diff options
Diffstat (limited to 'sound/soc')
57 files changed, 952 insertions, 1858 deletions
diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 9f521a55d610..b5e41df6bb3a 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1051,6 +1051,11 @@ static int acp_audio_probe(struct platform_device *pdev) struct resource *res; const u32 *pdata = pdev->dev.platform_data; + if (!pdata) { + dev_err(&pdev->dev, "Missing platform data\n"); + return -ENODEV; + } + audio_drv_data = devm_kzalloc(&pdev->dev, sizeof(struct audio_drv_data), GFP_KERNEL); if (audio_drv_data == NULL) @@ -1058,6 +1063,8 @@ static int acp_audio_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); audio_drv_data->acp_mmio = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(audio_drv_data->acp_mmio)) + return PTR_ERR(audio_drv_data->acp_mmio); /* The following members gets populated in device 'open' * function. Till then interrupts are disabled in 'acp_init' diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 4a56f3dfba51..dcee145dd179 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -64,7 +64,7 @@ config SND_AT91_SOC_SAM9X5_WM8731 config SND_ATMEL_SOC_CLASSD tristate "Atmel ASoC driver for boards using CLASSD" depends on ARCH_AT91 || COMPILE_TEST - select SND_ATMEL_SOC_DMA + select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO help Say Y if you want to add support for Atmel ASoC driver for boards using diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a42ddbc93f3d..3ed2b985b38b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -133,7 +133,6 @@ 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 @@ -818,9 +817,6 @@ 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 0001069ce2a7..ae25cbe85d1d 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -140,7 +140,6 @@ 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/da7218.c b/sound/soc/codecs/da7218.c index b2d42ec1dcd9..56564ce90cb6 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -2520,7 +2520,7 @@ static struct da7218_pdata *da7218_of_to_pdata(struct snd_soc_codec *codec) } if (da7218->dev_id == DA7218_DEV_ID) { - hpldet_np = of_find_node_by_name(np, "da7218_hpldet"); + hpldet_np = of_get_child_by_name(np, "da7218_hpldet"); if (!hpldet_np) return pdata; diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 5f3c42c4f74a..066ea2f4ce7b 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -267,7 +267,7 @@ #define MSM8916_WCD_ANALOG_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000) #define MSM8916_WCD_ANALOG_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S32_LE) static int btn_mask = SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_BTN_4; diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index a10a724eb448..13354d6304a8 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -194,7 +194,7 @@ SNDRV_PCM_RATE_32000 | \ SNDRV_PCM_RATE_48000) #define MSM8916_WCD_DIGITAL_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S32_LE) struct msm8916_wcd_digital_priv { struct clk *ahbclk, *mclk; @@ -645,7 +645,7 @@ static int msm8916_wcd_digital_hw_params(struct snd_pcm_substream *substream, RX_I2S_CTL_RX_I2S_MODE_MASK, RX_I2S_CTL_RX_I2S_MODE_16); break; - case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: snd_soc_update_bits(dai->codec, LPASS_CDC_CLK_TX_I2S_CTL, TX_I2S_CTL_TX_I2S_MODE_MASK, TX_I2S_CTL_TX_I2S_MODE_32); diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 714ce17da717..e853a6dfd33b 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -905,6 +905,7 @@ static int nau8825_adc_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: + msleep(125); regmap_update_bits(nau8825->regmap, NAU8825_REG_ENA_CTRL, NAU8825_ENABLE_ADC, NAU8825_ENABLE_ADC); break; diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 2df91db765ac..64bf26cec20d 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -289,6 +289,8 @@ static int rt5514_spi_pcm_probe(struct snd_soc_platform *platform) dev_err(&rt5514_spi->dev, "%s Failed to reguest IRQ: %d\n", __func__, ret); + else + device_init_wakeup(rt5514_dsp->dev, true); } return 0; @@ -456,8 +458,6 @@ static int rt5514_spi_probe(struct spi_device *spi) return ret; } - device_init_wakeup(&spi->dev, true); - return 0; } @@ -482,10 +482,13 @@ static int __maybe_unused rt5514_resume(struct device *dev) if (device_may_wakeup(dev)) disable_irq_wake(irq); - if (rt5514_dsp->substream) { - rt5514_spi_burst_read(RT5514_IRQ_CTRL, (u8 *)&buf, sizeof(buf)); - if (buf[0] & RT5514_IRQ_STATUS_BIT) - rt5514_schedule_copy(rt5514_dsp); + if (rt5514_dsp) { + if (rt5514_dsp->substream) { + rt5514_spi_burst_read(RT5514_IRQ_CTRL, (u8 *)&buf, + sizeof(buf)); + if (buf[0] & RT5514_IRQ_STATUS_BIT) + rt5514_schedule_copy(rt5514_dsp); + } } return 0; diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 2a5b5d74e697..2dd6e9f990a4 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -496,7 +496,7 @@ static const struct snd_soc_dapm_widget rt5514_dapm_widgets[] = { SND_SOC_DAPM_PGA("DMIC1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("DMIC2", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_SUPPLY_S("DMIC CLK", 1, SND_SOC_NOPM, 0, 0, rt5514_set_dmic_clk, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_SUPPLY("ADC CLK", RT5514_CLK_CTRL1, diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index f020d2d1eef4..edc152c8a1fe 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3823,6 +3823,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, regmap_read(regmap, RT5645_VENDOR_ID, &val); rt5645->v_id = val & 0xff; + regmap_write(rt5645->regmap, RT5645_AD_DA_MIXER, 0x8080); + ret = regmap_register_patch(rt5645->regmap, init_list, ARRAY_SIZE(init_list)); if (ret != 0) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index b036c9dc0c8c..d329bf719d80 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -1560,6 +1560,10 @@ static int rt5663_jack_detect(struct snd_soc_codec *codec, int jack_insert) RT5663_IRQ_POW_SAV_MASK, RT5663_IRQ_POW_SAV_EN); snd_soc_update_bits(codec, RT5663_IRQ_1, RT5663_EN_IRQ_JD1_MASK, RT5663_EN_IRQ_JD1_EN); + snd_soc_update_bits(codec, RT5663_EM_JACK_TYPE_1, + RT5663_EM_JD_MASK, RT5663_EM_JD_RST); + snd_soc_update_bits(codec, RT5663_EM_JACK_TYPE_1, + RT5663_EM_JD_MASK, RT5663_EM_JD_NOR); while (true) { regmap_read(rt5663->regmap, RT5663_INT_ST_2, &val); diff --git a/sound/soc/codecs/rt5663.h b/sound/soc/codecs/rt5663.h index c5a9b69579ad..03adc8004ba9 100644 --- a/sound/soc/codecs/rt5663.h +++ b/sound/soc/codecs/rt5663.h @@ -1029,6 +1029,10 @@ #define RT5663_POL_EXT_JD_SHIFT 10 #define RT5663_POL_EXT_JD_EN (0x1 << 10) #define RT5663_POL_EXT_JD_DIS (0x0 << 10) +#define RT5663_EM_JD_MASK (0x1 << 7) +#define RT5663_EM_JD_SHIFT 7 +#define RT5663_EM_JD_NOR (0x1 << 7) +#define RT5663_EM_JD_RST (0x0 << 7) /* DACREF LDO Control (0x0112)*/ #define RT5663_PWR_LDO_DACREFL_MASK (0x1 << 9) 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/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 730fb2058869..1ff3edb7bbb6 