Merge tag 'asoc-fix-v5.19-rc4' of https://git.kernel.org/pub/scm/linux/kernel/git...

ASoC: Fixes for v5.19

Quite a large batch due to things building up for a couple of weeks but
all driver specific apart from Marek's documentation fix.

diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml
index e9a5330..ef18a57 100644 (file)
--- a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml
+++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml
@@ -25,12 +25,12 @@ properties:
       - qcom,sc7280-lpass-cpu
 
   reg:
-    minItems: 2
+    minItems: 1
     maxItems: 6
     description: LPAIF core registers
 
   reg-names:
-    minItems: 2
+    minItems: 1
     maxItems: 6
 
   clocks:
@@ -42,12 +42,12 @@ properties:
     maxItems: 10
 
   interrupts:
-    minItems: 2
+    minItems: 1
     maxItems: 4
     description: LPAIF DMA buffer interrupt
 
   interrupt-names:
-    minItems: 2
+    minItems: 1
     maxItems: 4
 
   qcom,adsp:

diff --git a/Documentation/sound/soc/dai.rst b/Documentation/sound/soc/dai.rst
index 009b07e..bf84313 100644 (file)
--- a/Documentation/sound/soc/dai.rst
+++ b/Documentation/sound/soc/dai.rst
@@ -10,7 +10,7 @@ AC97
 ====
 
 AC97 is a five wire interface commonly found on many PC sound cards. It is
-now also popular in many portable devices. This DAI has a reset line and time
+now also popular in many portable devices. This DAI has a RESET line and time
 multiplexes its data on its SDATA_OUT (playback) and SDATA_IN (capture) lines.
 The bit clock (BCLK) is always driven by the CODEC (usually 12.288MHz) and the
 frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97

diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index e32871b..7434aee 100644 (file)
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -1760,8 +1760,8 @@ static bool arizona_aif_cfg_changed(struct snd_soc_component *component,
        if (bclk != (val & ARIZONA_AIF1_BCLK_FREQ_MASK))
                return true;
 
-       val = snd_soc_component_read(component, base + ARIZONA_AIF_TX_BCLK_RATE);
-       if (lrclk != (val & ARIZONA_AIF1TX_BCPF_MASK))
+       val = snd_soc_component_read(component, base + ARIZONA_AIF_RX_BCLK_RATE);
+       if (lrclk != (val & ARIZONA_AIF1RX_BCPF_MASK))
                return true;
 
        val = snd_soc_component_read(component, base + ARIZONA_AIF_FRAME_CTRL_1);

diff --git a/sound/soc/codecs/cs47l92.c b/sound/soc/codecs/cs47l92.c
index a1b8dcd..444026b 100644 (file)
--- a/sound/soc/codecs/cs47l92.c
+++ b/sound/soc/codecs/cs47l92.c
@@ -119,7 +119,13 @@ static int cs47l92_put_demux(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;
 }
 
 static SOC_ENUM_SINGLE_DECL(cs47l92_outdemux_enum,

diff --git a/sound/soc/codecs/max98396.c b/sound/soc/codecs/max98396.c
index 56eb62b..34db388 100644 (file)
--- a/sound/soc/codecs/max98396.c
+++ b/sound/soc/codecs/max98396.c
@@ -342,12 +342,15 @@ static int max98396_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 {
        struct snd_soc_component *component = codec_dai->component;
        struct max98396_priv *max98396 = snd_soc_component_get_drvdata(component);
-       unsigned int format = 0;
+       unsigned int format_mask, format = 0;
        unsigned int bclk_pol = 0;
        int ret, status;
        int reg;
        bool update = false;
 
+       format_mask = MAX98396_PCM_MODE_CFG_FORMAT_MASK |
+                     MAX98396_PCM_MODE_CFG_LRCLKEDGE;
+
        dev_dbg(component->dev, "%s: fmt 0x%08X\n", __func__, fmt);
 
        switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
@@ -395,7 +398,7 @@ static int max98396_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
                ret = regmap_read(max98396->regmap, MAX98396_R2041_PCM_MODE_CFG, &reg);
                if (ret < 0)
                        return -EINVAL;
-               if (format != (reg & MAX98396_PCM_BCLKEDGE_BSEL_MASK)) {
+               if (format != (reg & format_mask)) {
                        update = true;
                } else {
                        ret = regmap_read(max98396->regmap,
@@ -412,8 +415,7 @@ static int max98396_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 
        regmap_update_bits(max98396->regmap,
                           MAX98396_R2041_PCM_MODE_CFG,
-                          MAX98396_PCM_BCLKEDGE_BSEL_MASK,
-                          format);
+                          format_mask, format);
 
