diff options
Diffstat (limited to 'sound')
26 files changed, 688 insertions, 1736 deletions
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b228cc13191a..2b331f7266ab 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -136,7 +136,6 @@ 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 @@ -842,9 +841,6 @@ 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" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fcc5073f6d61..da1571336f1e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -144,7 +144,6 @@ 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 diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index b88a1ee66f80..c88f974ebe3e 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -107,8 +107,30 @@ static const struct snd_soc_codec_driver soc_dmic = { static int dmic_dev_probe(struct platform_device *pdev) { + int err; + u32 chans; + struct snd_soc_dai_driver *dai_drv = &dmic_dai; + + if (pdev->dev.of_node) { + err = of_property_read_u32(pdev->dev.of_node, "num-channels", &chans); + if (err && (err != -ENOENT)) + return err; + + if (!err) { + if (chans < 1 || chans > 8) + return -EINVAL; + + dai_drv = devm_kzalloc(&pdev->dev, sizeof(*dai_drv), GFP_KERNEL); + if (!dai_drv) + return -ENOMEM; + + memcpy(dai_drv, &dmic_dai, sizeof(*dai_drv)); + dai_drv->capture.channels_max = chans; + } + } + return snd_soc_register_codec(&pdev->dev, - &soc_dmic, &dmic_dai, 1); + &soc_dmic, dai_drv, 1); } static int dmic_dev_remove(struct platform_device *pdev) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 9af0d985d6e9..31b0864583e8 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -176,6 +176,7 @@ static int max98373_get_bclk_sel(int bclk) } return 0; } + static int max98373_set_clock(struct snd_soc_codec *codec, struct snd_pcm_hw_params *params) { @@ -270,6 +271,7 @@ static int max98373_dai_hw_params(struct snd_pcm_substream *substream, params_rate(params)); goto err; } + /* set DAI_SR to correct LRCLK frequency */ regmap_update_bits(max98373->regmap, MAX98373_R2027_PCM_SR_SETUP_1, @@ -309,7 +311,10 @@ static int max98373_dai_tdm_slot(struct snd_soc_dai *dai, unsigned int mask; int x, slot_found; - max98373->tdm_mode = true; + if (!tx_mask && !rx_mask && !slots && !slot_width) + max98373->tdm_mode = false; + else + max98373->tdm_mode = true; /* BCLK configuration */ bsel = max98373_get_bclk_sel(slots * slot_width); @@ -606,13 +611,13 @@ SOC_ENUM("Output Voltage", max98373_out_volt_enum), /* Dynamic Headroom Tracking */ SOC_SINGLE("DHT Switch", MAX98373_R20D4_DHT_EN, MAX98373_DHT_EN_SHIFT, 1, 0), -SOC_SINGLE_TLV("DHT Gain Min", MAX98373_R20D1_DHT_CFG, +SOC_SINGLE_TLV("DHT Min Volume", MAX98373_R20D1_DHT_CFG, MAX98373_DHT_SPK_GAIN_MIN_SHIFT, 9, 0, max98373_dht_spkgain_min_tlv), -SOC_SINGLE_TLV("DHT Rot Pnt", MAX98373_R20D1_DHT_CFG, +SOC_SINGLE_TLV("DHT Rot Pnt Volume", MAX98373_R20D1_DHT_CFG, MAX98373_DHT_ROT_PNT_SHIFT, 15, 0, max98373_dht_rotation_point_tlv), -SOC_SINGLE_TLV("DHT Attack Step", MAX98373_R20D2_DHT_ATTACK_CFG, +SOC_SINGLE_TLV("DHT Attack Step Volume", MAX98373_R20D2_DHT_ATTACK_CFG, MAX98373_DHT_ATTACK_STEP_SHIFT, 4, 0, max98373_dht_step_size_tlv), -SOC_SINGLE_TLV("DHT Release Step", MAX98373_R20D3_DHT_RELEASE_CFG, +SOC_SINGLE_TLV("DHT Release Step Volume", MAX98373_R20D3_DHT_RELEASE_CFG, MAX98373_DHT_RELEASE_STEP_SHIFT, 4, 0, max98373_dht_step_size_tlv), SOC_ENUM("DHT Attack Rate", max98373_dht_attack_rate_enum), SOC_ENUM("DHT Release Rate", max98373_dht_release_rate_enum), @@ -645,36 +650,36 @@ SOC_SINGLE("BDE Thresh Hysteresis", MAX98373_R209B_BDE_THRESH_HYST, 0, 0xFF, 0), SOC_SINGLE("BDE Hold Time", MAX98373_R2090_BDE_LVL_HOLD, 0, 0xFF, 0), SOC_SINGLE("BDE Attack Rate", MAX98373_R2091_BDE_GAIN_ATK_REL_RATE, 4, 0xF, 0), SOC_SINGLE("BDE Release Rate", MAX98373_R2091_BDE_GAIN_ATK_REL_RATE, 0, 0xF, 0), -SOC_SINGLE_TLV("BDE