_2 = 149, /* In-call recording RX */
SLIM_4_TX_1 = 150, /* In-call musid delivery TX */
};
static u32 top_spk_pamp_gpio = PM8921_GPIO_PM_TO_SYS(18);
static u32 bottom_spk_pamp_gpio = PM8921_GPIO_PM_TO_SYS(19);
static int msm_spk_control;
static int msm_ext_bottom_spk_pamp;
static int msm_ext_top_spk_pamp;
static int msm_slim_0_rx_ch = 1;
static int msm_slim_0_tx_ch = 1;
static int msm_btsco_rate = BTSCO_RATE_8KHZ;
static int msm_headset_gpios_configured;
static struct snd_soc_jack hs_jack;
static struct snd_soc_jack button_jack;
static struct mutex cdc_mclk_mutex;
static struct q_clkdiv *codec_clk;
static int clk_users;
static atomic_t auxpcm_rsc_ref;
static void msm_enable_ext_spk_amp_gpio(u32 spk_amp_gpio)
{
int ret = 0;
struct pm_gpio param = {
.direction = PM_GPIO_DIR_OUT,
.output_buffer = PM_GPIO_OUT_BUF_CMOS,
.output_value = 1,
.pull = PM_GPIO_PULL_NO,
.vin_sel = PM_GPIO_VIN_S4,
.out_strength = PM_GPIO_STRENGTH_MED,
.
function = PM_GPIO_FUNC_NORMAL,
};
if (spk_amp_gpio == bottom_spk_pamp_gpio) {
ret = gpio_request(bottom_spk_pamp_gpio, "BOTTOM_SPK_AMP");
if (ret) {
pr_err("%s: Error requesting BOTTOM SPK AMP GPIO %u\n",
__func__, bottom_spk_pamp_gpio);
return;
}
ret = pm8xxx_gpio_config(bottom_spk_pamp_gpio, &param);
if (ret)
pr_err("%s: Failed to configure Bottom Spk Ampl gpio %u\n",
__func__, bottom_spk_pamp_gpio);
else {
pr_debug("%s: enable Bottom spkr amp gpio\n", __func__);
gpio_direction_output(bottom_spk_pamp_gpio, 1);
}
} else if (spk_amp_gpio == top_spk_pamp_gpio) {
ret = gpio_request(top_spk_pamp_gpio, "TOP_SPK_AMP");
if (ret) {
pr_err("%s: Error requesting GPIO %d\n", __func__,
top_spk_pamp_gpio);
return;
}
ret = pm8xxx_gpio_config(top_spk_pamp_gpio, &param);
if (ret)
pr_err("%s: Failed to configure Top Spk Ampl gpio %u\n",
__func__, top_spk_pamp_gpio);
else {
pr_debug("%s: enable Top spkr amp gpio\n", __func__);
gpio_direction_output(top_spk_pamp_gpio, 1);
}
} else {
pr_err("%s: ERROR : Invalid External Speaker Ampl GPIO. gpio = %u\n",
__func__, spk_amp_gpio);
return;
}
}
static void msm_ext_spk_power_amp_on(u32 spk)
{
if (spk & (BOTTOM_SPK_AMP_POS | BOTTOM_SPK_AMP_NEG)) {
if ((msm_ext_bottom_spk_pamp & BOTTOM_SPK_AMP_POS) &&
(msm_ext_bottom_spk_pamp & BOTTOM_SPK_AMP_NEG)) {
pr_debug("%s() External Bottom Speaker Ampl already turned on. spk = 0x%08x\n",
__func__, spk);
return;
}
msm_ext_bottom_spk_pamp |= spk;
if ((msm_ext_bottom_spk_pamp & BOTTOM_SPK_AMP_POS) &&
(msm_ext_bottom_spk_pamp & BOTTOM_SPK_AMP_NEG)) {
msm_enable_ext_spk_amp_gpio(bottom_spk_pamp_gpio);
pr_debug("%s: slepping 4 ms after turning on external Bottom Speaker Ampl\n",
__func__);
usleep_range(4000, 4000);
}
} else if (spk & (TOP_SPK_AMP_POS | TOP_SPK_AMP_NEG)) {
if ((msm_ext_top_spk_pamp & TOP_SPK_AMP_POS) &&
(msm_ext_top_spk_pamp & TOP_SPK_AMP_NEG)) {
pr_debug("%s() External Top Speaker Ampl already turned on. spk = 0x%08x\n",