        regmap_update_bits(max98396->regmap,
                           MAX98396_R2042_PCM_CLK_SETUP,

diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 69c80d8..18b3da9 100644 (file)
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -1984,7 +1984,12 @@ static int rt5640_set_bias_level(struct snd_soc_component *component,
                snd_soc_component_write(component, RT5640_PWR_DIG2, 0x0000);
                snd_soc_component_write(component, RT5640_PWR_VOL, 0x0000);
                snd_soc_component_write(component, RT5640_PWR_MIXER, 0x0000);
-               snd_soc_component_write(component, RT5640_PWR_ANLG1, 0x0000);
+               if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER)
+                       snd_soc_component_write(component, RT5640_PWR_ANLG1,
+                               0x0018);
+               else
+                       snd_soc_component_write(component, RT5640_PWR_ANLG1,
+                               0x0000);
                snd_soc_component_write(component, RT5640_PWR_ANLG2, 0x0000);
                break;
 
@@ -2393,9 +2398,15 @@ static void rt5640_jack_work(struct work_struct *work)
 static irqreturn_t rt5640_irq(int irq, void *data)
 {
        struct rt5640_priv *rt5640 = data;
+       int delay = 0;
+
+       if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) {
+               cancel_delayed_work_sync(&rt5640->jack_work);
+               delay = 100;
+       }
 
        if (rt5640->jack)
-               queue_delayed_work(system_long_wq, &rt5640->jack_work, 0);
+               queue_delayed_work(system_long_wq, &rt5640->jack_work, delay);
 
        return IRQ_HANDLED;
 }
@@ -2580,6 +2591,12 @@ static void rt5640_enable_hda_jack_detect(
 
        snd_soc_component_update_bits(component, RT5640_DUMMY1, 0x400, 0x0);
 
+       snd_soc_component_update_bits(component, RT5640_PWR_ANLG1,
+               RT5640_PWR_VREF2, RT5640_PWR_VREF2);
+       usleep_range(10000, 15000);
+       snd_soc_component_update_bits(component, RT5640_PWR_ANLG1,
+               RT5640_PWR_FV2, RT5640_PWR_FV2);
+
        rt5640->jack = jack;
 
        ret = request_irq(rt5640->irq, rt5640_irq,
@@ -2696,16 +2713,13 @@ static int rt5640_probe(struct snd_soc_component *component)
 
        if (device_property_read_u32(component->dev,
                                     "realtek,jack-detect-source", &val) == 0) {
-               if (val <= RT5640_JD_SRC_GPIO4) {
+               if (val <= RT5640_JD_SRC_GPIO4)
                        rt5640->jd_src = val << RT5640_JD_SFT;
-               } else if (val == RT5640_JD_SRC_HDA_HEADER) {
+               else if (val == RT5640_JD_SRC_HDA_HEADER)
                        rt5640->jd_src = RT5640_JD_SRC_HDA_HEADER;
-                       snd_soc_component_update_bits(component, RT5640_DUMMY1,
-                               0x0300, 0x0);
-               } else {
+               else
                        dev_warn(component->dev, "Warning: Invalid jack-detect-source value: %d, leaving jack-detect disabled\n",
                                 val);
-               }
        }
 
        if (!device_property_read_bool(component->dev, "realtek,jack-detect-not-inverted"))

diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 2aa48ae..3363d16 100644 (file)
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1795,6 +1795,9 @@ 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);
@@ -1802,6 +1805,11 @@ 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},
        {},
@@ -1822,6 +1830,7 @@ static struct i2c_driver sgtl5000_i2c_driver = {
        },
        .probe_new = 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 56ec586..3a808c7 100644 (file)
--- 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/tas2764.c b/sound/soc/codecs/tas2764.c
index d395fef..4cb788f 100644 (file)
--- a/sound/soc/codecs/tas2764.c
+++ b/sound/soc/codecs/tas2764.c
@@ -42,10 +42,12 @@ static void tas2764_reset(struct tas2764_priv *tas2764)
                gpiod_set_value_cansleep(tas2764->reset_gpio, 0);
                msleep(20);
                gpiod_set_value_cansleep(tas2764->reset_gpio, 1);
+               usleep_range(1000, 2000);
        }
 
        snd_soc_component_write(tas2764->component, TAS2764_SW_RST,
                                TAS2764_RST);
+       usleep_range(1000, 2000);
 }
 
 static int tas2764_set_bias_level(struct snd_soc_component *component,
@@ -107,8 +109,10 @@ static int tas2764_codec_resume(struct snd_soc_component *component)
        struct tas2764_priv *tas2764 = snd_soc_component_get_drvdata(component);
        int ret;
 