LVL1 Clip Thresh", MAX98373_R20A9_BDE_L1_CFG_2, +SOC_SINGLE_TLV("BDE LVL1 Clip Thresh Volume", MAX98373_R20A9_BDE_L1_CFG_2, 0, 0x3C, 0, max98373_bde_gain_tlv), -SOC_SINGLE_TLV("BDE LVL2 Clip Thresh", MAX98373_R20AC_BDE_L2_CFG_2, +SOC_SINGLE_TLV("BDE LVL2 Clip Thresh Volume", MAX98373_R20AC_BDE_L2_CFG_2, 0, 0x3C, 0, max98373_bde_gain_tlv), -SOC_SINGLE_TLV("BDE LVL3 Clip Thresh", MAX98373_R20AF_BDE_L3_CFG_2, +SOC_SINGLE_TLV("BDE LVL3 Clip Thresh Volume", MAX98373_R20AF_BDE_L3_CFG_2, 0, 0x3C, 0, max98373_bde_gain_tlv), -SOC_SINGLE_TLV("BDE LVL4 Clip Thresh", MAX98373_R20B2_BDE_L4_CFG_2, +SOC_SINGLE_TLV("BDE LVL4 Clip Thresh Volume", MAX98373_R20B2_BDE_L4_CFG_2, 0, 0x3C, 0, max98373_bde_gain_tlv), -SOC_SINGLE_TLV("BDE LVL1 Clip Gain Reduct", MAX98373_R20AA_BDE_L1_CFG_3, +SOC_SINGLE_TLV("BDE LVL1 Clip Reduction Volume", MAX98373_R20AA_BDE_L1_CFG_3, 0, 0x3C, 0, max98373_bde_gain_tlv), -SOC_SINGLE_TLV("BDE LVL2 Clip Gain Reduct", MAX98373_R20AD_BDE_L2_CFG_3, +SOC_SINGLE_TLV("BDE LVL2 Clip Reduction Volume", MAX98373_R20AD_BDE_L2_CFG_3, 0, 0x3C, 0, max98373_bde_gain_tlv), -SOC_SINGLE_TLV("BDE LVL3 Clip Gain Reduct", MAX98373_R20B0_BDE_L3_CFG_3, +SOC_SINGLE_TLV("BDE LVL3 Clip Reduction Volume", MAX98373_R20B0_BDE_L3_CFG_3, 0, 0x3C, 0, max98373_bde_gain_tlv), -SOC_SINGLE_TLV("BDE LVL4 Clip Gain Reduct", MAX98373_R20B3_BDE_L4_CFG_3, +SOC_SINGLE_TLV("BDE LVL4 Clip Reduction Volume", MAX98373_R20B3_BDE_L4_CFG_3, 0, 0x3C, 0, max98373_bde_gain_tlv), -SOC_SINGLE_TLV("BDE LVL1 Limiter Thresh", MAX98373_R20A8_BDE_L1_CFG_1, +SOC_SINGLE_TLV("BDE LVL1 Limiter Thresh Volume", MAX98373_R20A8_BDE_L1_CFG_1, 0, 0xF, 0, max98373_limiter_thresh_tlv), -SOC_SINGLE_TLV("BDE LVL2 Limiter Thresh", MAX98373_R20AB_BDE_L2_CFG_1, +SOC_SINGLE_TLV("BDE LVL2 Limiter Thresh Volume", MAX98373_R20AB_BDE_L2_CFG_1, 0, 0xF, 0, max98373_limiter_thresh_tlv), -SOC_SINGLE_TLV("BDE LVL3 Limiter Thresh", MAX98373_R20AE_BDE_L3_CFG_1, +SOC_SINGLE_TLV("BDE LVL3 Limiter Thresh Volume", MAX98373_R20AE_BDE_L3_CFG_1, 0, 0xF, 0, max98373_limiter_thresh_tlv), -SOC_SINGLE_TLV("BDE LVL4 Limiter Thresh", MAX98373_R20B1_BDE_L4_CFG_1, +SOC_SINGLE_TLV("BDE LVL4 Limiter Thresh Volume", MAX98373_R20B1_BDE_L4_CFG_1, 0, 0xF, 0, max98373_limiter_thresh_tlv), /* Limiter */ SOC_SINGLE("Limiter Switch", MAX98373_R20E2_LIMITER_EN, MAX98373_LIMITER_EN_SHIFT, 1, 0), SOC_SINGLE("Limiter Src Switch", MAX98373_R20E0_LIMITER_THRESH_CFG, MAX98373_LIMITER_THRESH_SRC_SHIFT, 1, 0), -SOC_SINGLE_TLV("Limiter Thresh", MAX98373_R20E0_LIMITER_THRESH_CFG, +SOC_SINGLE_TLV("Limiter Thresh Volume", MAX98373_R20E0_LIMITER_THRESH_CFG, MAX98373_LIMITER_THRESH_SHIFT, 15, 0, max98373_limiter_thresh_tlv), SOC_ENUM("Limiter Attack Rate", max98373_limiter_attack_rate_enum), SOC_ENUM("Limiter Release Rate", max98373_limiter_release_rate_enum), diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 789346cb30b9..8f140c8b93ac 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3739,6 +3739,17 @@ static const struct dmi_system_id dmi_platform_data[] = { { } }; +static bool rt5645_check_dp(struct device *dev) +{ + if (device_property_present(dev, "realtek,in2-differential") || + device_property_present(dev, "realtek,dmic1-data-pin") || + device_property_present(dev, "realtek,dmic2-data-pin") || + device_property_present(dev, "realtek,jd-mode")) + return true; + + return false; +} + static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device *dev) { rt5645->pdata.in2_diff = device_property_read_bool(dev, @@ -3779,8 +3790,10 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (pdata) rt5645->pdata = *pdata; - else + else if (rt5645_check_dp(&i2c->dev)) rt5645_parse_dt(rt5645, &i2c->dev); + else + rt5645->pdata = jd_mode3_platform_data; if (quirk != -1) { rt5645->pdata.in2_diff = QUIRK_IN2_DIFF(quirk); diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c