Merge tag 'sound-5.15-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai...

Pull sound fixes from Takashi Iwai:
 "This became a slightly large collection of changes, partly because
  I've been off in the last weeks. Most of changes are small and
  scattered while a bit big change is found in HD-audio Realtek codec
  driver; it's a very device-specific fix that has been long wanted, so
  I decided to pick up although it's in the middle RC.

  Some highlights:

   - A new guard ioctl for ALSA rawmidi API to avoid the misuse of the
     new timestamp framing mode; it's for a regression fix

   - HD-audio: a revert of the 5.15 change that might work badly, new
     quirks for Lenovo Legion & co, a follow-up fix for CS8409

   - ASoC: lots of SOF-related fixes, fsl component fixes, corrections
     of mediatek drivers

   - USB-audio: fix for the PM resume

   - FireWire: oxfw and motu fixes"

* tag 'sound-5.15-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (25 commits)
  ALSA: pcsp: Make hrtimer forwarding more robust
  ALSA: rawmidi: introduce SNDRV_RAWMIDI_IOCTL_USER_PVERSION
  ALSA: firewire-motu: fix truncated bytes in message tracepoints
  ASoC: SOF: trace: Omit error print when waking up trace sleepers
  ASoC: mediatek: mt8195: remove wrong fixup assignment on HDMITX
  ASoC: SOF: loader: Re-phrase the missing firmware error to avoid duplication
  ASoC: SOF: loader: release_firmware() on load failure to avoid batching
  ALSA: hda/cs8409: Setup Dolphin Headset Mic as Phantom Jack
  ALSA: pcxhr: "fix" PCXHR_REG_TO_PORT definition
  ASoC: SOF: imx: imx8m: Bar index is only valid for IRAM and SRAM types
  ASoC: SOF: imx: imx8: Bar index is only valid for IRAM and SRAM types
  ASoC: SOF: Fix DSP oops stack dump output contents
  ALSA: hda/realtek: Quirks to enable speaker output for Lenovo Legion 7i 15IMHG05, Yoga 7i 14ITL5/15ITL5, and 13s Gen2 laptops.
  ALSA: usb-audio: Unify mixer resume and reset_resume procedure
  Revert "ALSA: hda: Drop workaround for a hang at shutdown again"
  ALSA: oxfw: fix transmission method for Loud models based on OXFW971
  ASoC: mediatek: common: handle NULL case in suspend/resume function
  ASoC: fsl_xcvr: register platform component before registering cpu dai
  ASoC: fsl_spdif: register platform component before registering cpu dai
  ASoC: fsl_micfil: register platform component before registering cpu dai
  ...

diff --git a/MAINTAINERS b/MAINTAINERS
index 5b33791..bc68a13 100644 (file)
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -17883,7 +17883,8 @@ M:      Olivier Moysan <olivier.moysan@foss.st.com>
 M:     Arnaud Pouliquen <arnaud.pouliquen@foss.st.com>
 L:     alsa-devel@alsa-project.org (moderated for non-subscribers)
 S:     Maintained
-F:     Documentation/devicetree/bindings/iio/adc/st,stm32-*.yaml
+F:     Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml
+F:     Documentation/devicetree/bindings/sound/st,stm32-*.yaml
 F:     sound/soc/stm/
 
 STM32 TIMER/LPTIMER DRIVERS

diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h
index 989e151..7a08ed2 100644 (file)
--- a/include/sound/rawmidi.h
+++ b/include/sound/rawmidi.h
@@ -98,6 +98,7 @@ struct snd_rawmidi_file {
        struct snd_rawmidi *rmidi;
        struct snd_rawmidi_substream *input;
        struct snd_rawmidi_substream *output;
+       unsigned int user_pversion;     /* supported protocol version */
 };
 
 struct snd_rawmidi_str {

diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h
index 1d84ec9..5859ca0 100644 (file)
--- a/include/uapi/sound/asound.h
+++ b/include/uapi/sound/asound.h
@@ -784,6 +784,7 @@ struct snd_rawmidi_status {
 
