diff options
Diffstat (limited to 'sound/soc')
74 files changed, 555 insertions, 282 deletions
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 71f2d42188c4..51e75b781968 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -11,7 +11,6 @@ if SND_ATMEL_SOC config SND_ATMEL_SOC_PDC bool - depends on HAS_DMA config SND_ATMEL_SOC_DMA bool diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index bbe2b638abb5..d870f56c44cf 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -200,6 +200,7 @@ struct atmel_i2s_dev { unsigned int fmt; const struct atmel_i2s_gck_param *gck_param; const struct atmel_i2s_caps *caps; + int clk_use_no; }; static irqreturn_t atmel_i2s_interrupt(int irq, void *dev_id) @@ -321,9 +322,16 @@ static int atmel_i2s_hw_params(struct snd_pcm_substream *substream, { struct atmel_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); bool is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); - unsigned int mr = 0; + unsigned int mr = 0, mr_mask; int ret; + mr_mask = ATMEL_I2SC_MR_FORMAT_MASK | ATMEL_I2SC_MR_MODE_MASK | + ATMEL_I2SC_MR_DATALENGTH_MASK; + if (is_playback) + mr_mask |= ATMEL_I2SC_MR_TXMONO; + else + mr_mask |= ATMEL_I2SC_MR_RXMONO; + switch (dev->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: mr |= ATMEL_I2SC_MR_FORMAT_I2S; @@ -402,7 +410,7 @@ static int atmel_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - return regmap_write(dev->regmap, ATMEL_I2SC_MR, mr); + return regmap_update_bits(dev->regmap, ATMEL_I2SC_MR, mr_mask, mr); } static int atmel_i2s_switch_mck_generator(struct atmel_i2s_dev *dev, @@ -495,18 +503,28 @@ static int atmel_i2s_trigger(struct snd_pcm_substream *substream, int cmd, is_master = (mr & ATMEL_I2SC_MR_MODE_MASK) == ATMEL_I2SC_MR_MODE_MASTER; /* If master starts, enable the audio clock. */ - if (is_master && mck_enabled) - err = atmel_i2s_switch_mck_generator(dev, true); - if (err) - return err; + if (is_master && mck_enabled) { + if (!dev->clk_use_no) { + err = atmel_i2s_switch_mck_generator(dev, true); + if (err) + return err; + } + dev->clk_use_no++; + } err = regmap_write(dev->regmap, ATMEL_I2SC_CR, cr); if (err) return err; /* If master stops, disable the audio clock. */ - if (is_master && !mck_enabled) - err = atmel_i2s_switch_mck_generator(dev, false); + if (is_master && !mck_enabled) { + if (dev->clk_use_no == 1) { + err = atmel_i2s_switch_mck_generator(dev, false); + if (err) + return err; + } + dev->clk_use_no--; + } return err; } diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index ca603397651c..1e0973322cd0 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -280,7 +280,10 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, /* Enable PMC peripheral clock for this SSC */ pr_debug("atmel_ssc_dai: Starting clock\n"); - clk_enable(ssc_p->ssc->clk); + ret = clk_enable(ssc_p->ssc->clk); + if (ret) + return ret; + ssc_p->mck_rate = clk_get_rate(ssc_p->ssc->clk); /* Reset the SSC unless initialized to keep it in a clean state */ diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index b1bef2bf142d..d1579896f3a1 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -46,35 +46,6 @@ */ #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), @@ -135,7 +106,6 @@ 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), @@ -148,7 +118,6 @@ 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; @@ -162,31 +131,6 @@ 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 */ @@ -214,6 +158,7 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0); if (!cpu_np) { dev_err(&pdev->dev, "dai and pcm info missing\n"); + of_node_put(codec_np); return -EINVAL; } at91sam9g20ek_dai.cpus->of_node = cpu_np; @@ -229,9 +174,6 @@ 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; @@ -241,8 +183,6 @@ 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 12008d3f38a7..e83c333a81cb 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -590,21 +590,26 @@ config SND_SOC_CS4349 config SND_SOC_CS47L15 tristate + depends on MFD_CS47L15 config SND_SOC_CS47L24 tristate config SND_SOC_CS47L35 tristate + depends on MFD_CS47L35 config SND_SOC_CS47L85 tristate + depends on MFD_CS47L85 config SND_SOC_CS47L90 tristate + depends on MFD_CS47L90 config SND_SOC_CS47L92 tristate + depends on MFD_CS47L92 # Cirrus Logic Quad-Channel ADC config SND_SOC_CS53L30 diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c index 1902689c5ea2..acd88fe38cd4 100644 --- a/sound/soc/codecs/cpcap.c +++ b/sound/soc/codecs/cpcap.c @@ -1541,6 +1541,8 @@ static int cpcap_codec_probe(struct platform_device *pdev) { struct device_node *codec_node = of_get_child_by_name(pdev->dev.parent->of_node, "audio-codec"); + if (!codec_node) + return -ENODEV; pdev->dev.of_node = codec_node; diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 2fb65f246b0c..77af5b67b9bb 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -150,7 +150,6 @@ static const struct snd_kcontrol_new cs4265_snd_controls[] = { SOC_SINGLE("E to F Buffer Disable Switch", CS4265_SPDIF_CTL1, 6, 1, 0), SOC_ENUM("C Data Access", cam_mode_enum), - SOC_SINGLE("SPDIF Switch", CS4265_SPDIF_CTL2, 5, 1, 1), SOC_SINGLE("Validity Bit Control Switch", CS4265_SPDIF_CTL2, 3, 1, 0), SOC_ENUM("SPDIF Mono/Stereo", spdif_mono_stereo_enum), @@ -186,7 +185,7 @@ static const struct snd_soc_dapm_widget cs4265_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("Loopback", SND_SOC_NOPM, 0, 0, &loopback_ctl), - SND_SOC_DAPM_SWITCH("SPDIF", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_SWITCH("SPDIF", CS4265_SPDIF_CTL2, 5, 1, &spdif_switch), SND_SOC_DAPM_SWITCH("DAC", CS4265_PWRCTL, 1, 1, &dac_switch), diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 5faf8877137a..ebee58eca4d5 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -91,7 +91,7 @@ static const struct reg_default cs42l42_reg_defaults[] = { { CS42L42_ASP_RX_INT_MASK, 0x1F }, { CS42L42_ASP_TX_INT_MASK, 0x0F }, { CS42L42_CODEC_INT_MASK, 0x03 }, - { CS42L42_SRCPL_INT_MASK, 0xFF }, + { CS42L42_SRCPL_INT_MASK, 0x7F }, { CS42L42_VPMON_INT_MASK, 0x01 }, { CS42L42_PLL_LOCK_INT_MASK, 0x01 }, { CS42L42_TSRS_PLUG_INT_MASK, 0x0F }, @@ -128,7 +128,7 @@ static const struct reg_default cs42l42_reg_defaults[] = { { CS42L42_MIXER_CHA_VOL, 0x3F }, { CS42L42_MIXER_ADC_VOL, 0x3F }, { CS42L42_MIXER_CHB_VOL, 0x3F }, - { CS42L42_EQ_COEF_IN0, 0x22 }, + { CS42L42_EQ_COEF_IN0, 0x00 }, { CS42L42_EQ_COEF_IN1, 0x00 }, { CS42L42_EQ_COEF_IN2, 0x00 }, { CS42L42_EQ_COEF_IN3, 0x00 }, @@ -403,7 +403,7 @@ static const struct regmap_config cs42l42_regmap = { .use_single_write = true, }; -static DECLARE_TLV_DB_SCALE(adc_tlv, -9600, 100, false); +static DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 100, true); static DECLARE_TLV_DB_SCALE(mixer_tlv, -6300, 100, true); static const char * const cs42l42_hpf_freq_text[] = { @@ -423,34 +423,23 @@ static SOC_ENUM_SINGLE_DECL(cs42l42_wnf3_freq_enum, CS42L42_ADC_WNF_HPF_CTL, CS42L42_ADC_WNF_CF_SHIFT, cs42l42_wnf3_freq_text); -static const char * const cs42l42_wnf05_freq_text[] = { - "280Hz", "315Hz", "350Hz", "385Hz", - "420Hz", "455Hz", "490Hz", "525Hz" -}; - -static SOC_ENUM_SINGLE_DECL(cs42l42_wnf05_freq_enum, CS42L42_ADC_WNF_HPF_CTL, - CS42L42_ADC_WNF_CF_SHIFT, - cs42l42_wnf05_freq_text); - static const struct snd_kcontrol_new cs42l42_snd_controls[] = { /* ADC Volume and Filter Controls */ SOC_SINGLE("ADC Notch Switch", CS42L42_ADC_CTL, - CS42L42_ADC_NOTCH_DIS_SHIFT, true, false), + CS42L42_ADC_NOTCH_DIS_SHIFT, true, true), SOC_SINGLE("ADC Weak Force Switch", CS42L42_ADC_CTL, CS42L42_ADC_FORCE_WEAK_VCM_SHIFT, true, false), SOC_SINGLE("ADC Invert Switch", CS42L42_ADC_CTL, CS42L42_ADC_INV_SHIFT, true, false), SOC_SINGLE("ADC Boost Switch", CS42L42_ADC_CTL, CS42L42_ADC_DIG_BOOST_SHIFT, true, false), - SOC_SINGLE_SX_TLV("ADC Volume", CS42L42_ADC_VOLUME, - CS42L42_ADC_VOL_SHIFT, 0xA0, 0x6C, adc_tlv), + SOC_SINGLE_S8_TLV("ADC Volume", CS42L42_ADC_VOLUME, -97, 12, adc_tlv), SOC_SINGLE("ADC WNF Switch", CS42L42_ADC_WNF_HPF_CTL, CS42L42_ADC_WNF_EN_SHIFT, true, false), SOC_SINGLE("ADC HPF Switch", CS42L42_ADC_WNF_HPF_CTL, CS42L42_ADC_HPF_EN_SHIFT, true, false), SOC_ENUM("HPF Corner Freq", cs42l42_hpf_freq_enum), SOC_ENUM("WNF 3dB Freq", cs42l42_wnf3_freq_enum), - SOC_ENUM("WNF 05dB Freq", cs42l42_wnf05_freq_enum), /* DAC Volume and Filter Controls */ SOC_SINGLE("DACA Invert Switch", CS42L42_DAC_CTL1, @@ -669,15 +658,6 @@ static int cs42l42_pll_config(struct