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -116,7 +116,7 @@ struct aic31xx_pdata { /* INT2 interrupt control */ #define AIC31XX_INT2CTRL AIC31XX_REG(0, 49) /* GPIO1 control */ -#define AIC31XX_GPIO1 AIC31XX_REG(0, 50) +#define AIC31XX_GPIO1 AIC31XX_REG(0, 51) #define AIC31XX_DACPRB AIC31XX_REG(0, 60) /* ADC Instruction Set Register */ diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index c482b2e7a7d2..cfe72b9d4356 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -232,7 +232,7 @@ static struct twl4030_codec_data *twl4030_get_pdata(struct snd_soc_codec *codec) struct twl4030_codec_data *pdata = dev_get_platdata(codec->dev); struct device_node *twl4030_codec_node = NULL; - twl4030_codec_node = of_find_node_by_name(codec->dev->parent->of_node, + twl4030_codec_node = of_get_child_by_name(codec->dev->parent->of_node, "codec"); if (!pdata && twl4030_codec_node) { @@ -241,9 +241,11 @@ static struct twl4030_codec_data *twl4030_get_pdata(struct snd_soc_codec *codec) GFP_KERNEL); if (!pdata) { dev_err(codec->dev, "Can not allocate memory\n"); + of_node_put(twl4030_codec_node); return NULL; } twl4030_setup_pdata_of(pdata, twl4030_codec_node); + of_node_put(twl4030_codec_node); } return pdata; diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 65c059b5ffd7..66e32f5d2917 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1733,7 +1733,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) le64_to_cpu(footer->timestamp)); while (pos < firmware->size && - pos - firmware->size > sizeof(*region)) { + sizeof(*region) < firmware->size - pos) { region = (void *)&(firmware->data[pos]); region_name = "Unknown"; reg = 0; @@ -1782,8 +1782,8 @@ static int wm_adsp_load(struct wm_adsp *dsp) regions, le32_to_cpu(region->len), offset, region_name); - if ((pos + le32_to_cpu(region->len) + sizeof(*region)) > - firmware->size) { + if (le32_to_cpu(region->len) > + firmware->size - pos - sizeof(*region)) { adsp_err(dsp, "%s.%d: %s region len %d bytes exceeds file length %zu\n", file, regions, region_name, @@ -2253,7 +2253,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) blocks = 0; while (pos < firmware->size && - pos - firmware->size > sizeof(*blk)) { + sizeof(*blk) < firmware->size - pos) { blk = (void *)(&firmware->data[pos]); type = le16_to_cpu(blk->type); @@ -2327,8 +2327,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) } if (reg) { - if ((pos + le32_to_cpu(blk->len) + sizeof(*blk)) > - firmware->size) { + if (le32_to_cpu(blk->len) > + firmware->size - pos - sizeof(*blk)) { adsp_err(dsp, "%s.%d: %s region len %d bytes exceeds file length %zu\n", file, blocks, region_name, diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h index 0f163abe4ba3..52c27a358933 100644 --- a/sound/soc/fsl/fsl_asrc.h +++ b/sound/soc/fsl/fsl_asrc.h @@ -260,8 +260,8 @@ #define ASRFSTi_OUTPUT_FIFO_SHIFT 12 #define ASRFSTi_OUTPUT_FIFO_MASK (((1 << ASRFSTi_OUTPUT_FIFO_WIDTH) - 1) << ASRFSTi_OUTPUT_FIFO_SHIFT) #define ASRFSTi_IAEi_SHIFT 11 -#define ASRFSTi_IAEi_MASK (1 << ASRFSTi_OAFi_SHIFT) -#define ASRFSTi_IAEi (1 << ASRFSTi_OAFi_SHIFT) +#define ASRFSTi_IAEi_MASK (1 << ASRFSTi_IAEi_SHIFT) +#define ASRFSTi_IAEi (1 << ASRFSTi_IAEi_SHIFT) #define ASRFSTi_INPUT_FIFO_WIDTH 7 #define ASRFSTi_INPUT_FIFO_SHIFT 0 #define ASRFSTi_INPUT_FIFO_MASK ((1 << ASRFSTi_INPUT_FIFO_WIDTH) - 1) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f2f51e06e22c..424bafaf51ef 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -38,6 +38,7 @@ #include <linux/ctype.h> #include <linux/device.h> #include <linux/delay.h> +#include <linux/mutex.h> #include <linux/slab.h> #include <linux/spinlock.h> #include <linux/of.h> @@ -265,6 +266,8 @@ struct fsl_ssi_private { u32 fifo_watermark; u32 dma_maxburst; + + struct mutex ac97_reg_lock; }; /* @@ -1260,11 +1263,13 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, if (reg > 0x7f) return; + mutex_lock(&fsl_ac97_data->ac97_reg_lock); + ret = clk_prepare_enable(fsl_ac97_data->clk); if (ret) { pr_err("ac97 write clk_prepare_enable failed: %d\n", ret); - return; + goto ret_unlock; } lreg = reg << 12; @@ -1278,6 +1283,9 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, udelay(100); clk_disable_unprepare(fsl_ac97_data->clk); + +ret_unlock: + mutex_unlock(&fsl_ac97_data->ac97_reg_lock); } static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, @@ -1285,16 +1293,18 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, { struct regmap *regs = fsl_ac97_data->regs; - unsigned short val = -1; + unsigned short val = 0; u32 reg_val; unsigned int lreg; int ret; + mutex_lock(&fsl_ac97_data->ac97_reg_lock); + ret = clk_prepare_enable(fsl_ac97_data->clk); if (ret) { pr_err("ac97 read clk_prepare_enable failed: %d\n", ret); - return -1; + goto ret_unlock; } lreg = (reg & 0x7f) << 12; @@ -1309,6 +1319,8 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, clk_disable_unprepare(fsl_ac97_data->clk); +ret_unlock: + mutex_unlock(&fsl_ac97_data->ac97_reg_lock); return val; } @@ -1458,12 +1470,6 @@ static int fsl_ssi_probe(struct platform_device *pdev) sizeof(fsl_ssi_ac97_dai)); fsl_ac97_data = ssi_private; - - ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); - if (ret) { - dev_err(&pdev->dev, "could not set AC'97 ops\n"); - return ret; - } } else { /* Initialize this copy of the CPU DAI driver structure */ memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, @@ -1574,6 +1580,15 @@ static int fsl_ssi_probe(struct platform_device *pdev) return ret; } + if (fsl_ssi_is_ac97(ssi_private)) { + mutex_init(&ssi_private->ac97_reg_lock); + ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + if (ret) { + dev_err(&pdev->dev, "could not set AC'97 ops\n"); + goto error_ac97_ops; + } + } + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_ssi_component, &ssi_private->cpu_dai_drv, 1); if (ret) { @@ -1657,6 +1672,13 @@ error_sound_card: fsl_ssi_debugfs_remove(&ssi_private->dbg_stats); error_asoc_register: + if (fsl_ssi_is_ac97(ssi_private)) + snd_soc_set_ac97_ops(NULL); + +error_ac97_ops: + if (fsl_ssi_is_ac97(ssi_private)) + mutex_destroy(&ssi_private->ac97_reg_lock); + if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); @@ -1675,8 +1697,10 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); - if (fsl_ssi_is_ac97(ssi_private)) + if (fsl_ssi_is_ac97(ssi_private)) { snd_soc_set_ac97_ops(NULL); + mutex_destroy(&ssi_private->ac97_reg_lock); + } return 0; } 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_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 32d6e02e2104..6cd481bec275 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -236,6 +236,9 @@ static int sst_platform_get_resources(struct intel_sst_drv *ctx) /* Find the IRQ */ ctx->irq_num = platform_get_irq(pdev, ctx->pdata->res_info->acpi_ipc_irq_index); + if (ctx->irq_num <= 0) + return ctx->irq_num < 0 ? ctx->irq_num : -EIO; + return 0; } 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..d4e103615f51 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -1,183 +1,183 @@ -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_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 +185,12 @@ 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 +198,12 @@ 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 +212,12 @@ 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 +225,12 @@ 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 +238,13 @@ 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 +253,8 @@ 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/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c index c4d82ad41bd7..6219c04d4731 100644 --- a/sound/soc/intel/boards/bytcht_da7213.c +++ b/sound/soc/intel/boards/bytcht_da7213.c @@ -219,7 +219,7 @@ static struct snd_soc_card bytcht_da7213_card = { .num_dapm_routes = ARRAY_SIZE(audio_map), }; -static char codec_name[16]; /* i2c-<HID>:00 with HID being 8 chars */ +static char codec_name[SND_ACPI_I2C_ID_LEN]; static int bytcht_da7213_probe(struct platform_device *pdev) { diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 8088396717e3..079f35cd4eaf 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -232,15 +232,39 @@ static struct snd_soc_card byt_cht_es8316_card = { .fully_routed = true, }; +static char codec_name[SND_ACPI_I2C_ID_LEN]; + static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) { - int ret = 0; struct byt_cht_es8316_private *priv; + struct snd_soc_acpi_mach *mach; + const char *i2c_name = NULL; + int dai_index = 0; + int i; + int ret = 0; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_ATOMIC); if (!priv) return -ENOMEM; + mach = (&pdev->dev)->platform_data; + /* fix index of codec dai */ + for (i = 0; i < ARRAY_SIZE(byt_cht_es8316_dais); i++) { + if (!strcmp(byt_cht_es8316_dais[i].codec_name, + "i2c-ESSX8316:00")) { + dai_index = i; + break; + } + } + + /* fixup codec name based on HID */ + i2c_name = snd_soc_acpi_find_name_from_hid(mach->id); + if (i2c_name) { + snprintf(codec_name, sizeof(codec_name), + "%s%s", "i2c-", i2c_name); + byt_cht_es8316_dais[dai_index].codec_name = codec_name; + } + /* register the soc card */ byt_cht_es8316_card.dev = &pdev->dev; snd_soc_card_set_drvdata(&byt_cht_es8316_card, priv); diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index f2c0fc415e52..4548f75498d0 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -713,7 +713,7 @@ static struct snd_soc_card byt_rt5640_card = { .fully_routed = true, }; -static char byt_rt5640_codec_name[16]; /* i2c-<HID>:00 with HID being 8 chars */ +static char byt_rt5640_codec_name[SND_ACPI_I2C_ID_LEN]; static char byt_rt5640_codec_aif_name[12]; /* = "rt5640-aif[1|2]" */ static char byt_rt5640_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index d955836c6870..5a6b7dedb773 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -38,6 +38,8 @@ enum { BYT_RT5651_DMIC_MAP, 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)) @@ -62,6 +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) @@ -127,6 +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), @@ -138,6 +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"}, @@ -151,6 +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[] = { @@ -171,11 +180,25 @@ static const struct snd_soc_dapm_route byt_rt5651_intmic_in2_map[] = { {"IN2P", NULL, "Internal Mic"}, }; +static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_in2_map[] = { + {"Internal Mic", NULL, "micbias1"}, + {"IN1P", NULL, "Internal Mic"}, + {"IN2P", NULL, "Internal Mic"}, + {"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[] = { @@ -247,8 +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, @@ -256,7 +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_IN2_MAP), + .driver_data = (void *)(BYT_RT5651_MCLK_EN | + BYT_RT5651_IN1_IN2_MAP), }, {} }; @@ -281,6 +313,14 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) custom_map = byt_rt5651_intmic_in2_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_in2_map); break; + case BYT_RT5651_IN1_IN2_MAP: + 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); @@ -469,7 +509,7 @@ static struct snd_soc_card byt_rt5651_card = { .fully_routed = true, }; -static char byt_rt5651_codec_name[16]; /* i2c-<HID>:00 with HID being 8 chars */ +static char byt_rt5651_codec_name[SND_ACPI_I2C_ID_LEN]; static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) { diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 18d129caa974..cef6a8c31c8d 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -49,7 +49,7 @@ struct cht_acpi_card { struct cht_mc_private { struct snd_soc_jack jack; struct cht_acpi_card *acpi_card; - char codec_name[16]; + char codec_name[SND_ACPI_I2C_ID_LEN]; struct clk *mclk; }; @@ -499,7 +499,7 @@ static struct cht_acpi_card snd_soc_cards[] = { {"10EC5650", CODEC_TYPE_RT5650, &snd_soc_card_chtrt5650}, }; -static char cht_rt5645_codec_name[16]; /* i2c-<HID>:00 with HID being 8 chars */ +static char cht_rt5645_codec_name[SND_ACPI_I2C_ID_LEN]; static char cht_rt5645_codec_aif_name[12]; /* = "rt5645-aif[1|2]" */ static char cht_rt5645_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index f8f21eee9b2d..1f3d38dc4fcb 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -35,7 +35,7 @@ struct cht_mc_private { struct snd_soc_jack headset; - char codec_name[16]; + char codec_name[SND_ACPI_I2C_ID_LEN]; struct clk *mclk; }; diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index 5e1ea0371c90..3c5160779204 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -76,7 +76,7 @@ static int haswell_rt5640_hw_params(struct snd_pcm_substream *substream, } /* set correct codec filter for DAI format and clock config */ - snd_soc_update_bits(rtd->codec, 0x83, 0xffff, 0x8000); + snd_soc_component_update_bits(codec_dai->component, 0x83, 0xffff, 0x8000); return ret; } diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 6f9a8bcf20f3..bf7014ca486f 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -101,7 +101,7 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "ssp0 Tx", NULL, "spk_out" }, { "AIF Playback", NULL, "ssp1 Tx" }, - { "ssp1 Tx", NULL, "hs_out" }, + { "ssp1 Tx", NULL, "codec1_out" }, { "hs_in", NULL, "ssp1 Rx" }, { "ssp1 Rx", NULL, "AIF Capture" }, @@ -225,7 +225,7 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) } jack = &ctx->kabylake_headset; - snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 6072164f2d43..90ea98f01c4c 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -109,7 +109,7 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "ssp0 Tx", NULL, "spk_out" }, { "AIF Playback", NULL, "ssp1 Tx" }, - { "ssp1 Tx", NULL, "hs_out" }, + { "ssp1 Tx", NULL, "codec1_out" }, { "hs_in", NULL, "ssp1 Rx" }, { "ssp1 Rx", NULL, "AIF Capture" }, @@ -195,7 +195,7 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) } jack = &ctx->kabylake_headset; - snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); diff --git a/sound/soc/intel/boards/mfld_machine.c b/sound/soc/intel/boards/mfld_machine.c deleted file mode 