 #define SNDRV_RAWMIDI_IOCTL_PVERSION   _IOR('W', 0x00, int)
 #define SNDRV_RAWMIDI_IOCTL_INFO       _IOR('W', 0x01, struct snd_rawmidi_info)
+#define SNDRV_RAWMIDI_IOCTL_USER_PVERSION _IOW('W', 0x02, int)
 #define SNDRV_RAWMIDI_IOCTL_PARAMS     _IOWR('W', 0x10, struct snd_rawmidi_params)
 #define SNDRV_RAWMIDI_IOCTL_STATUS     _IOWR('W', 0x20, struct snd_rawmidi_status)
 #define SNDRV_RAWMIDI_IOCTL_DROP       _IOW('W', 0x30, int)

diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index 6c0a4a6..6f30231 100644 (file)
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -873,12 +873,21 @@ static long snd_rawmidi_ioctl(struct file *file, unsigned int cmd, unsigned long
                        return -EINVAL;
                }
        }
+       case SNDRV_RAWMIDI_IOCTL_USER_PVERSION:
+               if (get_user(rfile->user_pversion, (unsigned int __user *)arg))
+                       return -EFAULT;
+               return 0;
+
        case SNDRV_RAWMIDI_IOCTL_PARAMS:
        {
                struct snd_rawmidi_params params;
 
                if (copy_from_user(&params, argp, sizeof(struct snd_rawmidi_params)))
                        return -EFAULT;
+               if (rfile->user_pversion < SNDRV_PROTOCOL_VERSION(2, 0, 2)) {
+                       params.mode = 0;
+                       memset(params.reserved, 0, sizeof(params.reserved));
+               }
                switch (params.stream) {
                case SNDRV_RAWMIDI_STREAM_OUTPUT:
                        if (rfile->output == NULL)

diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c
index ed40d0f..773db4b 100644 (file)
--- a/sound/drivers/pcsp/pcsp_lib.c
+++ b/sound/drivers/pcsp/pcsp_lib.c
@@ -143,7 +143,7 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
        if (pointer_update)
                pcsp_pointer_update(chip);
 
-       hrtimer_forward(handle, hrtimer_get_expires(handle), ns_to_ktime(ns));
+       hrtimer_forward_now(handle, ns_to_ktime(ns));
 
        return HRTIMER_RESTART;
 }

diff --git a/sound/firewire/motu/amdtp-motu.c b/sound/firewire/motu/amdtp-motu.c
index 5388b85..a18c2c0 100644 (file)
--- a/sound/firewire/motu/amdtp-motu.c
+++ b/sound/firewire/motu/amdtp-motu.c
@@ -276,10 +276,11 @@ static void __maybe_unused copy_message(u64 *frames, __be32 *buffer,
 
        /* This is just for v2/v3 protocol. */
        for (i = 0; i < data_blocks; ++i) {
-               *frames = (be32_to_cpu(buffer[1]) << 16) |
-                         (be32_to_cpu(buffer[2]) >> 16);
+               *frames = be32_to_cpu(buffer[1]);
+               *frames <<= 16;
+               *frames |= be32_to_cpu(buffer[2]) >> 16;
+               ++frames;
                buffer += data_block_quadlets;
-               frames++;
        }
 }
 

diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c
index cb5b5e3..daf7313 100644 (file)
--- a/sound/firewire/oxfw/oxfw.c
+++ b/sound/firewire/oxfw/oxfw.c
@@ -184,13 +184,16 @@ static int detect_quirks(struct snd_oxfw *oxfw, const struct ieee1394_device_id
                        model = val;
        }
 
-       /*
-        * Mackie Onyx Satellite with base station has a quirk to report a wrong
-        * value in 'dbs' field of CIP header against its format information.
-        */
-       if (vendor == VENDOR_LOUD && model == MODEL_SATELLITE)
+       if (vendor == VENDOR_LOUD) {
+               // Mackie Onyx Satellite with base station has a quirk to report a wrong
+               // value in 'dbs' field of CIP header against its format information.
                oxfw->quirks |= SND_OXFW_QUIRK_WRONG_DBS;
 
+               // OXFW971-based models may transfer events by blocking method.
+               if (!(oxfw->quirks & SND_OXFW_QUIRK_JUMBO_PAYLOAD))
+                       oxfw->quirks |= SND_OXFW_QUIRK_BLOCKING_TRANSMISSION;
+       }
+
        return 0;
 }
 