-       if (tas2764->sdz_gpio)
+       if (tas2764->sdz_gpio) {
                gpiod_set_value_cansleep(tas2764->sdz_gpio, 1);
+               usleep_range(1000, 2000);
+       }
 
        ret = snd_soc_component_update_bits(component, TAS2764_PWR_CTRL,
                                            TAS2764_PWR_CTRL_MASK,
@@ -131,7 +135,8 @@ static const char * const tas2764_ASI1_src[] = {
 };
 
 static SOC_ENUM_SINGLE_DECL(
-       tas2764_ASI1_src_enum, TAS2764_TDM_CFG2, 4, tas2764_ASI1_src);
+       tas2764_ASI1_src_enum, TAS2764_TDM_CFG2, TAS2764_TDM_CFG2_SCFG_SHIFT,
+       tas2764_ASI1_src);
 
 static const struct snd_kcontrol_new tas2764_asi1_mux =
        SOC_DAPM_ENUM("ASI1 Source", tas2764_ASI1_src_enum);
@@ -329,20 +334,22 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
        struct snd_soc_component *component = dai->component;
        struct tas2764_priv *tas2764 = snd_soc_component_get_drvdata(component);
-       u8 tdm_rx_start_slot = 0, asi_cfg_1 = 0;
-       int iface;
+       u8 tdm_rx_start_slot = 0, asi_cfg_0 = 0, asi_cfg_1 = 0;
        int ret;
 
        switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+       case SND_SOC_DAIFMT_NB_IF:
+               asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START;
+               fallthrough;
        case SND_SOC_DAIFMT_NB_NF:
                asi_cfg_1 = TAS2764_TDM_CFG1_RX_RISING;
                break;
+       case SND_SOC_DAIFMT_IB_IF:
+               asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START;
+               fallthrough;
        case SND_SOC_DAIFMT_IB_NF:
                asi_cfg_1 = TAS2764_TDM_CFG1_RX_FALLING;
                break;
-       default:
-               dev_err(tas2764->dev, "ASI format Inverse is not found\n");
-               return -EINVAL;
        }
 
        ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG1,
@@ -353,13 +360,13 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 
        switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
        case SND_SOC_DAIFMT_I2S:
+               asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START;
+               fallthrough;
        case SND_SOC_DAIFMT_DSP_A:
-               iface = TAS2764_TDM_CFG2_SCFG_I2S;
                tdm_rx_start_slot = 1;
                break;
        case SND_SOC_DAIFMT_DSP_B:
        case SND_SOC_DAIFMT_LEFT_J:
-               iface = TAS2764_TDM_CFG2_SCFG_LEFT_J;
                tdm_rx_start_slot = 0;
                break;
        default:
@@ -368,14 +375,15 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
                return -EINVAL;
        }
 
-       ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG1,
-                                           TAS2764_TDM_CFG1_MASK,
-                                           (tdm_rx_start_slot << TAS2764_TDM_CFG1_51_SHIFT));
+       ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG0,
+                                           TAS2764_TDM_CFG0_FRAME_START,
+                                           asi_cfg_0);
        if (ret < 0)
                return ret;
 
-       ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG2,
-                                           TAS2764_TDM_CFG2_SCFG_MASK, iface);
+       ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG1,
+                                           TAS2764_TDM_CFG1_MASK,
+                                           (tdm_rx_start_slot << TAS2764_TDM_CFG1_51_SHIFT));
        if (ret < 0)
                return ret;
 
@@ -501,8 +509,10 @@ static int tas2764_codec_probe(struct snd_soc_component *component)
 
        tas2764->component = component;
 
-       if (tas2764->sdz_gpio)
+       if (tas2764->sdz_gpio) {
                gpiod_set_value_cansleep(tas2764->sdz_gpio, 1);
+               usleep_range(1000, 2000);
+       }
 
        tas2764_reset(tas2764);
 
@@ -526,12 +536,12 @@ static int tas2764_codec_probe(struct snd_soc_component *component)
 }
 
 static DECLARE_TLV_DB_SCALE(tas2764_digital_tlv, 1100, 50, 0);
-static DECLARE_TLV_DB_SCALE(tas2764_playback_volume, -10000, 50, 0);
+static DECLARE_TLV_DB_SCALE(tas2764_playback_volume, -10050, 50, 1);
 
 static const struct snd_kcontrol_new tas2764_snd_controls[] = {
        SOC_SINGLE_TLV("Speaker Volume", TAS2764_DVC, 0,
                       TAS2764_DVC_MAX, 1, tas2764_playback_volume),
-       SOC_SINGLE_TLV("Amp Gain Volume", TAS2764_CHNL_0, 0, 0x14, 0,
+       SOC_SINGLE_TLV("Amp Gain Volume", TAS2764_CHNL_0, 1, 0x14, 0,
                       tas2764_digital_tlv),
 };
 