100644 index 6f44acfb4aae..000000000000 --- a/sound/soc/intel/boards/mfld_machine.c +++ /dev/null @@ -1,428 +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); - - /* 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/intel/common/sst-dsp.c b/sound/soc/intel/common/sst-dsp.c index 11c0805393ff..fd82f4b1d4a0 100644 --- a/sound/soc/intel/common/sst-dsp.c +++ b/sound/soc/intel/common/sst-dsp.c @@ -269,7 +269,7 @@ int sst_dsp_register_poll(struct sst_dsp *ctx, u32 offset, u32 mask, */ timeout = jiffies + msecs_to_jiffies(time); - while (((sst_dsp_shim_read_unlocked(ctx, offset) & mask) != target) + while ((((reg = sst_dsp_shim_read_unlocked(ctx, offset)) & mask) != target) && time_before(jiffies, timeout)) { k++; if (k > 10) @@ -278,8 +278,6 @@ int sst_dsp_register_poll(struct sst_dsp *ctx, u32 offset, u32 mask, usleep_range(s, 2*s); } - reg = sst_dsp_shim_read_unlocked(ctx, offset); - if ((reg & mask) == target) { dev_dbg(ctx->dev, "FW Poll Status: reg=%#x %s successful\n", reg, operation); diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index 4524211960e4..440bca7afbf1 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -595,7 +595,7 @@ int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, INIT_DELAYED_WORK(&skl->d0i3.work, bxt_set_dsp_D0i3); skl->d0i3.state = SKL_DSP_D0I3_NONE; - return 0; + return skl_dsp_acquire_irq(sst); } EXPORT_SYMBOL_GPL(bxt_sst_dsp_init); diff --git a/sound/soc/intel/skylake/cnl-sst.c b/sound/soc/intel/skylake/cnl-sst.c index 387de388ce29..245df1067ba8 100644 --- a/sound/soc/intel/skylake/cnl-sst.c +++ b/sound/soc/intel/skylake/cnl-sst.c @@ -458,7 +458,7 @@ int cnl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, cnl->boot_complete = false; init_waitqueue_head(&cnl->boot_wait); - return 0; + return skl_dsp_acquire_irq(sst); } EXPORT_SYMBOL_GPL(cnl_sst_dsp_init); diff --git a/sound/soc/intel/skylake/skl-i2s.h b/sound/soc/intel/skylake/skl-i2s.h new file mode 100644 index 000000000000..dcf819bc688f --- /dev/null +++ b/sound/soc/intel/skylake/skl-i2s.h @@ -0,0 +1,64 @@ +/* + * skl-i2s.h - i2s blob mapping + * + * Copyright (C) 2017 Intel Corp + * Author: Subhransu S. Prusty < subhransu.s.prusty@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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + */ + +#ifndef __SOUND_SOC_SKL_I2S_H +#define __SOUND_SOC_SKL_I2S_H + +#define SKL_I2S_MAX_TIME_SLOTS 8 +#define SKL_MCLK_DIV_CLK_SRC_MASK GENMASK(17, 16) + +#define SKL_MNDSS_DIV_CLK_SRC_MASK GENMASK(21, 20) +#define SKL_SHIFT(x) (ffs(x) - 1) +#define SKL_MCLK_DIV_RATIO_MASK GENMASK(11, 0) + +struct skl_i2s_config { + u32 ssc0; + u32 ssc1; + u32 sscto; + u32 sspsp; + u32 sstsa; + u32 ssrsa; + u32 ssc2; + u32 sspsp2; + u32 ssc3; + u32 ssioc; +} __packed; + +struct skl_i2s_config_mclk { + u32 mdivctrl; + u32 mdivr; +}; + +/** + * struct skl_i2s_config_blob_legacy - Structure defines I2S Gateway + * configuration legacy blob + * + * @gtw_attr: Gateway attribute for the I2S Gateway + * @tdm_ts_group: TDM slot mapping against channels in the Gateway. + * @i2s_cfg: I2S HW registers + * @mclk: MCLK clock source and divider values + */ +struct skl_i2s_config_blob_legacy { + u32 gtw_attr; + u32 tdm_ts_group[SKL_I2S_MAX_TIME_SLOTS]; + struct skl_i2s_config i2s_cfg; + struct skl_i2s_config_mclk mclk; +}; + +#endif /* __SOUND_SOC_SKL_I2S_H */ diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index 61b5bfa79d13..8cbf080c38b3 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -55,6 +55,19 @@ static int skl_free_dma_buf(struct device *dev, struct snd_dma_buffer *dmab) return 0; } +#define SKL_ASTATE_PARAM_ID 4 + +void skl_dsp_set_astate_cfg(struct skl_sst *ctx, u32 cnt, void *data) +{ + struct skl_ipc_large_config_msg msg = {0}; + + msg.large_param_id = SKL_ASTATE_PARAM_ID; + msg.param_data_size = (cnt * sizeof(struct skl_astate_param) + + sizeof(cnt)); + + skl_ipc_set_large_config(&ctx->ipc, &msg, data); +} + #define NOTIFICATION_PARAM_ID 3 #define NOTIFICATION_MASK 0xf @@ -404,11 +417,20 @@ int skl_resume_dsp(struct skl *skl) if (skl->skl_sst->is_first_boot == true) return 0; + /* disable dynamic clock gating during fw and lib download */ + ctx->enable_miscbdcge(ctx->dev, false); + ret = skl_dsp_wake(ctx->dsp); + ctx->enable_miscbdcge(ctx->dev, true); if (ret < 0) return ret; skl_dsp_enable_notification(skl->skl_sst, false); + + if (skl->cfg.astate_cfg != NULL) { + skl_dsp_set_astate_cfg(skl->skl_sst, skl->cfg.astate_cfg->count, + skl->cfg.astate_cfg); + } return ret; } diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index d14c50a60289..3b1d2b828c1b 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -19,6 +19,7 @@ */ #include <linux/pci.h> #include "skl.h" +#include "skl-i2s.h" #define NHLT_ACPI_HEADER_SIG "NHLT" @@ -43,7 +44,8 @@ struct nhlt_acpi_table *skl_nhlt_init(struct device *dev) obj = acpi_evaluate_dsm(handle, &osc_guid, 1, 1, NULL); if (obj && obj->type == ACPI_TYPE_BUFFER) { nhlt_ptr = (struct nhlt_resource_desc *)obj->buffer.pointer; - nhlt_table = (struct nhlt_acpi_table *) + if (nhlt_ptr->length) + nhlt_table = (struct nhlt_acpi_table *) memremap(nhlt_ptr->min_addr, nhlt_ptr->length, MEMREMAP_WB); ACPI_FREE(obj); @@ -119,11 +121,16 @@ static bool skl_check_ep_match(struct device *dev, struct nhlt_endpoint *epnt, if ((epnt->virtual_bus_id == instance_id) && (epnt->linktype == link_type) && - (epnt->direction == dirn) && - (epnt->device_type == dev_type)) - return true; - else - return false; + (epnt->direction == dirn)) { + /* do not check dev_type for DMIC link type */ + if (epnt->linktype == NHLT_LINK_DMIC) + return true; + + if (epnt->device_type == dev_type) + return true; + } + + return false; } struct nhlt_specific_cfg @@ -271,3 +278,157 @@ void skl_nhlt_remove_sysfs(struct skl *skl) sysfs_remove_file(&dev->kobj, &dev_attr_platform_id.attr); } + +/* + * Queries NHLT for all the fmt configuration for a particular endpoint and + * stores all possible rates supported in a rate table for the corresponding + * sclk/sclkfs. + */ +static void skl_get_ssp_clks(struct skl *skl, struct skl_ssp_clk *ssp_clks, + struct nhlt_fmt *fmt, u8 id) +{ + struct skl_i2s_config_blob_legacy *i2s_config; + struct skl_clk_parent_src *parent; + struct skl_ssp_clk *sclk, *sclkfs; + struct nhlt_fmt_cfg *fmt_cfg; + struct wav_fmt_ext *wav_fmt; + unsigned long rate = 0; + bool present = false; + int rate_index = 0; + u16 channels, bps; + u8 clk_src; + int i, j; + u32 fs; + + sclk = &ssp_clks[SKL_SCLK_OFS]; + sclkfs = &ssp_clks[SKL_SCLKFS_OFS]; + + if (fmt->fmt_count == 0) + return; + + for (i = 0; i < fmt->fmt_count; i++) { + fmt_cfg = &fmt->fmt_config[i]; + wav_fmt = &fmt_cfg->fmt_ext; + + channels = wav_fmt->fmt.channels; + bps = wav_fmt->fmt.bits_per_sample; + fs = wav_fmt->fmt.samples_per_sec; + + /* + * In case of TDM configuration on a ssp, there can + * be more than one blob in which channel masks are + * different for each usecase for a specific rate and bps. + * But the sclk rate will