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 3aa432d..4777743 100644 (file)
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -883,10 +883,11 @@ static unsigned int azx_get_pos_skl(struct azx *chip, struct azx_dev *azx_dev)
        return azx_get_pos_posbuf(chip, azx_dev);
 }
 
-static void azx_shutdown_chip(struct azx *chip)
+static void __azx_shutdown_chip(struct azx *chip, bool skip_link_reset)
 {
        azx_stop_chip(chip);
-       azx_enter_link_reset(chip);
+       if (!skip_link_reset)
+               azx_enter_link_reset(chip);
        azx_clear_irq_pending(chip);
        display_power(chip, false);
 }
@@ -895,6 +896,11 @@ static void azx_shutdown_chip(struct azx *chip)
 static DEFINE_MUTEX(card_list_lock);
 static LIST_HEAD(card_list);
 
+static void azx_shutdown_chip(struct azx *chip)
+{
+       __azx_shutdown_chip(chip, false);
+}
+
 static void azx_add_card_list(struct azx *chip)
 {
        struct hda_intel *hda = container_of(chip, struct hda_intel, chip);
@@ -2357,7 +2363,7 @@ static void azx_shutdown(struct pci_dev *pci)
                return;
        chip = card->private_data;
        if (chip && chip->running)
-               azx_shutdown_chip(chip);
+               __azx_shutdown_chip(chip, true);
 }
 
 /* PCI IDs */

diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c
index 3c7ef55..31ff11a 100644 (file)
--- a/sound/pci/hda/patch_cs8409.c
+++ b/sound/pci/hda/patch_cs8409.c
@@ -1207,6 +1207,9 @@ void dolphin_fixups(struct hda_codec *codec, const struct hda_fixup *fix, int ac
                snd_hda_jack_add_kctl(codec, DOLPHIN_LO_PIN_NID, "Line Out", true,
                                      SND_JACK_HEADPHONE, NULL);
 
+               snd_hda_jack_add_kctl(codec, DOLPHIN_AMIC_PIN_NID, "Microphone", true,
+                                     SND_JACK_MICROPHONE, NULL);
+
                cs8409_fix_caps(codec, DOLPHIN_HP_PIN_NID);
                cs8409_fix_caps(codec, DOLPHIN_LO_PIN_NID);
                cs8409_fix_caps(codec, DOLPHIN_AMIC_PIN_NID);

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 8b7a389..4407f7d 100644 (file)
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6429,6 +6429,20 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec,
        hda_fixup_thinkpad_acpi(codec, fix, action);
 }
 
+/* Fixup for Lenovo Legion 15IMHg05 speaker output on headset removal. */
+static void alc287_fixup_legion_15imhg05_speakers(struct hda_codec *codec,
+                                                 const struct hda_fixup *fix,
+                                                 int action)
+{
+       struct alc_spec *spec = codec->spec;
+
+       switch (action) {
+       case HDA_FIXUP_ACT_PRE_PROBE:
+               spec->gen.suppress_auto_mute = 1;
+               break;
+       }
+}
+
 /* for alc295_fixup_hp_top_speakers */
 #include "hp_x360_helper.c"
 