__func__, spk);
return;
}
msm_ext_top_spk_pamp |= spk;
if ((msm_ext_top_spk_pamp & TOP_SPK_AMP_POS) &&
(msm_ext_top_spk_pamp & TOP_SPK_AMP_NEG)) {
msm_enable_ext_spk_amp_gpio(top_spk_pamp_gpio);
pr_debug("%s: sleeping 4 ms after turning on external Top Speaker Ampl\n",
__func__);
usleep_range(4000, 4000);
}
} else {
pr_err("%s: ERROR : Invalid External Speaker Ampl. spk = 0x%08x\n",
__func__, spk);
return;
}
}
static void msm_ext_spk_power_amp_off(u32 spk)
{
if (spk & (BOTTOM_SPK_AMP_POS | BOTTOM_SPK_AMP_NEG)) {
if (!msm_ext_bottom_spk_pamp)
return;
gpio_direction_output(bottom_spk_pamp_gpio, 0);
gpio_free(bottom_spk_pamp_gpio);
msm_ext_bottom_spk_pamp = 0;
pr_debug("%s: sleeping 4 ms after turning off external Bottom Speaker Ampl\n",
__func__);
usleep_range(4000, 4000);
} else if (spk & (TOP_SPK_AMP_POS | TOP_SPK_AMP_NEG)) {
if (!msm_ext_top_spk_pamp)
return;
gpio_direction_output(top_spk_pamp_gpio, 0);
gpio_free(top_spk_pamp_gpio);
msm_ext_top_spk_pamp = 0;
pr_debug("%s: sleeping 4 ms after turning off external Top Spkaker Ampl\n",
__func__);
usleep_range(4000, 4000);
} else {
pr_err("%s: ERROR : Invalid Ext Spk Ampl. spk = 0x%08x\n",
__func__, spk);
return;
}
}
static void msm_ext_control(struct snd_soc_codec *codec)
{
struct snd_soc_dapm_context *dapm = &codec->dapm;
mutex_lock(&dapm->codec->mutex);
pr_debug("%s: msm_spk_control = %d", __func__, msm_spk_control);
if (msm_spk_control == MSM8974_SPK_ON) {
snd_soc_dapm_enable_pin(dapm, "Ext Spk Bottom Pos");
snd_soc_dapm_enable_pin(dapm, "Ext Spk Bottom Neg");
snd_soc_dapm_enable_pin(dapm, "Ext Spk Top Pos");
snd_soc_dapm_enable_pin(dapm, "Ext Spk Top Neg");
} else {
snd_soc_dapm_disable_pin(dapm, "Ext Spk Bottom Pos");
snd_soc_dapm_disable_pin(dapm, "Ext Spk Bottom Neg");
snd_soc_dapm_disable_pin(dapm, "Ext Spk Top Pos");
snd_soc_dapm_disable_pin(dapm, "Ext Spk Top Neg");
}
snd_soc_dapm_sync(dapm);
mutex_unlock(&dapm->codec->mutex);
}
static int msm_get_spk(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
pr_debug("%s: msm_spk_control = %d", __func__, msm_spk_control);
ucontrol->value.integer.value[0] = msm_spk_control;
return 0;
}
static int msm_set_spk(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
pr_debug("%s()\n", __func__);
if (msm_spk_control == ucontrol->value.integer.value[0])
return 0;
msm_spk_control = ucontrol->value.integer.value[0];
msm_ext_control(codec);
return 1;
}
static int msm_spkramp_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
pr_debug("%s() %x\n", __func__, SND_SOC_DAPM_EVENT_ON(event));
if (SND_SOC_DAPM_EVENT_ON(event)) {
if (!strncmp(w->name, "Ext Spk Bottom Pos", 18))
msm_ext_spk_power_amp_on(BOTTOM_SPK_AMP_POS);
else if (!strncmp(w->name, "Ext Spk Bottom Neg", 18))
msm_ext_spk_power_amp_on(BOTTOM_SPK_AMP_NEG);
else if (!strncmp(w->name, "Ext Spk Top Pos", 15))
msm_ext_spk_power_amp_on(TOP_SPK_AMP_POS);
else if (!strncmp(w->name, "Ext Spk Top Neg", 15))
