diff options
Diffstat (limited to 'sound/soc')
102 files changed, 1175 insertions, 572 deletions
diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index d870f56c44cf..0341b3119767 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -163,11 +163,14 @@ struct atmel_i2s_gck_param { #define I2S_MCK_12M288 12288000UL #define I2S_MCK_11M2896 11289600UL +#define I2S_MCK_6M144 6144000UL /* mck = (32 * (imckfs+1) / (imckdiv+1)) * fs */ static const struct atmel_i2s_gck_param gck_params[] = { + /* mck = 6.144Mhz */ + { 8000, I2S_MCK_6M144, 1, 47}, /* mck = 768 fs */ + /* mck = 12.288MHz */ - { 8000, I2S_MCK_12M288, 0, 47}, /* mck = 1536 fs */ { 16000, I2S_MCK_12M288, 1, 47}, /* mck = 768 fs */ { 24000, I2S_MCK_12M288, 3, 63}, /* mck = 512 fs */ { 32000, I2S_MCK_12M288, 3, 47}, /* mck = 384 fs */ diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index d1579896f3a1..05277a88e20d 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -46,6 +46,35 @@ */ #undef ENABLE_MIC_INPUT +static struct clk *mclk; + +static int at91sam9g20ek_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + static int mclk_on; + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + if (!mclk_on) + ret = clk_enable(mclk); + if (ret == 0) + mclk_on = 1; + break; + + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_STANDBY: + if (mclk_on) + clk_disable(mclk); + mclk_on = 0; + break; + } + + return ret; +} + static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = { SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_SPK("Ext Spk", NULL), @@ -106,6 +135,7 @@ static struct snd_soc_card snd_soc_at91sam9g20ek = { .owner = THIS_MODULE, .dai_link = &at91sam9g20ek_dai, .num_links = 1, + .set_bias_level = at91sam9g20ek_set_bias_level, .dapm_widgets = at91sam9g20ek_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(at91sam9g20ek_dapm_widgets), @@ -118,6 +148,7 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; struct device_node *codec_np, *cpu_np; + struct clk *pllb; struct snd_soc_card *card = &snd_soc_at91sam9g20ek; int ret; @@ -131,6 +162,31 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) return -EINVAL; } + /* + * Codec MCLK is supplied by PCK0 - set it up. + */ + mclk = clk_get(NULL, "pck0"); + if (IS_ERR(mclk)) { + dev_err(&pdev->dev, "Failed to get MCLK\n"); + ret = PTR_ERR(mclk); + goto err; + } + + pllb = clk_get(NULL, "pllb"); + if (IS_ERR(pllb)) { + dev_err(&pdev->dev, "Failed to get PLLB\n"); + ret = PTR_ERR(pllb); + goto err_mclk; + } + ret = clk_set_parent(mclk, pllb); + clk_put(pllb); + if (ret != 0) { + dev_err(&pdev->dev, "Failed to set MCLK parent\n"); + goto err_mclk; + } + + clk_set_rate(mclk, MCLK_RATE); + card->dev = &pdev->dev; /* Parse device node info */ @@ -174,6 +230,9 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) return ret; +err_mclk: + clk_put(mclk); + mclk = NULL; err: atmel_ssc_put_audio(0); return ret; @@ -183,6 +242,8 @@ static int at91sam9g20ek_audio_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); + clk_disable(mclk); + mclk = NULL; snd_soc_unregister_card(card); atmel_ssc_put_audio(0); diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 466dc67799f4..dfc536cd9d2f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -759,7 +759,6 @@ config SND_SOC_MAX98095 config SND_SOC_MAX98357A tristate "Maxim MAX98357A CODEC" - depends on GPIOLIB config SND_SOC_MAX98371 tristate diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c index 8894369e329a..87b299d24bd8 100644 --- a/sound/soc/codecs/cs35l33.c +++ b/sound/soc/codecs/cs35l33.c @@ -22,13 +22,11 @@ #include <sound/soc-dapm.h> #include <sound/initval.h> #include <sound/tlv.h> -#include <linux/gpio.h> #include <linux/gpio/consumer.h> #include <sound/cs35l33.h> #include <linux/pm_runtime.h> #include <linux/regulator/consumer.h> #include <linux/regulator/machine.h> -#include <linux/of_gpio.h> #include <linux/of.h> #include <linux/of_device.h> #include <linux/of_irq.h> @@ -1168,7 +1166,7 @@ static int cs35l33_i2c_probe(struct i2c_client *i2c_client, /* We could issue !RST or skip it based on AMP topology */ cs35l33->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, - "reset-gpios", GPIOD_OUT_HIGH); + "reset", GPIOD_OUT_HIGH); if (IS_ERR(cs35l33->reset_gpio)) { dev_err(&i2c_client->dev, "%s ERROR: Can't get reset GPIO\n", __func__); diff --git a/sound/soc/codecs/cs35l34.c b/sound/soc/codecs/cs35l34.c index b792c006e530..d9f975b52b21 100644 --- a/sound/soc/codecs/cs35l34.c +++ b/sound/soc/codecs/cs35l34.c @@ -20,14 +20,12 @@ #include <linux/regulator/machine.h> #include <linux/pm_runtime.h> #include <linux/of_device.h> -#include <linux/of_gpio.h> #include <linux/of_irq.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-dapm.h> -#include <linux/gpio.h> #include <linux/gpio/consumer.h> #include <sound/initval.h> #include <sound/tlv.h> @@ -1058,7 +1056,7 @@ static int cs35l34_i2c_probe(struct i2c_client *i2c_client, dev_err(&i2c_client->dev, "Failed to request IRQ: %d\n", ret); cs35l34->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, - "reset-gpios", GPIOD_OUT_LOW); + "reset", GPIOD_OUT_LOW); if (IS_ERR(cs35l34->reset_gpio)) return PTR_ERR(cs35l34->reset_gpio); diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index e9b5f76f27a8..aa32b8c26578 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -444,7 +444,8 @@ static bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) } } -static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 25, 0); +static const DECLARE_TLV_DB_RANGE(dig_vol_tlv, 0, 912, + TLV_DB_MINMAX_ITEM(-10200, 1200)); static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); static const char * const cs35l36_pcm_sftramp_text[] = { diff --git a/sound/soc/codecs/cs42l51-i2c.c b/sound/soc/codecs/cs42l51-i2c.c index 70260e0a8f09..3ff73367897d 100644 --- a/sound/soc/codecs/cs42l51-i2c.c +++ b/sound/soc/codecs/cs42l51-i2c.c @@ -19,6 +19,12 @@ static struct i2c_device_id cs42l51_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, cs42l51_i2c_id); +const struct of_device_id cs42l51_of_match[] = { + { .compatible = "cirrus,cs42l51", }, + { } +}; +MODULE_DEVICE_TABLE(of, cs42l51_of_match); + static int cs42l51_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index cdd7ae90c2b5..07371e32167c 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -811,13 +811,6 @@ int __maybe_unused cs42l51_resume(struct device *dev) } EXPORT_SYMBOL_GPL(cs42l51_resume); -const struct of_device_id cs42l51_of_match[] = { - { .compatible = "cirrus,cs42l51", }, - { } -}; -MODULE_DEVICE_TABLE(of, cs42l51_of_match); -EXPORT_SYMBOL_GPL(cs42l51_of_match); - MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>"); MODULE_DESCRIPTION("Cirrus Logic CS42L51 ALSA SoC Codec Driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l51.h b/sound/soc/codecs/cs42l51.h index 9d06cf7f8876..4f13c38484b7 100644 --- a/sound/soc/codecs/cs42l51.h +++ b/sound/soc/codecs/cs42l51.h @@ -16,7 +16,6 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap); int cs42l51_remove(struct device *dev); int __maybe_unused cs42l51_suspend(struct device *dev); int __maybe_unused cs42l51_resume(struct device *dev); -extern const struct of_device_id cs42l51_of_match[]; #define CS42L51_CHIP_ID 0x1B #define CS42L51_CHIP_REV_A 0x00 diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 2ea4cba3be2a..6c054b357205 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,7 +137,9 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); -static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(pass_tlv, -6000, 50, 0); + +static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -351,7 +353,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv), SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, - CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv), + CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pass_tlv), SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), @@ -364,7 +366,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 0, 0x19, 0x7F, ipd_tlv), + 0, 0x19, 0x7F, mix_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 51d7a87ab4c3..732405587c5a 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -391,9 +391,9 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1), SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, - CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_HPB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, - CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_LOB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, 0, 0x00, 1, tone_tlv), @@ -1192,18 +1192,12 @@ static int cs42l56_i2c_probe(struct i2c_client *i2c_client, if (pdata) { cs42l56->pdata = *pdata; } else { - pdata = devm_kzalloc(&i2c_client->dev, sizeof(*pdata), - GFP_KERNEL); - if (!pdata) - return -ENOMEM; - if (i2c_client->dev.of_node) { ret = cs42l56_handle_of_data(i2c_client, &cs42l56->pdata); if (ret != 0) return ret; } - cs42l56->pdata = *pdata; } if (cs42l56->pdata.gpio_nreset) { diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 8f70dee95878..02fb9317b697 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -578,7 +578,7 @@ static int cs43130_set_sp_fmt(int dai_id, unsigned int bitwidth_sclk, break; case SND_SOC_DAIFMT_LEFT_J: hi_size = bitwidth_sclk; - frm_delay = 2; + frm_delay = 0; frm_phase = 1; break; case SND_SOC_DAIFMT_DSP_A: @@ -1683,7 +1683,7 @@ static ssize_t cs43130_show_dc_r(struct device *dev, return cs43130_show_dc(dev, buf, HP_RIGHT); } -static u16 const cs43130_ac_freq[CS43130_AC_FREQ] = { +static const u16 cs43130_ac_freq[CS43130_AC_FREQ] = { 24, 43, 93, @@ -2364,7 +2364,7 @@ static const struct regmap_config cs43130_regmap = { .use_single_write = true, }; -static u16 const cs43130_dc_threshold[CS43130_DC_THRESHOLD] = { +static const u16 cs43130_dc_threshold[CS43130_DC_THRESHOLD] = { 50, 120, }; diff --git a/sound/soc/codecs/cs47l15.c b/sound/soc/codecs/cs47l15.c index ece1276f38eb..1f7148794a5a 100644 --- a/sound/soc/codecs/cs47l15.c +++ b/sound/soc/codecs/cs47l15.c @@ -122,6 +122,9 @@ static int cs47l15_in1_adc_put(struct snd_kcontrol *kcontrol, snd_soc_kcontrol_component(kcontrol); struct cs47l15 *cs47l15 = snd_soc_component_get_drvdata(component); + if (!!ucontrol->value.integer.value[0] == cs47l15->in1_lp_mode) + return 0; + switch (ucontrol->value.integer.value[0]) { case 0: /* Set IN1 to normal mode */ @@ -150,7 +153,7 @@ static int cs47l15_in1_adc_put(struct snd_kcontrol *kcontrol, break; } - return 0; + return 1; } static const struct snd_kcontrol_new cs47l15_snd_controls[] = { diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index ed22361b35c1..a5a383b92305 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -347,22 +347,22 @@ static const struct snd_kcontrol_new cs53l30_snd_controls[] = { SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), SOC_SINGLE_SX_TLV("ADC1A PGA Volume", - CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1B PGA Volume", - CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2A PGA Volume", - CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2B PGA Volume", - CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1A Digital Volume", - CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC1B Digital Volume", - CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2A Digital Volume", - CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2B Digital Volume", - CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), }; static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index e172913d04a4..efc5049c0796 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1333,6 +1333,8 @@ static int __init da7210_modinit(void) int ret = 0; #if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&da7210_i2c_driver); + if (ret) + return ret; #endif #if defined(CONFIG_SPI_MASTER) ret = spi_register_driver(&da7210_spi_driver); diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 4f2a96e9fd45..e4e314604c0a 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -347,11 +347,15 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); u8 events[DA7219_AAD_IRQ_REG_MAX]; u8 statusa; - int i, report = 0, mask = 0; + int i, ret, report = 0, mask = 0; /* Read current IRQ events */ - regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, - events, DA7219_AAD_IRQ_REG_MAX); + ret = regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, + events, DA7219_AAD_IRQ_REG_MAX); + if (ret) { + dev_warn_ratelimited(component->dev, "Failed to read IRQ events: %d\n", ret); + return IRQ_NONE; + } if (!events[DA7219_AAD_IRQ_REG_A] && !events[DA7219_AAD_IRQ_REG_B]) return IRQ_NONE; @@ -651,7 +655,7 @@ static struct da7219_aad_pdata *da7219_aad_fw_to_pdata(struct snd_soc_component aad_pdata->mic_det_thr = da7219_aad_fw_mic_det_thr(component, fw_val32); else - aad_pdata->mic_det_thr = DA7219_AAD_MIC_DET_THR_500_OHMS; + aad_pdata->mic_det_thr = DA7219_AAD_MIC_DET_THR_200_OHMS; if (fwnode_property_read_u32(aad_np, "dlg,jack-ins-deb", &fw_val32) >= 0) aad_pdata->jack_ins_deb = @@ -855,6 +859,8 @@ void da7219_aad_suspend(struct snd_soc_component *component) } } } + + synchronize_irq(da7219_aad->irq); } void da7219_aad_resume(struct snd_soc_component *component) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index f83a6eaba12c..ef8bd9e04637 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -446,7 +446,7 @@ static int da7219_tonegen_freq_put(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mixer_ctrl = (struct soc_mixer_control *) kcontrol->private_value; unsigned int reg = mixer_ctrl->reg; - __le16 val; + __le16 val_new, val_old; int ret; /* @@ -454,13 +454,19 @@ static int da7219_tonegen_freq_put(struct snd_kcontrol *kcontrol, * Therefore we need to convert to little endian here to align with * HW registers. */ - val = cpu_to_le16(ucontrol->value.integer.value[0]); + val_new = cpu_to_le16(ucontrol->value.integer.value[0]); mutex_lock(&da7219->ctrl_lock); - ret = regmap_raw_write(da7219->regmap, reg, &val, sizeof(val)); + ret = regmap_raw_read(da7219->regmap, reg, &val_old, sizeof(val_old)); + if (ret == 0 && (val_old != val_new)) + ret = regmap_raw_write(da7219->regmap, reg, + &val_new, sizeof(val_new)); mutex_unlock(&da7219->ctrl_lock); - return ret; + if (ret < 0) + return ret; + + return val_old != val_new; } diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index b781b28de012..dd2df9a903e0 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -52,7 +52,12 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(dac_vol_tlv, -9600, 50, 1); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9600, 50, 1); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_max_gain_tlv, -650, 150, 0); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_min_gain_tlv, -1200, 150, 0); -static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_target_tlv, -1650, 150, 0); + +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(alc_target_tlv, + 0, 10, TLV_DB_SCALE_ITEM(-1650, 150, 0), + 11, 11, TLV_DB_SCALE_ITEM(-150, 0, 0), +); + static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpmixer_gain_tlv, 0, 4, TLV_DB_SCALE_ITEM(-1200, 150, 0), 8, 11, TLV_DB_SCALE_ITEM(-450, 150, 0), @@ -115,7 +120,7 @@ static const struct snd_kcontrol_new es8316_snd_controls[] = { alc_max_gain_tlv), SOC_SINGLE_TLV("ALC Capture Min Volume", ES8316_ADC_ALC2, 0, 28, 0, alc_min_gain_tlv), - SOC_SINGLE_TLV("ALC Capture Target