@@ -556,7 +566,7 @@ static const struct reg_default tas2764_reg_defaults[] = {
        { TAS2764_SW_RST, 0x00 },
        { TAS2764_PWR_CTRL, 0x1a },
        { TAS2764_DVC, 0x00 },
-       { TAS2764_CHNL_0, 0x00 },
+       { TAS2764_CHNL_0, 0x28 },
        { TAS2764_TDM_CFG0, 0x09 },
        { TAS2764_TDM_CFG1, 0x02 },
        { TAS2764_TDM_CFG2, 0x0a },

diff --git a/sound/soc/codecs/tas2764.h b/sound/soc/codecs/tas2764.h
index 67d6fd9..f015f22 100644 (file)
--- a/sound/soc/codecs/tas2764.h
+++ b/sound/soc/codecs/tas2764.h
@@ -47,6 +47,7 @@
 #define TAS2764_TDM_CFG0_MASK          GENMASK(3, 1)
 #define TAS2764_TDM_CFG0_44_1_48KHZ    BIT(3)
 #define TAS2764_TDM_CFG0_88_2_96KHZ    (BIT(3) | BIT(1))
+#define TAS2764_TDM_CFG0_FRAME_START   BIT(0)
 
 /* TDM Configuration Reg1 */
 #define TAS2764_TDM_CFG1               TAS2764_REG(0X0, 0x09)
@@ -66,10 +67,7 @@
 #define TAS2764_TDM_CFG2_RXS_16BITS    0x0
 #define TAS2764_TDM_CFG2_RXS_24BITS    BIT(0)
 #define TAS2764_TDM_CFG2_RXS_32BITS    BIT(1)
-#define TAS2764_TDM_CFG2_SCFG_MASK     GENMASK(5, 4)
-#define TAS2764_TDM_CFG2_SCFG_I2S      0x0
-#define TAS2764_TDM_CFG2_SCFG_LEFT_J   BIT(4)
-#define TAS2764_TDM_CFG2_SCFG_RIGHT_J  BIT(5)
+#define TAS2764_TDM_CFG2_SCFG_SHIFT    4
 
 /* TDM Configuration Reg3 */
 #define TAS2764_TDM_CFG3               TAS2764_REG(0X0, 0x0c)

diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c
index b55f0b8..0b72965 100644 (file)
--- a/sound/soc/codecs/tlv320adcx140.c
+++ b/sound/soc/codecs/tlv320adcx140.c
@@ -33,7 +33,6 @@ struct adcx140_priv {
        bool micbias_vg;
 
        unsigned int dai_fmt;
-       unsigned int tdm_delay;
        unsigned int slot_width;
 };
 
@@ -792,12 +791,13 @@ static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai,
 {
        struct snd_soc_component *component = codec_dai->component;
        struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component);
-       unsigned int lsb;
 
-       /* TDM based on DSP mode requires slots to be adjacent */
-       lsb = __ffs(tx_mask);
-       if ((lsb + 1) != __fls(tx_mask)) {
-               dev_err(component->dev, "Invalid mask, slots must be adjacent\n");
+       /*
+        * The chip itself supports arbitrary masks, but the driver currently
+        * only supports adjacent slots beginning at the first slot.
+        */
+       if (tx_mask != GENMASK(__fls(tx_mask), 0)) {
+               dev_err(component->dev, "Only lower adjacent slots are supported\n");
                return -EINVAL;
        }
 
@@ -812,7 +812,6 @@ static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai,
                return -EINVAL;
        }
 