snd_soc_component *component) CS42L42_FSYNC_PULSE_WIDTH_MASK, CS42L42_FRAC1_VAL(fsync - 1) << CS42L42_FSYNC_PULSE_WIDTH_SHIFT); - snd_soc_component_update_bits(component, - CS42L42_ASP_FRM_CFG, - CS42L42_ASP_5050_MASK, - CS42L42_ASP_5050_MASK); - /* Set the frame delay to 1.0 SCLK clocks */ - snd_soc_component_update_bits(component, CS42L42_ASP_FRM_CFG, - CS42L42_ASP_FSD_MASK, - CS42L42_ASP_FSD_1_0 << - CS42L42_ASP_FSD_SHIFT); /* Set the sample rates (96k or lower) */ snd_soc_component_update_bits(component, CS42L42_FS_RATE_EN, CS42L42_FS_EN_MASK, @@ -773,7 +753,18 @@ static int cs42l42_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - case SND_SOC_DAIFMT_LEFT_J: + /* + * 5050 mode, frame starts on falling edge of LRCLK, + * frame delayed by 1.0 SCLKs + */ + snd_soc_component_update_bits(component, + CS42L42_ASP_FRM_CFG, + CS42L42_ASP_STP_MASK | + CS42L42_ASP_5050_MASK | + CS42L42_ASP_FSD_MASK, + CS42L42_ASP_5050_MASK | + (CS42L42_ASP_FSD_1_0 << + CS42L42_ASP_FSD_SHIFT)); break; default: return -EINVAL; @@ -1807,8 +1798,9 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client, NULL, cs42l42_irq_thread, IRQF_ONESHOT | IRQF_TRIGGER_LOW, "cs42l42", cs42l42); - - if (ret != 0) + if (ret == -EPROBE_DEFER) + goto err_disable; + else if (ret != 0) dev_err(&i2c_client->dev, "Failed to request IRQ: %d\n", ret); diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 866d7c873e3c..ca2019732013 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -77,7 +77,7 @@ #define CS42L42_HP_PDN_SHIFT 3 #define CS42L42_HP_PDN_MASK (1 << CS42L42_HP_PDN_SHIFT) #define CS42L42_ADC_PDN_SHIFT 2 -#define CS42L42_ADC_PDN_MASK (1 << CS42L42_HP_PDN_SHIFT) +#define CS42L42_ADC_PDN_MASK (1 << CS42L42_ADC_PDN_SHIFT) #define CS42L42_PDN_ALL_SHIFT 0 #define CS42L42_PDN_ALL_MASK (1 << CS42L42_PDN_ALL_SHIFT) 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/max9759.c b/sound/soc/codecs/max9759.c index 00e9d4fd1651..0c261335c8a1 100644 --- a/sound/soc/codecs/max9759.c +++ b/sound/soc/codecs/max9759.c @@ -64,7 +64,8 @@ static int speaker_gain_control_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *c = snd_soc_kcontrol_component(kcontrol); struct max9759 *priv = snd_soc_component_get_drvdata(c); - if (ucontrol->value.integer.value[0] > 3) + if (ucontrol->value.integer.value[0] < 0 || + ucontrol->value.integer.value[0] > 3) return -EINVAL; priv->gain = ucontrol->value.integer.value[0]; diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index cf6516693e4e..5a8eedea6be0 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -1196,8 +1196,8 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev) irq = platform_get_irq_byname(pdev, "mbhc_switch_int"); if (irq < 0) { - dev_err(dev, "failed to get mbhc switch irq\n"); - return irq; + ret = irq; + goto err_disable_clk; } ret = devm_request_threaded_irq(dev, irq, NULL, @@ -1211,8 +1211,8 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev) if (priv->mbhc_btn_enabled) { irq = platform_get_irq_byname(pdev, "mbhc_but_press_det"); if (irq < 0) { - dev_err(dev, "failed to get button press irq\n"); - return irq; + ret = irq; + goto err_disable_clk; } ret = devm_request_threaded_irq(dev, irq, NULL, @@ -1225,8 +1225,8 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev) irq = platform_get_irq_byname(pdev, "mbhc_but_rel_det"); if (irq < 0) { - dev_err(dev, "failed to get button release irq\n"); - return irq; + ret = irq; + goto err_disable_clk; } ret = devm_request_threaded_irq(dev, irq, NULL, @@ -1244,6 +1244,10 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev) return devm_snd_soc_register_component(dev, &pm8916_wcd_analog, pm8916_wcd_analog_dai, ARRAY_SIZE(pm8916_wcd_analog_dai)); + +err_disable_clk: + clk_disable_unprepare(priv->mclk); + return ret; } static int pm8916_wcd_analog_spmi_remove(struct platform_device *pdev) diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index 09fccacadd6b..e4cde214b7b2 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -1201,14 +1201,24 @@ static int msm8916_wcd_digital_probe(struct platform_device *pdev) ret = clk_prepare_enable(priv->mclk); if (ret < 0) { dev_err(dev, "failed to enable mclk %d\n", ret); - return ret; + goto err_clk; } dev_set_drvdata(dev, priv); - return devm_snd_soc_register_component(dev, &msm8916_wcd_digital, + ret = devm_snd_soc_register_component(dev, &msm8916_wcd_digital, msm8916_wcd_digital_dai, ARRAY_SIZE(msm8916_wcd_digital_dai)); + if (ret) + goto err_mclk; + + return 0; + +err_mclk: + clk_disable_unprepare(priv->mclk); +err_clk: + clk_disable_unprepare(priv->ahbclk); + return ret; } static int msm8916_wcd_digital_remove(struct platform_device *pdev) diff --git a/sound/soc/codecs/mt6358.c b/sound/soc/codecs/mt6358.c index bb737fd678cc..494ba0eeb433 100644 --- a/sound/soc/codecs/mt6358.c +++ b/sound/soc/codecs/mt6358.c @@ -103,6 +103,7 @@ int mt6358_set_mtkaif_protocol(struct snd_soc_component *cmpnt, priv->mtkaif_protocol = mtkaif_protocol; return 0; } +EXPORT_SYMBOL_GPL(mt6358_set_mtkaif_protocol); static void playback_gpio_set(struct mt6358_priv *priv) { @@ -269,6 +270,7 @@ int mt6358_mtkaif_calibration_enable(struct snd_soc_component *cmpnt) 1 << RG_AUD_PAD_TOP_DAT_MISO_LOOPBACK_SFT); return 0; } +EXPORT_SYMBOL_GPL(mt6358_mtkaif_calibration_enable); int mt6358_mtkaif_calibration_disable(struct snd_soc_component *cmpnt) { @@ -292,6 +294,7 @@ int mt6358_mtkaif_calibration_disable(struct snd_soc_component *cmpnt) capture_gpio_reset(priv); return 0; } +EXPORT_SYMBOL_GPL(mt6358_mtkaif_calibration_disable); int mt6358_set_mtkaif_calibration_phase(struct snd_soc_component *cmpnt, int phase_1, int phase_2) @@ -306,6 +309,7 @@ int mt6358_set_mtkaif_calibration_phase(struct snd_soc_component *cmpnt, phase_2 << RG_AUD_PAD_TOP_PHASE_MODE2_SFT); return 0; } +EXPORT_SYMBOL_GPL(mt6358_set_mtkaif_calibration_phase); /* dl pga gain */ enum { diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 15bd8335f667..c8ccfa2fff84 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -8,6 +8,7 @@ #include <linux/module.h> #include <linux/delay.h> +#include <linux/dmi.h> #include <linux/init.h> #include <linux/i2c.h> #include <linux/regmap.h> @@ -27,6 +28,12 @@ #include "nau8824.h" +#define NAU8824_JD_ACTIVE_HIGH BIT(0) + +static int nau8824_quirk; +static int quirk_override = -1; +module_param_named(quirk, quirk_override, uint, 0444); +MODULE_PARM_DESC(quirk, "Board-specific quirk override"); static int nau8824_config_sysclk(struct nau8824 *nau8824, int clk_id, unsigned int freq); @@ -1875,6 +1882,34 @@ static int nau8824_read_device_properties(struct device *dev, return 0; } +/* Please keep this list alphabetically sorted */ +static const struct dmi_system_id nau8824_quirk_table[] = { + { + /* Cyberbook T116 rugged tablet */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "Default string"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "Cherry Trail CR"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "20170531"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, + {} +}; + +static void nau8824_check_quirks(void) +{ + const struct dmi_system_id *dmi_id; + + if (quirk_override != -1) { + nau8824_quirk = quirk_override; + return; + } + + dmi_id = dmi_first_match(nau8824_quirk_table); + if (dmi_id) + nau8824_quirk = (unsigned long)dmi_id->driver_data; +} + static int nau8824_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1899,6 +1934,11 @@ static int nau8824_i2c_probe(struct i2c_client *i2c, nau8824->irq = i2c->irq; sema_init(&nau8824->jd_sem, 1); + nau8824_check_quirks(); + + if (nau8824_quirk & NAU8824_JD_ACTIVE_HIGH) + nau8824->jkdet_polarity = 0; + nau8824_print_device_properties(nau8824); ret = regmap_read(nau8824->regmap, NAU8824_REG_I2C_DEVICE_ID, &value); diff --git a/sound/soc/codecs/rk3328_codec.c b/sound/soc/codecs/rk3328_codec.c index 287c962ba00d..514ebe16bbfa 100644 --- a/sound/soc/codecs/rk3328_codec.c +++ b/sound/soc/codecs/rk3328_codec.c @@ -472,7 +472,8 @@ static int rk3328_platform_probe(struct platform_device *pdev) rk3328->pclk = devm_clk_get(&pdev->dev, "pclk"); if (IS_ERR(rk3328->pclk)) { dev_err(&pdev->dev, "can't get acodec pclk\n"); - return PTR_ERR(rk3328->pclk); + ret = PTR_ERR(rk3328->pclk); + goto err_unprepare_mclk; } ret = clk_prepare_enable(rk3328->pclk); @@ -482,19 +483,34 @@ static int rk3328_platform_probe(struct platform_device *pdev) } base = devm_platform_ioremap_resource(pdev, 0); - if (IS_ERR(base)) - return PTR_ERR(base); + if (IS_ERR(base)) { + ret = PTR_ERR(base); + goto err_unprepare_pclk; + } rk3328->regmap = devm_regmap_init_mmio(&pdev->dev, base, &rk3328_codec_regmap_config); - if (IS_ERR(rk3328->regmap)) - return PTR_ERR(rk3328->regmap); + if (IS_ERR(rk3328->regmap)) { + ret = PTR_ERR(rk3328->regmap); + goto err_unprepare_pclk; + } platform_set_drvdata(pdev, rk3328); - return devm_snd_soc_register_component(&pdev->dev, &soc_codec_rk3328, + ret = devm_snd_soc_register_component(&pdev->dev, &soc_codec_rk3328, rk3328_dai, ARRAY_SIZE(rk3328_dai)); + if (ret) + goto err_unprepare_pclk; + + return 0; + +err_unprepare_pclk: + clk_disable_unprepare(rk3328->pclk); + +err_unprepare_mclk: + clk_disable_unprepare(rk3328->mclk); + return ret; } static const struct of_device_id rk3328_codec_of_match[] = { diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index f70b9f7e68bb..281957a8fa86 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1691,6 +1691,8 @@ static const struct regmap_config rt5631_regmap_config = { .reg_defaults = rt5631_reg, .num_reg_defaults = ARRAY_SIZE(rt5631_reg), .