be generated for the total + * number of channels used for that endpoint. + * + * So for the given fs and bps, choose blob which has + * the superset of all channels for that endpoint and + * derive the rate. + */ + for (j = i; j < fmt->fmt_count; j++) { + fmt_cfg = &fmt->fmt_config[j]; + wav_fmt = &fmt_cfg->fmt_ext; + if ((fs == wav_fmt->fmt.samples_per_sec) && + (bps == wav_fmt->fmt.bits_per_sample)) + channels = max_t(u16, channels, + wav_fmt->fmt.channels); + } + + rate = channels * bps * fs; + + /* check if the rate is added already to the given SSP's sclk */ + for (j = 0; (j < SKL_MAX_CLK_RATES) && + (sclk[id].rate_cfg[j].rate != 0); j++) { + if (sclk[id].rate_cfg[j].rate == rate) { + present = true; + break; + } + } + + /* Fill rate and parent for sclk/sclkfs */ + if (!present) { + /* MCLK Divider Source Select */ + i2s_config = (struct skl_i2s_config_blob_legacy *) + fmt->fmt_config[0].config.caps; + clk_src = ((i2s_config->mclk.mdivctrl) + & SKL_MNDSS_DIV_CLK_SRC_MASK) >> + SKL_SHIFT(SKL_MNDSS_DIV_CLK_SRC_MASK); + + parent = skl_get_parent_clk(clk_src); + + /* + * Do not copy the config data if there is no parent + * clock available for this clock source select + */ + if (!parent) + continue; + + sclk[id].rate_cfg[rate_index].rate = rate; + sclk[id].rate_cfg[rate_index].config = fmt_cfg; + sclkfs[id].rate_cfg[rate_index].rate = rate; + sclkfs[id].rate_cfg[rate_index].config = fmt_cfg; + sclk[id].parent_name = parent->name; + sclkfs[id].parent_name = parent->name; + + rate_index++; + } + } +} + +static void skl_get_mclk(struct skl *skl, struct skl_ssp_clk *mclk, + struct nhlt_fmt *fmt, u8 id) +{ + struct skl_i2s_config_blob_legacy *i2s_config; + struct nhlt_specific_cfg *fmt_cfg; + struct skl_clk_parent_src *parent; + u32 clkdiv, div_ratio; + u8 clk_src; + + fmt_cfg = &fmt->fmt_config[0].config; + i2s_config = (struct skl_i2s_config_blob_legacy *)fmt_cfg->caps; + + /* MCLK Divider Source Select */ + clk_src = ((i2s_config->mclk.mdivctrl) & SKL_MCLK_DIV_CLK_SRC_MASK) >> + SKL_SHIFT(SKL_MCLK_DIV_CLK_SRC_MASK); + + clkdiv = i2s_config->mclk.mdivr & SKL_MCLK_DIV_RATIO_MASK; + + /* bypass divider */ + div_ratio = 1; + + if (clkdiv != SKL_MCLK_DIV_RATIO_MASK) + /* Divider is 2 + clkdiv */ + div_ratio = clkdiv + 2; + + /* Calculate MCLK rate from source using div value */ + parent = skl_get_parent_clk(clk_src); + if (!parent) + return; + + mclk[id].rate_cfg[0].rate = parent->rate/div_ratio; + mclk[id].rate_cfg[0].config = &fmt->fmt_config[0]; + mclk[id].parent_name = parent->name; +} + +void skl_get_clks(struct skl *skl, struct skl_ssp_clk *ssp_clks) +{ + struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt; + struct nhlt_endpoint *epnt; + struct nhlt_fmt *fmt; + int i; + u8 id; + + epnt = (struct nhlt_endpoint *)nhlt->desc; + for (i = 0; i < nhlt->endpoint_count; i++) { + if (epnt->linktype == NHLT_LINK_SSP) { + id = epnt->virtual_bus_id; + + fmt = (struct nhlt_fmt *)(epnt->config.caps + + epnt->config.size); + + skl_get_ssp_clks(skl, ssp_clks, fmt, id); + skl_get_mclk(skl, ssp_clks, fmt, id); + } + epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length); + } +} diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 1dd97479e0c0..e46828533826 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -537,7 +537,7 @@ static int skl_link_hw_params(struct snd_pcm_substream *substream, snd_soc_dai_set_dma_data(dai, substream, (void *)link_dev); - link = snd_hdac_ext_bus_get_link(ebus, rtd->codec->component.name); + link = snd_hdac_ext_bus_get_link(ebus, codec_dai->component->name); if (!link) return -EINVAL; @@ -620,7 +620,7 @@ static int skl_link_hw_free(struct snd_pcm_substream *substream, link_dev->link_prepared = 0; - link = snd_hdac_ext_bus_get_link(ebus, rtd->codec->component.name); + link = snd_hdac_ext_bus_get_link(ebus, rtd->codec_dai->component->name); if (!link) return -EINVAL; @@ -1343,7 +1343,11 @@ static int skl_platform_soc_probe(struct snd_soc_platform *platform) return -EIO; } + /* disable dynamic clock gating during fw and lib download */ + skl->skl_sst->enable_miscbdcge(platform->dev, false); + ret = ops->init_fw(platform->dev, skl->skl_sst); + skl->skl_sst->enable_miscbdcge(platform->dev, true); if (ret < 0) { dev_err(platform->dev, "Failed to boot first fw: %d\n", ret); return ret; @@ -1351,6 +1355,12 @@ static int skl_platform_soc_probe(struct snd_soc_platform *platform) skl_populate_modules(skl); skl->skl_sst->update_d0i3c = skl_update_d0i3c; skl_dsp_enable_notification(skl->skl_sst, false); + + if (skl->cfg.astate_cfg != NULL) { + skl_dsp_set_astate_cfg(skl->skl_sst, + skl->cfg.astate_cfg->count, + skl->cfg.astate_cfg); + } } pm_runtime_mark_last_busy(platform->dev); pm_runtime_put_autosuspend(platform->dev); diff --git a/sound/soc/intel/skylake/skl-ssp-clk.h b/sound/soc/intel/skylake/skl-ssp-clk.h new file mode 100644 index 000000000000..c9ea84004260 --- /dev/null +++ b/sound/soc/intel/skylake/skl-ssp-clk.h @@ -0,0 +1,79 @@ +/* + * skl-ssp-clk.h - Skylake ssp clock information and ipc structure + * + * Copyright (C) 2017 Intel Corp + * Author: Jaikrishna Nemallapudi <jaikrishnax.nemallapudi@intel.com> + * Author: Subhransu S. Prusty <subhransu.s.prusty@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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + */ + +#ifndef SOUND_SOC_SKL_SSP_CLK_H +#define SOUND_SOC_SKL_SSP_CLK_H + +#define SKL_MAX_SSP 6 +/* xtal/cardinal/pll, parent of ssp clocks and mclk */ +#define SKL_MAX_CLK_SRC 3 +#define SKL_MAX_SSP_CLK_TYPES 3 /* mclk, sclk, sclkfs */ + +#define SKL_MAX_CLK_CNT (SKL_MAX_SSP * SKL_MAX_SSP_CLK_TYPES) + +/* Max number of configurations supported for each clock */ +#define SKL_MAX_CLK_RATES 10 + +#define SKL_SCLK_OFS SKL_MAX_SSP +#define SKL_SCLKFS_OFS (SKL_SCLK_OFS + SKL_MAX_SSP) + +enum skl_clk_type { + SKL_MCLK, + SKL_SCLK, + SKL_SCLK_FS, +}; + +enum skl_clk_src_type { + SKL_XTAL, + SKL_CARDINAL, + SKL_PLL, +}; + +struct skl_clk_parent_src { + u8 clk_id; + const char *name; + unsigned long rate; + const char *parent_name; +}; + +struct skl_clk_rate_cfg_table { + unsigned long rate; + void *config; +}; + +/* + * rate for mclk will be in rates[0]. For sclk and sclkfs, rates[] store + * all possible clocks ssp can generate for that platform. + */ +struct skl_ssp_clk { + const char *name; + const char *parent_name; + struct skl_clk_rate_cfg_table rate_cfg[SKL_MAX_CLK_RATES]; +}; + +struct skl_clk_pdata { + struct skl_clk_parent_src *parent_clks; + int num_clks; + struct skl_ssp_clk *ssp_clks; + void *pvt_data; +}; + +#endif /* SOUND_SOC_SKL_SSP_CLK_H */ diff --git a/sound/soc/intel/skylake/skl-sst-dsp.c b/sound/soc/intel/skylake/skl-sst-dsp.c index 19ee1d4f3bdf..71e31ad0bb3f 100644 --- a/sound/soc/intel/skylake/skl-sst-dsp.c +++ b/sound/soc/intel/skylake/skl-sst-dsp.c @@ -435,16 +435,22 @@ struct sst_dsp *skl_dsp_ctx_init(struct device *dev, return NULL; } + return sst; +} + +int skl_dsp_acquire_irq(struct sst_dsp *sst) +{ + struct