-       adcx140->tdm_delay = lsb;
        adcx140->slot_width = slot_width;
 
        return 0;

diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c
index d9f1352..3cb7a3e 100644 (file)
--- a/sound/soc/codecs/wcd9335.c
+++ b/sound/soc/codecs/wcd9335.c
@@ -342,7 +342,7 @@ struct wcd9335_codec {
        struct regulator_bulk_data supplies[WCD9335_MAX_SUPPLY];
 
        unsigned int rx_port_value[WCD9335_RX_MAX];
-       unsigned int tx_port_value;
+       unsigned int tx_port_value[WCD9335_TX_MAX];
        int hph_l_gain;
        int hph_r_gain;
        u32 rx_bias_count;
@@ -1334,8 +1334,13 @@ static int slim_tx_mixer_get(struct snd_kcontrol *kc,
 
        struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kc);
        struct wcd9335_codec *wcd = dev_get_drvdata(dapm->dev);
+       struct snd_soc_dapm_widget *widget = snd_soc_dapm_kcontrol_widget(kc);
+       struct soc_mixer_control *mixer =
+                       (struct soc_mixer_control *)kc->private_value;
+       int dai_id = widget->shift;
+       int port_id = mixer->shift;
 
-       ucontrol->value.integer.value[0] = wcd->tx_port_value;
+       ucontrol->value.integer.value[0] = wcd->tx_port_value[port_id] == dai_id;
 
        return 0;
 }
@@ -1358,12 +1363,12 @@ static int slim_tx_mixer_put(struct snd_kcontrol *kc,
        case AIF2_CAP:
        case AIF3_CAP:
                /* only add to the list if value not set */
-               if (enable && !(wcd->tx_port_value & BIT(port_id))) {
-                       wcd->tx_port_value |= BIT(port_id);
+               if (enable && wcd->tx_port_value[port_id] != dai_id) {
+                       wcd->tx_port_value[port_id] = dai_id;
                        list_add_tail(&wcd->tx_chs[port_id].list,
                                        &wcd->dai[dai_id].slim_ch_list);
-               } else if (!enable && (wcd->tx_port_value & BIT(port_id))) {
-                       wcd->tx_port_value &= ~BIT(port_id);
+               } else if (!enable && wcd->tx_port_value[port_id] == dai_id) {
+                       wcd->tx_port_value[port_id] = -1;
                        list_del_init(&wcd->tx_chs[port_id].list);
                }
                break;

diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index da2f899..b034df4 100644 (file)
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -680,12 +680,17 @@ static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol,
 {
        struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
        struct arizona *arizona = dev_get_drvdata(component->dev->parent);
+       uint16_t dac_comp_coeff = get_unaligned_be16(ucontrol->value.bytes.data);
+       int ret = 0;
 
        mutex_lock(&arizona->dac_comp_lock);
-       arizona->dac_comp_coeff = get_unaligned_be16(ucontrol->value.bytes.data);
+       if (arizona->dac_comp_coeff != dac_comp_coeff) {
+               arizona->dac_comp_coeff = dac_comp_coeff;
+               ret = 1;
+       }
        mutex_unlock(&arizona->dac_comp_lock);
 
-       return 0;
+       return ret;
 }
 
 static int wm5102_out_comp_switch_get(struct snd_kcontrol *kcontrol,
@@ -706,12 +711,20 @@ static int wm5102_out_comp_switch_put(struct snd_kcontrol *kcontrol,
 {
        struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
        struct arizona *arizona = dev_get_drvdata(component->dev->parent);
+       struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value;
+       int ret = 0;
+
+       if (ucontrol->value.integer.value[0] > mc->max)
+               return -EINVAL;
 
        mutex_lock(&arizona->dac_comp_lock);
-       arizona->dac_comp_enabled = ucontrol->value.integer.value[0];
+       if (arizona->dac_comp_enabled != ucontrol->value.integer.value[0]) {
+               arizona->dac_comp_enabled = ucontrol->value.integer.value[0];
+               ret = 1;
+       }
        mutex_unlock(&arizona->dac_comp_lock);
 
-       return 0;
+       return ret;
 }
 
 static const char * const wm5102_osr_text[] = {

diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c
index 00b59fc..ab54811 100644 (file)
--- a/sound/soc/codecs/wm8998.c
+++ b/sound/soc/codecs/wm8998.c
@@ -108,6 +108,7 @@ static int wm8998_inmux_put(struct snd_kcontrol *kcontrol,
        struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
        unsigned int mode_reg, mode_index;
        unsigned int mux, inmode, src_val, mode_val;
+       int change, ret;
 
        mux = ucontrol->value.enumerated.item[0];
        if (mux > 1)
@@ -137,14 +138,20 @@ static int wm8998_inmux_put(struct snd_kcontrol *kcontrol,
        snd_soc_component_update_bits(component, mode_reg,
                                      ARIZONA_IN1_MODE_MASK, mode_val);
 
