Merge tag 'sound-5.13-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai...

Pull sound fixes from Takashi Iwai:
 "A bit more commits than expected at this time, but likely it's the
  last shot before the final.

  Many of changes are device-specific fix-ups for various ASoC drivers,
  while a few usual HD-audio quirks and a FireWire fix, as well as a
  couple of ALSA / ASoC core fixes.

  All look nice and small, and nothing to scare much"

* tag 'sound-5.13-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: seq: Fix race of snd_seq_timer_open()
  ALSA: hda/realtek: fix mute/micmute LEDs for HP ZBook Power G8
  ALSA: hda/realtek: headphone and mic don't work on an Acer laptop
  ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin
  ALSA: firewire-lib: fix the context to call snd_pcm_stop_xrun()
  ALSA: hda/realtek: fix mute/micmute LEDs for HP EliteBook 840 Aero G8
  ALSA: hda/realtek: fix mute/micmute LEDs and speaker for HP EliteBook x360 1040 G8
  ALSA: hda/realtek: fix mute/micmute LEDs and speaker for HP Elite Dragonfly G2
  ASoC: rt5682: Fix the fast discharge for headset unplugging in soundwire mode
  ASoC: tas2562: Fix TDM_CFG0_SAMPRATE values
  ASoC: meson: gx-card: fix sound-dai dt schema
  ASoC: AMD Renoir: Remove fix for DMI entry on Lenovo 2020 platforms
  ASoC: AMD Renoir - add DMI entry for Lenovo 2020 AMD platforms
  ASoC: SOF: reset enabled_cores state at suspend
  ASoC: fsl-asoc-card: Set .owner attribute when registering card.
  ASoC: topology: Fix spelling mistake "vesion" -> "version"
  ASoC: rt5659: Fix the lost powers for the HDA header
  ASoC: core: Fix Null-point-dereference in fmt_single_name()

diff --git a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml
index db61f07..2e35aea 100644 (file)
--- a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml
+++ b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml
@@ -57,7 +57,7 @@ patternProperties:
           rate
 
       sound-dai:
-        $ref: /schemas/types.yaml#/definitions/phandle
+        $ref: /schemas/types.yaml#/definitions/phandle-array
         description: phandle of the CPU DAI
 
     patternProperties:
@@ -71,7 +71,7 @@ patternProperties:
 
         properties:
           sound-dai:
-            $ref: /schemas/types.yaml#/definitions/phandle
+            $ref: /schemas/types.yaml#/definitions/phandle-array
             description: phandle of the codec DAI
 
         required:

diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c
index 1645e41..9863be6 100644 (file)
--- a/sound/core/seq/seq_timer.c
+++ b/sound/core/seq/seq_timer.c
@@ -297,8 +297,16 @@ int snd_seq_timer_open(struct snd_seq_queue *q)
                return err;
        }
        spin_lock_irq(&tmr->lock);
-       tmr->timeri = t;
+       if (tmr->timeri)
+               err = -EBUSY;
+       else
+               tmr->timeri = t;
        spin_unlock_irq(&tmr->lock);
+       if (err < 0) {
+               snd_timer_close(t);
+               snd_timer_instance_free(t);
+               return err;
+       }
        return 0;
 }
 

diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index e0faa66..5805c5d 100644 (file)
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -804,7 +804,7 @@ static void generate_pkt_descs(struct amdtp_stream *s, struct pkt_desc *descs,
 static inline void cancel_stream(struct amdtp_stream *s)
 {
        s->packet_index = -1;
-       if (current_work() == &s->period_work)
+       if (in_interrupt())
                amdtp_stream_pcm_abort(s);
        WRITE_ONCE(s->pcm_buffer_pointer, SNDRV_PCM_POS_XRUN);
 }