@@ -6646,6 +6660,10 @@ enum {
        ALC623_FIXUP_LENOVO_THINKSTATION_P340,
        ALC255_FIXUP_ACER_HEADPHONE_AND_MIC,
        ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST,
+       ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS,
+       ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE,
+       ALC287_FIXUP_YOGA7_14ITL_SPEAKERS,
+       ALC287_FIXUP_13S_GEN2_SPEAKERS
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -8236,6 +8254,113 @@ static const struct hda_fixup alc269_fixups[] = {
                .chained = true,
                .chain_id = ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF,
        },
+       [ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS] = {
+               .type = HDA_FIXUP_VERBS,
+               //.v.verbs = legion_15imhg05_coefs,
+               .v.verbs = (const struct hda_verb[]) {
+                        // set left speaker Legion 7i.
+                        { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0x41 },
+
+                        { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0xc },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0x1a },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+
+                        { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0x2 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+
+                        // set right speaker Legion 7i.
+                        { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0x42 },
+
+                        { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0xc },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0x2a },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+
+                        { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0x2 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+                        {}
+               },
+               .chained = true,
+               .chain_id = ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE,
+       },
+       [ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE] = {
+               .type = HDA_FIXUP_FUNC,
+               .v.func = alc287_fixup_legion_15imhg05_speakers,
+               .chained = true,
+               .chain_id = ALC269_FIXUP_HEADSET_MODE,
+       },
+       [ALC287_FIXUP_YOGA7_14ITL_SPEAKERS] = {
+               .type = HDA_FIXUP_VERBS,
+               .v.verbs = (const struct hda_verb[]) {
+                        // set left speaker Yoga 7i.
+                        { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0x41 },
+
+                        { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0xc },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0x1a },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+
+                        { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0x2 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+
+                        // set right speaker Yoga 7i.
+                        { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0x46 },
+
+                        { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0xc },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0x2a },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+
+                        { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0x2 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+                        { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+                        {}
+               },
+               .chained = true,
+               .chain_id = ALC269_FIXUP_HEADSET_MODE,
+       },
+       [ALC287_FIXUP_13S_GEN2_SPEAKERS] = {
+               .type = HDA_FIXUP_VERBS,
+               .v.verbs = (const struct hda_verb[]) {
+                       { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 },
+                       { 0x20, AC_VERB_SET_PROC_COEF, 0x41 },
+                       { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+                       { 0x20, AC_VERB_SET_PROC_COEF, 0x2 },
+                       { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+                       { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+                       { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+                       { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 },
+                       { 0x20, AC_VERB_SET_PROC_COEF, 0x42 },
+                       { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+                       { 0x20, AC_VERB_SET_PROC_COEF, 0x2 },
+                       { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+                       { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+                       { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+                       {}
+               },
+               .chained = true,
+               .chain_id = ALC269_FIXUP_HEADSET_MODE,
+       },
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -8630,6 +8755,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
        SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME),
        SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF),
        SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP),
+       SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS),
+       SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS),
+       SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS),
+       SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS),
        SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI),
        SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC),
        SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI),

diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c
index 87d2422..23f253e 100644 (file)
--- a/sound/pci/pcxhr/pcxhr_core.c
+++ b/sound/pci/pcxhr/pcxhr_core.c
@@ -52,7 +52,7 @@
 #define PCXHR_DSP 2
 
 #if (PCXHR_DSP_OFFSET_MAX > PCXHR_PLX_OFFSET_MIN)
-#undef  PCXHR_REG_TO_PORT(x)
+#error  PCXHR_REG_TO_PORT(x)
 #else
 #define PCXHR_REG_TO_PORT(x)   ((x)>PCXHR_DSP_OFFSET_MAX ? PCXHR_PLX : PCXHR_DSP)
 #endif

diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index a961f83..bda66b3 100644 (file)
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -1073,6 +1073,16 @@ static int fsl_esai_probe(struct platform_device *pdev)
        if (ret < 0)
                goto err_pm_get_sync;
 
+       /*
+        * Register platform component before registering cpu dai for there
+        * is not defer probe for platform component in snd_soc_add_pcm_runtime().
+        */
+       ret = imx_pcm_dma_init(pdev, IMX_ESAI_DMABUF_SIZE);
+       if (ret) {
+               dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret);
+               goto err_pm_get_sync;
+       }
+
        ret = devm_snd_soc_register_component(&pdev->dev, &fsl_esai_component,
                                              &fsl_esai_dai, 1);
        if (ret) {
@@ -1082,12 +1092,6 @@ static int fsl_esai_probe(struct platform_device *pdev)
 
        INIT_WORK(&esai_priv->work, fsl_esai_hw_reset);
 