Volume", ES8316_ADC_ALC3, 4, 10, 0, + SOC_SINGLE_TLV("ALC Capture Target Volume", ES8316_ADC_ALC3, 4, 11, 0, alc_target_tlv), SOC_SINGLE("ALC Capture Hold Time", ES8316_ADC_ALC3, 0, 10, 0), SOC_SINGLE("ALC Capture Decay Time", ES8316_ADC_ALC4, 4, 10, 0), @@ -148,7 +153,7 @@ static const char * const es8316_dmic_txt[] = { "dmic data at high level", "dmic data at low level", }; -static const unsigned int es8316_dmic_values[] = { 0, 1, 2 }; +static const unsigned int es8316_dmic_values[] = { 0, 2, 3 }; static const struct soc_enum es8316_dmic_src_enum = SOC_VALUE_ENUM_SINGLE(ES8316_ADC_DMIC, 0, 3, ARRAY_SIZE(es8316_dmic_txt), @@ -364,13 +369,11 @@ static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai, int count = 0; es8316->sysclk = freq; + es8316->sysclk_constraints.list = NULL; + es8316->sysclk_constraints.count = 0; - if (freq == 0) { - es8316->sysclk_constraints.list = NULL; - es8316->sysclk_constraints.count = 0; - + if (freq == 0) return 0; - } ret = clk_set_rate(es8316->mclk, freq); if (ret) @@ -386,8 +389,10 @@ static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai, es8316->allowed_rates[count++] = freq / ratio; } - es8316->sysclk_constraints.list = es8316->allowed_rates; - es8316->sysclk_constraints.count = count; + if (count) { + es8316->sysclk_constraints.list = es8316->allowed_rates; + es8316->sysclk_constraints.count = count; + } return 0; } @@ -806,15 +811,14 @@ static int es8316_i2c_probe(struct i2c_client *i2c_client, es8316->irq = i2c_client->irq; mutex_init(&es8316->lock); - ret = devm_request_threaded_irq(dev, es8316->irq, NULL, es8316_irq, - IRQF_TRIGGER_HIGH | IRQF_ONESHOT, - "es8316", es8316); - if (ret == 0) { - /* Gets re-enabled by es8316_set_jack() */ - disable_irq(es8316->irq); - } else { - dev_warn(dev, "Failed to get IRQ %d: %d\n", es8316->irq, ret); - es8316->irq = -ENXIO; + if (es8316->irq > 0) { + ret = devm_request_threaded_irq(dev, es8316->irq, NULL, es8316_irq, + IRQF_TRIGGER_HIGH | IRQF_ONESHOT | IRQF_NO_AUTOEN, + "es8316", es8316); + if (ret) { + dev_warn(dev, "Failed to get IRQ %d: %d\n", es8316->irq, ret); + es8316->irq = -ENXIO; + } } return devm_snd_soc_register_component(&i2c_client->dev, diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index fdf64c29f563..4117ab6e9b6f 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -161,13 +161,16 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, if (deemph > 1) return -EINVAL; + if (es8328->deemph == deemph) + return 0; + ret = es8328_set_deemph(component); if (ret < 0) return ret; es8328->deemph = deemph; - return 0; + return 1; } diff --git a/sound/soc/codecs/jz4725b.c b/sound/soc/codecs/jz4725b.c index 2567a5d15b55..48c40a44b763 100644 --- a/sound/soc/codecs/jz4725b.c +++ b/sound/soc/codecs/jz4725b.c @@ -136,14 +136,17 @@ enum { #define REG_CGR3_GO1L_OFFSET 0 #define REG_CGR3_GO1L_MASK (0x1f << REG_CGR3_GO1L_OFFSET) +#define REG_CGR10_GIL_OFFSET 0 +#define REG_CGR10_GIR_OFFSET 4 + struct jz_icdc { struct regmap *regmap; void __iomem *base; struct clk *clk; }; -static const SNDRV_CTL_TLVD_DECLARE_DB_LINEAR(jz4725b_dac_tlv, -2250, 0); -static const SNDRV_CTL_TLVD_DECLARE_DB_LINEAR(jz4725b_line_tlv, -1500, 600); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(jz4725b_adc_tlv, 0, 150, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(jz4725b_dac_tlv, -2250, 150, 0); static const struct snd_kcontrol_new jz4725b_codec_controls[] = { SOC_DOUBLE_TLV("Master Playback Volume", @@ -151,11 +154,11 @@ static const struct snd_kcontrol_new jz4725b_codec_controls[] = { REG_CGR1_GODL_OFFSET, REG_CGR1_GODR_OFFSET, 0xf, 1, jz4725b_dac_tlv), - SOC_DOUBLE_R_TLV("Master Capture Volume", - JZ4725B_CODEC_REG_CGR3, - JZ4725B_CODEC_REG_CGR2, - REG_CGR2_GO1R_OFFSET, - 0x1f, 1, jz4725b_line_tlv), + SOC_DOUBLE_TLV("Master Capture Volume", + JZ4725B_CODEC_REG_CGR10, + REG_CGR10_GIL_OFFSET, + REG_CGR10_GIR_OFFSET, + 0xf, 0, jz4725b_adc_tlv), SOC_SINGLE("Master Playback Switch", JZ4725B_CODEC_REG_CR1, REG_CR1_DAC_MUTE_OFFSET, 1, 1), @@ -180,7 +183,7 @@ static SOC_VALUE_ENUM_SINGLE_DECL(jz4725b_codec_adc_src_enum, jz4725b_codec_adc_src_texts, jz4725b_codec_adc_src_values); static const struct snd_kcontrol_new jz4725b_codec_adc_src_ctrl = - SOC_DAPM_ENUM("Route", jz4725b_codec_adc_src_enum); + SOC_DAPM_ENUM("ADC Source Capture Route", jz4725b_codec_adc_src_enum); static const struct snd_kcontrol_new jz4725b_codec_mixer_controls[] = { SOC_DAPM_SINGLE("Line In Bypass", JZ4725B_CODEC_REG_CR1, @@ -225,7 +228,7 @@ static const struct snd_soc_dapm_widget jz4725b_codec_dapm_widgets[] = { SND_SOC_DAPM_ADC("ADC", "Capture", JZ4725B_CODEC_REG_PMR1, REG_PMR1_SB_ADC_OFFSET, 1), - SND_SOC_DAPM_MUX("ADC Source", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_MUX("ADC Source Capture Route", SND_SOC_NOPM, 0, 0, &jz4725b_codec_adc_src_ctrl), /* Mixer */ @@ -236,7 +239,8 @@ static const struct snd_soc_dapm_widget jz4725b_codec_dapm_widgets[] = { SND_SOC_DAPM_MIXER("DAC to Mixer", JZ4725B_CODEC_REG_CR1, REG_CR1_DACSEL_OFFSET, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Line In", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Line In", JZ4725B_CODEC_REG_PMR1, + REG_PMR1_SB_LIN_OFFSET, 1, NULL, 0), SND_SOC_DAPM_MIXER("HP Out", JZ4725B_CODEC_REG_CR1, REG_CR1_HP_DIS_OFFSET, 1, NULL, 0), @@ -283,11 +287,11 @@ static const struct snd_soc_dapm_route jz4725b_codec_dapm_routes[] = { {"Mixer", NULL, "DAC to Mixer"}, {"Mixer to ADC", NULL, "Mixer"}, - {"ADC Source", "Mixer", "Mixer to ADC"}, - {"ADC Source", "Line In", "Line In"}, - {"ADC Source", "Mic 1", "Mic 1"}, - {"ADC Source", "Mic 2", "Mic 2"}, - {"ADC", NULL, "ADC Source"}, + {"ADC Source Capture Route", "Mixer", "Mixer to ADC"}, + {"ADC Source Capture Route", "Line In", "Line In"}, + {"ADC Source Capture Route", "Mic 1", "Mic 1"}, + {"ADC Source Capture Route", "Mic 2", "Mic 2"}, + {"ADC", NULL, "ADC Source Capture Route"}, {"Out Stage", NULL, "Mixer"}, {"HP Out", NULL, "Out Stage"}, diff --git a/sound/soc/codecs/madera.c b/sound/soc/codecs/madera.c index 52639811cc52..4a56082a4c43 100644 --- a/sound/soc/codecs/madera.c +++ b/sound/soc/codecs/madera.c @@ -568,7 +568,13 @@ int madera_out1_demux_put(struct snd_kcontrol *kcontrol, end: snd_soc_dapm_mutex_unlock(dapm); - return snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + ret = snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + if (ret < 0) { + dev_err(madera->dev, "Failed to update demux power state: %d\n", ret); + return ret; + } + + return change; } EXPORT_SYMBOL_GPL(madera_out1_demux_put); @@ -847,7 +853,7 @@ static int madera_adsp_rate_put(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; const int adsp_num = e->shift_l; const unsigned int item = ucontrol->value.enumerated.item[0]; - int ret; + int ret = 0; if (item >= e->items) return -EINVAL; @@ -864,10 +870,10 @@ static int madera_adsp_rate_put(struct snd_kcontrol *kcontrol, "Cannot change '%s' while in use by active audio paths\n", kcontrol->id.name); ret = -EBUSY; - } else { + } else if (priv->adsp_rate_cache[adsp_num] != e->values[item]) { /* Volatile register so defer until the codec is powered up */ priv->adsp_rate_cache[adsp_num] = e->values[item]; - ret = 0; + ret = 1; } mutex_unlock(&priv->rate_lock); diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 6b9d326e11b0..ce9f99dd3e87 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -413,6 +413,9 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, val = (val >> mc->shift) & mask; + if (sel < 0 || sel > mc->max) + return -EINVAL; + *select = sel; /* Setting a volume is only valid if it is already On */ @@ -427,7 +430,7 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, mask << mc->shift, sel << mc->shift); - return 0; + return *select != val; } static const char *max98090_perf_pwr_text[] = diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index e4cde214b7b2..6e5bce4f5eb2 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -328,8 +328,8 @@ static const struct snd_kcontrol_new rx1_mix2_inp1_mux = SOC_DAPM_ENUM( static const struct snd_kcontrol_new rx2_mix2_inp1_mux = SOC_DAPM_ENUM( "RX2 MIX2 INP1 Mux", rx2_mix2_inp1_chain_enum); -/* Digital Gain control -38.4 dB to +38.4 dB in 0.3 dB steps */ -static const DECLARE_TLV_DB_SCALE(digital_gain, -3840, 30, 0); +/* Digital Gain control -84 dB to +40 dB in 1 dB steps */ +static const DECLARE_TLV_DB_SCALE(digital_gain, -8400, 100, -8400); /* Cutoff Freq for High Pass Filter at -3dB */ static const char * const hpf_cutoff_text[] = { @@ -510,15 +510,15 @@ static int wcd_iir_filter_info(struct snd_kcontrol *kcontrol, static const struct snd_kcontrol_new msm8916_wcd_digital_snd_controls[] = { SOC_SINGLE_S8_TLV("RX1 Digital Volume", LPASS_CDC_RX1_VOL_CTL_B2_CTL, - -128, 127, digital_gain), + -84, 40, digital_gain), SOC_SINGLE_S8_TLV("RX2 Digital Volume", LPASS_CDC_RX2_VOL_CTL_B2_CTL, - -128, 127, digital_gain), + -84, 40, digital_gain), SOC_SINGLE_S8_TLV("RX3 Digital Volume", LPASS_CDC_RX3_VOL_CTL_B2_CTL, - -128, 127, digital_gain), + -84, 40, digital_gain), SOC_SINGLE_S8_TLV("TX1 Digital Volume", LPASS_CDC_TX1_VOL_CTL_GAIN, - -128, 127, digital_gain), + -84, 40, digital_gain), SOC_SINGLE_S8_TLV("TX2 Digital Volume", LPASS_CDC_TX2_VOL_CTL_GAIN, - -128, 127, digital_gain), + -84, 40, digital_gain), SOC_ENUM("TX1 HPF Cutoff", tx1_hpf_cutoff_enum), SOC_ENUM("TX2 HPF Cutoff", tx2_hpf_cutoff_enum), SOC_SINGLE("TX1 HPF Switch", LPASS_CDC_TX1_MUX_CTL, 3, 1, 0), @@ -553,22 +553,22 @@ static const struct snd_kcontrol_new msm8916_wcd_digital_snd_controls[] = { WCD_IIR_FILTER_CTL("IIR2 Band3", IIR2, BAND3), WCD_IIR_FILTER_CTL("IIR2 Band4", IIR2, BAND4), WCD_IIR_FILTER_CTL("IIR2 Band5", IIR2, BAND5), - SOC_SINGLE_SX_TLV("IIR1 INP1 Volume", LPASS_CDC_IIR1_GAIN_B1_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("IIR1 INP2 Volume", LPASS_CDC_IIR1_GAIN_B2_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("IIR1 INP3 Volume", LPASS_CDC_IIR1_GAIN_B3_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("IIR1 INP4 Volume", LPASS_CDC_IIR1_GAIN_B4_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("IIR2 INP1 Volume", LPASS_CDC_IIR2_GAIN_B1_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("IIR2 INP2 Volume", LPASS_CDC_IIR2_GAIN_B2_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("IIR2 INP3 Volume", LPASS_CDC_IIR2_GAIN_B3_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("IIR2 INP4 Volume", LPASS_CDC_IIR2_GAIN_B4_CTL, - 0, -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("IIR1 INP1 Volume", LPASS_CDC_IIR1_GAIN_B1_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("IIR1 INP2 Volume", LPASS_CDC_IIR1_GAIN_B2_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("IIR1 INP3 Volume", LPASS_CDC_IIR1_GAIN_B3_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("IIR1 INP4 Volume", LPASS_CDC_IIR1_GAIN_B4_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("IIR2 INP1 Volume", LPASS_CDC_IIR2_GAIN_B1_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("IIR2 INP2 Volume", LPASS_CDC_IIR2_GAIN_B2_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("IIR2 INP3 Volume", LPASS_CDC_IIR2_GAIN_B3_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("IIR2 INP4 Volume", LPASS_CDC_IIR2_GAIN_B4_CTL, + -84, 40, digital_gain), }; diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 78db3bd0b3bc..1450a84df4e8 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -184,6 +184,7 @@ static int nau8822_eq_get(struct snd_kcontrol *kcontrol, struct soc_bytes_ext *params = (void *)kcontrol->private_value; int i, reg; u16 reg_val, *val; + __be16 tmp; val = (u16 *)ucontrol->value.bytes.data; reg = NAU8822_REG_EQ1; @@ -192,8 +193,8 @@ static int nau8822_eq_get(struct snd_kcontrol *kcontrol, /* conversion of 16-bit integers between native CPU format * and big endian format */ - reg_val = cpu_to_be16(reg_val); - memcpy(val + i, ®_val, sizeof(reg_val)); + tmp = cpu_to_be16(reg_val); + memcpy(val + i, &tmp, sizeof(tmp)); } return 0; @@ -216,6 +217,7 @@ static int nau8822_eq_put(struct snd_kcontrol *kcontrol, void *data; u16 *val, value; int i, reg, ret; + __be16 *tmp; data = kmemdup(ucontrol->value.bytes.data, params->max, GFP_KERNEL | GFP_DMA); @@ -228,7 +230,8 @@ static int nau8822_eq_put(struct snd_kcontrol *kcontrol, /* conversion of 16-bit integers between native CPU format * and big endian format */ - value = be16_to_cpu(*(val + i)); + tmp = (__be16 *)(val + i); + value = be16_to_cpup(tmp); ret = snd_soc_component_write(component, reg + i, value); if (ret) { dev_err(component->dev, @@ -741,6 +744,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->mclk_scaler, pll_param->pre_factor); snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_OFF); + snd_soc_component_update_bits(component, NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK, (pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) | pll_param->pll_int); @@ -757,6 +762,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT); snd_soc_component_update_bits(component, NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_ON); return 0; } diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h index 489191ff187e..b45d42c15de6 100644 --- a/sound/soc/codecs/nau8822.h +++ b/sound/soc/codecs/nau8822.h @@ -90,6 +90,9 @@ #define NAU8822_REFIMP_3K 0x3 #define NAU8822_IOBUF_EN (0x1 << 2) #define NAU8822_ABIAS_EN (0x1 << 3) +#define NAU8822_PLL_EN_MASK (0x1 << 5) +#define NAU8822_PLL_ON (0x1 << 5) +#define NAU8822_PLL_OFF (0x0 << 5) /* NAU8822_REG_AUDIO_INTERFACE (0x4) */ #define NAU8822_AIFMT_MASK (0x3 << 3) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index c8ccfa2fff84..9b22219a7693 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1072,6 +1072,7 @@ static int nau8824_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct nau8824 *nau8824 = snd_soc_component_get_drvdata(component); unsigned int val_len = 0, osr, ctrl_val, bclk_fs, bclk_div; + int err = -EINVAL; nau8824_sema_acquire(nau8824, HZ); @@ -1088,7 +1089,7 @@ static int nau8824_hw_params(struct snd_pcm_substream *substream, osr &= NAU8824_DAC_OVERSAMPLE_MASK; if (nau8824_clock_check(nau8824, substream->stream, nau8824->fs, osr)) - return -EINVAL; + goto error; regmap_update_bits(nau8824->regmap, NAU8824_REG_CLK_DIVIDER, NAU8824_CLK_DAC_SRC_MASK, osr_dac_sel[osr].clk_src << NAU8824_CLK_DAC_SRC_SFT); @@ -1098,7 +1099,7 @@ static int nau8824_hw_params(struct snd_pcm_substream *substream, osr &= NAU8824_ADC_SYNC_DOWN_MASK; if (nau8824_clock_check(nau8824, substream->stream, nau8824->fs, osr)) - return -EINVAL; + goto error; regmap_update_bits(nau8824->regmap, NAU8824_REG_CLK_DIVIDER, NAU8824_CLK_ADC_SRC_MASK, osr_adc_sel[osr].clk_src << NAU8824_CLK_ADC_SRC_SFT); @@ -1119,7 +1120,7 @@ static int nau8824_hw_params(struct snd_pcm_substream *substream, else if (bclk_fs <= 256) bclk_div = 0; else - return -EINVAL; + goto error; regmap_update_bits(nau8824->regmap, NAU8824_REG_PORT0_I2S_PCM_CTRL_2, NAU8824_I2S_LRC_DIV_MASK | NAU8824_I2S_BLK_DIV_MASK, @@ -1140,15 +1141,17 @@ static int nau8824_hw_params(struct snd_pcm_substream *substream, val_len |= NAU8824_I2S_DL_32; break; default: - return -EINVAL; + goto error; } regmap_update_bits(nau8824->regmap, NAU8824_REG_PORT0_I2S_PCM_CTRL_1, NAU8824_I2S_DL_MASK, val_len); + err = 0; + error: nau8824_sema_release(nau8824); - return 0; + return err; } static int nau8824_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) @@ -1157,8 +1160,6 @@ static int nau8824_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct nau8824 *nau8824 = snd_soc_component_get_drvdata(component); unsigned int ctrl1_val = 0, ctrl2_val = 0; - nau8824_sema_acquire(nau8824, HZ); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: ctrl2_val |= NAU8824_I2S_MS_MASTER; @@ -1200,6 +1201,8 @@ static int nau8824_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } + nau8824_sema_acquire(nau8824, HZ); + regmap_update_bits(nau8824->regmap, NAU8824_REG_PORT0_I2S_PCM_CTRL_1, NAU8824_I2S_DF_MASK | NAU8824_I2S_BP_MASK | NAU8824_I2S_PCMB_EN, ctrl1_val); @@ -1893,6 +1896,30 @@ static const struct dmi_system_id nau8824_quirk_table[] = { }, .