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 43e3714..ab5113c 100644 (file)
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6568,6 +6568,7 @@ enum {
        ALC285_FIXUP_HP_SPECTRE_X360,
        ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP,
        ALC623_FIXUP_LENOVO_THINKSTATION_P340,
+       ALC255_FIXUP_ACER_HEADPHONE_AND_MIC,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -8146,6 +8147,15 @@ static const struct hda_fixup alc269_fixups[] = {
                .chained = true,
                .chain_id = ALC283_FIXUP_HEADSET_MIC,
        },
+       [ALC255_FIXUP_ACER_HEADPHONE_AND_MIC] = {
+               .type = HDA_FIXUP_PINS,
+               .v.pins = (const struct hda_pintbl[]) {
+                       { 0x21, 0x03211030 }, /* Change the Headphone location to Left */
+                       { }
+               },
+               .chained = true,
+               .chain_id = ALC255_FIXUP_XIAOMI_HEADSET_MIC
+       },
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -8182,6 +8192,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
        SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC),
        SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC),
        SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE),
+       SND_PCI_QUIRK(0x1025, 0x1466, "Acer Aspire A515-56", ALC255_FIXUP_ACER_HEADPHONE_AND_MIC),
        SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
        SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS),
        SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X),
@@ -8310,6 +8321,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
        SND_PCI_QUIRK(0x103c, 0x8519, "HP Spectre x360 15-df0xxx", ALC285_FIXUP_HP_SPECTRE_X360),
        SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED),
        SND_PCI_QUIRK(0x103c, 0x86c7, "HP Envy AiO 32", ALC274_FIXUP_HP_ENVY_GPIO),
+       SND_PCI_QUIRK(0x103c, 0x8716, "HP Elite Dragonfly G2 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT),
+       SND_PCI_QUIRK(0x103c, 0x8720, "HP EliteBook x360 1040 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT),
        SND_PCI_QUIRK(0x103c, 0x8724, "HP EliteBook 850 G7", ALC285_FIXUP_HP_GPIO_LED),
        SND_PCI_QUIRK(0x103c, 0x8729, "HP", ALC285_FIXUP_HP_GPIO_LED),
        SND_PCI_QUIRK(0x103c, 0x8730, "HP ProBook 445 G7", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
@@ -8328,10 +8341,12 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
        SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED),
        SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP),
        SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED),
+       SND_PCI_QUIRK(0x103c, 0x884b, "HP EliteBook 840 Aero G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED),
        SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED),
        SND_PCI_QUIRK(0x103c, 0x886d, "HP ZBook Fury 17.3 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT),
        SND_PCI_QUIRK(0x103c, 0x8870, "HP ZBook Fury 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT),
        SND_PCI_QUIRK(0x103c, 0x8873, "HP ZBook Studio 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT),
+       SND_PCI_QUIRK(0x103c, 0x888d, "HP ZBook Power 15.6 inch G8 Mobile Workstation PC", ALC236_FIXUP_HP_GPIO_LED),
        SND_PCI_QUIRK(0x103c, 0x8896, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_MUTE_LED),
        SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST),
        SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
@@ -8737,6 +8752,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
        {.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"},
        {.id = ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, .name = "alc287-ideapad-bass-spk-amp"},
        {.id = ALC623_FIXUP_LENOVO_THINKSTATION_P340, .name = "alc623-lenovo-thinkstation-p340"},
+       {.id = ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, .name = "alc255-acer-headphone-and-mic"},
        {}
 };
 #define ALC225_STANDARD_PINS \

diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c
index 87f5709..4a50b16 100644 (file)
--- a/sound/soc/codecs/rt5659.c
+++ b/sound/soc/codecs/rt5659.c
@@ -2433,13 +2433,18 @@ static int set_dmic_power(struct snd_soc_dapm_widget *w,
        return 0;
 }
 
-static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = {
+static const struct snd_soc_dapm_widget rt5659_particular_dapm_widgets[] = {
        SND_SOC_DAPM_SUPPLY("LDO2", RT5659_PWR_ANLG_3, RT5659_PWR_LDO2_BIT, 0,
                NULL, 0),
-       SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0,
-               NULL, 0),
+       SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT,
+               0, NULL, 0),
        SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5659_PWR_VOL,
                RT5659_PWR_MIC_DET_BIT, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = {
+       SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0,
+               NULL, 0),
        SND_SOC_DAPM_SUPPLY("Mono Vref", RT5659_PWR_ANLG_1,
                RT5659_PWR_VREF3_BIT, 0, NULL, 0),
 
@@ -2464,8 +2469,6 @@ static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = {
                RT5659_ADC_MONO_R_ASRC_SFT, 0, NULL, 0),
 
        /* Input Side */
-       SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT,
-               0, NULL, 0),
        SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5659_PWR_ANLG_2, RT5659_PWR_MB2_BIT,
                0, NULL, 0),
        SND_SOC_DAPM_SUPPLY("MICBIAS3", RT5659_PWR_ANLG_2, RT5659_PWR_MB3_BIT,
@@ -3660,10 +3663,23 @@ static int rt5659_set_bias_level(struct snd_soc_component *component,
 
 static int rt5659_probe(struct snd_soc_component *component)
 {
+       struct snd_soc_dapm_context *dapm =
+               snd_soc_component_get_dapm(component);
        struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component);
 
        rt5659->component = component;
 