-       ret = imx_pcm_dma_init(pdev, IMX_ESAI_DMABUF_SIZE);
-       if (ret) {
-               dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret);
-               goto err_pm_get_sync;
-       }
-
        return ret;
 
 err_pm_get_sync:

diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c
index 8c0c75c..9f90989 100644 (file)
--- a/sound/soc/fsl/fsl_micfil.c
+++ b/sound/soc/fsl/fsl_micfil.c
@@ -737,18 +737,23 @@ static int fsl_micfil_probe(struct platform_device *pdev)
        pm_runtime_enable(&pdev->dev);
        regcache_cache_only(micfil->regmap, true);
 
+       /*
+        * Register platform component before registering cpu dai for there
+        * is not defer probe for platform component in snd_soc_add_pcm_runtime().
+        */
+       ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
+       if (ret) {
+               dev_err(&pdev->dev, "failed to pcm register\n");
+               return ret;
+       }
+
        ret = devm_snd_soc_register_component(&pdev->dev, &fsl_micfil_component,
                                              &fsl_micfil_dai, 1);
        if (ret) {
                dev_err(&pdev->dev, "failed to register component %s\n",
                        fsl_micfil_component.name);
-               return ret;
        }
 
-       ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
-       if (ret)
-               dev_err(&pdev->dev, "failed to pcm register\n");
-
        return ret;
 }
 

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 223fcd1..38f6362 100644 (file)
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -1152,11 +1152,10 @@ static int fsl_sai_probe(struct platform_device *pdev)
        if (ret < 0)
                goto err_pm_get_sync;
 
-       ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component,
-                                             &sai->cpu_dai_drv, 1);
-       if (ret)
-               goto err_pm_get_sync;
-
+       /*
+        * Register platform component before registering cpu dai for there
+        * is not defer probe for platform component in snd_soc_add_pcm_runtime().
+        */
        if (sai->soc_data->use_imx_pcm) {
                ret = imx_pcm_dma_init(pdev, IMX_SAI_DMABUF_SIZE);
                if (ret)
@@ -1167,6 +1166,11 @@ static int fsl_sai_probe(struct platform_device *pdev)
                        goto err_pm_get_sync;
        }
 
+       ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component,
+                                             &sai->cpu_dai_drv, 1);
+       if (ret)
+               goto err_pm_get_sync;
+
        return ret;
 
 err_pm_get_sync:

diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 8ffb1a6..1c53719 100644 (file)
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -1434,16 +1434,20 @@ static int fsl_spdif_probe(struct platform_device *pdev)
        pm_runtime_enable(&pdev->dev);
        regcache_cache_only(spdif_priv->regmap, true);
 
-       ret = devm_snd_soc_register_component(&pdev->dev, &fsl_spdif_component,
-                                             &spdif_priv->cpu_dai_drv, 1);
+       /*
+        * Register platform component before registering cpu dai for there
+        * is not defer probe for platform component in snd_soc_add_pcm_runtime().
+        */
+       ret = imx_pcm_dma_init(pdev, IMX_SPDIF_DMABUF_SIZE);
        if (ret) {
-               dev_err(&pdev->dev, "failed to register DAI: %d\n", ret);
+               dev_err_probe(&pdev->dev, ret, "imx_pcm_dma_init failed\n");
                goto err_pm_disable;
        }
 
-       ret = imx_pcm_dma_init(pdev, IMX_SPDIF_DMABUF_SIZE);
+       ret = devm_snd_soc_register_component(&pdev->dev, &fsl_spdif_component,
+                                             &spdif_priv->cpu_dai_drv, 1);
        if (ret) {
-               dev_err_probe(&pdev->dev, ret, "imx_pcm_dma_init failed\n");
+               dev_err(&pdev->dev, "failed to register DAI: %d\n", ret);
                goto err_pm_disable;
        }
 

diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c
index 31c5ee6..7ba2fd1 100644 (file)
--- a/sound/soc/fsl/fsl_xcvr.c
+++ b/sound/soc/fsl/fsl_xcvr.c
@@ -1215,18 +1215,23 @@ static int fsl_xcvr_probe(struct platform_device *pdev)
        pm_runtime_enable(dev);
        regcache_cache_only(xcvr->regmap, true);
 