sst_dsp_device *sst_dev = sst->sst_dev; + int ret; + /* Register the ISR */ ret = request_threaded_irq(sst->irq, sst->ops->irq_handler, sst_dev->thread, IRQF_SHARED, "AudioDSP", sst); - if (ret) { + if (ret) dev_err(sst->dev, "unable to grab threaded IRQ %d, disabling device\n", sst->irq); - return NULL; - } - return sst; + return ret; } void skl_dsp_free(struct sst_dsp *dsp) diff --git a/sound/soc/intel/skylake/skl-sst-dsp.h b/sound/soc/intel/skylake/skl-sst-dsp.h index eba20d37ba8c..12fc9a73dc8a 100644 --- a/sound/soc/intel/skylake/skl-sst-dsp.h +++ b/sound/soc/intel/skylake/skl-sst-dsp.h @@ -206,6 +206,7 @@ int skl_cldma_wait_interruptible(struct sst_dsp *ctx); void skl_dsp_set_state_locked(struct sst_dsp *ctx, int state); struct sst_dsp *skl_dsp_ctx_init(struct device *dev, struct sst_dsp_device *sst_dev, int irq); +int skl_dsp_acquire_irq(struct sst_dsp *sst); bool is_skl_dsp_running(struct sst_dsp *ctx); unsigned int skl_dsp_get_enabled_cores(struct sst_dsp *ctx); @@ -251,6 +252,9 @@ void skl_freeup_uuid_list(struct skl_sst *ctx); int skl_dsp_strip_extended_manifest(struct firmware *fw); void skl_dsp_enable_notification(struct skl_sst *ctx, bool enable); + +void skl_dsp_set_astate_cfg(struct skl_sst *ctx, u32 cnt, void *data); + int skl_sst_ctx_init(struct device *dev, int irq, const char *fw_name, struct skl_dsp_loader_ops dsp_ops, struct skl_sst **dsp, struct sst_dsp_device *skl_dev); diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c index 8ff89280d9fd..2ae405617876 100644 --- a/sound/soc/intel/skylake/skl-sst-utils.c +++ b/sound/soc/intel/skylake/skl-sst-utils.c @@ -178,7 +178,8 @@ static inline int skl_pvtid_128(struct uuid_module *module) * skl_get_pvt_id: generate a private id for use as module id * * @ctx: driver context - * @mconfig: module configuration data + * @uuid_mod: module's uuid + * @instance_id: module's instance id * * This generates a 128 bit private unique id for a module TYPE so that * module instance is unique @@ -208,7 +209,8 @@ EXPORT_SYMBOL_GPL(skl_get_pvt_id); * skl_put_pvt_id: free up the private id allocated * * @ctx: driver context - * @mconfig: module configuration data + * @uuid_mod: module's uuid + * @pvt_id: module pvt id * * This frees a 128 bit private unique id previously generated */ diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index a436abf2fe3f..5a7e41b65ef3 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -569,7 +569,7 @@ int skl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, sst->fw_ops = skl_fw_ops; - return 0; + return skl_dsp_acquire_irq(sst); } EXPORT_SYMBOL_GPL(skl_sst_dsp_init); diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index a072bcf209d2..28bc16a8e09a 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -2908,7 +2908,7 @@ static int skl_tplg_control_load(struct snd_soc_component *cmpnt, break; default: - dev_warn(bus->dev, "Control load not supported %d:%d:%d\n", + dev_dbg(bus->dev, "Control load not supported %d:%d:%d\n", hdr->ops.get, hdr->ops.put, hdr->ops.info); break; } @@ -3056,11 +3056,13 @@ static int skl_tplg_get_int_tkn(struct device *dev, struct snd_soc_tplg_vendor_value_elem *tkn_elem, struct skl *skl) { - int tkn_count = 0, ret; + int tkn_count = 0, ret, size; static int mod_idx, res_val_idx, intf_val_idx, dir, pin_idx; struct skl_module_res *res = NULL; struct skl_module_iface *fmt = NULL; struct skl_module *mod = NULL; + static struct skl_astate_param *astate_table; + static int astate_cfg_idx, count; int i; if (skl->modules) { @@ -3093,6 +3095,46 @@ static int skl_tplg_get_int_tkn(struct device *dev, mod_idx = tkn_elem->value; break; + case SKL_TKN_U32_ASTATE_COUNT: + if (astate_table != NULL) { + dev_err(dev, "More than one entry for A-State count"); + return -EINVAL; + } + + if (tkn_elem->value > SKL_MAX_ASTATE_CFG) { + dev_err(dev, "Invalid A-State count %d\n", + tkn_elem->value); + return -EINVAL; + } + + size = tkn_elem->value * sizeof(struct skl_astate_param) + + sizeof(count); + skl->cfg.astate_cfg = devm_kzalloc(dev, size, GFP_KERNEL); + if (!skl->cfg.astate_cfg) + return -ENOMEM; + + astate_table = skl->cfg.astate_cfg->astate_table; + count = skl->cfg.astate_cfg->count = tkn_elem->value; + break; + + case SKL_TKN_U32_ASTATE_IDX: + if (tkn_elem->value >= count) { + dev_err(dev, "Invalid A-State index %d\n", + tkn_elem->value); + return -EINVAL; + } + + astate_cfg_idx = tkn_elem->value; + break; + + case SKL_TKN_U32_ASTATE_KCPS: + astate_table[astate_cfg_idx].kcps = tkn_elem->value; + break; + + case SKL_TKN_U32_ASTATE_CLK_SRC: + astate_table[astate_cfg_idx].clk_src = tkn_elem->value; + break; + case SKL_TKN_U8_IN_PIN_TYPE: case SKL_TKN_U8_OUT_PIN_TYPE: case SKL_TKN_U8_IN_QUEUE_COUNT: diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 31d8634e8aa1..32ce64c6b2dc 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -355,6 +355,7 @@ static int skl_resume(struct device *dev) if (ebus->cmd_dma_state) snd_hdac_bus_init_cmd_io(&ebus->bus); + ret = 0; } else { ret = _skl_resume(ebus); @@ -435,19 +436,51 @@ static int skl_free(struct hdac_ext_bus *ebus) return 0; } -static int skl_machine_device_register(struct skl *skl, void *driver_data) +/* + * For each ssp there are 3 clocks (mclk/sclk/sclkfs). + * e.g. for ssp0, clocks will be named as + * "ssp0_mclk", "ssp0_sclk", "ssp0_sclkfs" + * So for skl+, there are 6 ssps, so 18 clocks will be created. + */ +static struct skl_ssp_clk skl_ssp_clks[] = { + {.name = "ssp0_mclk"}, {.name = "ssp1_mclk"}, {.name = "ssp2_mclk"}, + {.name = "ssp3_mclk"}, {.name = "ssp4_mclk"}, {.name = "ssp5_mclk"}, + {.name = "ssp0_sclk"}, {.name = "ssp1_sclk"}, {.name = "ssp2_sclk"}, + {.name = "ssp3_sclk"}, {.name = "ssp4_sclk"}, {.name = "ssp5_sclk"}, + {.name = "ssp0_sclkfs"}, {.name = "ssp1_sclkfs"}, + {.name = "ssp2_sclkfs"}, + {.name = "ssp3_sclkfs"}, {.name = "ssp4_sclkfs"}, + {.name = "ssp5_sclkfs"}, +}; + +static int skl_find_machine(struct skl *skl, void *driver_data) { - struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); - struct platform_device *pdev; struct snd_soc_acpi_mach *mach = driver_data; - int ret; + struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); + struct skl_machine_pdata *pdata; mach = snd_soc_acpi_find_machine(mach); if (mach == NULL) { dev_err(bus->dev, "No matching machine driver found\n"); return -ENODEV; } + + skl->mach = mach; skl->fw_name = mach->fw_filename; + pdata = skl->mach->pdata; + + if (mach->pdata) + skl->use_tplg_pcm = pdata->use_tplg_pcm; + + return 0; +} + +static int skl_machine_device_register(struct skl *skl) +{ + struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); + struct snd_soc_acpi_mach *mach = skl->mach; + struct platform_device *pdev; + int ret; pdev = platform_device_alloc(mach->drv_name, -1); if (pdev == NULL) { @@ -462,11 +495,8 @@ static int skl_machine_device_register(struct skl *skl, void *driver_data) return -EIO; } - if (mach->pdata) { - skl->use_tplg_pcm = - ((struct skl_machine_pdata *)mach->pdata)->use_tplg_pcm; + if (mach->pdata) dev_set_drvdata(&pdev->dev, mach->pdata); - } skl->i2s_dev = pdev; @@ -509,6 +539,74 @@ static void skl_dmic_device_unregister(struct skl *skl) platform_device_unregister(skl->dmic_dev); } +static struct skl_clk_parent_src skl_clk_src[] = { + { .clk_id = SKL_XTAL, .name = "xtal" }, + { .clk_id = SKL_CARDINAL, .name = "cardinal", .rate = 24576000 }, + { .clk_id = SKL_PLL, .name = "pll", .rate = 96000000 }, +}; + +struct skl_clk_parent_src *skl_get_parent_clk(u8 clk_id) +{ + unsigned int i; + + for (i = 0; i < ARRAY_SIZE(skl_clk_src); i++) { + if (skl_clk_src[i].clk_id == clk_id) + return &skl_clk_src[i]; + } + + return NULL; +} + +static void init_skl_xtal_rate(int pci_id) +{ + switch (pci_id) { + case 0x9d70: + case 0x9d71: + skl_clk_src[0].rate = 24000000; + return; + + default: + skl_clk_src[0].rate = 19200000; + return; + } +} + +static int skl_clock_device_register(struct skl *skl) +{ + struct platform_device_info pdevinfo = {NULL}; + struct skl_clk_pdata *clk_pdata; + + clk_pdata = devm_kzalloc(&skl->pci->dev, sizeof(*clk_pdata), + GFP_KERNEL); + if (!clk_pdata) + return -ENOMEM; + + init_skl_xtal_rate(skl->pci->device); + + clk_pdata->parent_clks = skl_clk_src; + clk_pdata->ssp_clks = skl_ssp_clks; + clk_pdata->num_clks = ARRAY_SIZE(skl_ssp_clks); + + /* Query NHLT to fill the rates and parent */ + skl_get_clks(skl, clk_pdata->ssp_clks); + clk_pdata->pvt_data = skl; + + /* Register Platform device */ + pdevinfo.parent = &skl->pci->dev; + pdevinfo.id = -1; + pdevinfo.name = "skl-ssp-clk"; + pdevinfo.data = clk_pdata; + pdevinfo.size_data = sizeof(*clk_pdata); + skl->clk_dev = platform_device_register_full(&pdevinfo); + return PTR_ERR_OR_ZERO(skl->clk_dev); +} + +static void skl_clock_device_unregister(struct skl *skl) +{ + if (skl->clk_dev) + platform_device_unregister(skl->clk_dev); +} + /* * Probe the given codec address */ @@ -615,18 +713,30 @@ static void skl_probe_work(struct work_struct *work) /* create codec instances */ skl_codec_create(ebus); + /* register platform dai and controls */ + err = skl_platform_register(bus->dev); + if (err < 0) { + dev_err(bus->dev, "platform register failed: %d\n", err); + return; + } + + if (bus->ppcap) { + err = skl_machine_device_register(skl); + if (err < 0) { + dev_err(bus->dev, "machine register failed: %d\n", err); + goto out_err; + } + } + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { err = snd_hdac_display_power(bus, false); if (err < 0) { dev_err(bus->dev, "Cannot turn off display power on i915\n"); + skl_machine_device_unregister(skl); return; } } - /* register platform dai and controls */ - err = skl_platform_register(bus->dev); - if (err < 0) - return; /* * we are done probing so decrement link counts */ @@ -791,18 +901,21 @@ static int skl_probe(struct pci_dev *pci, /* check if dsp is there */ if (bus->ppcap) { - err = skl_machine_device_register(skl, - (void *)pci_id->driver_data); + /* create device for dsp clk */ + err = skl_clock_device_register(skl); + if (err < 0) + goto out_clk_free; + + err = skl_find_machine(skl, (void *)pci_id->driver_data); if (err < 0) goto out_nhlt_free; err = skl_init_dsp(skl); if (err < 0) { dev_dbg(bus->dev, "error failed to register dsp\n"); - goto out_mach_free; + goto out_nhlt_free; } skl->skl_sst->enable_miscbdcge = skl_enable_miscbdcge; - } if (bus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(ebus); @@ -820,8 +933,8 @@ static int skl_probe(struct pci_dev *pci, out_dsp_free: skl_free_dsp(skl); -out_mach_free: - skl_machine_device_unregister(skl); +out_clk_free: + skl_clock_device_unregister(skl); out_nhlt_free: skl_nhlt_free(skl->nhlt); out_free: @@ -872,6 +985,7 @@ static void skl_remove(struct pci_dev *pci) skl_free_dsp(skl); skl_machine_device_unregister(skl); skl_dmic_device_unregister(skl); + skl_clock_device_unregister(skl); skl_nhlt_remove_sysfs(skl); skl_nhlt_free(skl->nhlt); skl_free(ebus); diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index e00cde8200dd..f411579bc713 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -25,9 +25,12 @@ #include <sound/hdaudio_ext.h> #include <sound/soc.h> #include "skl-nhlt.h" +#include "skl-ssp-clk.h" #define SKL_SUSPEND_DELAY 2000 +#define SKL_MAX_ASTATE_CFG 3 + #define AZX_PCIREG_PGCTL 0x44 #define AZX_PGCTL_LSRMD_MASK (1 << 4) #define AZX_PCIREG_CGCTL 0x48 @@ -45,6 +48,20 @@ struct skl_dsp_resource { struct skl_debug; +struct skl_astate_param { + u32 kcps; + u32 clk_src; +}; + +struct skl_astate_config { + u32 count; + struct skl_astate_param astate_table[0]; +}; + +struct skl_fw_config { + struct skl_astate_config *astate_cfg; +}; + struct skl { struct hdac_ext_bus ebus; struct pci_dev *pci; @@ -52,6 +69,7 @@ struct skl { unsigned int init_done:1; /* delayed init status */ struct platform_device *dmic_dev; struct platform_device *i2s_dev; + struct platform_device *clk_dev; struct snd_soc_platform *platform; struct snd_soc_dai_driver *dais; @@ -75,6 +93,8 @@ struct skl { u8 nr_modules; struct skl_module **modules; bool use_tplg_pcm; + struct skl_fw_config cfg; + struct snd_soc_acpi_mach *mach; }; #define skl_to_ebus(s) (&(s)->ebus) @@ -125,6 +145,8 @@ const struct skl_dsp_ops *skl_get_dsp_ops(int pci_id); void skl_update_d0i3c(struct device *dev, bool enable); int skl_nhlt_create_sysfs(struct skl *skl); void skl_nhlt_remove_sysfs(struct skl *skl); +void skl_get_clks(struct skl *skl, struct skl_ssp_clk *ssp_clks); +struct skl_clk_parent_src *skl_get_parent_clk(u8 clk_id); struct skl_module_cfg; diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index ee5055d47d13..a89fe9b6463b 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -322,26 +322,30 @@ static int rk_spdif_probe(struct platform_device *pdev) spdif->mclk = devm_clk_get(&pdev->dev, "mclk"); if (IS_ERR(spdif->mclk)) { dev_err(&pdev->dev, "Can't retrieve rk_spdif master clock\n"); - return PTR_ERR(spdif->mclk); + ret = PTR_ERR(spdif->mclk); + goto err_disable_hclk; } ret = clk_prepare_enable(spdif->mclk); if (ret) { dev_err(spdif->dev, "clock enable failed %d\n", ret); - return ret; + goto err_disable_clocks; } res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(regs)) - return PTR_ERR(regs); + if (IS_ERR(regs)) { + ret = PTR_ERR(regs); + goto err_disable_clocks; + } spdif->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "hclk", regs, &rk_spdif_regmap_config); if (IS_ERR(spdif->regmap)) { dev_err(&pdev->dev, "Failed to initialise managed register map\n"); - return PTR_ERR(spdif->regmap); + ret = PTR_ERR(spdif->regmap); + goto err_disable_clocks; } spdif->playback_dma_data.addr = res->start + SPDIF_SMPDR; @@ -373,6 +377,10 @@ static int rk_spdif_probe(struct platform_device *pdev) err_pm_runtime: pm_runtime_disable(&pdev->dev); +err_disable_clocks: + clk_disable_unprepare(spdif->mclk); +err_disable_hclk: + clk_disable_unprepare(spdif->hclk); return ret; } diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 8ddb08714faa..4672688cac32 