+       switch (rt5659->pdata.jd_src) {
+       case RT5659_JD_HDA_HEADER:
+               break;
+
+       default:
+               snd_soc_dapm_new_controls(dapm,
+                       rt5659_particular_dapm_widgets,
+                       ARRAY_SIZE(rt5659_particular_dapm_widgets));
+               break;
+       }
+
        return 0;
 }
 

diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c
index fed80c8..e78ba3b 100644 (file)
--- a/sound/soc/codecs/rt5682-sdw.c
+++ b/sound/soc/codecs/rt5682-sdw.c
@@ -462,7 +462,8 @@ static int rt5682_io_init(struct device *dev, struct sdw_slave *slave)
 
        regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_2,
                RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL);
-       regmap_write(rt5682->regmap, RT5682_CBJ_CTRL_1, 0xd042);
+       regmap_write(rt5682->regmap, RT5682_CBJ_CTRL_1, 0xd142);
+       regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_5, 0x0700, 0x0600);
        regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_3,
                RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN);
        regmap_update_bits(rt5682->regmap, RT5682_SAR_IL_CMD_1,

diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h
index 81866ae..55b2a1f 100644 (file)
--- a/sound/soc/codecs/tas2562.h
+++ b/sound/soc/codecs/tas2562.h
@@ -57,13 +57,13 @@
 #define TAS2562_TDM_CFG0_RAMPRATE_MASK         BIT(5)
 #define TAS2562_TDM_CFG0_RAMPRATE_44_1         BIT(5)
 #define TAS2562_TDM_CFG0_SAMPRATE_MASK         GENMASK(3, 1)
-#define TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ    0x0
-#define TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ   0x1
-#define TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ  0x2
-#define TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ   0x3
-#define TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ   0x4
-#define TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ   0x5
-#define TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ 0x6
+#define TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ    (0x0 << 1)
+#define TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ   (0x1 << 1)
+#define TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ  (0x2 << 1)
+#define TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ   (0x3 << 1)
+#define TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ   (0x4 << 1)
+#define TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ   (0x5 << 1)
+#define TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ (0x6 << 1)
 
 #define TAS2562_TDM_CFG2_RIGHT_JUSTIFY BIT(6)
 

diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index c62bfd1..4f55b31 100644 (file)
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -744,6 +744,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
        /* Initialize sound card */
        priv->pdev = pdev;
        priv->card.dev = &pdev->dev;
+       priv->card.owner = THIS_MODULE;
        ret = snd_soc_of_parse_card_name(&priv->card, "model");
        if (ret) {
                snprintf(priv->name, sizeof(priv->name), "%s-audio",

diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c
index 28c7497..a6e95db 100644 (file)
--- a/sound/soc/qcom/lpass-cpu.c
+++ b/sound/soc/qcom/lpass-cpu.c
@@ -93,8 +93,30 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream,
                struct snd_soc_dai *dai)
 {
        struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai);
+       struct lpaif_i2sctl *i2sctl = drvdata->i2sctl;
+       unsigned int id = dai->driver->id;
 
        clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]);
+       /*
+        * Ensure LRCLK is disabled even in device node validation.
+        * Will not impact if disabled in lpass_cpu_daiops_trigger()
+        * suspend.
+        */
+       if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+               regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE);
+       else
+               regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_DISABLE);
+
+       /*
+        * BCLK may not be enabled if lpass_cpu_daiops_prepare is called before
+        * lpass_cpu_daiops_shutdown. It's paired with the clk_enable in
+        * lpass_cpu_daiops_prepare.
+        */
+       if (drvdata->mi2s_was_prepared[dai->driver->id]) {
+               drvdata->mi2s_was_prepared[dai->driver->id] = false;
+               clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]);
+       }
+
        clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]);
 }
 
@@ -275,6 +297,18 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream,
        case SNDRV_PCM_TRIGGER_START:
        case SNDRV_PCM_TRIGGER_RESUME:
        case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+               /*
+                * Ensure lpass BCLK/LRCLK is enabled during
+                * device resume as lpass_cpu_daiops_prepare() is not called
+                * after the device resumes. We don't check mi2s_was_prepared before
+                * enable/disable BCLK in trigger events because:
+                *  1. These trigger events are paired, so the BCLK
+                *     enable_count is balanced.
+                *  2. the BCLK can be shared (ex: headset and headset mic),
+                *     we need to increase the enable_count so that we don't
+                *     turn off the shared BCLK while other devices are using
+                *     it.
+                */
                if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
                        ret = regmap_fields_write(i2sctl->spken, id,
                                                 LPAIF_I2SCTL_SPKEN_ENABLE);
@@ -296,6 +330,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream,
        case SNDRV_PCM_TRIGGER_STOP:
        case SNDRV_PCM_TRIGGER_SUSPEND:
        case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+               /*
+                * To ensure lpass BCLK/LRCLK is disabled during
+                * device suspend.
+                */
                if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
                        ret = regmap_fields_write(i2sctl->spken, id,
                                                 LPAIF_I2SCTL_SPKEN_DISABLE);
@@ -315,12 +353,53 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream,
        return ret;
 }
 