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, }; static int rt5631_i2c_probe(struct i2c_client *i2c, diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 2943692f66ed..19e2f622718d 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3461,6 +3461,7 @@ static void rt5663_calibrate(struct rt5663_priv *rt5663) static int rt5663_parse_dp(struct rt5663_priv *rt5663, struct device *dev) { int table_size; + int ret; device_property_read_u32(dev, "realtek,dc_offset_l_manual", &rt5663->pdata.dc_offset_l_manual); @@ -3477,9 +3478,13 @@ static int rt5663_parse_dp(struct rt5663_priv *rt5663, struct device *dev) table_size = sizeof(struct impedance_mapping_table) * rt5663->pdata.impedance_sensing_num; rt5663->imp_table = devm_kzalloc(dev, table_size, GFP_KERNEL); - device_property_read_u32_array(dev, + if (!rt5663->imp_table) + return -ENOMEM; + ret = device_property_read_u32_array(dev, "realtek,impedance_sensing_table", (u32 *)rt5663->imp_table, table_size); + if (ret) + return ret; } return 0; @@ -3504,8 +3509,11 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, if (pdata) rt5663->pdata = *pdata; - else - rt5663_parse_dp(rt5663, &i2c->dev); + else { + ret = rt5663_parse_dp(rt5663, &i2c->dev); + if (ret) + return ret; + } for (i = 0; i < ARRAY_SIZE(rt5663->supplies); i++) rt5663->supplies[i].supply = rt5663_supply_names[i]; diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c index 5716cede99cb..acc2b34ca334 100644 --- a/sound/soc/codecs/rt5668.c +++ b/sound/soc/codecs/rt5668.c @@ -1022,11 +1022,13 @@ static void rt5668_jack_detect_handler(struct work_struct *work) container_of(work, struct rt5668_priv, jack_detect_work.work); int val, btn_type; - while (!rt5668->component) - usleep_range(10000, 15000); - - while (!rt5668->component->card->instantiated) - usleep_range(10000, 15000); + if (!rt5668->component || !rt5668->component->card || + !rt5668->component->card->instantiated) { + /* card not yet ready, try later */ + mod_delayed_work(system_power_efficient_wq, + &rt5668->jack_detect_work, msecs_to_jiffies(15)); + return; + } mutex_lock(&rt5668->calibrate_mutex); diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 05e883a65d7a..a8cf4c745130 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1052,11 +1052,13 @@ static void rt5682_jack_detect_handler(struct work_struct *work) container_of(work, struct rt5682_priv, jack_detect_work.work); int val, btn_type; - while (!rt5682->component) - usleep_range(10000, 15000); - - while (!rt5682->component->card->instantiated) - usleep_range(10000, 15000); + if (!rt5682->component || !rt5682->component->card || + !rt5682->component->card->instantiated) { + /* card not yet ready, try later */ + mod_delayed_work(system_power_efficient_wq, + &rt5682->jack_detect_work, msecs_to_jiffies(15)); + return; + } mutex_lock(&rt5682->calibrate_mutex); diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 130efc243b38..385a885dbc3d 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1814,6 +1814,9 @@ static int sgtl5000_i2c_remove(struct i2c_client *client) { struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client); + 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); @@ -1821,6 +1824,11 @@ 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}, {}, @@ -1841,6 +1849,7 @@ 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/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index cb024955c978..73c5f6c8ed69 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -151,8 +151,8 @@ struct aic31xx_pdata { #define AIC31XX_WORD_LEN_24BITS 0x02 #define AIC31XX_WORD_LEN_32BITS 0x03 #define AIC31XX_IFACE1_MASTER_MASK GENMASK(3, 2) -#define AIC31XX_BCLK_MASTER BIT(2) -#define AIC31XX_WCLK_MASTER BIT(3) +#define AIC31XX_BCLK_MASTER BIT(3) +#define AIC31XX_WCLK_MASTER BIT(2) /* AIC31XX_DATA_OFFSET */ #define AIC31XX_DATA_OFFSET_MASK GENMASK(7, 0) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 81906c25e4a8..016aff97e2fb 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -4076,6 +4076,16 @@ static int wcd9335_setup_irqs(struct wcd9335_codec *wcd) return ret; } +static void wcd9335_teardown_irqs(struct wcd9335_codec *wcd) +{ + int i; + + /* disable interrupts on all slave ports */ + for (i = 0; i < WCD9335_SLIM_NUM_PORT_REG; i++) + regmap_write(wcd->if_regmap, WCD9335_SLIM_PGD_PORT_INT_EN0 + i, + 0x00); +} + static void wcd9335_cdc_sido_ccl_enable(struct wcd9335_codec *wcd, bool ccl_flag) { @@ -4844,6 +4854,7 @@ static void wcd9335_codec_init(struct snd_soc_component *component) static int wcd9335_codec_probe(struct snd_soc_component *component) { struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); + int ret; int i; snd_soc_component_init_regmap(component, wcd->regmap); @@ -4861,7 +4872,15 @@ static int wcd9335_codec_probe(struct snd_soc_component *component) for (i = 0; i < NUM_CODEC_DAIS; i++) INIT_LIST_HEAD(&wcd->dai[i].slim_ch_list); - return wcd9335_setup_irqs(wcd); + ret = wcd9335_setup_irqs(wcd); + if (ret) + goto free_clsh_ctrl; + + return 0; + +free_clsh_ctrl: + wcd_clsh_ctrl_free(wcd->clsh_ctrl); + return ret; } static void wcd9335_codec_remove(struct snd_soc_component *comp) @@ -4869,7 +4888,7 @@ static void wcd9335_codec_remove(struct snd_soc_component *comp) struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); wcd_clsh_ctrl_free(wcd->clsh_ctrl); - free_irq(regmap_irq_get_virq(wcd->irq_data, WCD9335_IRQ_SLIMBUS), wcd); + wcd9335_teardown_irqs(wcd); } static int wcd9335_codec_set_sysclk(struct snd_soc_component *comp, diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index fe99584c917f..9cd91bb0a902 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1535,18 +1535,38 @@ static int wm8350_component_probe(struct snd_soc_component *component) wm8350_clear_bits(wm8350, WM8350_JACK_DETECT, WM8350_JDL_ENA | WM8350_JDR_ENA); - wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, + ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, wm8350_hpl_jack_handler, 0, "Left jack detect", priv); - wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, + if (ret != 0) + goto err; + + ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, wm8350_hpr_jack_handler, 0, "Right jack detect", priv); - wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, + if (ret != 0) + goto free_jck_det_l; + + ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, wm8350_mic_handler, 0, "Microphone short", priv); - wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICD, + if (ret != 0) + goto free_jck_det_r; + + ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICD, wm8350_mic_handler, 0, "Microphone detect", priv); + if (ret != 0) + goto free_micscd; return 0; + +free_micscd: + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, priv); +free_jck_det_r: + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, priv); +free_jck_det_l: + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, priv); +err: + return ret; } static void wm8350_component_remove(struct snd_soc_component *component) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 6fd1bef848ed..fa55d79b39b6 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -601,7 +601,7 @@ static int wm8731_hw_init(struct device *dev, struct wm8731_priv *wm8731) ret = wm8731_reset(wm8731->regmap); if (ret < 0) { dev_err(dev, "Failed to issue reset: %d\n", ret); - goto err_regulator_enable; + goto err; } /* Clear POWEROFF, keep everything else disabled */ @@ -618,10 +618,7 @@ static int wm8731_hw_init(struct device *dev, struct wm8731_priv *wm8731) regcache_mark_dirty(wm8731->regmap); -err_regulator_enable: - /* Regulators will be enabled by bias management */ - regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); - +err: return ret; } @@ -765,21 +762,27 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, ret = PTR_ERR(wm8731->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); - return ret; + goto err_regulator_enable; } ret = wm8731_hw_init(&i2c->dev, wm8731); if (ret != 0) - return ret; + goto err_regulator_enable; ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_wm8731, &wm8731_dai, 1); if (ret != 0) { dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); - return ret; + goto err_regulator_enable; } return 0; + +err_regulator_enable: + /* Regulators will be enabled by bias management */ + regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); + + return ret; } static int wm8731_i2c_remove(struct i2c_client *client) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index b174a9381c0c..149cfa594b76 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -697,6 +697,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, int dcs_mask; int dcs_l, dcs_r; int dcs_l_reg, dcs_r_reg; + int an_out_reg; int timeout; int pwr_reg; @@ -712,6 +713,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, dcs_mask = WM8904_DCS_ENA_CHAN_0 | WM8904_DCS_ENA_CHAN_1; dcs_r_reg = WM8904_DC_SERVO_8; dcs_l_reg = WM8904_DC_SERVO_9; + an_out_reg = WM8904_ANALOGUE_OUT1_LEFT; dcs_l = 0; dcs_r = 1; break; @@ -720,6 +722,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, dcs_mask = WM8904_DCS_ENA_CHAN_2 | WM8904_DCS_ENA_CHAN_3; dcs_r_reg = WM8904_DC_SERVO_6; dcs_l_reg = WM8904_DC_SERVO_7; + an_out_reg = WM8904_ANALOGUE_OUT2_LEFT; dcs_l = 2; dcs_r = 3; break; @@ -792,6 +795,10 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, snd_soc_component_update_bits(component, reg, WM8904_HPL_ENA_OUTP | WM8904_HPR_ENA_OUTP, WM8904_HPL_ENA_OUTP | WM8904_HPR_ENA_OUTP); + + /* Update volume, requires PGA to be powered */ + val = snd_soc_component_read32(component, an_out_reg); + snd_soc_component_write(component, an_out_reg, val); break; case SND_SOC_DAPM_POST_PMU: 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/wm8960.c b/sound/soc/codecs/wm8960.c index 5ead3633f794..cf338ad9cddd 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -730,9 +730,16 @@ static int wm8960_configure_clocking(struct snd_soc_component *component) int i, j, k; int ret; - if (!(iface1 & (1<<6))) { - dev_dbg(component->dev, - "Codec is slave mode, no need to configure clock\n"); + /* + * For Slave mode clocking should still be configured, + * so this if statement should be removed, but some platform + * may not work if the sysclk is not configured, to avoid such + * compatible issue, just add '!wm8960->sysclk' condition in + * this if statement. + */ + if (!(iface1 & (1 << 6)) && !wm8960->sysclk) { + dev_warn(component->dev, + "slave mode, but proceeding with no clock configuration\n"); return 0; } diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c index fad1eb6253d5..9e602c345619 100644 --- a/sound/soc/fsl/imx-es8328.c +++ b/sound/soc/fsl/imx-es8328.c @@ -87,6 +87,7 @@ static int imx_es8328_probe(struct platform_device *pdev) if (int_port > MUX_PORT_MAX || int_port == 0) { dev_err(dev, "mux-int-port: hardware only has %d mux ports\n", MUX_PORT_MAX); + ret = -EINVAL; goto fail; } diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index af3c3b90c0ac..83b4a22bf15a 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -93,16 +93,21 @@ static int pcm030_fabric_probe(struct platform_device *op) dev_err(&op->dev, "platform_device_alloc() failed\n"); ret = platform_device_add(pdata->codec_device); - if (ret) + if (ret) { dev_err(&op->dev, "platform_device_add() failed: %d\n", ret); + platform_device_put(pdata->codec_device); + } ret = snd_soc_register_card(card); - if (ret) + if (ret) { dev_err(&op->dev, "snd_soc_register_card() failed: %d\n", ret); + platform_device_del(pdata->codec_device); + platform_device_put(pdata->codec_device); + } platform_set_drvdata(op, pdata); - return ret; + } static int pcm030_fabric_remove(struct platform_device *op) diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index ab3b76d298b3..03470e8f3008 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -102,18 +102,15 @@ static int hi6210_i2s_startup(struct snd_pcm_substream *substream, for (n = 0; n < i2s->clocks; n++) { ret = clk_prepare_enable(i2s->clk[n]); - if (ret) { - while (n--) - clk_disable_unprepare(i2s->clk[n]); - return ret; - } + if (ret) + goto err_unprepare_clk; } ret = clk_set_rate(i2s->clk[CLK_I2S_BASE], 49152000); if (ret) { dev_err(i2s->dev, "%s: setting 49.152MHz base rate failed %d\n", __func__, ret); - return ret; + goto err_unprepare_clk; } /* enable clock before frequency division */ @@ -165,6 +162,11 @@ static int hi6210_i2s_startup(struct snd_pcm_substream *substream, hi6210_write_reg(i2s, HII2S_SW_RST_N, val); return 0; + +err_unprepare_clk: + while (n--) + clk_disable_unprepare(i2s->clk[n]); + return ret; } static void hi6210_i2s_shutdown(struct snd_pcm_substream *substream, diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c index bb668551dd4b..243f916355ee 100644 --- a/sound/soc/img/img-i2s-in.c +++ b/sound/soc/img/img-i2s-in.c @@ -464,7 +464,7 @@ static int img_i2s_in_probe(struct platform_device *pdev) if (ret) goto err_pm_disable; } - ret = pm_runtime_get_sync(&pdev->dev); + ret = pm_runtime_resume_and_get(&pdev->dev); if (ret < 0) goto err_suspend; diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index c3ff203c3f44..7d59846808b5 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -127,7 +127,7 @@ static void sst_fill_alloc_params(struct snd_pcm_substream *substream, snd_pcm_uframes_t period_size; ssize_t periodbytes; ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream); - u32 buffer_addr = virt_to_phys(substream->dma_buffer.area); + u32 buffer_addr = virt_to_phys(substream->runtime->dma_area); channels = substream->runtime->channels; period_size = substream->runtime->period_size; @@ -233,7 +233,6 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, /* set codec params and inform SST driver the same */ sst_fill_pcm_params(substream, ¶m); sst_fill_alloc_params(substream, &alloc_params); - substream->runtime->dma_area = substream->dma_buffer.area; str_params.sparams = param; str_params.aparams = alloc_params; str_params.codec = SST_CODEC_TYPE_PCM; diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index c67b86e2d0c0..7830d014d924 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -284,9 +284,6 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { {"Headphone", NULL, "Platform Clock"}, {"Headset Mic", NULL, "Platform Clock"}, - {"Internal Mic", NULL, "Platform Clock"}, - {"Speaker", NULL, "Platform Clock"}, - {"Headset Mic", NULL, "MICBIAS1"}, {"IN2P", NULL, "Headset Mic"}, {"Headphone", NULL, "HPOL"}, @@ -294,19 +291,23 @@ static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic1_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"DMIC1", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic2_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"DMIC2", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_in1_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"Internal Mic", NULL, "MICBIAS1"}, {"IN1P", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_in3_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"Internal Mic", NULL, "MICBIAS1"}, {"IN3P", NULL, "Internal Mic"}, }; @@ -348,6 +349,7 @@ static const struct snd_soc_dapm_route byt_rt5640_ssp0_aif2_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_stereo_spk_map[] = { + {"Speaker", NULL, "Platform Clock"}, {"Speaker", NULL, "SPOLP"}, {"Speaker", NULL, "SPOLN"}, {"Speaker", NULL, "SPORP"}, @@ -355,6 +357,7 @@ static const struct snd_soc_dapm_route byt_rt5640_stereo_spk_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_mono_spk_map[] = { + {"Speaker", NULL, "Platform Clock"}, {"Speaker", NULL, "SPOLP"}, {"Speaker", NULL, "SPOLN"}, }; diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 537a88932bb6..69362eae65be 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -607,7 +607,7 @@ static int kabylake_audio_probe(struct platform_device *pdev) static const struct platform_device_id kbl_board_ids[] = { { - .name = "kbl_da7219_max98357a", + .name = "kbl_da7219_mx98357a", .driver_data = (kernel_ulong_t)&kabylake_audio_card_da7219_m98357a, }, @@ -629,4 +629,4 @@ module_platform_driver(kabylake_audio) MODULE_DESCRIPTION("Audio Machine driver-DA7219 & MAX98357A in I2S mode"); MODULE_AUTHOR("Naveen Manohar <naveen.m@intel.com>"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:kbl_da7219_max98357a"); +MODULE_ALIAS("platform:kbl_da7219_mx98357a"); diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c index e200baa11011..df7f82e55a5a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c @@ -113,7 +113,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { }, { .id = "DLGS7219", - .drv_name = "kbl_da7219_max98373", + .drv_name = "kbl_da7219_mx98373", .fw_filename = "intel/dsp_fw_kbl.