deleted file mode 100644 index 887923e68849..000000000000 --- a/sound/soc/codecs/sn95031.c +++ /dev/null @@ -1,936 +0,0 @@ -/* - * 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"); diff --git a/sound/soc/codecs/sn95031.h b/sound/soc/codecs/sn95031.h deleted file mode 100644 index 7651fe4e6a45..000000000000 --- a/sound/soc/codecs/sn95031.h +++ /dev/null @@ -1,133 +0,0 @@ -/* - * 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 diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index eedd600875e5..e7661d0315e6 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -355,8 +355,8 @@ static int dapm_micb_event(struct snd_soc_dapm_widget *w, return 0; } -int pll_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int pll_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); int ret; @@ -369,8 +369,8 @@ int pll_event(struct snd_soc_dapm_widget *w, return ret; } -int dac_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int dac_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); struct tscs42xx *tscs42xx = snd_soc_codec_get_drvdata(codec); @@ -631,7 +631,7 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { 0, mic_boost_scale), /* Input Channel Map */ - SOC_ENUM("Input Channel Map Switch", ch_map_select_enum), + SOC_ENUM("Input Channel Map", ch_map_select_enum), /* Coefficient Ram */ COEFF_RAM_CTL("Cascade1L BiQuad1", BIQUAD_SIZE, 0x00), @@ -708,13 +708,13 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { /* EQ */ SOC_SINGLE("EQ1 Switch", R_CONFIG1, FB_CONFIG1_EQ1_EN, 1, 0), SOC_SINGLE("EQ2 Switch", R_CONFIG1, FB_CONFIG1_EQ2_EN, 1, 0), - SOC_ENUM("EQ1 Band Enable Switch", eq1_band_enable_enum), - SOC_ENUM("EQ2 Band Enable Switch", eq2_band_enable_enum), + SOC_ENUM("EQ1 Band Enable", eq1_band_enable_enum), + SOC_ENUM("EQ2 Band Enable", eq2_band_enable_enum), /* CLE */ - SOC_ENUM("CLE Level Detect Switch", + SOC_ENUM("CLE Level Detect", cle_level_detection_enum), - SOC_ENUM("CLE Level Detect Win Switch", + SOC_ENUM("CLE Level Detect Win", cle_level_detection_window_enum), SOC_SINGLE("Expander Switch", R_CLECTL, FB_CLECTL_EXP_EN, 1, 0), @@ -726,7 +726,7 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { R_MUGAIN, FB_MUGAIN_CLEMUG, 0x1f, 0, mugain_scale), SOC_SINGLE_TLV("Comp Thresh Playback Volume", R_COMPTH, FB_COMPTH, 0xff, 0, compth_scale), - SOC_ENUM("Comp Ratio Switch", compressor_ratio_enum), + SOC_ENUM("Comp Ratio", compressor_ratio_enum), SND_SOC_BYTES("Comp Atk Time", R_CATKTCL, 2), /* Effects */ @@ -740,50 +740,50 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { SOC_SINGLE("MBC Band1 Switch", R_DACMBCEN, FB_DACMBCEN_MBCEN1, 1, 0), SOC_SINGLE("MBC Band2 Switch", R_DACMBCEN, FB_DACMBCEN_MBCEN2, 1, 0), SOC_SINGLE("MBC Band3 Switch", R_DACMBCEN, FB_DACMBCEN_MBCEN3, 1, 0), - SOC_ENUM("MBC Band1 Level Detect Switch", + SOC_ENUM("MBC Band1 Level Detect", mbc_level_detection_enums[0]), - SOC_ENUM("MBC Band2 Level Detect Switch", + SOC_ENUM("MBC Band2 Level Detect", mbc_level_detection_enums[1]), - SOC_ENUM("MBC Band3 Level Detect Switch", + SOC_ENUM("MBC Band3 Level Detect", mbc_level_detection_enums[2]), - SOC_ENUM("MBC Band1 Level Detect Win Switch", + SOC_ENUM("MBC Band1 Level Detect Win", mbc_level_detection_window_enums[0]), - SOC_ENUM("MBC Band2 Level Detect Win Switch", + SOC_ENUM("MBC Band2 Level Detect Win", mbc_level_detection_window_enums[1]), - SOC_ENUM("MBC Band3 Level Detect Win Switch", + SOC_ENUM("MBC Band3 Level Detect Win", mbc_level_detection_window_enums[2]), - SOC_SINGLE("MBC1 Phase Invert", R_DACMBCMUG1, FB_DACMBCMUG1_PHASE, - 1, 0), + SOC_SINGLE("MBC1 Phase Invert Switch", + R_DACMBCMUG1, FB_DACMBCMUG1_PHASE, 1, 0), SOC_SINGLE_TLV("DAC MBC1 Make-Up Gain Playback Volume", R_DACMBCMUG1, FB_DACMBCMUG1_MUGAIN, 0x1f, 0, mugain_scale), SOC_SINGLE_TLV("DAC MBC1 Comp