+       /*
+        * Register platform component before registering cpu dai for there
+        * is not defer probe for platform component in snd_soc_add_pcm_runtime().
+        */
+       ret = devm_snd_dmaengine_pcm_register(dev, NULL, 0);
+       if (ret) {
+               dev_err(dev, "failed to pcm register\n");
+               return ret;
+       }
+
        ret = devm_snd_soc_register_component(dev, &fsl_xcvr_comp,
                                              &fsl_xcvr_dai, 1);
        if (ret) {
                dev_err(dev, "failed to register component %s\n",
                        fsl_xcvr_comp.name);
-               return ret;
        }
 
-       ret = devm_snd_dmaengine_pcm_register(dev, NULL, 0);
-       if (ret)
-               dev_err(dev, "failed to pcm register\n");
-
        return ret;
 }
 

diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 6602eda..6b06248 100644 (file)
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -929,6 +929,11 @@ static int create_sdw_dailink(struct snd_soc_card *card,
                              cpus + *cpu_id, cpu_dai_num,
                              codecs, codec_num,
                              NULL, &sdw_ops);
+               /*
+                * SoundWire DAILINKs use 'stream' functions and Bank Switch operations
+                * based on wait_for_completion(), tag them as 'nonatomic'.
+                */
+               dai_links[*be_index].nonatomic = true;
 
                ret = set_codec_init_func(card, link, dai_links + (*be_index)++,
                                          playback, group_id);

diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig
index 5a2f466..81ad2dc 100644 (file)
--- a/sound/soc/mediatek/Kconfig
+++ b/sound/soc/mediatek/Kconfig
@@ -1,6 +1,7 @@
 # SPDX-License-Identifier: GPL-2.0-only
 config SND_SOC_MEDIATEK
        tristate
+       select REGMAP_MMIO
 
 config SND_SOC_MT2701
        tristate "ASoC support for Mediatek MT2701 chip"
@@ -188,7 +189,9 @@ config SND_SOC_MT8192_MT6359_RT1015_RT5682
 config SND_SOC_MT8195
        tristate "ASoC support for Mediatek MT8195 chip"
        depends on ARCH_MEDIATEK || COMPILE_TEST
+       depends on COMMON_CLK
        select SND_SOC_MEDIATEK
+       select MFD_SYSCON if SND_SOC_MT6359
        help
          This adds ASoC platform driver support for Mediatek MT8195 chip
          that can be used with other codecs.

diff --git a/sound/soc/mediatek/common/mtk-afe-fe-dai.c b/sound/soc/mediatek/common/mtk-afe-fe-dai.c
index baaa588..e95c7c0 100644 (file)
--- a/sound/soc/mediatek/common/mtk-afe-fe-dai.c
+++ b/sound/soc/mediatek/common/mtk-afe-fe-dai.c
@@ -334,9 +334,11 @@ int mtk_afe_suspend(struct snd_soc_component *component)
                        devm_kcalloc(dev, afe->reg_back_up_list_num,
                                     sizeof(unsigned int), GFP_KERNEL);
 
-       for (i = 0; i < afe->reg_back_up_list_num; i++)
-               regmap_read(regmap, afe->reg_back_up_list[i],
-                           &afe->reg_back_up[i]);
+       if (afe->reg_back_up) {
+               for (i = 0; i < afe->reg_back_up_list_num; i++)
+                       regmap_read(regmap, afe->reg_back_up_list[i],
+                                   &afe->reg_back_up[i]);
+       }
 
        afe->suspended = true;
        afe->runtime_suspend(dev);
@@ -356,12 +358,13 @@ int mtk_afe_resume(struct snd_soc_component *component)
 
        afe->runtime_resume(dev);
 
-       if (!afe->reg_back_up)
+       if (!afe->reg_back_up) {
                dev_dbg(dev, "%s no reg_backup\n", __func__);
-
-       for (i = 0; i < afe->reg_back_up_list_num; i++)
-               mtk_regmap_write(regmap, afe->reg_back_up_list[i],
-                                afe->reg_back_up[i]);
+       } else {
+               for (i = 0; i < afe->reg_back_up_list_num; i++)
+                       mtk_regmap_write(regmap, afe->reg_back_up_list[i],
+                                        afe->reg_back_up[i]);
+       }
 
        afe->suspended = false;
        return 0;

diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c
index c97ace7..de09f67 100644 (file)
--- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c
+++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c
@@ -424,8 +424,8 @@ static int mt8195_hdmi_codec_init(struct snd_soc_pcm_runtime *rtd)
        return snd_soc_component_set_jack(cmpnt_codec, &priv->hdmi_jack, NULL);
 }
 