msm_ext_spk_power_amp_on(TOP_SPK_AMP_NEG);
else {
pr_err("%s() Invalid Speaker Widget = %s\n",
__func__, w->name);
return -EINVAL;
}
} else {
if (!strncmp(w->name, "Ext Spk Bottom Pos", 18))
msm_ext_spk_power_amp_off(BOTTOM_SPK_AMP_POS);
else if (!strncmp(w->name, "Ext Spk Bottom Neg", 18))
msm_ext_spk_power_amp_off(BOTTOM_SPK_AMP_NEG);
else if (!strncmp(w->name, "Ext Spk Top Pos", 15))
msm_ext_spk_power_amp_off(TOP_SPK_AMP_POS);
else if (!strncmp(w->name, "Ext Spk Top Neg", 15))
msm_ext_spk_power_amp_off(TOP_SPK_AMP_NEG);
else {
pr_err("%s() Invalid Speaker Widget = %s\n",
__func__, w->name);
return -EINVAL;
}
}
return 0;
}
static int msm_mclk_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
return 0;
}
static const struct snd_soc_dapm_widget msm_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY("MCLK", SND_SOC_NOPM, 0, 0,
msm_mclk_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_SPK("Ext Spk Bottom Pos", msm_spkramp_event),
SND_SOC_DAPM_SPK("Ext Spk Bottom Neg", msm_spkramp_event),
SND_SOC_DAPM_SPK("Ext Spk Top Pos", msm_spkramp_event),
SND_SOC_DAPM_SPK("Ext Spk Top Neg", msm_spkramp_event),
SND_SOC_DAPM_MIC("Handset Mic", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
SND_SOC_DAPM_MIC("Digital Mic1", NULL),
SND_SOC_DAPM_MIC("ANCRight Headset Mic", NULL),
SND_SOC_DAPM_MIC("ANCLeft Headset Mic", NULL),
SND_SOC_DAPM_MIC("Digital Mic1", NULL),
SND_SOC_DAPM_MIC("Digital Mic2", NULL),
SND_SOC_DAPM_MIC("Digital Mic3", NULL),
SND_SOC_DAPM_MIC("Digital Mic4", NULL),
SND_SOC_DAPM_MIC("Digital Mic5", NULL),
SND_SOC_DAPM_MIC("Digital Mic6", NULL),
};
static const struct snd_soc_dapm_route common_audio_map[] = {
{"RX_BIAS", NULL, "MCLK"},
{"LDO_H", NULL, "MCLK"},
/* Speaker path */
{"Ext Spk Bottom Pos", NULL, "LINEOUT1"},
{"Ext Spk Bottom Neg", NULL, "LINEOUT3"},
{"Ext Spk Top Pos", NULL, "LINEOUT2"},
{"Ext Spk Top Neg", NULL, "LINEOUT4"},
/* Microphone path */
{"AMIC1", NULL, "MIC BIAS1 Internal1"},
{"MIC BIAS1 Internal1", NULL, "Handset Mic"},
{"AMIC2", NULL, "MIC BIAS2 External"},
{"MIC BIAS2 External", NULL, "Headset Mic"},
/**
* AMIC3 and AMIC4 inputs are connected to ANC microphones
* These mics are biased differently on CDP and FLUID
* routing entries below are based on bias arrangement
* on FLUID.
*/
{"AMIC3", NULL, "MIC BIAS3 Internal1"},
{"MIC BIAS3 Internal1", NULL, "ANCRight Headset Mic"},
{"AMIC4", NULL, "MIC BIAS1 Internal2"},
{"MIC BIAS1 Internal2", NULL, "ANCLeft Headset Mic"},
{"HEADPHONE", NULL, "LDO_H"},
/**
* The digital Mic routes are setup considering
* fluid as default device.
*/
/**
* Digital Mic1. Front Bottom left Digital Mic on Fluid and MTP.
* Digital Mic GM5 on CDP mainboard.
* Conncted to DMIC2 Input on Tabla codec.
*/
{"DMIC2", NULL, "MIC BIAS1 External"},
{"MIC BIAS1 External", NULL, "Digital Mic1"},
/**
* Digital Mic2. Front Bottom right Digital Mic on Fluid and MTP.