+static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream,
+               struct snd_soc_dai *dai)
+{
+       struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai);
+       struct lpaif_i2sctl *i2sctl = drvdata->i2sctl;
+       unsigned int id = dai->driver->id;
+       int ret;
+
+       /*
+        * Ensure lpass BCLK/LRCLK is enabled bit before playback/capture
+        * data flow starts. This allows other codec to have some delay before
+        * the data flow.
+        * (ex: to drop start up pop noise before capture starts).
+        */
+       if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+               ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE);
+       else
+               ret = regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_ENABLE);
+
+       if (ret) {
+               dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret);
+               return ret;
+       }
+
+       /*
+        * Check mi2s_was_prepared before enabling BCLK as lpass_cpu_daiops_prepare can
+        * be called multiple times. It's paired with the clk_disable in
+        * lpass_cpu_daiops_shutdown.
+        */
+       if (!drvdata->mi2s_was_prepared[dai->driver->id]) {
+               ret = clk_enable(drvdata->mi2s_bit_clk[id]);
+               if (ret) {
+                       dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret);
+                       return ret;
+               }
+               drvdata->mi2s_was_prepared[dai->driver->id] = true;
+       }
+       return 0;
+}
+
 const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = {
        .set_sysclk     = lpass_cpu_daiops_set_sysclk,
        .startup        = lpass_cpu_daiops_startup,
        .shutdown       = lpass_cpu_daiops_shutdown,
        .hw_params      = lpass_cpu_daiops_hw_params,
        .trigger        = lpass_cpu_daiops_trigger,
+       .prepare        = lpass_cpu_daiops_prepare,
 };
 EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops);
 

diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h
index 83b2e08..7f72214 100644 (file)
--- a/sound/soc/qcom/lpass.h
+++ b/sound/soc/qcom/lpass.h
@@ -67,6 +67,10 @@ struct lpass_data {
        /* MI2S SD lines to use for playback/capture */
        unsigned int mi2s_playback_sd_mode[LPASS_MAX_MI2S_PORTS];
        unsigned int mi2s_capture_sd_mode[LPASS_MAX_MI2S_PORTS];
+
+       /* The state of MI2S prepare dai_ops was called */
+       bool mi2s_was_prepared[LPASS_MAX_MI2S_PORTS];
+
        int hdmi_port_enable;
 
        /* low-power audio interface (LPAIF) registers */

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 1c0904a..a76974c 100644 (file)
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2225,6 +2225,8 @@ static char *fmt_single_name(struct device *dev, int *id)
                return NULL;
 
        name = devm_kstrdup(dev, devname, GFP_KERNEL);
+       if (!name)
+               return NULL;
 
        /* are we a "%s.%d" name (platform and SPI components) */
        found = strstr(name, dev->driver->name);

diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 73076d4..4893a56 100644 (file)
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1901,7 +1901,7 @@ static void stream_caps_new_ver(struct snd_soc_tplg_stream_caps *dest,
  * @src: older version of pcm as a source
  * @pcm: latest version of pcm created from the source
  *
- * Support from vesion 4. User should free the returned pcm manually.
+ * Support from version 4. User should free the returned pcm manually.
  */
 static int pcm_new_ver(struct soc_tplg *tplg,
                       struct snd_soc_tplg_pcm *src,
@@ -2089,7 +2089,7 @@ static void set_link_hw_format(struct snd_soc_dai_link *link,
  * @src: old version of phyical link config as a source
  * @link: latest version of physical link config created from the source
  *
- * Support from vesion 4. User need free the returned link config manually.
+ * Support from version 4. User need free the returned link config manually.
  */
 static int link_new_ver(struct soc_tplg *tplg,
                        struct snd_soc_tplg_link_config *src,
@@ -2400,7 +2400,7 @@ static int soc_tplg_dai_elems_load(struct soc_tplg *tplg,
  * @src: old version of manifest as a source
  * @manifest: latest version of manifest created from the source
  *
- * Support from vesion 4. Users need free the returned manifest manually.
+ * Support from version 4. Users need free the returned manifest manually.
  */
 static int manifest_new_ver(struct soc_tplg *tplg,
                            struct snd_soc_tplg_manifest *src,

diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c
index fd26580..c83fb62 100644 (file)
--- a/sound/soc/sof/pm.c
+++ b/sound/soc/sof/pm.c
@@ -256,6 +256,7 @@ suspend:
 
        /* reset FW state */
        sdev->fw_state = SOF_FW_BOOT_NOT_STARTED;
+       sdev->enabled_cores_mask = 0;
 
        return ret;
 }