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), }, + { + /* Positivo CW14Q01P */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "CW14Q01P"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, + { + /* Positivo K1424G */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "K1424G"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, + { + /* Positivo N14ZP74G */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "N14ZP74G"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, {} }; diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 4cbef9affffd..feb590a20544 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -1598,7 +1598,7 @@ int pcm512x_probe(struct device *dev, struct regmap *regmap) if (val > 6) { dev_err(dev, "Invalid pll-in\n"); ret = -EINVAL; - goto err_clk; + goto err_pm; } pcm512x->pll_in = val; } @@ -1607,7 +1607,7 @@ int pcm512x_probe(struct device *dev, struct regmap *regmap) if (val > 6) { dev_err(dev, "Invalid pll-out\n"); ret = -EINVAL; - goto err_clk; + goto err_pm; } pcm512x->pll_out = val; } @@ -1616,12 +1616,12 @@ int pcm512x_probe(struct device *dev, struct regmap *regmap) dev_err(dev, "Error: both pll-in and pll-out, or none\n"); ret = -EINVAL; - goto err_clk; + goto err_pm; } if (pcm512x->pll_in && pcm512x->pll_in == pcm512x->pll_out) { dev_err(dev, "Error: pll-in == pll-out\n"); ret = -EINVAL; - goto err_clk; + goto err_pm; } } #endif diff --git a/sound/soc/codecs/rk3328_codec.c b/sound/soc/codecs/rk3328_codec.c index 514ebe16bbfa..4e71ecf54af7 100644 --- a/sound/soc/codecs/rk3328_codec.c +++ b/sound/soc/codecs/rk3328_codec.c @@ -479,7 +479,7 @@ static int rk3328_platform_probe(struct platform_device *pdev) ret = clk_prepare_enable(rk3328->pclk); if (ret < 0) { dev_err(&pdev->dev, "failed to enable acodec pclk\n"); - return ret; + goto err_unprepare_mclk; } base = devm_platform_ioremap_resource(pdev, 0); diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index f8c0f977206c..cc7eb34a641d 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1166,6 +1166,13 @@ static const struct dmi_system_id force_combo_jack_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Geminilake") } }, + { + .ident = "Intel Kabylake R RVP", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "Kabylake Client platform") + } + }, { } }; diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 7081142a355e..c444a56df95b 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -419,7 +419,7 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, } } - return 0; + return 1; } static const struct snd_kcontrol_new rt5514_snd_controls[] = { diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index c83f7f5da96b..c78d5833c9cd 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -421,6 +421,7 @@ struct rt5645_priv { struct regulator_bulk_data supplies[ARRAY_SIZE(rt5645_supply_names)]; struct rt5645_eq_param_s *eq_param; struct timer_list btn_check_timer; + struct mutex jd_mutex; int codec_type; int sysclk; @@ -3179,6 +3180,8 @@ static int rt5645_jack_detect(struct snd_soc_component *component, int jack_inse rt5645_enable_push_button_irq(component, true); } } else { + if (rt5645->en_button_func) + rt5645_enable_push_button_irq(component, false); snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); snd_soc_dapm_sync(dapm); rt5645->jack_type = SND_JACK_HEADPHONE; @@ -3241,6 +3244,8 @@ int rt5645_set_jack_detect(struct snd_soc_component *component, RT5645_GP1_PIN_IRQ, RT5645_GP1_PIN_IRQ); regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1, RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL); + regmap_update_bits(rt5645->regmap, RT5645_DEPOP_M1, + RT5645_HP_CB_MASK, RT5645_HP_CB_PU); } rt5645_irq(0, rt5645); @@ -3257,6 +3262,8 @@ static void rt5645_jack_detect_work(struct work_struct *work) if (!rt5645->component) return; + mutex_lock(&rt5645->jd_mutex); + switch (rt5645->pdata.jd_mode) { case 0: /* Not using rt5645 JD */ if (rt5645->gpiod_hp_det) { @@ -3269,6 +3276,7 @@ static void rt5645_jack_detect_work(struct work_struct *work) report, SND_JACK_HEADPHONE); snd_soc_jack_report(rt5645->mic_jack, report, SND_JACK_MICROPHONE); + mutex_unlock(&rt5645->jd_mutex); return; case 4: val = snd_soc_component_read32(rt5645->component, RT5645_A_JD_CTRL1) & 0x0020; @@ -3281,7 +3289,7 @@ static void rt5645_jack_detect_work(struct work_struct *work) if (!val && (rt5645->jack_type == 0)) { /* jack in */ report = rt5645_jack_detect(rt5645->component, 1); - } else if (!val && rt5645->jack_type != 0) { + } else if (!val && rt5645->jack_type == SND_JACK_HEADSET) { /* for push button and jack out */ btn_type = 0; if (snd_soc_component_read32(rt5645->component, RT5645_INT_IRQ_ST) & 0x4) { @@ -3337,6 +3345,8 @@ static void rt5645_jack_detect_work(struct work_struct *work) rt5645_jack_detect(rt5645->component, 0); } + mutex_unlock(&rt5645->jd_mutex); + snd_soc_jack_report(rt5645->hp_jack, report, SND_JACK_HEADPHONE); snd_soc_jack_report(rt5645->mic_jack, report, SND_JACK_MICROPHONE); if (rt5645->en_button_func) @@ -3756,6 +3766,16 @@ static const struct dmi_system_id dmi_platform_data[] = { DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), DMI_EXACT_MATCH(DMI_BOARD_NAME, "Cherry Trail CR"), DMI_EXACT_MATCH(DMI_BOARD_VERSION, "Default string"), + /* + * Above strings are too generic, LattePanda BIOS versions for + * all 4 hw revisions are: + * DF-BI-7-S70CR100-* + * DF-BI-7-S70CR110-* + * DF-BI-7-S70CR200-* + * LP-BS-7-S70CR700-* + * Do a partial match for S70CR to avoid false positive matches. + */ + DMI_MATCH(DMI_BIOS_VERSION, "S70CR"), }, .driver_data = (void *)&lattepanda_board_platform_data, }, @@ -4039,6 +4059,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, } timer_setup(&rt5645->btn_check_timer, rt5645_btn_check_callback, 0); + mutex_init(&rt5645->jd_mutex); INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work); INIT_DELAYED_WORK(&rt5645->rcclock_work, rt5645_rcclock_work); @@ -4074,9 +4095,14 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) if (i2c->irq) free_irq(i2c->irq, rt5645); + /* + * Since the rt5645_btn_check_callback() can queue jack_detect_work, + * the timer need to be delted first + */ + del_timer_sync(&rt5645->btn_check_timer); + cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); - del_timer_sync(&rt5645->btn_check_timer); regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 68299ce26d3e..648e0708007e 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4472,6 +4472,8 @@ static void rt5665_remove(struct snd_soc_component *component) struct rt5665_priv *rt5665 = snd_soc_component_get_drvdata(component); regmap_write(rt5665->regmap, RT5665_RESET, 0); + + regulator_bulk_disable(ARRAY_SIZE(rt5665->supplies), rt5665->supplies); } #ifdef CONFIG_PM diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index f21181734170..fefdd8cbd8f5 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -3185,8 +3185,6 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, if (ret < 0) goto err; - pm_runtime_put(&i2c->dev); - return 0; err: pm_runtime_disable(&i2c->dev); diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 8a1e485982d8..d2dfc53e30ff 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1788,6 +1788,10 @@ static int sgtl5000_i2c_remove(struct i2c_client *client) { struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client); + regmap_write(sgtl5000->regmap, SGTL5000_CHIP_CLK_CTRL, SGTL5000_CHIP_CLK_CTRL_DEFAULT); + regmap_write(sgtl5000->regmap, SGTL5000_CHIP_DIG_POWER, SGTL5000_DIG_POWER_DEFAULT); + regmap_write(sgtl5000->regmap, SGTL5000_CHIP_ANA_POWER, SGTL5000_ANA_POWER_DEFAULT); + clk_disable_unprepare(sgtl5000->mclk); regulator_bulk_disable(sgtl5000->num_supplies, sgtl5000->supplies); regulator_bulk_free(sgtl5000->num_supplies, sgtl5000->supplies); @@ -1795,6 +1799,11 @@ static int sgtl5000_i2c_remove(struct i2c_client *client) return 0; } +static void sgtl5000_i2c_shutdown(struct i2c_client *client) +{ + sgtl5000_i2c_remove(client); +} + static const struct i2c_device_id sgtl5000_id[] = { {"sgtl5000", 0}, {}, @@ -1815,6 +1824,7 @@ static struct i2c_driver sgtl5000_i2c_driver = { }, .probe = sgtl5000_i2c_probe, .remove = sgtl5000_i2c_remove, + .shutdown = sgtl5000_i2c_shutdown, .id_table = sgtl5000_id, }; diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index 56ec5863f250..3a808c762299 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -80,6 +80,7 @@ /* * SGTL5000_CHIP_DIG_POWER */ +#define SGTL5000_DIG_POWER_DEFAULT 0x0000 #define SGTL5000_ADC_EN 0x0040 #define SGTL5000_DAC_EN 0x0020 #define SGTL5000_DAP_POWERUP 0x0010 diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 464a4d7873bb..b797f620e352 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -53,6 +53,18 @@ static const struct reg_default ssm2602_reg[SSM2602_CACHEREGNUM] = { { .reg = 0x09, .def = 0x0000 } }; +/* + * ssm2602 register patch + * Workaround for playback distortions after power up: activates digital + * core, and then powers on output, DAC, and whole chip at the same time + */ + +static const struct reg_sequence ssm2602_patch[] = { + { SSM2602_ACTIVE, 0x01 }, + { SSM2602_PWR, 0x07 }, + { SSM2602_RESET, 0x00 }, +}; + /*Appending several "None"s just for OSS mixer use*/ static const char *ssm2602_input_select[] = { @@ -588,6 +600,9 @@ static int ssm260x_component_probe(struct snd_soc_component *component) return ret; } + regmap_register_patch(ssm2602->regmap, ssm2602_patch, + ARRAY_SIZE(ssm2602_patch)); + /* set the update bits */ regmap_update_bits(ssm2602->regmap, SSM2602_LINVOL, LINVOL_LRIN_BOTH, LINVOL_LRIN_BOTH); diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c index c3587af9985c..3d981441b8d1 100644 --- a/sound/soc/codecs/tscs454.c +++ b/sound/soc/codecs/tscs454.c @@ -3128,18 +3128,17 @@ static int set_aif_sample_format(struct snd_soc_component *component, unsigned int width; int ret; - switch (format) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (snd_pcm_format_width(format)) { + case 16: width = FV_WL_16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: width = FV_WL_20; break; - case SNDRV_PCM_FORMAT_S24_3LE: + case 24: width = FV_WL_24; break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S32_LE: + case 32: width = FV_WL_32; break; default: @@ -3337,6 +3336,7 @@ static const struct snd_soc_component_driver soc_component_dev_tscs454 = { .num_dapm_routes = ARRAY_SIZE(tscs454_intercon), .controls = tscs454_snd_controls, .num_controls = ARRAY_SIZE(tscs454_snd_controls), + .endianness = 1, }; #define TSCS454_RATES SNDRV_PCM_RATE_8000_96000 diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 016aff97e2fb..a952b9454513 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -1971,8 +1971,8 @@ static int wcd9335_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - slim_stream_unprepare(dai_data->sruntime); slim_stream_disable(dai_data->sruntime); + slim_stream_unprepare(dai_data->sruntime); break; default: break; @@ -2252,51 +2252,42 @@ static int wcd9335_rx_hph_mode_put(struct snd_kcontrol *kc, static const struct snd_kcontrol_new wcd9335_snd_controls[] = { /* -84dB min - 40dB max */ - SOC_SINGLE_SX_TLV("RX0 Digital Volume", WCD9335_CDC_RX0_RX_VOL_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX1 Digital Volume", WCD9335_CDC_RX1_RX_VOL_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX2 Digital Volume", WCD9335_CDC_RX2_RX_VOL_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX3 Digital Volume", WCD9335_CDC_RX3_RX_VOL_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX4 Digital Volume", WCD9335_CDC_RX4_RX_VOL_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX5 Digital Volume", WCD9335_CDC_RX5_RX_VOL_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX6 Digital Volume", WCD9335_CDC_RX6_RX_VOL_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX7 Digital Volume", WCD9335_CDC_RX7_RX_VOL_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX8 Digital Volume", WCD9335_CDC_RX8_RX_VOL_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX0 Mix Digital Volume", - WCD9335_CDC_RX0_RX_VOL_MIX_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX1 Mix Digital Volume", - WCD9335_CDC_RX1_RX_VOL_MIX_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX2 Mix Digital Volume", - WCD9335_CDC_RX2_RX_VOL_MIX_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX3 Mix Digital Volume", - WCD9335_CDC_RX3_RX_VOL_MIX_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX4 Mix Digital Volume", - WCD9335_CDC_RX4_RX_VOL_MIX_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX5 Mix Digital Volume", - WCD9335_CDC_RX5_RX_VOL_MIX_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX6 Mix Digital Volume", - WCD9335_CDC_RX6_RX_VOL_MIX_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX7 Mix Digital Volume", - WCD9335_CDC_RX7_RX_VOL_MIX_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX8 Mix Digital Volume", - WCD9335_CDC_RX8_RX_VOL_MIX_CTL, - 0, -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX0 Digital Volume", WCD9335_CDC_RX0_RX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX1 Digital Volume", WCD9335_CDC_RX1_RX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX2 Digital Volume", WCD9335_CDC_RX2_RX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX3 Digital Volume", WCD9335_CDC_RX3_RX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX4 Digital Volume", WCD9335_CDC_RX4_RX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX5 Digital Volume", WCD9335_CDC_RX5_RX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX6 Digital Volume", WCD9335_CDC_RX6_RX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX7 Digital Volume", WCD9335_CDC_RX7_RX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX8 Digital Volume", WCD9335_CDC_RX8_RX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX0 Mix Digital Volume", WCD9335_CDC_RX0_RX_VOL_MIX_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX1 Mix Digital Volume", WCD9335_CDC_RX1_RX_VOL_MIX_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX2 Mix Digital Volume", WCD9335_CDC_RX2_RX_VOL_MIX_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX3 Mix Digital Volume", WCD9335_CDC_RX3_RX_VOL_MIX_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX4 Mix Digital Volume", WCD9335_CDC_RX4_RX_VOL_MIX_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX5 Mix Digital Volume", WCD9335_CDC_RX5_RX_VOL_MIX_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX6 Mix Digital Volume", WCD9335_CDC_RX6_RX_VOL_MIX_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX7 Mix Digital Volume", WCD9335_CDC_RX7_RX_VOL_MIX_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX8 Mix Digital Volume", WCD9335_CDC_RX8_RX_VOL_MIX_CTL, + -84, 