* Digital Mic GM6 on CDP mainboard.
* Conncted to DMIC1 Input on Tabla codec.
*/
{"DMIC1", NULL, "MIC BIAS1 External"},
{"MIC BIAS1 External", NULL, "Digital Mic2"},
/**
* Digital Mic3. Back Bottom Digital Mic on Fluid.
* Digital Mic GM1 on CDP mainboard.
* Conncted to DMIC4 Input on Tabla codec.
*/
{"DMIC4", NULL, "MIC BIAS3 External"},
{"MIC BIAS3 External", NULL, "Digital Mic3"},
/**
* Digital Mic4. Back top Digital Mic on Fluid.
* Digital Mic GM2 on CDP mainboard.
* Conncted to DMIC3 Input on Tabla codec.
*/
{"DMIC3", NULL, "MIC BIAS3 External"},
{"MIC BIAS3 External", NULL, "Digital Mic4"},
/**
* Digital Mic5. Front top Digital Mic on Fluid.
* Digital Mic GM3 on CDP mainboard.
* Conncted to DMIC5 Input on Tabla codec.
*/
{"DMIC5", NULL, "MIC BIAS4 External"},
{"MIC BIAS4 External", NULL, "Digital Mic5"},
/* Tabla digital Mic6 - back bottom digital Mic on Liquid and
* bottom mic on CDP. FLUID/MTP do not have dmic6 installed.
*/
{"DMIC6", NULL, "MIC BIAS4 External"},
{"MIC BIAS4 External", NULL, "Digital Mic6"},
};
static const char *const spk_function[] = {"Off", "On"};
static const char *const slim0_rx_ch_text[] = {"One", "Two"};
static const char *const slim0_tx_ch_text[] = {"One", "Two", "Three", "Four"};
static const struct soc_enum msm_enum[] = {
SOC_ENUM_SINGLE_EXT(2, spk_function),
SOC_ENUM_SINGLE_EXT(2, slim0_rx_ch_text),
SOC_ENUM_SINGLE_EXT(4, slim0_tx_ch_text),
};
static const char *const btsco_rate_text[] = {"8000", "16000"};
static const struct soc_enum msm_btsco_enum[] = {
SOC_ENUM_SINGLE_EXT(2, btsco_rate_text),
};
static int msm_slim_0_rx_ch_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
pr_debug("%s: msm_slim_0_rx_ch = %d\n", __func__,
msm_slim_0_rx_ch);
ucontrol->value.integer.value[0] = msm_slim_0_rx_ch - 1;
return 0;
}
static int msm_slim_0_rx_ch_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
msm_slim_0_rx_ch = ucontrol->value.integer.value[0] + 1;
pr_debug("%s: msm_slim_0_rx_ch = %d\n", __func__,
msm_slim_0_rx_ch);
return 1;
}
static int msm_slim_0_tx_ch_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
pr_debug("%s: msm_slim_0_tx_ch = %d\n", __func__,
msm_slim_0_tx_ch);
ucontrol->value.integer.value[0] = msm_slim_0_tx_ch - 1;
return 0;
}
static int msm_slim_0_tx_ch_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
msm_slim_0_tx_ch = ucontrol->value.integer.value[0] + 1;
pr_debug("%s: msm_slim_0_tx_ch = %d\n", __func__, msm_slim_0_tx_ch);
return 1;
}
static int msm_btsco_rate_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
pr_debug("%s: msm_btsco_rate = %d", __func__, msm_btsco_rate);
ucontrol->value.integer.value[0] = msm_btsco_rate;
return 0;
}
static int msm_btsco_rate_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
switch (ucontrol->value.integer.value[0]) {
case 0:
msm_btsco_rate = BTSCO_RATE_8KHZ;
break;
case 1:
msm_btsco_rate = BTSCO_RATE_16KHZ;
break;
default:
msm_btsco_rate = BTSCO_RATE_8KHZ;
break;
}
pr_debug("%s: msm_btsco_rate = %d\n", __func__, msm_btsco_rate);
return 0;
}
static const struct snd_kcontrol_new int_btsco_rate_mixer_controls[] = {