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -222,7 +222,7 @@ int rsnd_adg_set_cmd_timsel_gen2(struct rsnd_mod *cmd_mod, NULL, &val, NULL); val = val << shift; - mask = 0xffff << shift; + mask = 0x0f1f << shift; rsnd_mod_bset(adg_mod, CMDOUT_TIMSEL, mask, val); @@ -250,7 +250,7 @@ int rsnd_adg_set_src_timesel_gen2(struct rsnd_mod *src_mod, in = in << shift; out = out << shift; - mask = 0xffff << shift; + mask = 0x0f1f << shift; switch (id / 2) { case 0: @@ -380,7 +380,7 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *ssi_mod, unsigned int rate) ckr = 0x80000000; } - rsnd_mod_bset(adg_mod, BRGCKR, 0x80FF0000, adg->ckr | ckr); + rsnd_mod_bset(adg_mod, BRGCKR, 0x80770000, adg->ckr | ckr); rsnd_mod_write(adg_mod, BRRA, adg->rbga); rsnd_mod_write(adg_mod, BRRB, adg->rbgb); diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index c70eb2097816..f12a88a21dfa 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1332,8 +1332,8 @@ static int rsnd_pcm_new(struct snd_soc_pcm_runtime *rtd) return snd_pcm_lib_preallocate_pages_for_all( rtd->pcm, - SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), + SNDRV_DMA_TYPE_DEV, + rtd->card->snd_card->dev, PREALLOC_BUFFER, PREALLOC_BUFFER_MAX); } diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index fd557abfe390..4d750bdf8e24 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -26,10 +26,7 @@ struct rsnd_dmaen { struct dma_chan *chan; dma_cookie_t cookie; - dma_addr_t dma_buf; unsigned int dma_len; - unsigned int dma_period; - unsigned int dma_cnt; }; struct rsnd_dmapp { @@ -71,38 +68,10 @@ static struct rsnd_mod mem = { /* * Audio DMAC */ -#define rsnd_dmaen_sync(dmaen, io, i) __rsnd_dmaen_sync(dmaen, io, i, 1) -#define rsnd_dmaen_unsync(dmaen, io, i) __rsnd_dmaen_sync(dmaen, io, i, 0) -static void __rsnd_dmaen_sync(struct rsnd_dmaen *dmaen, struct rsnd_dai_stream *io, - int i, int sync) -{ - struct device *dev = dmaen->chan->device->dev; - enum dma_data_direction dir; - int is_play = rsnd_io_is_play(io); - dma_addr_t buf; - int len, max; - size_t period; - - len = dmaen->dma_len; - period = dmaen->dma_period; - max = len / period; - i = i % max; - buf = dmaen->dma_buf + (period * i); - - dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; - - if (sync) - dma_sync_single_for_device(dev, buf, period, dir); - else - dma_sync_single_for_cpu(dev, buf, period, dir); -} - static void __rsnd_dmaen_complete(struct rsnd_mod *mod, struct rsnd_dai_stream *io) { struct rsnd_priv *priv = rsnd_mod_to_priv(mod); - struct rsnd_dma *dma = rsnd_mod_to_dma(mod); - struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); bool elapsed = false; unsigned long flags; @@ -115,22 +84,9 @@ static void __rsnd_dmaen_complete(struct rsnd_mod *mod, */ spin_lock_irqsave(&priv->lock, flags); - if (rsnd_io_is_working(io)) { - rsnd_dmaen_unsync(dmaen, io, dmaen->dma_cnt); - - /* - * Next period is already started. - * Let's sync Next Next period - * see - * rsnd_dmaen_start() - */ - rsnd_dmaen_sync(dmaen, io, dmaen->dma_cnt + 2); - + if (rsnd_io_is_working(io)) elapsed = true; - dmaen->dma_cnt++; - } - spin_unlock_irqrestore(&priv->lock, flags); if (elapsed) @@ -165,14 +121,8 @@ static int rsnd_dmaen_stop(struct rsnd_mod *mod, struct rsnd_dma *dma = rsnd_mod_to_dma(mod); struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); - if (dmaen->chan) { - int is_play = rsnd_io_is_play(io); - + if (dmaen->chan) dmaengine_terminate_all(dmaen->chan); - dma_unmap_single(dmaen->chan->device->dev, - dmaen->dma_buf, dmaen->dma_len, - is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE); - } return 0; } @@ -237,11 +187,7 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, struct device *dev = rsnd_priv_to_dev(priv); struct dma_async_tx_descriptor *desc; struct dma_slave_config cfg = {}; - dma_addr_t buf; - size_t len; - size_t period; int is_play = rsnd_io_is_play(io); - int i; int ret; cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; @@ -258,19 +204,10 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, if (ret < 0) return ret; - len = snd_pcm_lib_buffer_bytes(substream); - period = snd_pcm_lib_period_bytes(substream); - buf = dma_map_single(dmaen->chan->device->dev, - substream->runtime->dma_area, - len, - is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE); - if (dma_mapping_error(dmaen->chan->device->dev, buf)) { - dev_err(dev, "dma map failed\n"); - return -EIO; - } - desc = dmaengine_prep_dma_cyclic(dmaen->chan, - buf, len, period, + substream->runtime->dma_addr, + snd_pcm_lib_buffer_bytes(substream), + snd_pcm_lib_period_bytes(substream), is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM, DMA_PREP_INTERRUPT | DMA_CTRL_ACK); @@ -282,18 +219,7 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, desc->callback = rsnd_dmaen_complete; desc->callback_param = rsnd_mod_get(dma); - dmaen->dma_buf = buf; - dmaen->dma_len = len; - dmaen->dma_period = period; - dmaen->dma_cnt = 0; - - /* - * synchronize this and next period - * see - * __rsnd_dmaen_complete() - */ - for (i = 0; i < 2; i++) - rsnd_dmaen_sync(dmaen, io, i); + dmaen->dma_len = snd_pcm_lib_buffer_bytes(substream); dmaen->cookie = dmaengine_submit(desc); if (dmaen->cookie < 0) { diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index fece1e5f582f..cbf3bf312d23 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -446,25 +446,29 @@ static bool rsnd_ssi_pointer_update(struct rsnd_mod *mod, int byte) { struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + bool ret = false; + int byte_pos; - ssi->byte_pos += byte; + byte_pos = ssi->byte_pos + byte; - if (ssi->byte_pos >= ssi->next_period_byte) { + if (byte_pos >= ssi->next_period_byte) { struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); ssi->period_pos++; ssi->next_period_byte += ssi->byte_per_period; if (ssi->period_pos >= runtime->periods) { - ssi->byte_pos = 0; + byte_pos = 0; ssi->period_pos = 0; ssi->next_period_byte = ssi->byte_per_period; } - return true; + ret = true; } - return false; + WRITE_ONCE(ssi->byte_pos, byte_pos); + + return ret; } /* @@ -838,7 +842,7 @@ static int rsnd_ssi_pointer(struct rsnd_mod *mod, struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); - *pointer = bytes_to_frames(runtime, ssi->byte_pos); + *pointer = bytes_to_frames(runtime, READ_ONCE(ssi->byte_pos)); return 0; } diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 4d948757d300..6ff8a36c2c82 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -125,6 +125,7 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, { int hdmi = rsnd_ssi_hdmi_port(io); int ret; + u32 mode = 0; ret = rsnd_ssiu_init(mod, io, priv); if (ret < 0) @@ -136,9 +137,11 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, * see * rsnd_ssi_config_init() */ - rsnd_mod_write(mod, SSI_MODE, 0x1); + mode = 0x1; } + rsnd_mod_write(mod, SSI_MODE, mode); + if (rsnd_ssi_use_busif(io)) { rsnd_mod_write(mod, SSI_BUSIF_ADINR, rsnd_get_adinr_bit(mod, io) | 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; } |