Thresh Playback Volume", R_DACMBCTHR1, FB_DACMBCTHR1_THRESH, 0xff, 0, compth_scale), - SOC_ENUM("DAC MBC1 Comp Ratio Switch", + SOC_ENUM("DAC MBC1 Comp Ratio", dac_mbc1_compressor_ratio_enum), SND_SOC_BYTES("DAC MBC1 Comp Atk Time", R_DACMBCATK1L, 2), SND_SOC_BYTES("DAC MBC1 Comp Rel Time Const", R_DACMBCREL1L, 2), - SOC_SINGLE("MBC2 Phase Invert", R_DACMBCMUG2, FB_DACMBCMUG2_PHASE, - 1, 0), + SOC_SINGLE("MBC2 Phase Invert Switch", + R_DACMBCMUG2, FB_DACMBCMUG2_PHASE, 1, 0), SOC_SINGLE_TLV("DAC MBC2 Make-Up Gain Playback Volume", R_DACMBCMUG2, FB_DACMBCMUG2_MUGAIN, 0x1f, 0, mugain_scale), SOC_SINGLE_TLV("DAC MBC2 Comp Thresh Playback Volume", R_DACMBCTHR2, FB_DACMBCTHR2_THRESH, 0xff, 0, compth_scale), - SOC_ENUM("DAC MBC2 Comp Ratio Switch", + SOC_ENUM("DAC MBC2 Comp Ratio", dac_mbc2_compressor_ratio_enum), SND_SOC_BYTES("DAC MBC2 Comp Atk Time", R_DACMBCATK2L, 2), SND_SOC_BYTES("DAC MBC2 Comp Rel Time Const", R_DACMBCREL2L, 2), - SOC_SINGLE("MBC3 Phase Invert", R_DACMBCMUG3, FB_DACMBCMUG3_PHASE, - 1, 0), + SOC_SINGLE("MBC3 Phase Invert Switch", + R_DACMBCMUG3, FB_DACMBCMUG3_PHASE, 1, 0), SOC_SINGLE_TLV("DAC MBC3 Make-Up Gain Playback Volume", R_DACMBCMUG3, FB_DACMBCMUG3_MUGAIN, 0x1f, 0, mugain_scale), SOC_SINGLE_TLV("DAC MBC3 Comp Thresh Playback Volume", R_DACMBCTHR3, FB_DACMBCTHR3_THRESH, 0xff, 0, compth_scale), - SOC_ENUM("DAC MBC3 Comp Ratio Switch", + SOC_ENUM("DAC MBC3 Comp Ratio", dac_mbc3_compressor_ratio_enum), SND_SOC_BYTES("DAC MBC3 Comp Atk Time", R_DACMBCATK3L, 2), SND_SOC_BYTES("DAC MBC3 Comp Rel Time Const", diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 0c11f434a374..8c2981b70f64 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -913,8 +913,8 @@ static int fsl_soc_dma_probe(struct platform_device *pdev) dma->dai.pcm_free = fsl_dma_free_dma_buffers; /* Store the SSI-specific information that we need */ - dma->ssi_stx_phys = res.start + CCSR_SSI_STX0; - dma->ssi_srx_phys = res.start + CCSR_SSI_SRX0; + dma->ssi_stx_phys = res.start + REG_SSI_STX0; + dma->ssi_srx_phys = res.start + REG_SSI_SRX0; iprop = of_get_property(ssi_np, "fsl,fifo-depth", NULL); if (iprop) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 7b49d04e3c60..b0bd1938b71e 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -1,71 +1,123 @@ +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" diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index b973d457e834..8160520fd74c 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -1,6 +1,6 @@ # 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/ diff --git a/sound/soc/intel/atom/sst/sst_stream.c b/sound/soc/intel/atom/sst/sst_stream.c index 65e257b17a7e..7ee6aeb7e0af 100644 --- a/sound/soc/intel/atom/sst/sst_stream.c +++ b/sound/soc/intel/atom/sst/sst_stream.c @@ -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, diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 6f754708a48c..de598dcbef30 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -1,183 +1,182 @@ -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 diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 488ec48f296a..22c9cc5d135e 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -39,6 +39,7 @@ enum { 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)) @@ -63,6 +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) @@ -128,6 +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), @@ -139,6 +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"}, @@ -152,6 +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[] = { @@ -179,11 +187,18 @@ static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_in2_map[] = { {"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[] = { @@ -255,8 +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, @@ -264,7 +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_IN1_IN2_MAP), + .driver_data = (void *)(BYT_RT5651_MCLK_EN | + BYT_RT5651_IN1_IN2_MAP), }, {} }; @@ -293,6 +317,10 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) 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); diff --git a/sound/soc/intel/boards/mfld_machine.c b/sound/soc/intel/boards/mfld_machine.c deleted file mode 100644 index 7cb44fdde1ee..000000000000 --- a/sound/soc/intel/boards/mfld_machine.c +++ /dev/null @@ -1,430 +0,0 @@ -/* - * 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); - if (irq <= 0) - return irq < 0 ? irq : -ENODEV; - - /* 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"); diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c index f0cd08fa5c5d..5bc4e00a4a29 100644 --- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c +++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c @@ -1440,9 +1440,9 @@ static int mt2701_afe_pcm_dev_probe(struct platform_device *pdev) } afe->regmap = syscon_node_to_regmap(dev->parent->of_node); - if (!afe->regmap) { + if (IS_ERR(afe->regmap)) { dev_err(dev, "could not get regmap from parent\n"); - return -ENODEV; + return PTR_ERR(afe->regmap); } mutex_init(&afe->irq_alloc_lock); diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c index 99c15219dbc8..5a9a5482976e 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c @@ -37,8 +37,6 @@ static const struct snd_soc_dapm_route mt8173_rt5650_rt5514_routes[] = { {"Sub DMIC1R", NULL, "Int Mic"}, {"Headphone", NULL, "HPOL"}, {"Headphone", NULL, "HPOR"}, - {"Headset Mic", NULL, "micbias1"}, - {"Headset Mic", NULL, "micbias2"}, {"IN1P", NULL, "Headset Mic"}, {"IN1N", NULL, "Headset Mic"}, }; diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c index 42de84ca8c84..b7248085ca04 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c @@ -40,8 +40,6 @@ static const struct snd_soc_dapm_route mt8173_rt5650_rt5676_routes[] = { {"Headphone", NULL, "HPOL"}, {"Headphone", NULL, "HPOR"}, {"Headphone", NULL, "Sub AIF2TX"}, /* IF2 ADC to 5650 */ - {"Headset Mic", NULL, "micbias1"}, - {"Headset Mic", NULL, "micbias2"}, {"IN1P", NULL, "Headset Mic"}, {"IN1N", NULL, "Headset Mic"}, {"Sub AIF2RX", NULL, "Headset Mic"}, /* IF2 DAC from 5650 */ diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index e69c141d8ed4..40ebefd625c1 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -51,8 +51,6 @@ static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { {"DMIC R1", NULL, "Int Mic"}, {"Headphone", NULL, "HPOL"}, {"Headphone", NULL, "HPOR"}, - {"Headset Mic", NULL, "micbias1"}, - {"Headset Mic", NULL, "micbias2"}, {"IN1P", NULL, "Headset Mic"}, {"IN1N", NULL, "Headset Mic"}, }; diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 908211e1d6fc..950823d69e9c 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -328,6 +328,7 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, val |= I2S_CHN_4; break; case 2: + case 1: val |= I2S_CHN_2; break; default: @@ -460,7 +461,7 @@ static struct snd_soc_dai_driver rockchip_i2s_dai = { }, .capture = { .stream_name = "Capture", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_192000, .formats = (SNDRV_PCM_FMTBIT_S8 | @@ -504,6 +505,7 @@ static bool rockchip_i2s_rd_reg(struct device *dev, unsigned int reg) case I2S_INTCR: case I2S_XFER: case I2S_CLR: + case I2S_TXDR: case I2S_RXDR: case I2S_FIFOLR: case I2S_INTSR: @@ -518,6 +520,9 @@ static bool rockchip_i2s_volatile_reg(struct device *dev, unsigned int reg) switch (reg) { case I2S_INTSR: case I2S_CLR: + case I2S_FIFOLR: + case I2S_TXDR: + case I2S_RXDR: return true; default: return false; @@ -527,6 +532,8 @@ static bool rockchip_i2s_volatile_reg(struct device *dev, unsigned int reg) static bool rockchip_i2s_precious_reg(struct device *dev, unsigned int reg) { switch (reg) { + case I2S_RXDR: + return true; default: return false; } @@ -654,7 +661,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) } if (!of_property_read_u32(node, "rockchip,capture-channels", &val)) { - if (val >= 2 && val <= 8) + if (val >= 1 && val <= 8) soc_dai->capture.channels_max = val; } diff --git a/sound/soc/soc-acpi.c b/sound/soc/soc-acpi.c index f21df28bc28e..7f43c9bf3d09 100644 --- a/sound/soc/soc-acpi.c +++ b/sound/soc/soc-acpi.c @@ -49,46 +49,16 @@ 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 (mach->machine_quirk == NULL) - return mach; - - if (mach->machine_quirk(mach) != NULL) - return mach; + if (acpi_dev_present(mach->id, NULL, -1)) { + if (mach->machine_quirk) + mach = mach->machine_quirk(mach); + return mach; } } return NULL; @@ -163,7 +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; } diff --git a/sound/soc/stm/Kconfig b/sound/soc/stm/Kconfig index 3398e6c57f37..3ad881fc40a1 100644 --- a/sound/soc/stm/Kconfig +++ b/sound/soc/stm/Kconfig @@ -28,4 +28,16 @@ config SND_SOC_STM32_SPDIFRX help Say Y if you want to enable S/PDIF capture for STM32 +config SND_SOC_STM32_DFSDM + tristate "SoC Audio support for STM32 DFSDM" + depends on ARCH_STM32 || COMPILE_TEST + depends on SND_SOC + depends on STM32_DFSDM_ADC + select SND_SOC_GENERIC_DMAENGINE_PCM + select SND_SOC_DMIC + select IIO_BUFFER_CB + help + Select this option to enable the STM32 Digital Filter + for Sigma Delta Modulators (DFSDM) driver used + in various STM32 series for digital microphone capture. endmenu diff --git a/sound/soc/stm/Makefile b/sound/soc/stm/Makefile index 5b7f0fab0bd6..3143c0b47042 100644 --- a/sound/soc/stm/Makefile +++ b/sound/soc/stm/Makefile @@ -13,3 +13,6 @@ obj-$(CONFIG_SND_SOC_STM32_I2S) += snd-soc-stm32-i2s.o # SPDIFRX snd-soc-stm32-spdifrx-objs := stm32_spdifrx.o obj-$(CONFIG_SND_SOC_STM32_SPDIFRX) += snd-soc-stm32-spdifrx.o + +#DFSDM +obj-$(CONFIG_SND_SOC_STM32_DFSDM) += stm32_adfsdm.o diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c new file mode 100644 index 000000000000..7306e3eca9e1 --- /dev/null +++ b/sound/soc/stm/stm32_adfsdm.c @@ -0,0 +1,347 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * This file is part of STM32 DFSDM ASoC DAI driver + * + * Copyright (C) 2017, STMicroelectronics - All Rights Reserved + * Authors: Arnaud Pouliquen <arnaud.pouliquen@st.com> + * Olivier Moysan <olivier.moysan@st.com> + */ + +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> + +#include <linux/iio/iio.h> +#include <linux/iio/consumer.h> +#include <linux/iio/adc/stm32-dfsdm-adc.h> + +#include <sound/pcm.h> +#include <sound/soc.h> + +#define STM32_ADFSDM_DRV_NAME "stm32-adfsdm" + +#define DFSDM_MAX_PERIOD_SIZE (PAGE_SIZE / 2) +#define DFSDM_MAX_PERIODS 6 + +struct stm32_adfsdm_priv { + struct snd_soc_dai_driver dai_drv; + struct snd_pcm_substream *substream; + struct device *dev; + + /* IIO */ + struct iio_channel *iio_ch; + struct iio_cb_buffer *iio_cb; + bool iio_active; + + /* PCM buffer */ + unsigned char *pcm_buff; + unsigned int pos; +}; + +static const struct snd_pcm_hardware stm32_adfsdm_pcm_hw = { + .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + + .rate_min = 8000, + .rate_max = 32000, + + .channels_min = 1, + .channels_max = 1, + + .periods_min = 2, + .periods_max = DFSDM_MAX_PERIODS, + + .period_bytes_max = DFSDM_MAX_PERIOD_SIZE, + .buffer_bytes_max = DFSDM_MAX_PERIODS * DFSDM_MAX_PERIOD_SIZE +}; + +static void stm32_adfsdm_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct stm32_adfsdm_priv *priv = snd_soc_dai_get_drvdata(dai); + + if (priv->iio_active) { + iio_channel_stop_all_cb(priv->iio_cb); + priv->iio_active = false; + } +} + +static int stm32_adfsdm_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct stm32_adfsdm_priv *priv = snd_soc_dai_get_drvdata(dai); + int ret; + + ret = iio_write_channel_attribute(priv->iio_ch, + substream->runtime->rate, 