-       snd_soc_component_update_bits(component, e->reg,
-                                     ARIZONA_IN1L_SRC_MASK |
-                                     ARIZONA_IN1L_SRC_SE_MASK,
-                                     src_val);
+       change = snd_soc_component_update_bits(component, e->reg,
+                                              ARIZONA_IN1L_SRC_MASK |
+                                              ARIZONA_IN1L_SRC_SE_MASK,
+                                              src_val);
 
-       return snd_soc_dapm_mux_update_power(dapm, kcontrol,
-                                            ucontrol->value.enumerated.item[0],
-                                            e, NULL);
+       ret = snd_soc_dapm_mux_update_power(dapm, kcontrol,
+                                           ucontrol->value.enumerated.item[0],
+                                           e, NULL);
+       if (ret < 0) {
+               dev_err(arizona->dev, "Failed to update demux power state: %d\n", ret);
+               return ret;
+       }
+
+       return change;
 }
 
 static const char * const wm8998_inmux_texts[] = {

diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c
index 77ac405..d34b29a 100644 (file)
--- a/sound/soc/generic/audio-graph-card2.c
+++ b/sound/soc/generic/audio-graph-card2.c
@@ -90,12 +90,12 @@ links indicates connection part of CPU side (= A).
                        ports@0 {
 (X) (A)                        mcpu:   port@0 { mcpu0_ep: endpoint { remote-endpoint = <&mcodec0_ep>; }; };
 (y)                            port@1 { mcpu1_ep: endpoint { remote-endpoint = <&cpu1_ep>; }; };
-(y)                            port@1 { mcpu2_ep: endpoint { remote-endpoint = <&cpu2_ep>; }; };
+(y)                            port@2 { mcpu2_ep: endpoint { remote-endpoint = <&cpu2_ep>; }; };
                        };
                        ports@1 {
 (X)                            port@0 { mcodec0_ep: endpoint { remote-endpoint = <&mcpu0_ep>; }; };
-(y)                            port@0 { mcodec1_ep: endpoint { remote-endpoint = <&codec1_ep>; }; };
-(y)                            port@1 { mcodec2_ep: endpoint { remote-endpoint = <&codec2_ep>; }; };
+(y)                            port@1 { mcodec1_ep: endpoint { remote-endpoint = <&codec1_ep>; }; };
+(y)                            port@2 { mcodec2_ep: endpoint { remote-endpoint = <&codec2_ep>; }; };
                        };
                };
        };

diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index 5d67a2c..4a90a0a 100644 (file)
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -69,11 +69,10 @@ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN |
 
 static int is_legacy_cpu;
 
-static struct snd_soc_jack sof_hdmi[3];
-
 struct sof_hdmi_pcm {
        struct list_head head;
        struct snd_soc_dai *codec_dai;
+       struct snd_soc_jack hdmi_jack;
        int device;
 };
 
@@ -434,7 +433,6 @@ static int sof_card_late_probe(struct snd_soc_card *card)
        char jack_name[NAME_SIZE];
        struct sof_hdmi_pcm *pcm;
        int err;
-       int i = 0;
 
        /* HDMI is not supported by SOF on Baytrail/CherryTrail */
        if (is_legacy_cpu || !ctx->idisp_codec)
@@ -455,17 +453,15 @@ static int sof_card_late_probe(struct snd_soc_card *card)
                snprintf(jack_name, sizeof(jack_name),
                         "HDMI/DP, pcm=%d Jack", pcm->device);
                err = snd_soc_card_jack_new(card, jack_name,
-                                           SND_JACK_AVOUT, &sof_hdmi[i]);
+                                           SND_JACK_AVOUT, &pcm->hdmi_jack);
 
                if (err)
                        return err;
 
                err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device,
-                                         &sof_hdmi[i]);
+                                         &pcm->hdmi_jack);
                if (err < 0)
                        return err;
-
-               i++;
        }
 
        if (sof_rt5682_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) {

diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c
index 2439a57..deb7b82 100644 (file)
--- a/sound/soc/intel/skylake/skl-nhlt.c
+++ b/sound/soc/intel/skylake/skl-nhlt.c
@@ -99,7 +99,6 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
        struct nhlt_fmt_cfg *fmt_cfg;
        struct wav_fmt_ext *wav_fmt;
        unsigned long rate;
-       bool present = false;
        int rate_index = 0;
        u16 channels, bps;
        u8 clk_src;
@@ -112,9 +111,12 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
        if (fmt->fmt_count == 0)
                return;
 