SOC_ENUM_EXT("Internal BTSCO SampleRate", msm_btsco_enum[0],
msm_btsco_rate_get, msm_btsco_rate_put),
};
static int msm_auxpcm_be_params_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
struct snd_interval *rate =
hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval *channels =
hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
/* PCM only supports mono output with 8khz sample rate */
rate->min = rate->max = 8000;
channels->min = channels->max = 1;
return 0;
}
static int msm_aux_pcm_get_gpios(void)
{
int ret = 0;
pr_debug("%s\n", __func__);
ret = gpio_request(GPIO_AUX_PCM_DOUT, "AUX PCM DOUT");
if (ret < 0) {
pr_err("%s: Failed to request gpio(%d): AUX PCM DOUT",
__func__, GPIO_AUX_PCM_DOUT);
goto fail_dout;
}
ret = gpio_request(GPIO_AUX_PCM_DIN, "AUX PCM DIN");
if (ret < 0) {
pr_err("%s: Failed to request gpio(%d): AUX PCM DIN",
__func__, GPIO_AUX_PCM_DIN);
goto fail_din;
}
ret = gpio_request(GPIO_AUX_PCM_SYNC, "AUX PCM SYNC");
if (ret < 0) {
pr_err("%s: Failed to request gpio(%d): AUX PCM SYNC",
__func__, GPIO_AUX_PCM_SYNC);
goto fail_sync;
}
ret = gpio_request(GPIO_AUX_PCM_CLK, "AUX PCM CLK");
if (ret < 0) {
pr_err("%s: Failed to request gpio(%d): AUX PCM CLK",
__func__, GPIO_AUX_PCM_CLK);
goto fail_clk;
}
return 0;
fail_clk:
gpio_free(GPIO_AUX_PCM_SYNC);
fail_sync:
gpio_free(GPIO_AUX_PCM_DIN);
fail_din:
gpio_free(GPIO_AUX_PCM_DOUT);
fail_dout:
return ret;
}
static int msm_aux_pcm_free_gpios(void)
{
gpio_free(GPIO_AUX_PCM_DIN);
gpio_free(GPIO_AUX_PCM_DOUT);
gpio_free(GPIO_AUX_PCM_SYNC);
gpio_free(GPIO_AUX_PCM_CLK);
return 0;
}
static int msm_auxpcm_startup(struct snd_pcm_substream *substream)
{
int ret = 0;
pr_debug("%s(): substream = %s, auxpcm_rsc_ref counter = %d\n",
__func__, substream->name, atomic_read(&auxpcm_rsc_ref));
if (atomic_inc_return(&auxpcm_rsc_ref) == 1)
ret = msm_aux_pcm_get_gpios();
if (ret < 0) {
pr_err("%s: Aux PCM GPIO request failed\n", __func__);
return -EINVAL;
}
return ret;
}
static void msm_auxpcm_shutdown(struct snd_pcm_substream *substream)
{
pr_debug("%s(): substream = %s, auxpcm_rsc_ref counter = %d\n",
__func__, substream->name, atomic_read(&auxpcm_rsc_ref));
if (atomic_dec_return(&auxpcm_rsc_ref) == 0)
msm_aux_pcm_free_gpios();
}
static struct snd_soc_ops msm_auxpcm_be_ops = {
.startup = msm_auxpcm_startup,
.shutdown = msm_auxpcm_shutdown,
};
static int msm_slim_0_rx_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
struct snd_interval *rate = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval *channels =
hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
pr_debug("%s()\n", __func__);
rate->min = rate->max = 48000;
channels->min = channels->max = msm_slim_0_rx_ch;
return 0;
}
static int msm_slim_0_tx_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
struct snd_interval *rate = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval *channels = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
pr_debug("%s()\n", __func__);
rate->min = rate->max = 48000;
channels->min = channels->max = msm_slim_0_tx_ch;
return 0;
}