40, digital_gain), SOC_ENUM("RX INT0_1 HPF cut off", cf_int0_1_enum), SOC_ENUM("RX INT0_2 HPF cut off", cf_int0_2_enum), SOC_ENUM("RX INT1_1 HPF cut off", cf_int1_1_enum), diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 72e165cc6443..97ece3114b3d 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -536,7 +536,7 @@ static int wm2000_anc_transition(struct wm2000_priv *wm2000, { struct i2c_client *i2c = wm2000->i2c; int i, j; - int ret; + int ret = 0; if (wm2000->anc_mode == mode) return 0; @@ -566,13 +566,13 @@ static int wm2000_anc_transition(struct wm2000_priv *wm2000, ret = anc_transitions[i].step[j](i2c, anc_transitions[i].analogue); if (ret != 0) - return ret; + break; } if (anc_transitions[i].dest == ANC_OFF) clk_disable_unprepare(wm2000->mclk); - return 0; + return ret; } static int wm2000_anc_set_mode(struct wm2000_priv *wm2000) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 9dc215b5c504..06ec3f48c808 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -413,6 +413,7 @@ static int wm5110_put_dre(struct snd_kcontrol *kcontrol, unsigned int rnew = (!!ucontrol->value.integer.value[1]) << mc->rshift; unsigned int lold, rold; unsigned int lena, rena; + bool change = false; int ret; snd_soc_dapm_mutex_lock(dapm); @@ -440,8 +441,8 @@ static int wm5110_put_dre(struct snd_kcontrol *kcontrol, goto err; } - ret = regmap_update_bits(arizona->regmap, ARIZONA_DRE_ENABLE, - mask, lnew | rnew); + ret = regmap_update_bits_check(arizona->regmap, ARIZONA_DRE_ENABLE, + mask, lnew | rnew, &change); if (ret) { dev_err(arizona->dev, "Failed to set DRE: %d\n", ret); goto err; @@ -454,6 +455,9 @@ static int wm5110_put_dre(struct snd_kcontrol *kcontrol, if (!rnew && rold) wm5110_clear_pga_volume(arizona, mc->rshift); + if (change) + ret = 1; + err: snd_soc_dapm_mutex_unlock(dapm); diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 9e8c564f6e9c..9787257b69a9 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2276,6 +2276,9 @@ static int wm8904_i2c_probe(struct i2c_client *i2c, regmap_update_bits(wm8904->regmap, WM8904_BIAS_CONTROL_0, WM8904_POBCTRL, 0); + /* Fill the cache for the ADC test register */ + regmap_read(wm8904->regmap, WM8904_ADC_TEST_0, &val); + /* Can leave the device powered off until we need it */ regcache_cache_only(wm8904->regmap, true); regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 04f23477039a..c677c068b05e 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -534,7 +534,7 @@ static int wm8958_mbc_put(struct snd_kcontrol *kcontrol, wm8958_dsp_apply(component, mbc, wm8994->mbc_ena[mbc]); - return 0; + return 1; } #define WM8958_MBC_SWITCH(xname, xval) {\ @@ -660,7 +660,7 @@ static int wm8958_vss_put(struct snd_kcontrol *kcontrol, wm8958_dsp_apply(component, vss, wm8994->vss_ena[vss]); - return 0; + return 1; } @@ -734,7 +734,7 @@ static int wm8958_hpf_put(struct snd_kcontrol *kcontrol, wm8958_dsp_apply(component, hpf % 3, ucontrol->value.integer.value[0]); - return 0; + return 1; } #define WM8958_HPF_SWITCH(xname, xval) {\ @@ -828,7 +828,7 @@ static int wm8958_enh_eq_put(struct snd_kcontrol *kcontrol, wm8958_dsp_apply(component, eq, ucontrol->value.integer.value[0]); - return 0; + return 1; } #define WM8958_ENH_EQ_SWITCH(xname, xval) {\ diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index d9d59f45833f..15828ae62f9d 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1840,6 +1840,49 @@ SOC_SINGLE_TLV("SPKOUTR Mixer DACR Volume", WM8962_SPEAKER_MIXER_5, 4, 1, 0, inmix_tlv), }; +static int tp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + int ret, reg, val, mask; + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + + ret = pm_runtime_resume_and_get(component->dev); + if (ret < 0) { + dev_err(component->dev, "Failed to resume device: %d\n", ret); + return ret; + } + + reg = WM8962_ADDITIONAL_CONTROL_4; + + if (!strcmp(w->name, "TEMP_HP")) { + mask = WM8962_TEMP_ENA_HP_MASK; + val = WM8962_TEMP_ENA_HP; + } else if (!strcmp(w->name, "TEMP_SPK")) { + mask = WM8962_TEMP_ENA_SPK_MASK; + val = WM8962_TEMP_ENA_SPK; + } else { + pm_runtime_put(component->dev); + return -EINVAL; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMD: + val = 0; + fallthrough; + case SND_SOC_DAPM_POST_PMU: + ret = snd_soc_component_update_bits(component, reg, mask, val); + break; + default: + WARN(1, "Invalid event %d\n", event); + pm_runtime_put(component->dev); + return -EINVAL; + } + + pm_runtime_put(component->dev); + + return 0; +} + static int cp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -2132,8 +2175,10 @@ SND_SOC_DAPM_SUPPLY("TOCLK", WM8962_ADDITIONAL_CONTROL_1, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("DSP2", 1, WM8962_DSP2_POWER_MANAGEMENT, WM8962_DSP2_ENA_SHIFT, 0, dsp2_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), -SND_SOC_DAPM_SUPPLY("TEMP_HP", WM8962_ADDITIONAL_CONTROL_4, 2, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("TEMP_SPK", WM8962_ADDITIONAL_CONTROL_4, 1, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("TEMP_HP", SND_SOC_NOPM, 0, 0, tp_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("TEMP_SPK", SND_SOC_NOPM, 0, 0, tp_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MIXER("INPGAL", WM8962_LEFT_INPUT_PGA_CONTROL, 4, 0, inpgal, ARRAY_SIZE(inpgal)), @@ -2173,6 +2218,9 @@ SND_SOC_DAPM_PGA_E("HPOUT", SND_SOC_NOPM, 0, 0, NULL, 0, hp_event, SND_SOC_DAPM_OUTPUT("HPOUTL"), SND_SOC_DAPM_OUTPUT("HPOUTR"), + +SND_SOC_DAPM_PGA("SPKOUTL Output", WM8962_CLASS_D_CONTROL_1, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("SPKOUTR Output", WM8962_CLASS_D_CONTROL_1, 7, 0, NULL, 0), }; static const struct snd_soc_dapm_widget wm8962_dapm_spk_mono_widgets[] = { @@ -2180,7 +2228,6 @@ SND_SOC_DAPM_MIXER("Speaker Mixer", WM8962_MIXER_ENABLES, 1, 0, spkmixl, ARRAY_SIZE(spkmixl)), SND_SOC_DAPM_MUX_E("Speaker PGA", WM8962_PWR_MGMT_2, 4, 0, &spkoutl_mux, out_pga_event, SND_SOC_DAPM_POST_PMU), -SND_SOC_DAPM_PGA("Speaker Output", WM8962_CLASS_D_CONTROL_1, 7, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("SPKOUT"), }; @@ -2195,9 +2242,6 @@ SND_SOC_DAPM_MUX_E("SPKOUTL PGA", WM8962_PWR_MGMT_2, 4, 0, &spkoutl_mux, SND_SOC_DAPM_MUX_E("SPKOUTR PGA", WM8962_PWR_MGMT_2, 3, 0, &spkoutr_mux, out_pga_event, SND_SOC_DAPM_POST_PMU), -SND_SOC_DAPM_PGA("SPKOUTR Output", WM8962_CLASS_D_CONTROL_1, 7, 0, NULL, 0), -SND_SOC_DAPM_PGA("SPKOUTL Output", WM8962_CLASS_D_CONTROL_1, 6, 0, NULL, 0), - SND_SOC_DAPM_OUTPUT("SPKOUTL"), SND_SOC_DAPM_OUTPUT("SPKOUTR"), }; @@ -2307,12 +2351,18 @@ static const struct snd_soc_dapm_route wm8962_spk_mono_intercon[] = { { "Speaker PGA", "Mixer", "Speaker Mixer" }, { "Speaker PGA", "DAC", "DACL" }, - { "Speaker Output", NULL, "Speaker PGA" }, - { "Speaker Output", NULL, "SYSCLK" }, - { "Speaker Output", NULL, "TOCLK" }, - { "Speaker Output", NULL, "TEMP_SPK" }, + { "SPKOUTL Output", NULL, "Speaker PGA" }, + { "SPKOUTL Output", NULL, "SYSCLK" }, + { "SPKOUTL Output", NULL, "TOCLK" }, + { "SPKOUTL Output", NULL, "TEMP_SPK" }, - { "SPKOUT", NULL, "Speaker Output" }, + { "SPKOUTR Output", NULL, "Speaker PGA" }, + { "SPKOUTR Output", NULL, "SYSCLK" }, + { "SPKOUTR Output", NULL, "TOCLK" }, + { "SPKOUTR Output", NULL, "TEMP_SPK" }, + + { "SPKOUT", NULL, "SPKOUTL Output" }, + { "SPKOUT", NULL, "SPKOUTR Output" }, }; static const struct snd_soc_dapm_route wm8962_spk_stereo_intercon[] = { @@ -2844,8 +2894,12 @@ static int wm8962_set_fll(struct snd_soc_component *component, int fll_id, int s switch (fll_id) { case WM8962_FLL_MCLK: case WM8962_FLL_BCLK: + fll1 |= (fll_id - 1) << WM8962_FLL_REFCLK_SRC_SHIFT; + break; case WM8962_FLL_OSC: fll1 |= (fll_id - 1) << WM8962_FLL_REFCLK_SRC_SHIFT; + snd_soc_component_update_bits(component, WM8962_PLL2, + WM8962_OSC_ENA, WM8962_OSC_ENA); break; case WM8962_FLL_INT: snd_soc_component_update_bits(component, WM8962_FLL_CONTROL_1, @@ -2854,7 +2908,7 @@ static int wm8962_set_fll(struct snd_soc_component *component, int fll_id, int s WM8962_FLL_FRC_NCO, WM8962_FLL_FRC_NCO); break; default: - dev_err(component->dev, "Unknown FLL source %d\n", ret); + dev_err(component->dev, "Unknown FLL source %d\n", source); return -EINVAL; } @@ -3750,6 +3804,11 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, if (ret < 0) goto err_pm_runtime; + regmap_update_bits(wm8962->regmap, WM8962_ADDITIONAL_CONTROL_4, + WM8962_TEMP_ENA_HP_MASK, 0); + regmap_update_bits(wm8962->regmap, WM8962_ADDITIONAL_CONTROL_4, + WM8962_TEMP_ENA_SPK_MASK, 0); + regcache_cache_only(wm8962->regmap, true); /* The drivers should power up as needed */ @@ -3854,6 +3913,7 @@ static int wm8962_runtime_suspend(struct device *dev) #endif static const struct dev_pm_ops wm8962_pm = { + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) }; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 6dbab3fc6537..4ae7ec8d71cd 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3711,7 +3711,12 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) } else { dev_dbg(component->dev, "Jack not detected\n"); + /* Release wm8994->accdet_lock to avoid deadlock: + * cancel_delayed_work_sync() takes wm8994->mic_work internal + * lock and wm1811_mic_work takes wm8994->accdet_lock */ + mutex_unlock(&wm8994->accdet_lock); cancel_delayed_work_sync(&wm8994->mic_work); + mutex_lock(&wm8994->accdet_lock); snd_soc_component_update_bits(component, WM8958_MICBIAS2, WM8958_MICB2_DISCH, WM8958_MICB2_DISCH); diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 13672928da99..8df5f3bc6e97 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -791,7 +791,7 @@ int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); - int ret = 0; + int ret = 1; if (ucontrol->value.enumerated.item[0] == dsp[e->shift_l].fw) return 0; @@ -3649,12 +3649,12 @@ static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf) ret = wm_adsp_buffer_read(buf, caps->region_defs[i].base_offset, ®ion->base_addr); if (ret < 0) - return ret; + goto err; ret = wm_adsp_buffer_read(buf, caps->region_defs[i].size_offset, &offset); if (ret < 0) - return ret; + goto err; region->cumulative_size = offset; @@ -3665,6 +3665,10 @@ static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf) } return 0; + +err: + kfree(buf->regions); + return ret; } static void wm_adsp_buffer_clear(struct wm_adsp_compr_buf *buf) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 65112b9d8588..90b8814d7506 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -132,13 +132,13 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id) /* Error Handling: TX */ if (isr[i] & ISR_TXFO) { - dev_err(dev->dev, "TX overrun (ch_id=%d)\n", i); + dev_err_ratelimited(dev->dev, "TX overrun (ch_id=%d)\n", i); irq_valid = true; } /* Error Handling: TX */ if (isr[i] & ISR_RXFO) { - dev_err(dev->dev, "RX overrun (ch_id=%d)\n", i); + dev_err_ratelimited(dev->dev, "RX overrun (ch_id=%d)\n", i); irq_valid = true; } } diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 6f3b768489f6..bf3d3f0aa858 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -86,7 +86,7 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) int ret; int int_port = 0, ext_port; struct device_node *np = pdev->dev.of_node; - struct device_node *ssi_np = NULL, *codec_np = NULL; + struct device_node *ssi_np = NULL, *codec_np = NULL, *tmp_np = NULL; eukrea_tlv320.dev = &pdev->dev; if (np) { @@ -143,7 +143,7 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) } if (machine_is_eukrea_cpuimx27() || - of_find_compatible_node(NULL, NULL, "fsl,imx21-audmux")) { + (tmp_np = of_find_compatible_node(NULL, NULL, "fsl,imx21-audmux"))) { imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, IMX_AUDMUX_V1_PCR_SYN | IMX_AUDMUX_V1_PCR_TFSDIR | @@ -158,10 +158,11 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) IMX_AUDMUX_V1_PCR_SYN | IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0) ); + of_node_put(tmp_np); } else if (machine_is_eukrea_cpuimx25sd() || machine_is_eukrea_cpuimx35sd() || machine_is_eukrea_cpuimx51sd() || - of_find_compatible_node(NULL, NULL, "fsl,imx31-audmux")) { + (tmp_np = of_find_compatible_node(NULL, NULL, "fsl,imx31-audmux"))) { if (!np) ext_port = machine_is_eukrea_cpuimx25sd() ? 4 : 3; @@ -178,6 +179,7 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) IMX_AUDMUX_V2_PTCR_SYN, IMX_AUDMUX_V2_PDCR_RXDSEL(int_port) ); + of_node_put(tmp_np); } else { if (np) { /* The eukrea,asoc-tlv320 driver was explicitly diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 39ea9bda1394..db663e7d17a4 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -112,11 +112,11 @@ static const struct snd_soc_dapm_route audio_map[] = { static const struct snd_soc_dapm_route audio_map_ac97[] = { /* 1st half -- Normal DAPM routes */ - {"Playback", NULL, "AC97 Playback"}, - {"AC97 Capture", NULL, "Capture"}, + {"AC97 Playback", NULL, "CPU AC97 Playback"}, + {"CPU AC97 Capture", NULL, "AC97 Capture"}, /* 2nd half -- ASRC DAPM routes */ - {"AC97 Playback", NULL, "ASRC-Playback"}, - {"ASRC-Capture", NULL, "AC97 Capture"}, + {"CPU AC97 Playback", NULL, "ASRC-Playback"}, + {"ASRC-Capture", NULL, "CPU AC97 Capture"}, }; /* Add all possible widgets into here without being redundant */ diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index f7f2d29f1bfe..6285ee8f829e 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -87,21 +87,21 @@ static DECLARE_TLV_DB_SCALE(gain_tlv, 0, 100, 0); static const struct snd_kcontrol_new fsl_micfil_snd_controls[] = { SOC_SINGLE_SX_TLV("CH0 Volume", REG_MICFIL_OUT_CTRL, - MICFIL_OUTGAIN_CHX_SHIFT(0), 0xF, 0x7, gain_tlv), + MICFIL_OUTGAIN_CHX_SHIFT(0), 0x8, 0xF, gain_tlv), SOC_SINGLE_SX_TLV("CH1 Volume", REG_MICFIL_OUT_CTRL, - MICFIL_OUTGAIN_CHX_SHIFT(1), 0xF, 0x7, gain_tlv), + MICFIL_OUTGAIN_CHX_SHIFT(1), 0x8, 0xF, gain_tlv), SOC_SINGLE_SX_TLV("CH2 Volume", REG_MICFIL_OUT_CTRL, - MICFIL_OUTGAIN_CHX_SHIFT(2), 0xF, 0x7, gain_tlv), + MICFIL_OUTGAIN_CHX_SHIFT(2), 0x8, 0xF, gain_tlv), SOC_SINGLE_SX_TLV("CH3 Volume", REG_MICFIL_OUT_CTRL, - MICFIL_OUTGAIN_CHX_SHIFT(3), 0xF, 0x7, gain_tlv), + MICFIL_OUTGAIN_CHX_SHIFT(3), 0x8, 0xF, gain_tlv), SOC_SINGLE_SX_TLV("CH4 Volume", REG_MICFIL_OUT_CTRL, - MICFIL_OUTGAIN_CHX_SHIFT(4), 0xF, 0x7, gain_tlv), + MICFIL_OUTGAIN_CHX_SHIFT(4), 0x8, 0xF, gain_tlv), SOC_SINGLE_SX_TLV("CH5 Volume", REG_MICFIL_OUT_CTRL, - MICFIL_OUTGAIN_CHX_SHIFT(5), 0xF, 0x7, gain_tlv), + MICFIL_OUTGAIN_CHX_SHIFT(5), 0x8, 0xF, gain_tlv), SOC_SINGLE_SX_TLV("CH6 Volume", REG_MICFIL_OUT_CTRL, - MICFIL_OUTGAIN_CHX_SHIFT(6), 0xF, 0x7, gain_tlv), + MICFIL_OUTGAIN_CHX_SHIFT(6), 0x8, 0xF, gain_tlv), SOC_SINGLE_SX_TLV("CH7 Volume", REG_MICFIL_OUT_CTRL, - MICFIL_OUTGAIN_CHX_SHIFT(7), 0xF, 0x7, gain_tlv), + MICFIL_OUTGAIN_CHX_SHIFT(7), 0x8, 0xF, gain_tlv), SOC_ENUM_EXT("MICFIL Quality Select", fsl_micfil_quality_enum, snd_soc_get_enum_double, snd_soc_put_enum_double), @@ -740,18 +740,23 @@ static int fsl_micfil_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); + /* + * Register platform component before registering cpu dai for there + * is not defer probe for platform component in snd_soc_add_pcm_runtime(). + */ + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) { + dev_err(&pdev->dev, "failed to pcm register\n"); + return ret; + } + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_micfil_component, &fsl_micfil_dai, 1); if (ret) { dev_err(&pdev->dev, "failed to register component %s\n", fsl_micfil_component.name); - return