+       fmt_cfg = (struct nhlt_fmt_cfg *)fmt->fmt_config;
        for (i = 0; i < fmt->fmt_count; i++) {
-               fmt_cfg = &fmt->fmt_config[i];
-               wav_fmt = &fmt_cfg->fmt_ext;
+               struct nhlt_fmt_cfg *saved_fmt_cfg = fmt_cfg;
+               bool present = false;
+
+               wav_fmt = &saved_fmt_cfg->fmt_ext;
 
                channels = wav_fmt->fmt.channels;
                bps = wav_fmt->fmt.bits_per_sample;
@@ -132,12 +134,18 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
                 * derive the rate.
                 */
                for (j = i; j < fmt->fmt_count; j++) {
-                       fmt_cfg = &fmt->fmt_config[j];
-                       wav_fmt = &fmt_cfg->fmt_ext;
+                       struct nhlt_fmt_cfg *tmp_fmt_cfg = fmt_cfg;
+
+                       wav_fmt = &tmp_fmt_cfg->fmt_ext;
                        if ((fs == wav_fmt->fmt.samples_per_sec) &&
-                          (bps == wav_fmt->fmt.bits_per_sample))
+                          (bps == wav_fmt->fmt.bits_per_sample)) {
                                channels = max_t(u16, channels,
                                                wav_fmt->fmt.channels);
+                               saved_fmt_cfg = tmp_fmt_cfg;
+                       }
+                       /* Move to the next nhlt_fmt_cfg */
+                       tmp_fmt_cfg = (struct nhlt_fmt_cfg *)(tmp_fmt_cfg->config.caps +
+                                                             tmp_fmt_cfg->config.size);
                }
 
                rate = channels * bps * fs;
@@ -153,8 +161,11 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
 
                /* Fill rate and parent for sclk/sclkfs */
                if (!present) {
+                       struct nhlt_fmt_cfg *first_fmt_cfg;
+
+                       first_fmt_cfg = (struct nhlt_fmt_cfg *)fmt->fmt_config;
                        i2s_config_ext = (struct skl_i2s_config_blob_ext *)
-                                               fmt->fmt_config[0].config.caps;
+                                               first_fmt_cfg->config.caps;
 
                        /* MCLK Divider Source Select */
                        if (is_legacy_blob(i2s_config_ext->hdr.sig)) {
@@ -168,6 +179,9 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
 
                        parent = skl_get_parent_clk(clk_src);
 
+                       /* Move to the next nhlt_fmt_cfg */
+                       fmt_cfg = (struct nhlt_fmt_cfg *)(fmt_cfg->config.caps +
+                                                         fmt_cfg->config.size);
                        /*
                         * Do not copy the config data if there is no parent
                         * clock available for this clock source select
@@ -176,9 +190,9 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
                                continue;
 
                        sclk[id].rate_cfg[rate_index].rate = rate;
-                       sclk[id].rate_cfg[rate_index].config = fmt_cfg;
+                       sclk[id].rate_cfg[rate_index].config = saved_fmt_cfg;
                        sclkfs[id].rate_cfg[rate_index].rate = rate;
-                       sclkfs[id].rate_cfg[rate_index].config = fmt_cfg;
+                       sclkfs[id].rate_cfg[rate_index].config = saved_fmt_cfg;
                        sclk[id].parent_name = parent->name;
                        sclkfs[id].parent_name = parent->name;
 
@@ -192,13 +206,13 @@ static void skl_get_mclk(struct skl_dev *skl, struct skl_ssp_clk *mclk,
 {
        struct skl_i2s_config_blob_ext *i2s_config_ext;
        struct skl_i2s_config_blob_legacy *i2s_config;
-       struct nhlt_specific_cfg *fmt_cfg;
+       struct nhlt_fmt_cfg *fmt_cfg;
        struct skl_clk_parent_src *parent;
        u32 clkdiv, div_ratio;
        u8 clk_src;
 
-       fmt_cfg = &fmt->fmt_config[0].config;
-       i2s_config_ext = (struct skl_i2s_config_blob_ext *)fmt_cfg->caps;
+       fmt_cfg = (struct nhlt_fmt_cfg *)fmt->fmt_config;
+       i2s_config_ext = (struct skl_i2s_config_blob_ext *)fmt_cfg->config.caps;
 