static const struct soc_enum msm_snd_enum[] = {
SOC_ENUM_SINGLE_EXT(2, spk_function),
SOC_ENUM_SINGLE_EXT(2, slim0_rx_ch_text),
SOC_ENUM_SINGLE_EXT(4, slim0_tx_ch_text),
};
static const struct snd_kcontrol_new msm_snd_controls[] = {
SOC_ENUM_EXT("Speaker Function", msm_snd_enum[0], msm_get_spk,
msm_set_spk),
SOC_ENUM_EXT("SLIM_0_RX Channels", msm_snd_enum[1],
msm_slim_0_rx_ch_get, msm_slim_0_rx_ch_put),
SOC_ENUM_EXT("SLIM_0_TX Channels", msm_snd_enum[2],
msm_slim_0_tx_ch_get, msm_slim_0_tx_ch_put),
};
static int msm_audrx_init(struct snd_soc_pcm_runtime *rtd)
{
int err;
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
pr_info("%s(), dev_name%s\n", __func__, dev_name(cpu_dai->dev));
if (machine_is_msm8960_liquid()) {
top_spk_pamp_gpio = (PM8921_GPIO_PM_TO_SYS(19));
bottom_spk_pamp_gpio = (PM8921_GPIO_PM_TO_SYS(18));
}
rtd->pmdown_time = 0;
err = snd_soc_add_codec_controls(codec, msm_snd_controls,
ARRAY_SIZE(msm_snd_controls));
if (err < 0)
return err;
snd_soc_dapm_new_controls(dapm, msm_dapm_widgets,
ARRAY_SIZE(msm_dapm_widgets));
snd_soc_dapm_add_routes(dapm, common_audio_map,
ARRAY_SIZE(common_audio_map));
snd_soc_dapm_enable_pin(dapm, "Ext Spk Bottom Pos");
snd_soc_dapm_enable_pin(dapm, "Ext Spk Bottom Neg");
snd_soc_dapm_enable_pin(dapm, "Ext Spk Top Pos");
snd_soc_dapm_enable_pin(dapm, "Ext Spk Top Neg");
snd_soc_dapm_sync(dapm);
err = snd_soc_jack_new(codec, "Headset Jack",
(SND_JACK_HEADSET | SND_JACK_OC_HPHL |
SND_JACK_OC_HPHR | SND_JACK_UNSUPPORTED),
&hs_jack);
if (err) {
pr_err("failed to create new jack\n");
return err;
}
err = snd_soc_jack_new(codec, "Button Jack",
TAIKO_JACK_BUTTON_MASK, &button_jack);
if (err) {
pr_err("failed to create new jack\n");
return err;
}
return err;
}
static int msm_snd_startup(struct snd_pcm_substream *substream)
{
pr_debug("%s(): substream = %s stream = %d\n", __func__,
substream->name, substream->stream);
return 0;
}
static int msm_snd_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
int ret = 0;
unsigned int rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS];
unsigned int rx_ch_cnt = 0, tx_ch_cnt = 0;
unsigned int user_set_tx_ch = 0;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
pr_debug("%s: rx_0_ch=%d\n", __func__, msm_slim_0_rx_ch);
ret = snd_soc_dai_get_channel_map(codec_dai,
&tx_ch_cnt, tx_ch, &rx_ch_cnt , rx_ch);
if (ret < 0) {
pr_err("%s: failed to get codec chan map\n", __func__);
goto end;
}
ret = snd_soc_dai_set_channel_map(cpu_dai, 0, 0,
msm_slim_0_rx_ch, rx_ch);
if (ret < 0) {
pr_err("%s: failed to set cpu chan map\n", __func__);
goto end;
}
ret = snd_soc_dai_set_channel_map(codec_dai, 0, 0,
msm_slim_0_rx_ch, rx_ch);
if (ret < 0) {
pr_err("%s: failed to set codec channel map\n",
__func__);
goto end;
}
} else {
if (codec_dai->id == 2)
user_set_tx_ch = msm_slim_0_tx_ch;
else if (codec_dai->id == 4)
user_set_tx_ch = params_channels(params);
pr_debug("%s: %s_tx_dai_id_%d_ch=%d\n", __func__,
codec_dai->name, codec_dai->id, user_set_tx_ch);
ret = snd_soc_dai_get_channel_map(codec_dai,
&tx_ch_cnt, tx_ch, &rx_ch_cnt , rx_ch);
if (ret < 0) {