ret; } - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); - if (ret) - dev_err(&pdev->dev, "failed to pcm register\n"); - return ret; } diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 027259695551..fdbfaedda4ce 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -37,6 +37,24 @@ static const struct snd_pcm_hw_constraint_list fsl_sai_rate_constraints = { .list = fsl_sai_rates, }; +/** + * fsl_sai_dir_is_synced - Check if stream is synced by the opposite stream + * + * SAI supports synchronous mode using bit/frame clocks of either Transmitter's + * or Receiver's for both streams. This function is used to check if clocks of + * the stream's are synced by the opposite stream. + * + * @sai: SAI context + * @dir: stream direction + */ +static inline bool fsl_sai_dir_is_synced(struct fsl_sai *sai, int dir) +{ + int adir = (dir == TX) ? RX : TX; + + /* current dir in async mode while opposite dir in sync mode */ + return !sai->synchronous[dir] && sai->synchronous[adir]; +} + static irqreturn_t fsl_sai_isr(int irq, void *devid) { struct fsl_sai *sai = (struct fsl_sai *)devid; @@ -212,6 +230,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, if (!sai->is_lsb_first) val_cr4 |= FSL_SAI_CR4_MF; + sai->is_dsp_mode = false; /* DAI mode */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: @@ -522,6 +541,38 @@ static int fsl_sai_hw_free(struct snd_pcm_substream *substream, return 0; } +static void fsl_sai_config_disable(struct fsl_sai *sai, int dir) +{ + unsigned int ofs = sai->soc_data->reg_offset; + bool tx = dir == TX; + u32 xcsr, count = 100; + + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx, ofs), + FSL_SAI_CSR_TERE | FSL_SAI_CSR_BCE, 0); + + /* TERE will remain set till the end of current frame */ + do { + udelay(10); + regmap_read(sai->regmap, FSL_SAI_xCSR(tx, ofs), &xcsr); + } while (--count && xcsr & FSL_SAI_CSR_TERE); + + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx, ofs), + FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); + + /* + * For sai master mode, after several open/close sai, + * there will be no frame clock, and can't recover + * anymore. Add software reset to fix this issue. + * This is a hardware bug, and will be fix in the + * next sai version. + */ + if (!sai->is_slave_mode) { + /* Software Reset */ + regmap_write(sai->regmap, FSL_SAI_xCSR(tx, ofs), FSL_SAI_CSR_SR); + /* Clear SR bit to finish the reset */ + regmap_write(sai->regmap, FSL_SAI_xCSR(tx, ofs), 0); + } +} static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) @@ -530,7 +581,9 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, unsigned int ofs = sai->soc_data->reg_offset; bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - u32 xcsr, count = 100; + int adir = tx ? RX : TX; + int dir = tx ? TX : RX; + u32 xcsr; /* * Asynchronous mode: Clear SYNC for both Tx and Rx. @@ -553,10 +606,22 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx, ofs), FSL_SAI_CSR_FRDE, FSL_SAI_CSR_FRDE); - regmap_update_bits(sai->regmap, FSL_SAI_RCSR(ofs), - FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); - regmap_update_bits(sai->regmap, FSL_SAI_TCSR(ofs), + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx, ofs), FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); + /* + * Enable the opposite direction for synchronous mode + * 1. Tx sync with Rx: only set RE for Rx; set TE & RE for Tx + * 2. Rx sync with Tx: only set TE for Tx; set RE & TE for Rx + * + * RM recommends to enable RE after TE for case 1 and to enable + * TE after RE for case 2, but we here may not always guarantee + * that happens: "arecord 1.wav; aplay 2.wav" in case 1 enables + * TE after RE, which is against what RM recommends but should + * be safe to do, judging by years of testing results. + */ + if (fsl_sai_dir_is_synced(sai, adir)) + regmap_update_bits(sai->regmap, FSL_SAI_xCSR((!tx), ofs), + FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx, ofs), FSL_SAI_CSR_xIE_MASK, FSL_SAI_FLAGS); @@ -571,43 +636,23 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, /* Check if the opposite FRDE is also disabled */ regmap_read(sai->regmap, FSL_SAI_xCSR(!tx, ofs), &xcsr); - if (!(xcsr & FSL_SAI_CSR_FRDE)) { - /* Disable both directions and reset their FIFOs */ - regmap_update_bits(sai->regmap, FSL_SAI_TCSR(ofs), - FSL_SAI_CSR_TERE, 0); - regmap_update_bits(sai->regmap, FSL_SAI_RCSR(ofs), - FSL_SAI_CSR_TERE, 0); - - /* TERE will remain set till the end of current frame */ - do { - udelay(10); - regmap_read(sai->regmap, - FSL_SAI_xCSR(tx, ofs), &xcsr); - } while (--count && xcsr & FSL_SAI_CSR_TERE); - - regmap_update_bits(sai->regmap, FSL_SAI_TCSR(ofs), - FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); - regmap_update_bits(sai->regmap, FSL_SAI_RCSR(ofs), - FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); - - /* - * For sai master mode, after several open/close sai, - * there will be no frame clock, and can't recover - * anymore. Add software reset to fix this issue. - * This is a hardware bug, and will be fix in the - * next sai version. - */ - if (!sai->is_slave_mode) { - /* Software Reset for both Tx and Rx */ - regmap_write(sai->regmap, FSL_SAI_TCSR(ofs), - FSL_SAI_CSR_SR); - regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), - FSL_SAI_CSR_SR); - /* Clear SR bit to finish the reset */ - regmap_write(sai->regmap, FSL_SAI_TCSR(ofs), 0); - regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), 0); - } - } + + /* + * If opposite stream provides clocks for synchronous mode and + * it is inactive, disable it before disabling the current one + */ + if (fsl_sai_dir_is_synced(sai, adir) && !(xcsr & FSL_SAI_CSR_FRDE)) + fsl_sai_config_disable(sai, adir); + + /* + * Disable current stream if either of: + * 1. current stream doesn't provide clocks for synchronous mode + * 2. current stream provides clocks for synchronous mode but no + * more stream is active. + */ + if (!fsl_sai_dir_is_synced(sai, dir) || !(xcsr & FSL_SAI_CSR_FRDE)) + fsl_sai_config_disable(sai, dir); + break; default: return -EINVAL; @@ -765,6 +810,8 @@ static struct reg_default fsl_sai_reg_defaults_ofs8[] = { {FSL_SAI_RCR4(8), 0}, {FSL_SAI_RCR5(8), 0}, {FSL_SAI_RMR, 0}, + {FSL_SAI_MCTL, 0}, + {FSL_SAI_MDIV, 0}, }; static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg) @@ -805,6 +852,18 @@ static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg) case FSL_SAI_RFR6: case FSL_SAI_RFR7: case FSL_SAI_RMR: + case FSL_SAI_MCTL: + case FSL_SAI_MDIV: + case FSL_SAI_VERID: + case FSL_SAI_PARAM: + case FSL_SAI_TTCTN: + case FSL_SAI_RTCTN: + case FSL_SAI_TTCTL: + case FSL_SAI_TBCTN: + case FSL_SAI_TTCAP: + case FSL_SAI_RTCTL: + case FSL_SAI_RBCTN: + case FSL_SAI_RTCAP: return true; default: return false; @@ -819,6 +878,10 @@ static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg) if (reg == FSL_SAI_TCSR(ofs) || reg == FSL_SAI_RCSR(ofs)) return true; + /* Set VERID and PARAM be volatile for reading value in probe */ + if (ofs == 8 && (reg == FSL_SAI_VERID || reg == FSL_SAI_PARAM)) + return true; + switch (reg) { case FSL_SAI_TFR0: case FSL_SAI_TFR1: @@ -872,6 +935,10 @@ static bool fsl_sai_writeable_reg(struct device *dev, unsigned int reg) case FSL_SAI_TDR7: case FSL_SAI_TMR: case FSL_SAI_RMR: + case FSL_SAI_MCTL: + case FSL_SAI_MDIV: + case FSL_SAI_TTCTL: + case FSL_SAI_RTCTL: return true; default: return false; @@ -920,6 +987,7 @@ static int fsl_sai_probe(struct platform_device *pdev) if (sai->soc_data->reg_offset == 8) { fsl_sai_regmap_config.reg_defaults = fsl_sai_reg_defaults_ofs8; + fsl_sai_regmap_config.max_register = FSL_SAI_MDIV; fsl_sai_regmap_config.num_reg_defaults = ARRAY_SIZE(fsl_sai_reg_defaults_ofs8); } diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 677ecfc1ec68..771990396804 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -14,6 +14,8 @@ SNDRV_PCM_FMTBIT_S32_LE) /* SAI Register Map Register */ +#define FSL_SAI_VERID 0x00 /* SAI Version ID Register */ +#define FSL_SAI_PARAM 0x04 /* SAI Parameter Register */ #define FSL_SAI_TCSR(ofs) (0x00 + ofs) /* SAI Transmit Control */ #define FSL_SAI_TCR1(ofs) (0x04 + ofs) /* SAI Transmit Configuration 1 */ #define FSL_SAI_TCR2(ofs) (0x08 + ofs) /* SAI Transmit Configuration 2 */ @@ -37,6 +39,10 @@ #define FSL_SAI_TFR6 0x58 /* SAI Transmit FIFO 6 */ #define FSL_SAI_TFR7 0x5C /* SAI Transmit FIFO 7 */ #define FSL_SAI_TMR 0x60 /* SAI Transmit Mask */ +#define FSL_SAI_TTCTL 0x70 /* SAI Transmit Timestamp Control Register */ +#define FSL_SAI_TTCTN 0x74 /* SAI Transmit Timestamp Counter Register */ +#define FSL_SAI_TBCTN 0x78 /* SAI Transmit Bit Counter Register */ +#define FSL_SAI_TTCAP 0x7C /* SAI Transmit Timestamp Capture */ #define FSL_SAI_RCSR(ofs) (0x80 + ofs) /* SAI Receive Control */ #define FSL_SAI_RCR1(ofs) (0x84 + ofs)/* SAI Receive Configuration 1 */ #define FSL_SAI_RCR2(ofs) (0x88 + ofs) /* SAI Receive Configuration 2 */ @@ -60,6 +66,13 @@ #define FSL_SAI_RFR6 0xd8 /* SAI Receive FIFO 6 */ #define FSL_SAI_RFR7 0xdc /* SAI Receive FIFO 7 */ #define FSL_SAI_RMR 0xe0 /* SAI Receive Mask */ +#define FSL_SAI_RTCTL 0xf0 /* SAI Receive Timestamp Control Register */ +#define FSL_SAI_RTCTN 0xf4 /* SAI Receive Timestamp Counter Register */ +#define FSL_SAI_RBCTN 0xf8 /* SAI Receive Bit Counter Register */ +#define FSL_SAI_RTCAP 0xfc /* SAI Receive Timestamp Capture */ + +#define FSL_SAI_MCTL 0x100 /* SAI MCLK Control Register */ +#define FSL_SAI_MDIV 0x104 /* SAI MCLK Divide Register */ #define FSL_SAI_xCSR(tx, ofs) (tx ? FSL_SAI_TCSR(ofs) : FSL_SAI_RCSR(ofs)) #define FSL_SAI_xCR1(tx, ofs) (tx ? FSL_SAI_TCR1(ofs) : FSL_SAI_RCR1(ofs)) @@ -67,12 +80,14 @@ #define FSL_SAI_xCR3(tx, ofs) (tx ? FSL_SAI_TCR3(ofs) : FSL_SAI_RCR3(ofs)) #define FSL_SAI_xCR4(tx, ofs) (tx ? FSL_SAI_TCR4(ofs) : FSL_SAI_RCR4(ofs)) #define FSL_SAI_xCR5(tx, ofs) (tx ? FSL_SAI_TCR5(ofs) : FSL_SAI_RCR5(ofs)) -#define FSL_SAI_xDR(tx, ofs) (tx ? FSL_SAI_TDR(ofs) : FSL_SAI_RDR(ofs)) -#define FSL_SAI_xFR(tx, ofs) (tx ? FSL_SAI_TFR(ofs) : FSL_SAI_RFR(ofs)) +#define FSL_SAI_xDR0(tx) (tx ? FSL_SAI_TDR0 : FSL_SAI_RDR0) +#define FSL_SAI_xFR0(tx) (tx ? FSL_SAI_TFR0 : FSL_SAI_RFR0) #define FSL_SAI_xMR(tx) (tx ? FSL_SAI_TMR : FSL_SAI_RMR) /* SAI Transmit/Receive Control Register */ #define FSL_SAI_CSR_TERE BIT(31) +#define FSL_SAI_CSR_SE BIT(30) +#define FSL_SAI_CSR_BCE BIT(28) #define FSL_SAI_CSR_FR BIT(25) #define FSL_SAI_CSR_SR BIT(24) #define FSL_SAI_CSR_xF_SHIFT 16 @@ -106,6 +121,7 @@ #define FSL_SAI_CR2_MSEL(ID) ((ID) << 26) #define FSL_SAI_CR2_BCP BIT(25) #define FSL_SAI_CR2_BCD_MSTR BIT(24) +#define FSL_SAI_CR2_BYP BIT(23) /* BCLK bypass */ #define FSL_SAI_CR2_DIV_MASK 0xff /* SAI Transmit and Receive Configuration 3 Register */ @@ -115,6 +131,13 @@ #define FSL_SAI_CR3_WDFL_MASK 0x1f /* SAI Transmit and Receive Configuration 4 Register */ + +#define FSL_SAI_CR4_FCONT BIT(28) +#define FSL_SAI_CR4_FCOMB_SHIFT BIT(26) +#define FSL_SAI_CR4_FCOMB_SOFT BIT(27) +#define FSL_SAI_CR4_FCOMB_MASK (0x3 << 26) +#define FSL_SAI_CR4_FPACK_8 (0x2 << 24) +#define FSL_SAI_CR4_FPACK_16 (0x3 << 24) #define FSL_SAI_CR4_FRSZ(x) (((x) - 1) << 16) #define FSL_SAI_CR4_FRSZ_MASK (0x1f << 16) #define FSL_SAI_CR4_SYWD(x) (((x) - 1) << 8) @@ -132,6 +155,43 @@ #define FSL_SAI_CR5_FBT(x) ((x) << 8) #define FSL_SAI_CR5_FBT_MASK (0x1f << 8) +/* SAI MCLK Control Register */ +#define FSL_SAI_MCTL_MCLK_EN BIT(30) /* MCLK Enable */ +#define FSL_SAI_MCTL_MSEL_MASK (0x3 << 24) +#define FSL_SAI_MCTL_MSEL(ID) ((ID) << 24) +#define FSL_SAI_MCTL_MSEL_BUS 0 +#define FSL_SAI_MCTL_MSEL_MCLK1 BIT(24) +#define FSL_SAI_MCTL_MSEL_MCLK2 BIT(25) +#define FSL_SAI_MCTL_MSEL_MCLK3 (BIT(24) | BIT(25)) +#define FSL_SAI_MCTL_DIV_EN BIT(23) +#define FSL_SAI_MCTL_DIV_MASK 0xFF + +/* SAI VERID Register */ +#define FSL_SAI_VERID_MAJOR_SHIFT 24 +#define FSL_SAI_VERID_MAJOR_MASK GENMASK(31, 24) +#define FSL_SAI_VERID_MINOR_SHIFT 16 +#define FSL_SAI_VERID_MINOR_MASK GENMASK(23, 16) +#define FSL_SAI_VERID_FEATURE_SHIFT 0 +#define FSL_SAI_VERID_FEATURE_MASK GENMASK(15, 0) +#define FSL_SAI_VERID_EFIFO_EN BIT(0) +#define FSL_SAI_VERID_TSTMP_EN BIT(1) + +/* SAI PARAM Register */ +#define FSL_SAI_PARAM_SPF_SHIFT 16 +#define FSL_SAI_PARAM_SPF_MASK GENMASK(19, 16) +#define FSL_SAI_PARAM_WPF_SHIFT 8 +#define FSL_SAI_PARAM_WPF_MASK GENMASK(11, 8) +#define FSL_SAI_PARAM_DLN_MASK GENMASK(3, 0) + +/* SAI MCLK Divide Register */ +#define FSL_SAI_MDIV_MASK 0xFFFFF + +/* SAI timestamp and bitcounter */ +#define FSL_SAI_xTCTL_TSEN BIT(0) +#define FSL_SAI_xTCTL_TSINC BIT(1) +#define FSL_SAI_xTCTL_RTSC BIT(8) +#define FSL_SAI_xTCTL_RBC BIT(9) + /* SAI type */ #define FSL_SAI_DMA BIT(0) #define FSL_SAI_USE_AC97 BIT(1) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 7858a5499ac5..4fd4ba9972af 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -615,6 +615,8 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: regmap_update_bits(regmap, REG_SPDIF_SCR, dmaen, 0); regmap_update_bits(regmap, REG_SPDIF_SIE, intr, 0); + regmap_write(regmap, REG_SPDIF_STL, 0x0); + regmap_write(regmap, REG_SPDIF_STR, 0x0); break; default: return -EINVAL; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index ed18bc69e095..0ab35c3dc7d2 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1147,14 +1147,14 @@ static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { .symmetric_channels = 1, .probe = fsl_ssi_dai_probe, .playback = { - .stream_name = "AC97 Playback", + .stream_name = "CPU AC97 Playback", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S20, }, .capture = { - .stream_name = "AC97 Capture", + .stream_name = "CPU AC97 Capture", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_48000, diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index 71590ca6394b..119a3a9684f5 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -230,6 +230,8 @@ static int imx_audmix_probe(struct platform_device *pdev) dai_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s%s", fe_name_pref, args.np->full_name + 1); + if (!dai_name) + return -ENOMEM; dev_info(pdev->dev.parent, "DAI FE name:%s\n", dai_name); @@ -238,6 +240,8 @@ static int imx_audmix_probe(struct platform_device *pdev) capture_dai_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s %s", dai_name, "CPU-Capture"); + if (!capture_dai_name) + return -ENOMEM; } priv->dai[i].cpus = &dlc[0]; @@ -268,6 +272,8 @@ static int imx_audmix_probe(struct platform_device *pdev) "AUDMIX-Playback-%d", i); be_cp = devm_kasprintf(&pdev->dev, GFP_KERNEL, "AUDMIX-Capture-%d", i); + if (!be_name || !be_pb || !be_cp) + return -ENOMEM; priv->dai[num_dai + i].cpus = &dlc[3]; priv->dai[num_dai + i].codecs = &dlc[4]; @@ -295,6 +301,9 @@ static int imx_audmix_probe(struct platform_device *pdev) priv->dapm_routes[i].source = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s %s", dai_name, "CPU-Playback"); + if (!priv->dapm_routes[i].source) + return -ENOMEM; + priv->dapm_routes[i].sink = be_pb; priv->dapm_routes[num_dai + i].source = be_pb; priv->dapm_routes[num_dai + i].sink = be_cp; @@ -313,7 +322,7 @@ static int imx_audmix_probe(struct platform_device *pdev) if (IS_ERR(priv->cpu_mclk)) { ret = PTR_ERR(priv->cpu_mclk); dev_err(&cpu_pdev->dev, "failed to get DAI mclk1: %d\n", ret); - return -EINVAL; + return ret; } priv->audmix_pdev = audmix_pdev; diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 15e8b9343c35..7106d56a3346 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -120,19 +120,19 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); if (!data) { ret = -ENOMEM; - goto fail; + goto put_device; } comp = devm_kzalloc(&pdev->dev, 3 * sizeof(*comp), GFP_KERNEL); if (!comp) { ret = -ENOMEM; - goto fail; + goto put_device; } data->codec_clk = clk_get(&codec_dev->dev, NULL); if (IS_ERR(data->codec_clk)) { ret = PTR_ERR(data->codec_clk); - goto fail; + goto put_device; } data->clk_frequency = clk_get_rate(data->codec_clk); @@ -158,10 +158,10 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data->card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&data->card, "model"); if (ret) - goto fail; + goto put_device; ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); if (ret) - goto fail; + goto put_device; data->card.num_links = 1; data->card.owner = THIS_MODULE; data->card.dai_link = &data->dai; @@ -176,7 +176,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) if (ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - goto fail; + goto put_device; } of_node_put(ssi_np); @@ -184,6 +184,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) return 0; +put_device: + put_device(&codec_dev->dev); fail: if (data && !IS_ERR(data->codec_clk)) clk_put(data->codec_clk); diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 1bc498124689..a96a7cd8af6e 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -149,8 +149,10 @@ static int asoc_simple_parse_dai(struct device_node *ep, * if he unbinded CPU or Codec. */ ret = snd_soc_get_dai_name(&args, &dlc->dai_name); - if (ret < 0) + if (ret < 0) { + of_node_put(node); return ret; + } dlc->of_node = node; @@ -464,8 +466,10 @@ static int graph_for_each_link(struct asoc_simple_priv *priv, of_node_put(codec_ep); of_node_put(codec_port); - if (ret < 0) + if (ret < 0) { + of_node_put(cpu_ep); return ret; + } codec_port_old = codec_port; } diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index ed332177b0f9..57d6d0b48068 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -446,6 +446,13 @@ static const struct dmi_system_id byt_cht_es8316_quirk_table[] = { | BYT_CHT_ES8316_INTMIC_IN2_MAP | BYT_CHT_ES8316_JD_INVERTED), }, + { /* Nanote UMPC-01 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "RWC CO.,LTD"), + DMI_MATCH(DMI_PRODUCT_NAME, "UMPC-01"), + }, + .driver_data = (void *)BYT_CHT_ES8316_INTMIC_IN1_MAP, + }, { /* Teclast X98 Plus II */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "TECLAST"), diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 7830d014d924..c740dec00f83 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -391,6 +391,18 @@ static int byt_rt5640_aif1_hw_params(struct snd_pcm_substream *substream, /* Please keep this list alphabetically sorted */ static const struct dmi_system_id byt_rt5640_quirk_table[] = { + { /* Acer Iconia One 7 B1-750 */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Insyde"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "VESPA2"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD1_IN4P | + BYT_RT5640_OVCD_TH_1500UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* Acer Iconia Tab 8 W1-810 */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Acer"), @@ -429,6 +441,21 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_MCLK_EN), }, { + /* Advantech MICA-071 */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Advantech"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "MICA-071"), + }, + /* OVCD Th = 1500uA to reliable detect head-phones vs -set */ + .driver_data = (void *)(BYT_RT5640_IN3_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_1500UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_MCLK_EN), + }, + { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 80 Cesium"), @@ -499,6 +526,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* Chuwi Vi8 dual-boot (CWI506) */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Insyde"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "i86"), + /* The above are too generic, also match BIOS info */ + DMI_MATCH(DMI_BIOS_VERSION, "CHUWI2.D86JHBNR02"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* Chuwi Vi10 (CWI505) */ .matches = { diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 921c09cdb480..0c1c8628b991 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -919,7 +919,6 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) if (adev) { snprintf(byt_rt5651_codec_name, sizeof(byt_rt5651_codec_name), "i2c-%s", acpi_dev_name(adev)); - put_device(&adev->dev); byt_rt5651_dais[dai_index].codecs->name = byt_rt5651_codec_name; } else { dev_err(&pdev->dev, "Error cannot find '%s' dev\n", mach->id); @@ -928,6 +927,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) codec_dev = bus_find_device_by_name(&i2c_bus_type, NULL, byt_rt5651_codec_name); + acpi_dev_put(adev); if (!codec_dev) return -EPROBE_DEFER; diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 439dd4ba690c..3256f7c4eb74 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -260,8 +260,10 @@ static int skl_pcm_open(struct snd_pcm_substream *substream, snd_pcm_set_sync(substream); mconfig = skl_tplg_fe_get_cpr_module(dai, substream->stream); - if (!mconfig) + if (!mconfig) { + kfree(dma_params); return -EINVAL; + } skl_tplg_d0i3_get(skl, mconfig->d0i3_caps); @@ -1490,6 +1492,7 @@ int skl_platform_register(struct device *dev) dais = krealloc(skl->dais, sizeof(skl_fe_dai) + sizeof(skl_platform_dai), GFP_KERNEL); if (!dais) { + kfree(skl->dais); ret = -ENOMEM; goto err; } @@ -1502,8 +1505,10 @@ int skl_platform_register(struct device *dev) ret = devm_snd_soc_register_component(dev, &skl_component, skl->dais, num_dais); - if (ret) + if (ret) { + kfree(skl->dais); dev_err(dev, "soc component registration failed %d\n", ret); + } err: return ret; } diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c index 667cdddc289f..7286cbd0c46f 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.c +++ b/sound/soc/intel/skylake/skl-sst-ipc.c @@ -1003,8 +1003,10 @@ int skl_ipc_get_large_config(struct sst_generic_ipc *ipc, reply.size = (reply.header >> 32) & IPC_DATA_OFFSET_SZ_MASK; buf = krealloc(reply.data, reply.size, GFP_KERNEL); - if (!buf) + if (!buf) { + kfree(reply.data); return -ENOMEM; + } *payload = buf; *bytes = reply.size; diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c index d43cbf4a71ef..d4db64d72b2c 100644 --- a/sound/soc/intel/skylake/skl-sst-utils.c +++ b/sound/soc/intel/skylake/skl-sst-utils.c @@ -299,6 +299,7 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, module->instance_id = devm_kzalloc(ctx->dev, size, GFP_KERNEL); if (!module->instance_id) { ret = -ENOMEM; + kfree(module); goto free_uuid_list; } diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 2e5fbd220923..80cff74c23af 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -438,7 +438,7 @@ static int skl_free(struct hdac_bus *bus) skl->init_done = 0; /* to be sure */ - snd_hdac_ext_stop_streams(bus); + snd_hdac_stop_streams_and_chip(bus); if (bus->irq >= 0) free_irq(bus->irq, (void *)bus); @@ -1116,7 +1116,10 @@ static void skl_shutdown(struct pci_dev *pci) if (!skl->init_done) return; - snd_hdac_ext_stop_streams(bus); + snd_hdac_stop_streams(bus); + snd_hdac_ext_bus_link_power_down_all(bus); + skl_dsp_sleep(skl->dsp); + list_for_each_entry(s, &bus->stream_list, list) { stream = stream_to_hdac_ext_stream(s); snd_hdac_ext_stream_decouple(bus, stream, false); diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index d2d5c25bf550..0215d187bdff 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -86,7 +86,7 @@ kirkwood_dma_conf_mbus_windows(void __iomem *base, int win, /* try to find matching cs for current dma address */ for (i = 0; i < dram->num_cs; i++) { - const struct mbus_dram_window *cs = dram->cs + i; + const struct mbus_dram_window *cs = &dram->cs[i]; if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { writel(cs->base & 0xffff0000, base + KIRKWOOD_AUDIO_WIN_BASE_REG(win)); diff --git a/sound/soc/mediatek/common/mtk-btcvsd.c b/sound/soc/mediatek/common/mtk-btcvsd.c index b66f7dee1e14..f6ec6937a71b 100644 --- a/sound/soc/mediatek/common/mtk-btcvsd.c +++ b/sound/soc/mediatek/common/mtk-btcvsd.c @@ -1054,11 +1054,9 @@ static int mtk_pcm_btcvsd_copy(struct snd_pcm_substream *substream, struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(component); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - mtk_btcvsd_snd_write(bt, buf, count); + return mtk_btcvsd_snd_write(bt, buf, count); else - mtk_btcvsd_snd_read(bt, buf, count); - - return 0; + return mtk_btcvsd_snd_read(bt, buf, count); } static struct snd_pcm_ops mtk_btcvsd_ops = { diff --git a/sound/soc/mediatek/mt2701/mt2701-wm8960.c b/sound/soc/mediatek/mt2701/mt2701-wm8960.c index 8c4c89e4c616..b9ad42112ea1 100644 --- a/sound/soc/mediatek/mt2701/mt2701-wm8960.c +++ b/sound/soc/mediatek/mt2701/mt2701-wm8960.c @@ -129,7 +129,8 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev) if (!codec_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_platform_node; } for_each_card_prelinks(card, i, dai_link) { if (dai_link->codecs->name) @@ -140,7 +141,7 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev) ret = snd_soc_of_parse_audio_routing(card, "audio-routing"); if (ret) { dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret); - return ret; + goto put_codec_node; } ret = devm_snd_soc_register_card(&pdev->dev, card); @@ -148,6 +149,10 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); +put_codec_node: + of_node_put(codec_node); +put_platform_node: + of_node_put(platform_node); return ret; } diff --git a/sound/soc/mediatek/mt6797/mt6797-mt6351.c b/sound/soc/mediatek/mt6797/mt6797-mt6351.c index 496f32bcfb5e..d2f6213a6bfc 100644 --- a/sound/soc/mediatek/mt6797/mt6797-mt6351.c +++ b/sound/soc/mediatek/mt6797/mt6797-mt6351.c @@ -217,7 +217,8 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) if (!codec_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_platform_node; } for_each_card_prelinks(card, i, dai_link) { if (dai_link->codecs->name) @@ -230,6 +231,9 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); + of_node_put(codec_node); +put_platform_node: + of_node_put(platform_node); return ret; } diff --git a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c index 0ee29255e731..f3dbd8164b86 100644 --- a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c +++ b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c @@ -1073,16 +1073,6 @@ static int mt8173_afe_pcm_dev_probe(struct platform_device *pdev) afe->dev = &pdev->dev; - irq_id = platform_get_irq(pdev, 0); - if (irq_id <= 0) - return irq_id < 0 ? irq_id : -ENXIO; - ret = devm_request_irq(afe->dev, irq_id, mt8173_afe_irq_handler, - 0, "Afe_ISR_Handle", (void *)afe); - if (ret) { - dev_err(afe->dev, "could not request_irq\n"); - return ret; - } - afe->base_addr = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(afe->base_addr)) return PTR_ERR(afe->base_addr); @@ -1158,6 +1148,16 @@ static int mt8173_afe_pcm_dev_probe(struct platform_device *pdev) if (ret) goto err_pm_disable; + irq_id = platform_get_irq(pdev, 0); + if (irq_id <= 0) + return irq_id < 0 ? irq_id : -ENXIO; + ret = devm_request_irq(afe->dev, irq_id, mt8173_afe_irq_handler, + 0, "Afe_ISR_Handle", (void *)afe); + if (ret) { + dev_err(afe->dev, "could not request_irq\n"); + goto err_pm_disable; + } + dev_info(&pdev->dev, "MT8173 AFE driver initialized.\n"); return 0; diff --git a/sound/soc/mediatek/mt8173/mt8173-max98090.c b/sound/soc/mediatek/mt8173/mt8173-max98090.c index de1410c2c446..32df18180114 100644 --- a/sound/soc/mediatek/mt8173/mt8173-max98090.c +++ b/sound/soc/mediatek/mt8173/mt8173-max98090.c @@ -167,7 +167,8 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev) if (!codec_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_platform_node; } for_each_card_prelinks(card, i, dai_link) { if (dai_link->codecs->name) @@ -182,6 +183,8 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev) __func__, ret); of_node_put(codec_node); + +put_platform_node: of_node_put(platform_node); return ret; } diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c index 6f8542329bab..a21aefe1a4d1 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c @@ -200,14 +200,16 @@ static int mt8173_rt5650_rt5514_dev_probe(struct platform_device *pdev) if (!mt8173_rt5650_rt5514_dais[DAI_LINK_CODEC_I2S].codecs[0].of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto out; } mt8173_rt5650_rt5514_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node = of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 1); if (!mt8173_rt5650_rt5514_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto out; } mt8173_rt5650_rt5514_codec_conf[0].of_node = mt8173_rt5650_rt5514_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node; @@ -219,6 +221,7 @@ static int mt8173_rt5650_rt5514_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); +out: of_node_put(platform_node); return ret; } diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c index 727ff0f7f20b..8e1e60a9b45c 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c @@ -256,14 +256,16 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev) if (!mt8173_rt5650_rt5676_dais[DAI_LINK_CODEC_I2S].codecs[0].of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_node; } mt8173_rt5650_rt5676_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node = of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 1); if (!mt8173_rt5650_rt5676_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_node; } mt8173_rt5650_rt5676_codec_conf[0].of_node = mt8173_rt5650_rt5676_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node; @@ -276,7 +278,8 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev) if (!mt8173_rt5650_rt5676_dais[DAI_LINK_HDMI_I2S].codecs->of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_node; } card->dev = &pdev->dev; @@ -286,6 +289,7 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); +put_node: of_node_put(platform_node); return ret; } diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index 21e7d4d3ded5..cdfc697ad94e 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -266,7 +266,8 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) if (!mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[0].of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_platform_node; } mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node = mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[0].of_node; @@ -279,7 +280,7 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "%s codec_capture_dai name fail %d\n", __func__, ret); - return ret; + goto put_platform_node; } mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[1].dai_name = codec_capture_dai; @@ -301,7 +302,8 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) if (!mt8173_rt5650_dais[DAI_LINK_HDMI_I2S].codecs->of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_platform_node; } card->dev = &pdev->dev; @@ -310,6 +312,7 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); +put_platform_node: of_node_put(platform_node); return ret; } diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index e0d24592ebd7..f9188274f6b0 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -85,9 +85,13 @@ config SND_MESON_AXG_PDM Select Y or M to add support for PDM input embedded in the Amlogic AXG SoC family +config SND_MESON_CODEC_GLUE + tristate + config