diff options
| author | Takashi Iwai <tiwai@suse.de> | 2018-01-12 14:02:15 +0100 |
|---|---|---|
| committer | Takashi Iwai <tiwai@suse.de> | 2018-01-12 14:02:15 +0100 |
| commit | 8999bd3c6323a3f3815b5c45628023f3a994b7a8 (patch) | |
| tree | 6d4c4ef9de11d333d0ed0b3e0e1ea3b04e956c29 /sound | |
| parent | 8ac60e733f7c9c41e4c125619a2f8390aca9d4db (diff) | |
| parent | 4ac71d1b68365915bcde14d0ff8fda186ad377ac (diff) | |
Merge tag 'asoc-v4.16-2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v4.16
Some more updates for v4.16, the big things here are the ST DFSDM driver
and the IIO patches required to support that and even more in the
neverending series of code quality improvements for x86, including
Pierre's work to improve the Kconfig. The unused SN95031 driver and
associated board support are also removed, they haven't been buildable
for a considerable time without anyone noticing.
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"); |