0, + IIO_CHAN_INFO_SAMP_FREQ); + if (ret < 0) { + dev_err(dai->dev, "%s: Failed to set %d sampling rate\n", + __func__, substream->runtime->rate); + return ret; + } + + if (!priv->iio_active) { + ret = iio_channel_start_all_cb(priv->iio_cb); + if (!ret) + priv->iio_active = true; + else + dev_err(dai->dev, "%s: IIO channel start failed (%d)\n", + __func__, ret); + } + + return ret; +} + +static int stm32_adfsdm_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct stm32_adfsdm_priv *priv = snd_soc_dai_get_drvdata(dai); + ssize_t size; + char str_freq[10]; + + dev_dbg(dai->dev, "%s: Enter for freq %d\n", __func__, freq); + + /* Set IIO frequency if CODEC is master as clock comes from SPI_IN */ + + snprintf(str_freq, sizeof(str_freq), "%d\n", freq); + size = iio_write_channel_ext_info(priv->iio_ch, "spi_clk_freq", + str_freq, sizeof(str_freq)); + if (size != sizeof(str_freq)) { + dev_err(dai->dev, "%s: Failed to set SPI clock\n", + __func__); + return -EINVAL; + } + return 0; +} + +static const struct snd_soc_dai_ops stm32_adfsdm_dai_ops = { + .shutdown = stm32_adfsdm_shutdown, + .prepare = stm32_adfsdm_dai_prepare, + .set_sysclk = stm32_adfsdm_set_sysclk, +}; + +static const struct snd_soc_dai_driver stm32_adfsdm_dai = { + .capture = { + .channels_min = 1, + .channels_max = 1, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_32000), + }, + .ops = &stm32_adfsdm_dai_ops, +}; + +static const struct snd_soc_component_driver stm32_adfsdm_dai_component = { + .name = "stm32_dfsdm_audio", +}; + +static int stm32_afsdm_pcm_cb(const void *data, size_t size, void *private) +{ + struct stm32_adfsdm_priv *priv = private; + struct snd_soc_pcm_runtime *rtd = priv->substream->private_data; + u8 *pcm_buff = priv->pcm_buff; + u8 *src_buff = (u8 *)data; + unsigned int buff_size = snd_pcm_lib_buffer_bytes(priv->substream); + unsigned int period_size = snd_pcm_lib_period_bytes(priv->substream); + unsigned int old_pos = priv->pos; + unsigned int cur_size = size; + + dev_dbg(rtd->dev, "%s: buff_add :%p, pos = %d, size = %zu\n", + __func__, &pcm_buff[priv->pos], priv->pos, size); + + if ((priv->pos + size) > buff_size) { + memcpy(&pcm_buff[priv->pos], src_buff, buff_size - priv->pos); + cur_size -= buff_size - priv->pos; + priv->pos = 0; + } + + memcpy(&pcm_buff[priv->pos], &src_buff[size - cur_size], cur_size); + priv->pos = (priv->pos + cur_size) % buff_size; + + if (cur_size != size || (old_pos && (old_pos % period_size < size))) + snd_pcm_period_elapsed(priv->substream); + + return 0; +} + +static int stm32_adfsdm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct stm32_adfsdm_priv *priv = + snd_soc_dai_get_drvdata(rtd->cpu_dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + priv->pos = 0; + return stm32_dfsdm_get_buff_cb(priv->iio_ch->indio_dev, + stm32_afsdm_pcm_cb, priv); + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + return stm32_dfsdm_release_buff_cb(priv->iio_ch->indio_dev); + } + + return -EINVAL; +} + +static int stm32_adfsdm_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct stm32_adfsdm_priv *priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); + int ret; + + ret = snd_soc_set_runtime_hwparams(substream, &stm32_adfsdm_pcm_hw); + if (!ret) + priv->substream = substream; + + return ret; +} + +static int stm32_adfsdm_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct stm32_adfsdm_priv *priv = + snd_soc_dai_get_drvdata(rtd->cpu_dai); + + snd_pcm_lib_free_pages(substream); + priv->substream = NULL; + + return 0; +} + +static snd_pcm_uframes_t stm32_adfsdm_pcm_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct stm32_adfsdm_priv *priv = + snd_soc_dai_get_drvdata(rtd->cpu_dai); + + return bytes_to_frames(substream->runtime, priv->pos); +} + +static int stm32_adfsdm_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct stm32_adfsdm_priv *priv = + snd_soc_dai_get_drvdata(rtd->cpu_dai); + int ret; + + ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + if (ret < 0) + return ret; + priv->pcm_buff = substream->runtime->dma_area; + + return iio_channel_cb_set_buffer_watermark(priv->iio_cb, + params_period_size(params)); +} + +static int stm32_adfsdm_pcm_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_lib_free_pages(substream); + + return 0; +} + +static struct snd_pcm_ops stm32_adfsdm_pcm_ops = { + .open = stm32_adfsdm_pcm_open, + .close = stm32_adfsdm_pcm_close, + .hw_params = stm32_adfsdm_pcm_hw_params, + .hw_free = stm32_adfsdm_pcm_hw_free, + .trigger = stm32_adfsdm_trigger, + .pointer = stm32_adfsdm_pcm_pointer, +}; + +static int stm32_adfsdm_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + struct stm32_adfsdm_priv *priv = + snd_soc_dai_get_drvdata(rtd->cpu_dai); + unsigned int size = DFSDM_MAX_PERIODS * DFSDM_MAX_PERIOD_SIZE; + + return snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + priv->dev, size, size); +} + +static void stm32_adfsdm_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_soc_pcm_runtime *rtd; + struct stm32_adfsdm_priv *priv; + + substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; + if (substream) { + rtd = substream->private_data; + priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); + + snd_pcm_lib_preallocate_free_for_all(pcm); + } +} + +static struct snd_soc_platform_driver stm32_adfsdm_soc_platform = { + .ops = &stm32_adfsdm_pcm_ops, + .pcm_new = stm32_adfsdm_pcm_new, + .pcm_free = stm32_adfsdm_pcm_free, +}; + +static const struct of_device_id stm32_adfsdm_of_match[] = { + {.compatible = "st,stm32h7-dfsdm-dai"}, + {} +}; +MODULE_DEVICE_TABLE(of, stm32_adfsdm_of_match); + +static int stm32_adfsdm_probe(struct platform_device *pdev) +{ + struct stm32_adfsdm_priv *priv; + int ret; + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->dev = &pdev->dev; + priv->dai_drv = stm32_adfsdm_dai; + + dev_set_drvdata(&pdev->dev, priv); + + ret = devm_snd_soc_register_component(&pdev->dev, + &stm32_adfsdm_dai_component, + &priv->dai_drv, 1); + if (ret < 0) + return ret; + + /* Associate iio channel */ + priv->iio_ch = devm_iio_channel_get_all(&pdev->dev); + if (IS_ERR(priv->iio_ch)) + return PTR_ERR(priv->iio_ch); + + priv->iio_cb = iio_channel_get_all_cb(&pdev->dev, NULL, NULL); + if (IS_ERR(priv->iio_cb)) + return PTR_ERR(priv->iio_cb); + + ret = devm_snd_soc_register_platform(&pdev->dev, + &stm32_adfsdm_soc_platform); + if (ret < 0) + dev_err(&pdev->dev, "%s: Failed to register PCM platform\n", + __func__); + + return ret; +} + +static struct platform_driver stm32_adfsdm_driver = { + .driver = { + .name = STM32_ADFSDM_DRV_NAME, + .of_match_table = stm32_adfsdm_of_match, + }, + .probe = stm32_adfsdm_probe, +}; + +module_platform_driver(stm32_adfsdm_driver); + +MODULE_DESCRIPTION("stm32 DFSDM DAI driver"); +MODULE_AUTHOR("Arnaud Pouliquen <arnaud.pouliquen@st.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" STM32_ADFSDM_DRV_NAME); diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 070a6880980e..c60a57797640 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -163,3 +163,7 @@ static struct platform_driver snd_soc_mop500_driver = { }; module_platform_driver(snd_soc_mop500_driver); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("ASoC MOP500 board driver"); +MODULE_AUTHOR("Ola Lilja"); diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index f12c01dddc8d..d35ba7700f46 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -165,3 +165,8 @@ int ux500_pcm_unregister_platform(struct platform_device *pdev) return 0; } EXPORT_SYMBOL_GPL(ux500_pcm_unregister_platform); + +MODULE_AUTHOR("Ola Lilja"); +MODULE_AUTHOR("Roger Nilsson"); +MODULE_DESCRIPTION("ASoC UX500 driver"); +MODULE_LICENSE("GPL v2"); |