-static int mt8195_hdmitx_dptx_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
-                                             struct snd_pcm_hw_params *params)
+static int mt8195_dptx_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+                                      struct snd_pcm_hw_params *params)
 
 {
        /* fix BE i2s format to 32bit, clean param mask first */
@@ -902,7 +902,7 @@ static struct snd_soc_dai_link mt8195_mt6359_rt1019_rt5682_dai_links[] = {
                .no_pcm = 1,
                .dpcm_playback = 1,
                .ops = &mt8195_dptx_ops,
-               .be_hw_params_fixup = mt8195_hdmitx_dptx_hw_params_fixup,
+               .be_hw_params_fixup = mt8195_dptx_hw_params_fixup,
                SND_SOC_DAILINK_REG(DPTX_BE),
        },
        [DAI_LINK_ETDM1_IN_BE] = {
@@ -953,7 +953,6 @@ static struct snd_soc_dai_link mt8195_mt6359_rt1019_rt5682_dai_links[] = {
                        SND_SOC_DAIFMT_NB_NF |
                        SND_SOC_DAIFMT_CBS_CFS,
                .dpcm_playback = 1,
-               .be_hw_params_fixup = mt8195_hdmitx_dptx_hw_params_fixup,
                SND_SOC_DAILINK_REG(ETDM3_OUT_BE),
        },
        [DAI_LINK_PCM1_BE] = {

diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c
index 3e4dd4a..59d0d7b 100644 (file)
--- a/sound/soc/sof/core.c
+++ b/sound/soc/sof/core.c
@@ -371,7 +371,6 @@ int snd_sof_device_remove(struct device *dev)
                        dev_warn(dev, "error: %d failed to prepare DSP for device removal",
                                 ret);
 
-               snd_sof_fw_unload(sdev);
                snd_sof_ipc_free(sdev);
                snd_sof_free_debug(sdev);
                snd_sof_free_trace(sdev);
@@ -394,8 +393,7 @@ int snd_sof_device_remove(struct device *dev)
                snd_sof_remove(sdev);
 
        /* release firmware */
-       release_firmware(pdata->fw);
-       pdata->fw = NULL;
+       snd_sof_fw_unload(sdev);
 
        return 0;
 }

diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c
index 12fedf0..7e9723a 100644 (file)
--- a/sound/soc/sof/imx/imx8.c
+++ b/sound/soc/sof/imx/imx8.c
@@ -365,7 +365,14 @@ static int imx8_remove(struct snd_sof_dev *sdev)
 /* on i.MX8 there is 1 to 1 match between type and BAR idx */
 static int imx8_get_bar_index(struct snd_sof_dev *sdev, u32 type)
 {
-       return type;
+       /* Only IRAM and SRAM bars are valid */
+       switch (type) {
+       case SOF_FW_BLK_TYPE_IRAM:
+       case SOF_FW_BLK_TYPE_SRAM:
+               return type;
+       default:
+               return -EINVAL;
+       }
 }
 
 static void imx8_ipc_msg_data(struct snd_sof_dev *sdev,

diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c
index cb822d9..892e148 100644 (file)
--- a/sound/soc/sof/imx/imx8m.c
+++ b/sound/soc/sof/imx/imx8m.c
@@ -228,7 +228,14 @@ static int imx8m_remove(struct snd_sof_dev *sdev)
 /* on i.MX8 there is 1 to 1 match between type and BAR idx */
 static int imx8m_get_bar_index(struct snd_sof_dev *sdev, u32 type)
 {
-       return type;
+       /* Only IRAM and SRAM bars are valid */
+       switch (type) {
+       case SOF_FW_BLK_TYPE_IRAM:
+       case SOF_FW_BLK_TYPE_SRAM:
+               return type;
+       default:
+               return -EINVAL;
+       }
 }
 