SND_MESON_G12A_TOHDMITX tristate "Amlogic G12A To HDMI TX Control Support" select REGMAP_MMIO + select SND_MESON_CODEC_GLUE imply SND_SOC_HDMI_CODEC help Select Y or M to add support for HDMI audio on the g12a SoC diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index 1a8b1470ed84..529a807b3f37 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -11,6 +11,7 @@ snd-soc-meson-axg-sound-card-objs := axg-card.o snd-soc-meson-axg-spdifin-objs := axg-spdifin.o snd-soc-meson-axg-spdifout-objs := axg-spdifout.o snd-soc-meson-axg-pdm-objs := axg-pdm.o +snd-soc-meson-codec-glue-objs := meson-codec-glue.o snd-soc-meson-g12a-tohdmitx-objs := g12a-tohdmitx.o obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o @@ -24,4 +25,5 @@ obj-$(CONFIG_SND_MESON_AXG_SOUND_CARD) += snd-soc-meson-axg-sound-card.o obj-$(CONFIG_SND_MESON_AXG_SPDIFIN) += snd-soc-meson-axg-spdifin.o obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o obj-$(CONFIG_SND_MESON_AXG_PDM) += snd-soc-meson-axg-pdm.o +obj-$(CONFIG_SND_MESON_CODEC_GLUE) += snd-soc-meson-codec-glue.o obj-$(CONFIG_SND_MESON_G12A_TOHDMITX) += snd-soc-meson-g12a-tohdmitx.o diff --git a/sound/soc/meson/axg-spdifin.c b/sound/soc/meson/axg-spdifin.c index d0d09f945b48..7aaded1fc376 100644 --- a/sound/soc/meson/axg-spdifin.c +++ b/sound/soc/meson/axg-spdifin.c @@ -112,34 +112,6 @@ static int axg_spdifin_prepare(struct snd_pcm_substream *substream, return 0; } -static int axg_spdifin_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai); - int ret; - - ret = clk_prepare_enable(priv->refclk); - if (ret) { - dev_err(dai->dev, - "failed to enable spdifin reference clock\n"); - return ret; - } - - regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN, - SPDIFIN_CTRL0_EN); - - return 0; -} - -static void axg_spdifin_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai); - - regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN, 0); - clk_disable_unprepare(priv->refclk); -} - static void axg_spdifin_write_mode_param(struct regmap *map, int mode, unsigned int val, unsigned int num_per_reg, @@ -251,25 +223,38 @@ static int axg_spdifin_dai_probe(struct snd_soc_dai *dai) ret = axg_spdifin_sample_mode_config(dai, priv); if (ret) { dev_err(dai->dev, "mode configuration failed\n"); - clk_disable_unprepare(priv->pclk); - return ret; + goto pclk_err; } + ret = clk_prepare_enable(priv->refclk); + if (ret) { + dev_err(dai->dev, + "failed to enable spdifin reference clock\n"); + goto pclk_err; + } + + regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN, + SPDIFIN_CTRL0_EN); + return 0; + +pclk_err: + clk_disable_unprepare(priv->pclk); + return ret; } static int axg_spdifin_dai_remove(struct snd_soc_dai *dai) { struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai); + regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN, 0); + clk_disable_unprepare(priv->refclk); clk_disable_unprepare(priv->pclk); return 0; } static const struct snd_soc_dai_ops axg_spdifin_ops = { .prepare = axg_spdifin_prepare, - .startup = axg_spdifin_startup, - .shutdown = axg_spdifin_shutdown, }; static int axg_spdifin_iec958_info(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c index f7e8e9da68a0..981dbaaa6f3b 100644 --- a/sound/soc/meson/axg-tdm-formatter.c +++ b/sound/soc/meson/axg-tdm-formatter.c @@ -30,27 +30,32 @@ int axg_tdm_formatter_set_channel_masks(struct regmap *map, struct axg_tdm_stream *ts, unsigned int offset) { - unsigned int val, ch = ts->channels; - unsigned long mask; - int i, j; + unsigned int ch = ts->channels; + u32 val[AXG_TDM_NUM_LANES]; + int i, j, k; + + /* + * We need to mimick the slot distribution used by the HW to keep the + * channel placement consistent regardless of the number of channel + * in the stream. This is why the odd algorithm below is used. + */ + memset(val, 0, sizeof(*val) * AXG_TDM_NUM_LANES); /* * Distribute the channels of the stream over the available slots - * of each TDM lane + * of each TDM lane. We need to go over the 32 slots ... */ - for (i = 0; i < AXG_TDM_NUM_LANES; i++) { - val = 0; - mask = ts->mask[i]; - - for (j = find_first_bit(&mask, 32); - (j < 32) && ch; - j = find_next_bit(&mask, 32, j + 1)) { - val |= 1 << j; - ch -= 1; + for (i = 0; (i < 32) && ch; i += 2) { + /* ... of all the lanes ... */ + for (j = 0; j < AXG_TDM_NUM_LANES; j++) { + /* ... then distribute the channels in pairs */ + for (k = 0; k < 2; k++) { + if ((BIT(i + k) & ts->mask[j]) && ch) { + val[j] |= BIT(i + k); + ch -= 1; + } + } } - - regmap_write(map, offset, val); - offset += regmap_get_reg_stride(map); } /* @@ -63,6 +68,11 @@ int axg_tdm_formatter_set_channel_masks(struct regmap *map, return -EINVAL; } + for (i = 0; i < AXG_TDM_NUM_LANES; i++) { + regmap_write(map, offset, val[i]); + offset += regmap_get_reg_stride(map); + } + return 0; } EXPORT_SYMBOL_GPL(axg_tdm_formatter_set_channel_masks); diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index f5a431b8de6c..34aff050caf2 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -266,8 +266,8 @@ static int axg_tdm_iface_set_sclk(struct snd_soc_dai *dai, srate = iface->slots * iface->slot_width * params_rate(params); if (!iface->mclk_rate) { - /* If no specific mclk is requested, default to bit clock * 4 */ - clk_set_rate(iface->mclk, 4 * srate); + /* If no specific mclk is requested, default to bit clock * 2 */ + clk_set_rate(iface->mclk, 2 * srate); } else { /* Check if we can actually get the bit clock from mclk */ if (iface->mclk_rate % srate) { diff --git a/sound/soc/meson/g12a-tohdmitx.c b/sound/soc/meson/g12a-tohdmitx.c index 9cfbd343a00c..c875c350be07 100644 --- a/sound/soc/meson/g12a-tohdmitx.c +++ b/sound/soc/meson/g12a-tohdmitx.c @@ -12,112 +12,54 @@ #include <sound/soc-dai.h> #include <dt-bindings/sound/meson-g12a-tohdmitx.h> +#include "meson-codec-glue.h" #define G12A_TOHDMITX_DRV_NAME "g12a-tohdmitx" #define TOHDMITX_CTRL0 0x0 #define CTRL0_ENABLE_SHIFT 31 -#define CTRL0_I2S_DAT_SEL GENMASK(13, 12) +#define CTRL0_I2S_DAT_SEL_SHIFT 12 +#define CTRL0_I2S_DAT_SEL (0x3 << CTRL0_I2S_DAT_SEL_SHIFT) #define CTRL0_I2S_LRCLK_SEL GENMASK(9, 8) #define CTRL0_I2S_BLK_CAP_INV BIT(7) #define CTRL0_I2S_BCLK_O_INV BIT(6) #define CTRL0_I2S_BCLK_SEL GENMASK(5, 4) #define CTRL0_SPDIF_CLK_CAP_INV BIT(3) #define CTRL0_SPDIF_CLK_O_INV BIT(2) -#define CTRL0_SPDIF_SEL BIT(1) +#define CTRL0_SPDIF_SEL_SHIFT 1 +#define CTRL0_SPDIF_SEL (0x1 << CTRL0_SPDIF_SEL_SHIFT) #define CTRL0_SPDIF_CLK_SEL BIT(0) -struct g12a_tohdmitx_input { - struct snd_soc_pcm_stream params; - unsigned int fmt; -}; - -static struct snd_soc_dapm_widget * -g12a_tohdmitx_get_input(struct snd_soc_dapm_widget *w) -{ - struct snd_soc_dapm_path *p = NULL; - struct snd_soc_dapm_widget *in; - - snd_soc_dapm_widget_for_each_source_path(w, p) { - if (!p->connect) - continue; - - /* Check that we still are in the same component */ - if (snd_soc_dapm_to_component(w->dapm) != - snd_soc_dapm_to_component(p->source->dapm)) - continue; - - if (p->source->id == snd_soc_dapm_dai_in) - return p->source; - - in = g12a_tohdmitx_get_input(p->source); - if (in) - return in; - } - - return NULL; -} - -static struct g12a_tohdmitx_input * -g12a_tohdmitx_get_input_data(struct snd_soc_dapm_widget *w) -{ - struct snd_soc_dapm_widget *in = - g12a_tohdmitx_get_input(w); - struct snd_soc_dai *dai; - - if (WARN_ON(!in)) - return NULL; - - dai = in->priv; - - return dai->playback_dma_data; -} - static const char * const g12a_tohdmitx_i2s_mux_texts[] = { "I2S A", "I2S B", "I2S C", }; -static SOC_ENUM_SINGLE_EXT_DECL(g12a_tohdmitx_i2s_mux_enum, - g12a_tohdmitx_i2s_mux_texts); - -static int g12a_tohdmitx_get_input_val(struct snd_soc_component *component, - unsigned int mask) -{ - unsigned int val; - - snd_soc_component_read(component, TOHDMITX_CTRL0, &val); - return (val & mask) >> __ffs(mask); -} - -static int g12a_tohdmitx_i2s_mux_get_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = - snd_soc_dapm_kcontrol_component(kcontrol); - - ucontrol->value.enumerated.item[0] = - g12a_tohdmitx_get_input_val(component, CTRL0_I2S_DAT_SEL); - - return 0; -} - static int g12a_tohdmitx_i2s_mux_put_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = snd_soc_dapm_kcontrol_component(kcontrol); struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int mux = ucontrol->value.enumerated.item[0]; - unsigned int val = g12a_tohdmitx_get_input_val(component, - CTRL0_I2S_DAT_SEL); + unsigned int mux, changed; + + if (ucontrol->value.enumerated.item[0] >= e->items) + return -EINVAL; + + mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]); + changed = snd_soc_component_test_bits(component, e->reg, + CTRL0_I2S_DAT_SEL, + FIELD_PREP(CTRL0_I2S_DAT_SEL, + mux)); + + if (!changed) + return 0; /* Force disconnect of the mux while updating */ - if (val != mux) - snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL); + snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL); - snd_soc_component_update_bits(component, TOHDMITX_CTRL0, + snd_soc_component_update_bits(component, e->reg, CTRL0_I2S_DAT_SEL | CTRL0_I2S_LRCLK_SEL | CTRL0_I2S_BCLK_SEL, @@ -127,33 +69,22 @@ static int g12a_tohdmitx_i2s_mux_put_enum(struct snd_kcontrol *kcontrol, snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); - return 0; + return 1; } +static SOC_ENUM_SINGLE_DECL(g12a_tohdmitx_i2s_mux_enum, TOHDMITX_CTRL0, + CTRL0_I2S_DAT_SEL_SHIFT, + g12a_tohdmitx_i2s_mux_texts); + static const struct snd_kcontrol_new g12a_tohdmitx_i2s_mux = SOC_DAPM_ENUM_EXT("I2S Source", g12a_tohdmitx_i2s_mux_enum, - g12a_tohdmitx_i2s_mux_get_enum, + snd_soc_dapm_get_enum_double, g12a_tohdmitx_i2s_mux_put_enum); static const char * const g12a_tohdmitx_spdif_mux_texts[] = { "SPDIF A", "SPDIF B", }; -static SOC_ENUM_SINGLE_EXT_DECL(g12a_tohdmitx_spdif_mux_enum, - g12a_tohdmitx_spdif_mux_texts); - -static int g12a_tohdmitx_spdif_mux_get_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = - snd_soc_dapm_kcontrol_component(kcontrol); - - ucontrol->value.enumerated.item[0] = - g12a_tohdmitx_get_input_val(component, CTRL0_SPDIF_SEL); - - return 0; -} - static int g12a_tohdmitx_spdif_mux_put_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -162,13 +93,21 @@ static int g12a_tohdmitx_spdif_mux_put_enum(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int mux = ucontrol->value.enumerated.item[0]; - unsigned int val = g12a_tohdmitx_get_input_val(component, - CTRL0_SPDIF_SEL); + unsigned int mux, changed; + + if (ucontrol->value.enumerated.item[0] >= e->items) + return -EINVAL; + + mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]); + changed = snd_soc_component_test_bits(component, TOHDMITX_CTRL0, + CTRL0_SPDIF_SEL, + FIELD_PREP(CTRL0_SPDIF_SEL, mux)); + + if (!changed) + return 0; /* Force disconnect of the mux while updating */ - if (val != mux) - snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL); + snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL); snd_soc_component_update_bits(component, TOHDMITX_CTRL0, CTRL0_SPDIF_SEL | @@ -178,12 +117,16 @@ static int g12a_tohdmitx_spdif_mux_put_enum(struct snd_kcontrol *kcontrol, snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); - return 0; + return 1; } +static SOC_ENUM_SINGLE_DECL(g12a_tohdmitx_spdif_mux_enum, TOHDMITX_CTRL0, + CTRL0_SPDIF_SEL_SHIFT, + g12a_tohdmitx_spdif_mux_texts); + static const struct snd_kcontrol_new g12a_tohdmitx_spdif_mux = SOC_DAPM_ENUM_EXT("SPDIF Source", g12a_tohdmitx_spdif_mux_enum, - g12a_tohdmitx_spdif_mux_get_enum, + snd_soc_dapm_get_enum_double, g12a_tohdmitx_spdif_mux_put_enum); static const struct snd_kcontrol_new g12a_tohdmitx_out_enable = @@ -201,83 +144,13 @@ static const struct snd_soc_dapm_widget g12a_tohdmitx_widgets[] = { &g12a_tohdmitx_out_enable), }; -static int g12a_tohdmitx_input_probe(struct snd_soc_dai *dai) -{ - struct g12a_tohdmitx_input *data; - - data = kzalloc(sizeof(*data), GFP_KERNEL); - if (!data) - return -ENOMEM; - - dai->playback_dma_data = data; - return 0; -} - -static int g12a_tohdmitx_input_remove(struct snd_soc_dai *dai) -{ - kfree(dai->playback_dma_data); - return 0; -} - -static int g12a_tohdmitx_input_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct g12a_tohdmitx_input *data = dai->playback_dma_data; - - data->params.rates = snd_pcm_rate_to_rate_bit(params_rate(params)); - data->params.rate_min = params_rate(params); - data->params.rate_max = params_rate(params); - data->params.formats = 1 << params_format(params); - data->params.channels_min = params_channels(params); - data->params.channels_max = params_channels(params); - data->params.sig_bits = dai->driver->playback.sig_bits; - - return 0; -} - - -static int g12a_tohdmitx_input_set_fmt(struct snd_soc_dai *dai, - unsigned int fmt) -{ - struct g12a_tohdmitx_input *data = dai->playback_dma_data; - - /* Save the source stream format for the downstream link */ - data->fmt = fmt; - return 0; -} - -static int g12a_tohdmitx_output_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct g12a_tohdmitx_input *in_data = - g12a_tohdmitx_get_input_data(dai->capture_widget); - - if (!in_data) - return -ENODEV; - - if (WARN_ON(!rtd->dai_link->params)) { - dev_warn(dai->dev, "codec2codec link expected\n"); - return -EINVAL; - } - - /* Replace link params with the input params */ - rtd->dai_link->params = &in_data->params; - - if (!in_data->fmt) - return 0; - - return snd_soc_runtime_set_dai_fmt(rtd, in_data->fmt); -} - static const struct snd_soc_dai_ops g12a_tohdmitx_input_ops = { - .hw_params = g12a_tohdmitx_input_hw_params, - .set_fmt = g12a_tohdmitx_input_set_fmt, + .hw_params = meson_codec_glue_input_hw_params, + .set_fmt = meson_codec_glue_input_set_fmt, }; static const struct snd_soc_dai_ops g12a_tohdmitx_output_ops = { - .startup = g12a_tohdmitx_output_startup, + .startup = meson_codec_glue_output_startup, }; #define TOHDMITX_SPDIF_FORMATS \ @@ -304,8 +177,8 @@ static const struct snd_soc_dai_ops g12a_tohdmitx_output_ops = { .id = (xid), \ .playback = TOHDMITX_STREAM(xname, "Playback", xfmt, xchmax), \ .ops = &g12a_tohdmitx_input_ops, \ - .probe = g12a_tohdmitx_input_probe, \ - .remove = g12a_tohdmitx_input_remove, \ + .probe = meson_codec_glue_input_dai_probe, \ + .remove = meson_codec_glue_input_dai_remove, \ } #define TOHDMITX_OUT(xname, xid, xfmt, xchmax) { \ diff --git a/sound/soc/meson/meson-codec-glue.c b/sound/soc/meson/meson-codec-glue.c new file mode 100644 index 000000000000..524a33472337 --- /dev/null +++ b/sound/soc/meson/meson-codec-glue.c @@ -0,0 +1,149 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (c) 2019 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/module.