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &kbl_7219_98373_codecs, diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index 476ef1897961..79c6cf2c14bf 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -802,9 +802,12 @@ static u16 skl_get_module_param_size(struct skl_dev *skl, case SKL_MODULE_TYPE_BASE_OUTFMT: case SKL_MODULE_TYPE_MIC_SELECT: - case SKL_MODULE_TYPE_KPB: return sizeof(struct skl_base_outfmt_cfg); + case SKL_MODULE_TYPE_MIXER: + case SKL_MODULE_TYPE_KPB: + return sizeof(struct skl_base_cfg); + default: /* * return only base cfg when no specific module type is @@ -857,10 +860,14 @@ static int skl_set_module_format(struct skl_dev *skl, case SKL_MODULE_TYPE_BASE_OUTFMT: case SKL_MODULE_TYPE_MIC_SELECT: - case SKL_MODULE_TYPE_KPB: skl_set_base_outfmt_format(skl, module_config, *param_data); break; + case SKL_MODULE_TYPE_MIXER: + case SKL_MODULE_TYPE_KPB: + skl_set_base_module_format(skl, module_config, *param_data); + break; + default: skl_set_base_module_format(skl, module_config, *param_data); break; diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 7f287424af9b..439dd4ba690c 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1333,21 +1333,6 @@ static int skl_get_module_info(struct skl_dev *skl, return -EIO; } - list_for_each_entry(module, &skl->uuid_list, list) { - if (guid_equal(uuid_mod, &module->uuid)) { - mconfig->id.module_id = module->id; - if (mconfig->module) - mconfig->module->loadable = module->is_loadable; - ret = 0; - break; - } - } - - if (ret) - return ret; - - uuid_mod = &module->uuid; - ret = -EIO; for (i = 0; i < skl->nr_modules; i++) { skl_module = skl->modules[i]; uuid_tplg = &skl_module->uuid; @@ -1357,10 +1342,18 @@ static int skl_get_module_info(struct skl_dev *skl, break; } } + if (skl->nr_modules && ret) return ret; + ret = -EIO; list_for_each_entry(module, &skl->uuid_list, list) { + if (guid_equal(uuid_mod, &module->uuid)) { + mconfig->id.module_id = module->id; + mconfig->module->loadable = module->is_loadable; + ret = 0; + } + for (i = 0; i < MAX_IN_QUEUE; i++) { pin_id = &mconfig->m_in_pin[i].id; if (guid_equal(&pin_id->mod_uuid, &module->uuid)) @@ -1374,7 +1367,7 @@ static int skl_get_module_info(struct skl_dev *skl, } } - return 0; + return ret; } static int skl_populate_modules(struct skl_dev *skl) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 1940b17f27ef..254b796e635d 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -113,7 +113,7 @@ static int is_skl_dsp_widget_type(struct snd_soc_dapm_widget *w, static void skl_dump_mconfig(struct skl_dev *skl, struct skl_module_cfg *mcfg) { - struct skl_module_iface *iface = &mcfg->module->formats[0]; + struct skl_module_iface *iface = &mcfg->module->formats[mcfg->fmt_idx]; dev_dbg(skl->dev, "Dumping config\n"); dev_dbg(skl->dev, "Input Format:\n"); @@ -195,8 +195,8 @@ static void skl_tplg_update_params_fixup(struct skl_module_cfg *m_cfg, struct skl_module_fmt *in_fmt, *out_fmt; /* Fixups will be applied to pin 0 only */ - in_fmt = &m_cfg->module->formats[0].inputs[0].fmt; - out_fmt = &m_cfg->module->formats[0].outputs[0].fmt; + in_fmt = &m_cfg->module->formats[m_cfg->fmt_idx].inputs[0].fmt; + out_fmt = &m_cfg->module->formats[m_cfg->fmt_idx].outputs[0].fmt; if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) { if (is_fe) { @@ -239,9 +239,9 @@ static void skl_tplg_update_buffer_size(struct skl_dev *skl, /* Since fixups is applied to pin 0 only, ibs, obs needs * change for pin 0 only */ - res = &mcfg->module->resources[0]; - in_fmt = &mcfg->module->formats[0].inputs[0].fmt; - out_fmt = &mcfg->module->formats[0].outputs[0].fmt; + res = &mcfg->module->resources[mcfg->res_idx]; + in_fmt = &mcfg->module->formats[mcfg->fmt_idx].inputs[0].fmt; + out_fmt = &mcfg->module->formats[mcfg->fmt_idx].outputs[0].fmt; if (mcfg->m_type == SKL_MODULE_TYPE_SRCINT) multiplier = 5; @@ -1463,12 +1463,6 @@ static int skl_tplg_tlv_control_set(struct snd_kcontrol *kcontrol, struct skl_dev *skl = get_skl_ctx(w->dapm->dev); if (ac->params) { - /* - * Widget data is expected to be stripped of T and L - */ - size -= 2 * sizeof(unsigned int); - data += 2; - if (size > ac->max) return -EINVAL; ac->size = size; @@ -1637,11 +1631,12 @@ int skl_tplg_update_pipe_params(struct device *dev, struct skl_module_cfg *mconfig, struct skl_pipe_params *params) { - struct skl_module_res *res = &mconfig->module->resources[0]; + struct skl_module_res *res; struct skl_dev *skl = get_skl_ctx(dev); struct skl_module_fmt *format = NULL; u8 cfg_idx = mconfig->pipe->cur_config_idx; + res = &mconfig->module->resources[mconfig->res_idx]; skl_tplg_fill_dma_id(mconfig, params); mconfig->fmt_idx = mconfig->mod_cfg[cfg_idx].fmt_idx; mconfig->res_idx = mconfig->mod_cfg[cfg_idx].res_idx; @@ -1650,9 +1645,9 @@ int skl_tplg_update_pipe_params(struct device *dev, return 0; if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) - format = &mconfig->module->formats[0].inputs[0].fmt; + format = &mconfig->module->formats[mconfig->fmt_idx].inputs[0].fmt; else - format = &mconfig->module->formats[0].outputs[0].fmt; + format = &mconfig->module->formats[mconfig->fmt_idx].outputs[0].fmt; /* set the hw_params */ format->s_freq = params->s_freq; diff --git a/sound/soc/mediatek/common/mtk-btcvsd.c b/sound/soc/mediatek/common/mtk-btcvsd.c index c7a81c4be068..5b47cf5d7ead 100644 --- a/sound/soc/mediatek/common/mtk-btcvsd.c +++ b/sound/soc/mediatek/common/mtk-btcvsd.c @@ -1302,7 +1302,7 @@ static const struct snd_soc_component_driver mtk_btcvsd_snd_platform = { static int mtk_btcvsd_snd_probe(struct platform_device *pdev) { - int ret = 0; + int ret; int irq_id; u32 offset[5] = {0, 0, 0, 0, 0}; struct mtk_btcvsd_snd *btcvsd; @@ -1360,7 +1360,8 @@ static int mtk_btcvsd_snd_probe(struct platform_device *pdev) btcvsd->bt_sram_bank2_base = of_iomap(dev->of_node, 1); if (!btcvsd->bt_sram_bank2_base) { dev_err(dev, "iomap bt_sram_bank2_base fail\n"); - return -EIO; + ret = -EIO; + goto unmap_pkv_err; } btcvsd->infra = syscon_regmap_lookup_by_phandle(dev->of_node, @@ -1368,7 +1369,8 @@ static int mtk_btcvsd_snd_probe(struct platform_device *pdev) if (IS_ERR(btcvsd->infra)) { dev_err(dev, "cannot find infra controller: %ld\n", PTR_ERR(btcvsd->infra)); - return PTR_ERR(btcvsd->infra); + ret = PTR_ERR(btcvsd->infra); + goto unmap_bank2_err; } /* get offset */ @@ -1377,7 +1379,7 @@ static int mtk_btcvsd_snd_probe(struct platform_device *pdev) ARRAY_SIZE(offset)); if (ret) { dev_warn(dev, "%s(), get offset fail, ret %d\n", __func__, ret); - return ret; + goto unmap_bank2_err; } btcvsd->infra_misc_offset = offset[0]; btcvsd->conn_bt_cvsd_mask = offset[1]; @@ -1396,8 +1398,18 @@ static int mtk_btcvsd_snd_probe(struct platform_device *pdev) mtk_btcvsd_snd_set_state(btcvsd, btcvsd->tx, BT_SCO_STATE_IDLE); mtk_btcvsd_snd_set_state(btcvsd, btcvsd->rx, BT_SCO_STATE_IDLE); - return devm_snd_soc_register_component(dev, &mtk_btcvsd_snd_platform, - NULL, 0); + ret = devm_snd_soc_register_component(dev, &mtk_btcvsd_snd_platform, + NULL, 0); + if (ret) + goto unmap_bank2_err; + + return 0; + +unmap_bank2_err: + iounmap(btcvsd->bt_sram_bank2_base); +unmap_pkv_err: + iounmap(btcvsd->bt_pkv_base); + return ret; } static int mtk_btcvsd_snd_remove(struct platform_device *pdev) diff --git a/sound/soc/mediatek/mt8173/mt8173-max98090.c b/sound/soc/mediatek/mt8173/mt8173-max98090.c index 22c00600c999..de1410c2c446 