        /* MCLK Divider Source Select and divider */
        if (is_legacy_blob(i2s_config_ext->hdr.sig)) {
@@ -227,7 +241,7 @@ static void skl_get_mclk(struct skl_dev *skl, struct skl_ssp_clk *mclk,
                return;
 
        mclk[id].rate_cfg[0].rate = parent->rate/div_ratio;
-       mclk[id].rate_cfg[0].config = &fmt->fmt_config[0];
+       mclk[id].rate_cfg[0].config = fmt_cfg;
        mclk[id].parent_name = parent->name;
 }
 

diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c
index f424d7a..7940192 100644 (file)
--- a/sound/soc/qcom/qdsp6/q6apm.c
+++ b/sound/soc/qcom/qdsp6/q6apm.c
@@ -75,6 +75,7 @@ static struct audioreach_graph *q6apm_get_audioreach_graph(struct q6apm *apm, ui
        id = idr_alloc(&apm->graph_idr, graph, graph_id, graph_id + 1, GFP_KERNEL);
        if (id < 0) {
                dev_err(apm->dev, "Unable to allocate graph id (%d)\n", graph_id);
+               kfree(graph->graph);
                kfree(graph);
                mutex_unlock(&apm->lock);
                return ERR_PTR(id);

diff --git a/sound/soc/ti/omap-mcbsp-priv.h b/sound/soc/ti/omap-mcbsp-priv.h
index 7865cda..da519ea 100644 (file)
--- a/sound/soc/ti/omap-mcbsp-priv.h
+++ b/sound/soc/ti/omap-mcbsp-priv.h
@@ -316,8 +316,6 @@ static inline int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg,
 
 /* Sidetone specific API */
 int omap_mcbsp_st_init(struct platform_device *pdev);
-void omap_mcbsp_st_cleanup(struct platform_device *pdev);
-
 int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp);
 int omap_mcbsp_st_stop(struct omap_mcbsp *mcbsp);
 

diff --git a/sound/soc/ti/omap-mcbsp-st.c b/sound/soc/ti/omap-mcbsp-st.c
index 0bc7d26..7e8179c 100644 (file)
--- a/sound/soc/ti/omap-mcbsp-st.c
+++ b/sound/soc/ti/omap-mcbsp-st.c
@@ -347,7 +347,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev)
        if (!st_data)
                return -ENOMEM;
 
-       st_data->mcbsp_iclk = clk_get(mcbsp->dev, "ick");
+       st_data->mcbsp_iclk = devm_clk_get(mcbsp->dev, "ick");
        if (IS_ERR(st_data->mcbsp_iclk)) {
                dev_warn(mcbsp->dev,
                         "Failed to get ick, sidetone might be broken\n");
@@ -359,7 +359,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev)
        if (!st_data->io_base_st)
                return -ENOMEM;
 
-       ret = sysfs_create_group(&mcbsp->dev->kobj, &sidetone_attr_group);
+       ret = devm_device_add_group(mcbsp->dev, &sidetone_attr_group);
        if (ret)
                return ret;
 
@@ -368,16 +368,6 @@ int omap_mcbsp_st_init(struct platform_device *pdev)
        return 0;
 }
 
-void omap_mcbsp_st_cleanup(struct platform_device *pdev)
-{
-       struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev);
-
-       if (mcbsp->st_data) {
-               sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group);
-               clk_put(mcbsp->st_data->mcbsp_iclk);
-       }
-}
-
 static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol,
                                    struct snd_ctl_elem_info *uinfo)
 {

diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c
index 4479d74..9933b33 100644 (file)
--- a/sound/soc/ti/omap-mcbsp.c
+++ b/sound/soc/ti/omap-mcbsp.c
@@ -702,8 +702,7 @@ static int omap_mcbsp_init(struct platform_device *pdev)
                mcbsp->max_tx_thres = max_thres(mcbsp) - 0x10;
                mcbsp->max_rx_thres = max_thres(mcbsp) - 0x10;
 
-               ret = sysfs_create_group(&mcbsp->dev->kobj,
-                                        &additional_attr_group);
+               ret = devm_device_add_group(mcbsp->dev, &additional_attr_group);
                if (ret) {
                        dev_err(mcbsp->dev,
                                "Unable to create additional controls\n");
@@ -711,16 +710,7 @@ static int omap_mcbsp_init(struct platform_device *pdev)
                }
        }
 
-       ret = omap_mcbsp_st_init(pdev);
-       if (ret)
-               goto err_st;
-
-       return 0;
-
-err_st:
-       if (mcbsp->pdata->buffer_size)
-               sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group);
-       return ret;
+       return omap_mcbsp_st_init(pdev);
 }
 
 /*
@@ -1431,11 +1421,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev)
        if (cpu_latency_qos_request_active(&mcbsp->pm_qos_req))
                cpu_latency_qos_remove_request(&mcbsp->pm_qos_req);
 
-       if (mcbsp->pdata->buffer_size)
-               sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group);
-
-       omap_mcbsp_st_cleanup(pdev);
-
        return 0;
 }