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "meson-codec-glue.h" + +static struct snd_soc_dapm_widget * +meson_codec_glue_get_input(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_path *p = NULL; + struct snd_soc_dapm_widget *in; + + snd_soc_dapm_widget_for_each_source_path(w, p) { + if (!p->connect) + continue; + + /* Check that we still are in the same component */ + if (snd_soc_dapm_to_component(w->dapm) != + snd_soc_dapm_to_component(p->source->dapm)) + continue; + + if (p->source->id == snd_soc_dapm_dai_in) + return p->source; + + in = meson_codec_glue_get_input(p->source); + if (in) + return in; + } + + return NULL; +} + +static void meson_codec_glue_input_set_data(struct snd_soc_dai *dai, + struct meson_codec_glue_input *data) +{ + dai->playback_dma_data = data; +} + +struct meson_codec_glue_input * +meson_codec_glue_input_get_data(struct snd_soc_dai *dai) +{ + return dai->playback_dma_data; +} +EXPORT_SYMBOL_GPL(meson_codec_glue_input_get_data); + +static struct meson_codec_glue_input * +meson_codec_glue_output_get_input_data(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_widget *in = + meson_codec_glue_get_input(w); + struct snd_soc_dai *dai; + + if (WARN_ON(!in)) + return NULL; + + dai = in->priv; + + return meson_codec_glue_input_get_data(dai); +} + +int meson_codec_glue_input_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct meson_codec_glue_input *data = + meson_codec_glue_input_get_data(dai); + + data->params.rates = snd_pcm_rate_to_rate_bit(params_rate(params)); + data->params.rate_min = params_rate(params); + data->params.rate_max = params_rate(params); + data->params.formats = 1ULL << (__force int) params_format(params); + data->params.channels_min = params_channels(params); + data->params.channels_max = params_channels(params); + data->params.sig_bits = dai->driver->playback.sig_bits; + + return 0; +} +EXPORT_SYMBOL_GPL(meson_codec_glue_input_hw_params); + +int meson_codec_glue_input_set_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct meson_codec_glue_input *data = + meson_codec_glue_input_get_data(dai); + + /* Save the source stream format for the downstream link */ + data->fmt = fmt; + return 0; +} +EXPORT_SYMBOL_GPL(meson_codec_glue_input_set_fmt); + +int meson_codec_glue_output_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct meson_codec_glue_input *in_data = + meson_codec_glue_output_get_input_data(dai->capture_widget); + + if (!in_data) + return -ENODEV; + + if (WARN_ON(!rtd->dai_link->params)) { + dev_warn(dai->dev, "codec2codec link expected\n"); + return -EINVAL; + } + + /* Replace link params with the input params */ + rtd->dai_link->params = &in_data->params; + + if (!in_data->fmt) + return 0; + + return snd_soc_runtime_set_dai_fmt(rtd, in_data->fmt); +} +EXPORT_SYMBOL_GPL(meson_codec_glue_output_startup); + +int meson_codec_glue_input_dai_probe(struct snd_soc_dai *dai) +{ + struct meson_codec_glue_input *data; + + data = kzalloc(sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + meson_codec_glue_input_set_data(dai, data); + return 0; +} +EXPORT_SYMBOL_GPL(meson_codec_glue_input_dai_probe); + +int meson_codec_glue_input_dai_remove(struct snd_soc_dai *dai) +{ + struct meson_codec_glue_input *data = + meson_codec_glue_input_get_data(dai); + + kfree(data); + return 0; +} +EXPORT_SYMBOL_GPL(meson_codec_glue_input_dai_remove); + +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_DESCRIPTION("Amlogic Codec Glue Helpers"); +MODULE_LICENSE("GPL v2"); + diff --git a/sound/soc/meson/meson-codec-glue.h b/sound/soc/meson/meson-codec-glue.h new file mode 100644 index 000000000000..07f99446c0c6 --- /dev/null +++ b/sound/soc/meson/meson-codec-glue.h @@ -0,0 +1,32 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * Copyright (c) 2018 Baylibre SAS. + * Author: Jerome Brunet <jbrunet@baylibre.com> + */ + +#ifndef _MESON_CODEC_GLUE_H +#define _MESON_CODEC_GLUE_H + +#include <sound/soc.h> + +struct meson_codec_glue_input { + struct snd_soc_pcm_stream params; + unsigned int fmt; +}; + +/* Input helpers */ +struct meson_codec_glue_input * +meson_codec_glue_input_get_data(struct snd_soc_dai *dai); +int meson_codec_glue_input_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai); +int meson_codec_glue_input_set_fmt(struct snd_soc_dai *dai, + unsigned int fmt); +int meson_codec_glue_input_dai_probe(struct snd_soc_dai *dai); +int meson_codec_glue_input_dai_remove(struct snd_soc_dai *dai); + +/* Output helpers */ +int meson_codec_glue_output_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai); + +#endif /* _MESON_CODEC_GLUE_H */ diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index cb1b525cbe9d..c899a05e896f 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -767,6 +767,7 @@ static int mxs_saif_probe(struct platform_device *pdev) saif->master_id = saif->id; } else { ret = of_alias_get_id(master, "saif"); + of_node_put(master); if (ret < 0) return ret; else diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index 7096b5263e25..e9f9642e988f 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -85,7 +85,7 @@ static bool filter(struct dma_chan *chan, void *param) devname = kasprintf(GFP_KERNEL, "%s.%d", dma_data->dma_res->name, dma_data->ssp_id); - if ((strcmp(dev_name(chan->device->dev), devname) == 0) && + if (devname && (strcmp(dev_name(chan->device->dev), devname) == 0) && (chan->chan_id == dma_data->dma_res->start)) { found = true; } diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 5fdd1a24c232..ff3db623c476 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -797,7 +797,7 @@ static int pxa_ssp_probe(struct snd_soc_dai *dai) if (IS_ERR(priv->extclk)) { ret = PTR_ERR(priv->extclk); if (ret == -EPROBE_DEFER) - return ret; + goto err_priv; priv->extclk = NULL; } diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c index da242515e146..8e3539941fad 100644 --- a/sound/soc/qcom/qdsp6/q6adm.c +++ b/sound/soc/qcom/qdsp6/q6adm.c @@ -217,7 +217,7 @@ static struct q6copp *q6adm_alloc_copp(struct q6adm *adm, int port_idx) idx = find_first_zero_bit(&adm->copp_bitmap[port_idx], MAX_COPPS_PER_PORT); - if (idx > MAX_COPPS_PER_PORT) + if (idx >= MAX_COPPS_PER_PORT) return ERR_PTR(-EBUSY); c = kzalloc(sizeof(*c), GFP_ATOMIC); diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index 1707414cfa92..9f6cbaf3c0e9 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -368,6 +368,7 @@ static int rockchip_pdm_runtime_resume(struct device *dev) ret = clk_prepare_enable(pdm->hclk); if (ret) { + clk_disable_unprepare(pdm->clk); dev_err(pdm->dev, "hclock enable failed %d\n", ret); return ret; } diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index 6635145a26c4..b2b4e5b7739a 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -86,6 +86,7 @@ static int __maybe_unused rk_spdif_runtime_resume(struct device *dev) ret = clk_prepare_enable(spdif->hclk); if (ret) { + clk_disable_unprepare(spdif->mclk); dev_err(spdif->dev, "hclk clock enable failed %d\n", ret); return ret; } diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 7647b3d4c0ba..25a8cfc27433 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -171,7 +171,11 @@ static int rsnd_ctu_init(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - rsnd_mod_power_on(mod); + int ret; + + ret = rsnd_mod_power_on(mod); + if (ret < 0) + return ret; rsnd_ctu_activation(mod); diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 8d91c0eb0880..53b2ad01222b 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -186,7 +186,11 @@ static int rsnd_dvc_init(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - rsnd_mod_power_on(mod); + int ret; + + ret = rsnd_mod_power_on(mod); + if (ret < 0) + return ret; rsnd_dvc_activation(mod); diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c index a3e0370f5704..c6fe2595c373 100644 --- a/sound/soc/sh/rcar/mix.c +++ b/sound/soc/sh/rcar/mix.c @@ -146,7 +146,11 @@ static int rsnd_mix_init(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - rsnd_mod_power_on(mod); + int ret; + + ret = rsnd_mod_power_on(mod); + if (ret < 0) + return ret; rsnd_mix_activation(mod); diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 585ffba0244b..fd52e26a3808 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -454,11 +454,14 @@ static int rsnd_src_init(struct rsnd_mod *mod, struct rsnd_priv *priv) { struct rsnd_src *src = rsnd_mod_to_src(mod); + int ret; /* reset sync convert_rate */ src->sync.val = 0; - rsnd_mod_power_on(mod); + ret = rsnd_mod_power_on(mod); + if (ret < 0) + return ret; rsnd_src_activation(mod); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 09af402ca31f..f8960bad2bd1 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -518,7 +518,9 @@ static int rsnd_ssi_init(struct rsnd_mod *mod, ssi->usrcnt++; - rsnd_mod_power_on(mod); + ret = rsnd_mod_power_on(mod); + if (ret < 0) + return ret; rsnd_ssi_config_init(mod, io); diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index da6e40aef7b6..1e37bb7436ec 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -927,7 +927,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->fe_compr = 1; if (rtd->dai_link->dpcm_playback) be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; - else if (rtd->dai_link->dpcm_capture) + if (rtd->dai_link->dpcm_capture) be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops)); } else { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 093ab32ea2c3..2115fd412c78 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3512,10 +3512,23 @@ EXPORT_SYMBOL_GPL(snd_soc_of_get_dai_link_codecs); static int __init snd_soc_init(void) { + int ret; + snd_soc_debugfs_init(); - snd_soc_util_init(); + ret = snd_soc_util_init(); + if (ret) + goto err_util_init; - return platform_driver_register(&soc_driver); + ret = platform_driver_register(&soc_driver); + if (ret) + goto err_register; + return 0; + +err_register: + snd_soc_util_exit(); +err_util_init: + snd_soc_debugfs_exit(); + return ret; } module_init(snd_soc_init); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1c09dfb0c0f0..56c9c4189f26 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3421,7 +3421,6 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mixer_update_power(card, kcontrol, connect, rconnect); @@ -3527,7 +3526,6 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index ca4b17bd95d1..5552c66ca642 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -91,10 +91,10 @@ static int dmaengine_pcm_hw_params(struct snd_pcm_substream *substream, memset(&slave_config, 0, sizeof(slave_config)); - if (pcm->config && pcm->config->prepare_slave_config) - prepare_slave_config = pcm->config->prepare_slave_config; - else + if (!pcm->config) prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config; + else + prepare_slave_config = pcm->config->prepare_slave_config; if (prepare_slave_config) { ret = prepare_slave_config(substream, params, &slave_config); diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index c88bc6bb41cf..08ed973b2d97 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -252,7 +252,7 @@ int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; int min = mc->min; int sign_bit = mc->sign_bit; - unsigned int mask = (1 << fls(max)) - 1; + unsigned int mask = (1ULL << fls(max)) - 1; unsigned int invert = mc->invert; int val; int ret; @@ -445,7 +445,7 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, val = ucontrol->value.integer.value[0]; if (mc->platform_max && val > mc->platform_max) return -EINVAL; - if (val > max - min) + if (val > max) return -EINVAL; if (val < 0) return -EINVAL; @@ -458,8 +458,15 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, return err; if (snd_soc_volsw_is_stereo(mc)) { + val2 = ucontrol->value.integer.value[1]; + + if (mc->platform_max && val2 > mc->platform_max) + return -EINVAL; + if (val2 > max) + return -EINVAL; + val_mask = mask << rshift; - val2 = (ucontrol->value.integer.value[1] + min) & mask; + val2 = (val2 + min) & mask; val2 = val2 << rshift; err = snd_soc_component_update_bits(component, reg2, val_mask, @@ -523,7 +530,15 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; unsigned int val, val_mask; - int err, ret; + int err, ret, tmp; + + tmp = ucontrol->value.integer.value[0]; + if (tmp < 0) + return -EINVAL; + if (mc->platform_max && tmp > mc->platform_max) + return -EINVAL; + if (tmp > mc->max - mc->min) + return -EINVAL; if (invert) val = (max - ucontrol->value.integer.value[0]) & mask; @@ -538,6 +553,14 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, ret = err; if (snd_soc_volsw_is_stereo(mc)) { + tmp = ucontrol->value.integer.value[1]; + if (tmp < 0) + return -EINVAL; + if (mc->platform_max && tmp > mc->platform_max) + return -EINVAL; + if (tmp > mc->max - mc->min) + return -EINVAL; + if (invert) val = (max - ucontrol->value.integer.value[1]) & mask; else diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1196167364d4..2f1ab70a68fc 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1201,6 +1201,8 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, return; be_substream = snd_soc_dpcm_get_substream(be, stream); + if (!be_substream) + return; for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 54dcece52b0c..abcc5d97b134 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -174,7 +174,7 @@ int __init snd_soc_util_init(void) return ret; } -void __exit snd_soc_util_exit(void) +void snd_soc_util_exit(void) { platform_driver_unregister(&soc_dummy_driver); platform_device_unregister(soc_dummy_dev); diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index b3cdd10c83ae..c30e450fa970 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -211,6 +211,10 @@ static int hda_link_hw_params(struct snd_pcm_substream *substream, int stream_tag; int ret; + link = snd_hdac_ext_bus_get_link(bus, codec_dai->component->name); + if (!link) + return -EINVAL; + /* get stored dma data if resuming from system suspend */ link_dev = snd_soc_dai_get_dma_data(dai, substream); if (!link_dev) { @@ -231,10 +235,6 @@ static int hda_link_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - link = snd_hdac_ext_bus_get_link(bus, codec_dai->component->name); - if (!link) - return -EINVAL; - /* set the stream tag in the codec dai dma params */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) snd_soc_dai_set_tdm_slot(codec_dai, stream_tag, 0, 0, 0); diff --git a/sound/soc/sunxi/sun4i-spdif.c b/sound/soc/sunxi/sun4i-spdif.c index cbe598b0fb10..680d64e0d69f 100644 --- a/sound/soc/sunxi/sun4i-spdif.c +++ b/sound/soc/sunxi/sun4i-spdif.c @@ -464,6 +464,11 @@ static const struct of_device_id sun4i_spdif_of_match[] = { .compatible = "allwinner,sun50i-h6-spdif", .data = &sun50i_h6_spdif_quirks, }, + { + .compatible = "allwinner,sun50i-h616-spdif", + /* Essentially the same as the H6, but without RX */ + .data = &sun50i_h6_spdif_quirks, + }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, sun4i_spdif_of_match); diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c index 8e2fb81ad05c..84cbd7446b1e 100644 --- a/sound/soc/ti/ams-delta.c +++ b/sound/soc/ti/ams-delta.c @@ -303,7 +303,7 @@ static int cx81801_open(struct tty_struct *tty) static void cx81801_close(struct tty_struct *tty) { struct snd_soc_component *component = tty->disc_data; - struct snd_soc_dapm_context *dapm = &component->card->dapm; + struct snd_soc_dapm_context *dapm; del_timer_sync(&cx81801_timer); @@ -315,6 +315,8 @@ static void cx81801_close(struct tty_struct *tty) v253_ops.close(tty); + dapm = &component->card->dapm; + /* Revert back to default audio input/output constellation */ snd_soc_dapm_mutex_lock(dapm); diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index 3273b317fa3b..3e8ed05f3ebd 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -74,7 +74,8 @@ static int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) return -EINVAL; } - pm_runtime_put_sync(mcbsp->dev); + if (mcbsp->active) + pm_runtime_put_sync(mcbsp->dev); r = clk_set_parent(mcbsp->fclk, fck_src); if (r) { @@ -84,7 +85,8 @@ static int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) return r; } - pm_runtime_get_sync(mcbsp->dev); + if (mcbsp->active) + pm_runtime_get_sync(mcbsp->dev); clk_put(fck_src); |