100644 --- a/sound/soc/mediatek/mt8173/mt8173-max98090.c +++ b/sound/soc/mediatek/mt8173/mt8173-max98090.c @@ -180,6 +180,9 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev) if (ret) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); + + of_node_put(codec_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 8717e87bfe26..6f8542329bab 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c @@ -218,6 +218,8 @@ static int mt8173_rt5650_rt5514_dev_probe(struct platform_device *pdev) if (ret) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); + + 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 9d4dd9721154..727ff0f7f20b 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c @@ -285,6 +285,8 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev) if (ret) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); + + 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 ef6f23675286..21e7d4d3ded5 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -309,6 +309,8 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) if (ret) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); + + of_node_put(platform_node); return ret; } diff --git a/sound/soc/meson/g12a-tohdmitx.c b/sound/soc/meson/g12a-tohdmitx.c index 9cfbd343a00c..cbe47e0cae42 100644 --- a/sound/soc/meson/g12a-tohdmitx.c +++ b/sound/soc/meson/g12a-tohdmitx.c @@ -127,7 +127,7 @@ 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 const struct snd_kcontrol_new g12a_tohdmitx_i2s_mux = diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index a2c79426513b..d7d272bbebb2 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -455,7 +455,10 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream, * basic clock which should be fast enough for the internal * logic. */ - clk_enable(saif->clk); + ret = clk_enable(saif->clk); + if (ret) + return ret; + ret = clk_set_rate(saif->clk, 24000000); clk_disable(saif->clk); if (ret) diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 9841e1da9782..8282fe6d00dd 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -118,6 +118,9 @@ static int mxs_sgtl5000_probe(struct platform_device *pdev) codec_np = of_parse_phandle(np, "audio-codec", 0); if (!saif_np[0] || !saif_np[1] || !codec_np) { dev_err(&pdev->dev, "phandle missing or invalid\n"); + of_node_put(codec_np); + of_node_put(saif_np[0]); + of_node_put(saif_np[1]); return -EINVAL; } diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 745cc9dd14f3..bc65009be875 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -440,9 +440,15 @@ static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol, struct session_data *session = &data->sessions[session_id]; if (ucontrol->value.integer.value[0]) { + if (session->port_id == be_id) + return 0; + session->port_id = be_id; snd_soc_dapm_mixer_update_power(dapm, kcontrol, 1, update); } else { + if (session->port_id == -1 || session->port_id != be_id) + return 0; + session->port_id = -1; snd_soc_dapm_mixer_update_power(dapm, kcontrol, 0, update); } diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 61c984f10d8e..086c90e09577 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -186,7 +186,9 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, { struct rk_i2s_dev *i2s = to_info(cpu_dai); unsigned int mask = 0, val = 0; + int ret = 0; + pm_runtime_get_sync(cpu_dai->dev); mask = I2S_CKR_MSS_MASK; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: @@ -199,7 +201,8 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, i2s->is_master_mode = false; break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_CKR, mask, val); @@ -213,7 +216,8 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, val = I2S_CKR_CKP_POS; break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_CKR, mask, val); @@ -229,14 +233,15 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_I2S: val = I2S_TXCR_IBM_NORMAL; break; - case SND_SOC_DAIFMT_DSP_A: /* PCM no delay mode */ - val = I2S_TXCR_TFS_PCM; - break; - case SND_SOC_DAIFMT_DSP_B: /* PCM delay 1 mode */ + case SND_SOC_DAIFMT_DSP_A: /* PCM delay 1 bit mode */ val = I2S_TXCR_TFS_PCM | I2S_TXCR_PBM_MODE(1); break; + case SND_SOC_DAIFMT_DSP_B: /* PCM no delay mode */ + val = I2S_TXCR_TFS_PCM; + break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_TXCR, mask, val); @@ -252,19 +257,23 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_I2S: val = I2S_RXCR_IBM_NORMAL; break; - case SND_SOC_DAIFMT_DSP_A: /* PCM no delay mode */ - val = I2S_RXCR_TFS_PCM; - break; - case SND_SOC_DAIFMT_DSP_B: /* PCM delay 1 mode */ + case SND_SOC_DAIFMT_DSP_A: /* PCM delay 1 bit mode */ val = I2S_RXCR_TFS_PCM | I2S_RXCR_PBM_MODE(1); break; + case SND_SOC_DAIFMT_DSP_B: /* PCM no delay mode */ + val = I2S_RXCR_TFS_PCM; + break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_RXCR, mask, val); - return 0; +err_pm_put: + pm_runtime_put(cpu_dai->dev); + + return ret; } static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index 65497cd477a5..47f6f5d70853 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -363,6 +363,8 @@ static int preallocate_idma_buffer(struct snd_pcm *pcm, int stream) buf->addr = idma.lp_tx_addr; buf->bytes = idma_hardware.buffer_bytes_max; buf->area = (unsigned char * __force)ioremap(buf->addr, buf->bytes); + if (!buf->area) + return -ENOMEM; return 0; } diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 3447dbdba1f1..6ac7df30a289 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -816,14 +816,27 @@ static int fsi_clk_enable(struct device *dev, return ret; } - clk_enable(clock->xck); - clk_enable(clock->ick); - clk_enable(clock->div); + ret = clk_enable(clock->xck); + if (ret) + goto err; + ret = clk_enable(clock->ick); + if (ret) + goto disable_xck; + ret = clk_enable(clock->div); + if (ret) + goto disable_ick; clock->count++; } return ret; + +disable_ick: + clk_disable(clock->ick); +disable_xck: + clk_disable(clock->xck); +err: + return ret; } static int fsi_clk_disable(struct device *dev, diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index b9aacf3d3b29..7532ab27a48d 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -289,7 +289,6 @@ static void rsnd_adg_set_ssi_clk(struct rsnd_mod *ssi_mod, u32 val) int rsnd_adg_clk_query(struct rsnd_priv *priv, unsigned int rate) { struct rsnd_adg *adg = rsnd_priv_to_adg(priv); - struct clk *clk; int i; int sel_table[] = { [CLKA] = 0x1, @@ -302,10 +301,9 @@ int rsnd_adg_clk_query(struct rsnd_priv *priv, unsigned int rate) * find suitable clock from * AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC/AUDIO_CLKI. */ - for_each_rsnd_clk(clk, adg, i) { + for (i = 0; i < CLKMAX; i++) if (rate == adg->clk_rate[i]) return sel_table[i]; - } /* * find divided clock from BRGA/BRGB diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 9e54d8ae6d2c..da6e40aef7b6 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -871,6 +871,11 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) return -EINVAL; } + if (!codec_dai) { + dev_err(rtd->card->dev, "Missing codec\n"); + return -EINVAL; + } + /* check client and interface hw capabilities */ if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a856eabf5f99..66a99d6f9434 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3180,7 +3180,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, if (!routes) { dev_err(card->dev, "ASoC: Could not allocate DAPM route table\n"); - return -EINVAL; + return -ENOMEM; } for (i = 0; i < num_routes; i++) { @@ -3364,7 +3364,7 @@ int snd_soc_get_dai_name(struct of_phandle_args *args, for_each_component(pos) { component_of_node = soc_component_to_node(pos); - if (component_of_node != args->np) + if (component_of_node != args->np || !pos->num_dai) continue; ret = snd_soc_component_of_xlate_dai_name(pos, args, dai_name); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7c4d5963692d..1c09dfb0c0f0 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1676,8 +1676,7 @@ static void dapm_seq_run(struct snd_soc_card *card, switch (w->id) { case snd_soc_dapm_pre: if (!w->event) - list_for_each_entry_safe_continue(w, n, list, - power_list); + continue; if (event == SND_SOC_DAPM_STREAM_START) ret = w->event(w, @@ -1689,8 +1688,7 @@ static void dapm_seq_run(struct snd_soc_card *card, case snd_soc_dapm_post: if (!w->event) - list_for_each_entry_safe_continue(w, n, list, - power_list); + continue; if (event == SND_SOC_DAPM_STREAM_START) ret = w->event(w, @@ -2542,10 +2540,16 @@ static struct snd_soc_dapm_widget *dapm_find_widget( return NULL; } -static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, - const char *pin, int status) +/* + * set the DAPM pin status: + * returns 1 when the value has been updated, 0 when unchanged, or a negative + * error code; called from kcontrol put callback + */ +static int __snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, + const char *pin, int status) { struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true); + int ret = 0; dapm_assert_locked(dapm); @@ -2558,13 +2562,26 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, dapm_mark_dirty(w, "pin configuration"); dapm_widget_invalidate_input_paths(w); dapm_widget_invalidate_output_paths(w); + ret = 1; } w->connected = status; if (status == 0) w->force = 0; - return 0; + return ret; +} + +/* + * similar as __snd_soc_dapm_set_pin(), but returns 0 when successful; + * called from several API functions below + */ +static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, + const char *pin, int status) +{ + int ret = __snd_soc_dapm_set_pin(dapm, pin, status); + + return ret < 0 ? ret : 0; } /** @@ -3580,14 +3597,15 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, { struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); const char *pin = (const char *)kcontrol->private_value; + int ret; - if (ucontrol->value.integer.value[0]) - snd_soc_dapm_enable_pin(&card->dapm, pin); - else - snd_soc_dapm_disable_pin(&card->dapm, pin); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + ret = __snd_soc_dapm_set_pin(&card->dapm, pin, + !!ucontrol->value.integer.value[0]); + mutex_unlock(&card->dapm_mutex); snd_soc_dapm_sync(&card->dapm); - return 0; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch); @@ -4029,7 +4047,7 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, rtd->params_select = ucontrol->value.enumerated.item[0]; - return 0; + return 1; } static void diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 95fc24580f85..c88bc6bb41cf 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -314,7 +314,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, unsigned int sign_bit = mc->sign_bit; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - int err; + int err, ret; bool type_2r = false; unsigned int val2 = 0; unsigned int val, val_mask; @@ -322,13 +322,27 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, if (sign_bit) mask = BIT(sign_bit + 1) - 1; - val = ((ucontrol->value.integer.value[0] + min) & mask); + val = ucontrol->value.integer.value[0]; + if (mc->platform_max && ((int)val + min) > mc->platform_max) + return -EINVAL; + if (val > max - min) + return -EINVAL; + if (val < 0) + return -EINVAL; + val = (val + min) & mask; if (invert) val = max - val; val_mask = mask << shift; val = val << shift; if (snd_soc_volsw_is_stereo(mc)) { - val2 = ((ucontrol->value.integer.value[1] + min) & mask); + val2 = ucontrol->value.integer.value[1]; + if (mc->platform_max && ((int)val2 + min) > mc->platform_max) + return -EINVAL; + if (val2 > max - min) + return -EINVAL; + if (val2 < 0) + return -EINVAL; + val2 = (val2 + min) & mask; if (invert) val2 = max - val2; if (reg == reg2) { @@ -342,12 +356,18 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, err = snd_soc_component_update_bits(component, reg, val_mask, val); if (err < 0) return err; + ret = err; - if (type_2r) + if (type_2r) { err = snd_soc_component_update_bits(component, reg2, val_mask, - val2); + val2); + /* Don't discard any error code or drop change flag */ + if (ret == 0 || err < 0) { + ret = err; + } + } - return err; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_put_volsw); @@ -422,8 +442,15 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, int err = 0; unsigned int val, val_mask, val2 = 0; + val = ucontrol->value.integer.value[0]; + if (mc->platform_max && val > mc->platform_max) + return -EINVAL; + if (val > max - min) + return -EINVAL; + if (val < 0) + return -EINVAL; val_mask = mask << shift; - val = (ucontrol->value.integer.value[0] + min) & mask; + val = (val + min) & mask; val = val << shift; err = snd_soc_component_update_bits(component, reg, val_mask, val); @@ -496,7 +523,7 @@ 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 ret; + int err, ret; if (invert) val = (max - ucontrol->value.integer.value[0]) & mask; @@ -505,9 +532,10 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, val_mask = mask << shift; val = val << shift; - ret = snd_soc_component_update_bits(component, reg, val_mask, val); - if (ret < 0) - return ret; + err = snd_soc_component_update_bits(component, reg, val_mask, val); + if (err < 0) + return err; + ret = err; if (snd_soc_volsw_is_stereo(mc)) { if (invert) @@ -517,8 +545,12 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, val_mask = mask << shift; val = val << shift; - ret = snd_soc_component_update_bits(component, rreg, val_mask, + err = snd_soc_component_update_bits(component, rreg, val_mask, val); + /* Don't discard any error code or drop change flag */ + if (ret == 0 || err < 0) { + ret = err; + } } return ret; @@ -889,6 +921,8 @@ int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, unsigned int i, regval, regmask; int err; + if (val < mc->min || val > mc->max) + return -EINVAL; if (invert) val = max - val; val &= mask; diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index c367609433bf..870b00229353 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -587,7 +587,8 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, if (le32_to_cpu(hdr->ops.info) == SND_SOC_TPLG_CTL_BYTES && k->iface & SNDRV_CTL_ELEM_IFACE_MIXER - && k->access & SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE + && (k->access & SNDRV_CTL_ELEM_ACCESS_TLV_READ + || k->access & SNDRV_CTL_ELEM_ACCESS_TLV_WRITE) && k->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { struct soc_bytes_ext *sbe; struct snd_soc_tplg_bytes_control *be; @@ -2777,6 +2778,7 @@ EXPORT_SYMBOL_GPL(snd_soc_tplg_widget_remove_all); /* remove dynamic controls from the component driver */ int snd_soc_tplg_component_remove(struct snd_soc_component *comp, u32 index) { + struct snd_card *card = comp->card->snd_card; struct snd_soc_dobj *dobj, *next_dobj; int pass = SOC_TPLG_PASS_END; @@ -2784,6 +2786,7 @@ int snd_soc_tplg_component_remove(struct snd_soc_component *comp, u32 index) while (pass >= SOC_TPLG_PASS_START) { /* remove mixer controls */ + down_write(&card->controls_rwsem); list_for_each_entry_safe(dobj, next_dobj, &comp->dobj_list, list) { @@ -2827,6 +2830,7 @@ int snd_soc_tplg_component_remove(struct snd_soc_component *comp, u32 index) break; } } + up_write(&card->controls_rwsem); pass--; } diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 3f645200d3a5..b3cdd10c83ae 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -67,6 +67,7 @@ static struct hdac_ext_stream * return NULL; } + spin_lock_irq(&bus->reg_lock); list_for_each_entry(stream, &bus->stream_list, list) { struct hdac_ext_stream *hstream = stream_to_hdac_ext_stream(stream); @@ -106,12 +107,12 @@ static struct hdac_ext_stream * * is updated in snd_hdac_ext_stream_decouple(). */ if (!res->decoupled) - snd_hdac_ext_stream_decouple(bus, res, true); - spin_lock_irq(&bus->reg_lock); + snd_hdac_ext_stream_decouple_locked(bus, res, true); + res->link_locked = 1; res->link_substream = substream; - spin_unlock_irq(&bus->reg_lock); } + spin_unlock_irq(&bus->reg_lock); return res; } diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 356bb134ae93..7573f3f9f0f2 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -50,7 +50,7 @@ static int cl_stream_prepare(struct snd_sof_dev *sdev, unsigned int format, ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, &pci->dev, size, dmab); if (ret < 0) { dev_err(sdev->dev, "error: memory alloc failed: %x\n", ret); - goto error; + goto out_put; } hstream->period_bytes = 0;/* initialize period_bytes */ @@ -60,16 +60,17 @@ static int cl_stream_prepare(struct snd_sof_dev *sdev, unsigned int format, ret = hda_dsp_stream_hw_params(sdev, dsp_stream, dmab, NULL); if (ret < 0) { dev_err(sdev->dev, "error: hdac prepare failed: %x\n", ret); - goto error; + goto out_free; } hda_dsp_stream_spib_config(sdev, dsp_stream, HDA_DSP_SPIB_ENABLE, size); return hstream->stream_tag; -error: - hda_dsp_stream_put(sdev, direction, hstream->stream_tag); +out_free: snd_dma_free_pages(dmab); +out_put: + hda_dsp_stream_put(sdev, direction, hstream->stream_tag); return ret; } diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index 2ed92c990b97..dd9013c47664 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -91,7 +91,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) SET_UNIPERIF_ITM_BCLR_FIFO_ERROR(player); /* Stop the player */ - snd_pcm_stop_xrun(player->substream); + snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); } ret = IRQ_HANDLED; @@ -105,7 +105,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) SET_UNIPERIF_ITM_BCLR_DMA_ERROR(player); /* Stop the player */ - snd_pcm_stop_xrun(player->substream); + snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); ret = IRQ_HANDLED; } @@ -138,7 +138,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) dev_err(player->dev, "Underflow recovery failed\n"); /* Stop the player */ - snd_pcm_stop_xrun(player->substream); + snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); ret = IRQ_HANDLED; } diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index 136059331211..065c5f0d1f5f 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -65,7 +65,7 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id) if (unlikely(status & UNIPERIF_ITS_FIFO_ERROR_MASK(reader))) { dev_err(reader->dev, "FIFO error detected\n"); - snd_pcm_stop_xrun(reader->substream); + snd_pcm_stop(reader->substream, SNDRV_PCM_STATE_XRUN); ret = IRQ_HANDLED; } diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 9e8b1497efd3..a281ceb3c67e 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -139,6 +139,7 @@ static struct snd_soc_dai_link tegra_alc5632_dai = { static struct snd_soc_card snd_soc_tegra_alc5632 = { .name = "tegra-alc5632", + .driver_name = "tegra", .owner = THIS_MODULE, .dai_link = &tegra_alc5632_dai, .num_links = 1, diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index 4954a33ff46b..30edd70e8183 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -182,6 +182,7 @@ static struct snd_soc_dai_link tegra_max98090_dai = { static struct snd_soc_card snd_soc_tegra_max98090 = { .name = "tegra-max98090", + .driver_name = "tegra", .owner = THIS_MODULE, .dai_link = &tegra_max98090_dai, .num_links = 1, diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index d46915a3ec4c..3d68a41040ed 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -132,6 +132,7 @@ static struct snd_soc_dai_link tegra_rt5640_dai = { static struct snd_soc_card snd_soc_tegra_rt5640 = { .name = "tegra-rt5640", + .driver_name = "tegra", .owner = THIS_MODULE, .dai_link = &tegra_rt5640_dai, .num_links = 1, diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c index 81cb6cc6236e..ae150ade9441 100644 --- a/sound/soc/tegra/tegra_rt5677.c +++ b/sound/soc/tegra/tegra_rt5677.c @@ -175,6 +175,7 @@ static struct snd_soc_dai_link tegra_rt5677_dai = { static struct snd_soc_card snd_soc_tegra_rt5677 = { .name = "tegra-rt5677", + .driver_name = "tegra", .owner = THIS_MODULE, .dai_link = &tegra_rt5677_dai, .num_links = 1, diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c index e13b81d29cf3..fe21d9eff8c0 100644 --- a/sound/soc/tegra/tegra_sgtl5000.c +++ b/sound/soc/tegra/tegra_sgtl5000.c @@ -97,6 +97,7 @@ static struct snd_soc_dai_link tegra_sgtl5000_dai = { static struct snd_soc_card snd_soc_tegra_sgtl5000 = { .name = "tegra-sgtl5000", + .driver_name = "tegra", .owner = THIS_MODULE, .dai_link = &tegra_sgtl5000_dai, .num_links = 1, diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c index f6dd790dad71..a2362a2189dc 100644 --- a/sound/soc/tegra/tegra_wm8753.c +++ b/sound/soc/tegra/tegra_wm8753.c @@ -101,6 +101,7 @@ static struct snd_soc_dai_link tegra_wm8753_dai = { static struct snd_soc_card snd_soc_tegra_wm8753 = { .name = "tegra-wm8753", + .driver_name = "tegra", .owner = THIS_MODULE, .dai_link = &tegra_wm8753_dai, .num_links = 1, diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 0fa01cacfec9..08bcc94dcff8 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -217,6 +217,7 @@ static struct snd_soc_dai_link tegra_wm8903_dai = { static struct snd_soc_card snd_soc_tegra_wm8903 = { .name = "tegra-wm8903", + .driver_name = "tegra", .owner = THIS_MODULE, .dai_link = &tegra_wm8903_dai, .num_links = 1, diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c index b85bd9f89073..232eac58373a 100644 --- a/sound/soc/tegra/tegra_wm9712.c +++ b/sound/soc/tegra/tegra_wm9712.c @@ -54,6 +54,7 @@ static struct snd_soc_dai_link tegra_wm9712_dai = { static struct snd_soc_card snd_soc_tegra_wm9712 = { .name = "tegra-wm9712", + .driver_name = "tegra", .owner = THIS_MODULE, .dai_link = &tegra_wm9712_dai, .num_links = 1, diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 3f67ddd13674..5086bc2446d2 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -94,6 +94,7 @@ static struct snd_soc_dai_link trimslice_tlv320aic23_dai = { static struct snd_soc_card snd_soc_trimslice = { .name = "tegra-trimslice", + .driver_name = "tegra", .owner = THIS_MODULE, .dai_link = &trimslice_tlv320aic23_dai, .num_links = 1, diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index d89b5c928c4d..b2b2dcdb05d4 100644 --- a/sound/soc/ti/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c @@ -708,7 +708,9 @@ static int davinci_i2s_probe(struct platform_device *pdev) dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) return -ENODEV; - clk_enable(dev->clk); + ret = clk_enable(dev->clk); + if (ret) + goto err_put_clk; dev->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, dev); @@ -730,6 +732,7 @@ err_unregister_component: snd_soc_unregister_component(&pdev->dev); err_release_clk: clk_disable(dev->clk); +err_put_clk: clk_put(dev->clk); return ret; } diff --git a/sound/soc/uniphier/Kconfig b/sound/soc/uniphier/Kconfig index aa3592ee1358..ddfa6424c656 100644 --- a/sound/soc/uniphier/Kconfig +++ b/sound/soc/uniphier/Kconfig @@ -23,7 +23,6 @@ config SND_SOC_UNIPHIER_LD11 tristate "UniPhier LD11/LD20 Device Driver" depends on SND_SOC_UNIPHIER select SND_SOC_UNIPHIER_AIO - select SND_SOC_UNIPHIER_AIO_DMA help This adds ASoC driver for Socionext UniPhier LD11/LD20 input and output that can be used with other codecs. @@ -34,7 +33,6 @@ config SND_SOC_UNIPHIER_PXS2 tristate "UniPhier PXs2 Device Driver" depends on SND_SOC_UNIPHIER select SND_SOC_UNIPHIER_AIO - select SND_SOC_UNIPHIER_AIO_DMA help This adds ASoC driver for Socionext UniPhier PXs2 input and output that can be used with other codecs. diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c index dc8721f4f56b..f6b3a5bdbcea 100644 --- a/sound/soc/xilinx/xlnx_formatter_pcm.c +++ b/sound/soc/xilinx/xlnx_formatter_pcm.c @@ -37,6 +37,7 @@ #define XLNX_AUD_XFER_COUNT 0x28 #define XLNX_AUD_CH_STS_START 0x2C #define XLNX_BYTES_PER_CH 0x44 +#define XLNX_AUD_ALIGN_BYTES 64 #define AUD_STS_IOC_IRQ_MASK BIT(31) #define AUD_STS_CH_STS_MASK BIT(29) @@ -370,12 +371,32 @@ static int xlnx_formatter_pcm_open(struct snd_pcm_substream *substream) snd_soc_set_runtime_hwparams(substream, &xlnx_pcm_hardware); runtime->private_data = stream_data; - /* Resize the period size divisible by 64 */ + /* Resize the period bytes as divisible by 64 */ err = snd_pcm_hw_constraint_step(runtime, 0, - SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64); + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + XLNX_AUD_ALIGN_BYTES); if (err) { dev_err(component->dev, - "unable to set constraint on period bytes\n"); + "Unable to set constraint on period bytes\n"); + return err; + } + + /* Resize the buffer bytes as divisible by 64 */ + err = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + XLNX_AUD_ALIGN_BYTES); + if (err) { + dev_err(component->dev, + "Unable to set constraint on buffer bytes\n"); + return err; + } + + /* Set periods as integer multiple */ + err = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (err < 0) { + dev_err(component->dev, + "Unable to set constraint on periods to be integer\n"); return err; } @@ -461,8 +482,8 @@ static int xlnx_formatter_pcm_hw_params(struct snd_pcm_substream *substream, stream_data->buffer_size = size; - low = lower_32_bits(substream->dma_buffer.addr); - high = upper_32_bits(substream->dma_buffer.addr); + low = lower_32_bits(runtime->dma_addr); + high = upper_32_bits(runtime->dma_addr); writel(low, stream_data->mmio + XLNX_AUD_BUFF_ADDR_LSB); writel(high, stream_data->mmio + XLNX_AUD_BUFF_ADDR_MSB); |