pr_err("%s: failed to get codec chan map\n", __func__);
goto end;
}
ret = snd_soc_dai_set_channel_map(cpu_dai,
user_set_tx_ch, tx_ch, 0 , 0);
if (ret < 0) {
pr_err("%s: failed to set cpu chan map\n", __func__);
goto end;
}
ret = snd_soc_dai_set_channel_map(codec_dai,
user_set_tx_ch, tx_ch, 0, 0);
if (ret < 0) {
pr_err("%s: failed to set codec channel map\n",
__func__);
goto end;
}
}
end:
return ret;
}
static void msm_snd_shutdown(struct snd_pcm_substream *substream)
{
pr_debug("%s(): substream = %s stream = %d\n", __func__,
substream->name, substream->stream);
}
static struct snd_soc_ops msm8974_be_ops = {
.startup = msm_snd_startup,
.hw_params = msm_snd_hw_params,
.shutdown = msm_snd_shutdown,
};
/* Digital audio interface glue - connects codec <---> CPU */
static struct snd_soc_dai_link msm_dai[] = {
/* FrontEnd DAI Links */
{
.name = "MSM8974 Media1",
.stream_name = "MultiMedia1",
.cpu_dai_name = "MultiMedia1",
.platform_name = "msm-pcm-dsp",
.dynamic = 1,
.trigger = {SND_SOC_DPCM_TRIGGER_POST,
SND_SOC_DPCM_TRIGGER_POST},
.codec_dai_name = "snd-soc-dummy-dai",
.codec_name = "snd-soc-dummy",
.ignore_suspend = 1,
/* this dainlink has playback support */
.ignore_pmdown_time = 1,
.be_id = MSM_FRONTEND_DAI_MULTIMEDIA1
},
{
.name = "MSM VoIP",
.stream_name = "VoIP",
.cpu_dai_name = "VoIP",
.platform_name = "msm-voip-dsp",
.dynamic = 1,
.trigger = {SND_SOC_DPCM_TRIGGER_POST,
SND_SOC_DPCM_TRIGGER_POST},
.codec_dai_name = "snd-soc-dummy-dai",
.codec_name = "snd-soc-dummy",
.ignore_suspend = 1,
/* this dainlink has playback support */
.ignore_pmdown_time = 1,
.be_id = MSM_FRONTEND_DAI_VOIP,
},
{
.name = "MSM8974 LPA",
.stream_name = "LPA",
.cpu_dai_name = "MultiMedia3",
.platform_name = "msm-pcm-lpa",
.dynamic = 1,
.trigger = {SND_SOC_DPCM_TRIGGER_POST,
SND_SOC_DPCM_TRIGGER_POST},
.codec_dai_name = "snd-soc-dummy-dai",
.codec_name = "snd-soc-dummy",
.ignore_suspend = 1,
/* this dainlink has playback support */
.ignore_pmdown_time = 1,
.be_id = MSM_FRONTEND_DAI_MULTIMEDIA3,
},
/* Hostless PCM purpose */
{
.name = "SLIMBUS_0 Hostless",
.stream_name = "SLIMBUS_0 Hostless",
.cpu_dai_name = "SLIMBUS0_HOSTLESS",
.platform_name = "msm-pcm-hostless",
.dynamic = 1,
.trigger = {SND_SOC_DPCM_TRIGGER_POST,
SND_SOC_DPCM_TRIGGER_POST},
.no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
.ignore_suspend = 1,
.ignore_pmdown_time = 1, /* dai link has playback support */
.codec_dai_name = "snd-soc-dummy-dai",
.codec_name = "snd-soc-dummy",
},
{
.name = "MSM8974 Compr",
.stream_name = "COMPR",
.cpu_dai_name = "MultiMedia4",
.platform_name = "msm-compr-dsp",
.dynamic = 1,
.trigger = {SND_SOC_DPCM_TRIGGER_POST,
SND_SOC_DPCM_TRIGGER_POST},
.codec_dai_name = "snd-soc-dummy-dai",
.codec_name = "snd-soc-dummy",
.ignore_suspend = 1,
.ignore_pmdown_time = 1,
/* this dainlink has playback support */
.be_id = MSM_FRONTEND_DAI_MULTIMEDIA4,
},
{
.name = "AUXPCM Hostless",
.stream_name = "AUXPCM Hostless",
.cpu_dai_name = "AUXPCM_HOSTLESS",
.platform_name = "msm-pcm-hostless",
.dynamic = 1,
.trigger = {SND_SOC_DPCM_TRIGGER_POST,
SND_SOC_DPCM_TRIGGER_POST},
.no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
.ignore_suspend = 1,
/* this dainlink has playback support */
.ignore_pmdown_time = 1,
.codec_dai_name = "snd-soc-dummy-dai",
.codec_name = "snd-soc-dummy",
},
/* AUX PCM Backend DAI Links */
{
.name = LPASS_BE_AUXPCM_RX,
.stream_name = "AUX PCM Playback",
.cpu_dai_name = "msm-dai-q6.4106",
.platform_name = "msm-pcm-routing",
.codec_name = "msm-stub-codec.1",
.codec_dai_name = "msm-stub-rx",
.no_pcm = 1,
.be_id = MSM_BACKEND_DAI_AUXPCM_RX,
.be_hw_params_fixup = msm_auxpcm_be_params_fixup,
.ops = &msm_auxpcm_be_ops,
.ignore_pmdown_time = 1,
/* this dainlink has playback support */
},
{
.name = LPASS_BE_AUXPCM_TX,
.stream_name = "AUX PCM Capture",
.cpu_dai_name = "msm-dai-q6.4107",
.platform_name = "msm-pcm-routing",
.codec_name = "msm-stub-codec.1",
.codec_dai_name = "msm-stub-tx",
.no_pcm = 1,
.be_id = MSM_BACKEND_DAI_AUXPCM_TX,
.be_hw_params_fixup = msm_auxpcm_be_params_fixup,
.ops = &msm_auxpcm_be_ops,
},
/* Backend DAI Links */
{
.name = LPASS_BE_SLIMBUS_0_RX,
.stream_name = "Slimbus Playback",
.cpu_dai_name = "msm-dai-q6-dev.16384",
.platform_name = "msm-pcm-routing",
.codec_name = "taiko_codec",
.codec_dai_name = "taiko_rx1",
.no_pcm = 1,
.be_id = MSM_BACKEND_DAI_SLIMBUS_0_RX,
.init = &msm_audrx_init,
.be_hw_params_fixup = msm_slim_0_rx_be_hw_params_fixup,
.ops = &msm8974_be_ops,
.ignore_pmdown_time = 1, /* dai link has playback support */
},
{
.name = LPASS_BE_SLIMBUS_0_TX,
.stream_name = "Slimbus Capture",
.cpu_dai_name = "msm-dai-q6-dev.16385",
.platform_name = "msm-pcm-routing",
.codec_name = "taiko_codec",
.codec_dai_name = "taiko_tx1",
.no_pcm = 1,
.be_id = MSM_BACKEND_DAI_SLIMBUS_0_TX,
.be_hw_params_fixup = msm_slim_0_tx_be_hw_params_fixup,
.ops = &msm8974_be_ops,
},
};
static struct snd_soc_card snd_soc_card_msm = {
.name = "msm8974-taiko-snd-card",
.dai_link = msm_dai,
.num_links = ARRAY_SIZE(msm_dai),
};
static struct platform_device *msm_snd_device;
static void msm_free_headset_mic_gpios(void)
{
if (msm_headset_gpios_configured) {
gpio_free(PM8921_GPIO_PM_TO_SYS(23));
gpio_free(PM8921_GPIO_PM_TO_SYS(35));
}
}
static int __init msm_audio_init(void)
{
int ret = 0;
mutex_init(&cdc_mclk_mutex);
if (!machine_is_msm8974_sim()) {
pr_info("%s: Not msm8974 machine type\n", __func__);
return -ENODEV;
}
msm_snd_device = platform_device_alloc("soc-audio", 0);
if (!msm_snd_device) {
pr_err("Platform device allocation failed\n");
return -ENOMEM;
}
platform_set_drvdata(msm_snd_device, &snd_soc_card_msm);
ret = platform_device_add(msm_snd_device);
if (ret) {
platform_device_put(msm_snd_device);
return ret;
}
mutex_init(&cdc_mclk_mutex);
atomic_set(&auxpcm_rsc_ref, 0);
return 0;
err:
devm_kfree(&pdev->dev, pdata);
return ret;
}
module_init(msm_audio_init);
static void __exit msm_audio_exit(void)
{
if (!machine_is_msm8974_sim()) {
pr_err("%s: Not the right machine type\n", __func__);
return ;
}
msm_free_headset_mic_gpios();
platform_device_unregister(msm_snd_device);
}
module_exit(msm_audio_exit);
MODULE_DESCRIPTION("ALSA SoC msm");
MODULE_LICENSE("GPL v2");