 static void imx8m_ipc_msg_data(struct snd_sof_dev *sdev,

diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c
index 2b38a77..bb79c77 100644 (file)
--- a/sound/soc/sof/loader.c
+++ b/sound/soc/sof/loader.c
@@ -729,10 +729,10 @@ int snd_sof_load_firmware_raw(struct snd_sof_dev *sdev)
        ret = request_firmware(&plat_data->fw, fw_filename, sdev->dev);
 
        if (ret < 0) {
-               dev_err(sdev->dev, "error: request firmware %s failed err: %d\n",
-                       fw_filename, ret);
                dev_err(sdev->dev,
-                       "you may need to download the firmware from https://github.com/thesofproject/sof-bin/\n");
+                       "error: sof firmware file is missing, you might need to\n");
+               dev_err(sdev->dev,
+                       "       download it from https://github.com/thesofproject/sof-bin/\n");
                goto err;
        } else {
                dev_dbg(sdev->dev, "request_firmware %s successful\n",
@@ -880,5 +880,7 @@ EXPORT_SYMBOL(snd_sof_run_firmware);
 void snd_sof_fw_unload(struct snd_sof_dev *sdev)
 {
        /* TODO: support module unloading at runtime */
+       release_firmware(sdev->pdata->fw);
+       sdev->pdata->fw = NULL;
 }
 EXPORT_SYMBOL(snd_sof_fw_unload);

diff --git a/sound/soc/sof/trace.c b/sound/soc/sof/trace.c
index f72a6e8..58f6ca5 100644 (file)
--- a/sound/soc/sof/trace.c
+++ b/sound/soc/sof/trace.c
@@ -530,7 +530,6 @@ void snd_sof_trace_notify_for_error(struct snd_sof_dev *sdev)
                return;
 
        if (sdev->dtrace_is_enabled) {
-               dev_err(sdev->dev, "error: waking up any trace sleepers\n");
                sdev->dtrace_error = true;
                wake_up(&sdev->trace_sleep);
        }

diff --git a/sound/soc/sof/xtensa/core.c b/sound/soc/sof/xtensa/core.c
index bbb9a22..f6e3411 100644 (file)
--- a/sound/soc/sof/xtensa/core.c
+++ b/sound/soc/sof/xtensa/core.c
@@ -122,9 +122,9 @@ static void xtensa_stack(struct snd_sof_dev *sdev, void *oops, u32 *stack,
         * 0x0049fbb0: 8000f2d0 0049fc00 6f6c6c61 00632e63
         */
        for (i = 0; i < stack_words; i += 4) {
-               hex_dump_to_buffer(stack + i * 4, 16, 16, 4,
+               hex_dump_to_buffer(stack + i, 16, 16, 4,
                                   buf, sizeof(buf), false);
-               dev_err(sdev->dev, "0x%08x: %s\n", stack_ptr + i, buf);
+               dev_err(sdev->dev, "0x%08x: %s\n", stack_ptr + i * 4, buf);
        }
 }
 

diff --git a/sound/usb/card.c b/sound/usb/card.c
index fd570a4..1764b93 100644 (file)
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -1054,7 +1054,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message)
        return 0;
 }
 
-static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume)
+static int usb_audio_resume(struct usb_interface *intf)
 {
        struct snd_usb_audio *chip = usb_get_intfdata(intf);
        struct snd_usb_stream *as;
@@ -1080,7 +1080,7 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume)
         * we just notify and restart the mixers
         */
        list_for_each_entry(mixer, &chip->mixer_list, list) {
-               err = snd_usb_mixer_resume(mixer, reset_resume);
+               err = snd_usb_mixer_resume(mixer);
                if (err < 0)
                        goto err_out;
        }
@@ -1100,20 +1100,10 @@ err_out:
        atomic_dec(&chip->active); /* allow autopm after this point */
        return err;
 }
-
-static int usb_audio_resume(struct usb_interface *intf)
-{
-       return __usb_audio_resume(intf, false);
-}
-
-static int usb_audio_reset_resume(struct usb_interface *intf)
-{
-       return __usb_audio_resume(intf, true);
-}
 #else
 #define usb_audio_suspend      NULL
 #define usb_audio_resume       NULL
-#define usb_audio_reset_resume NULL
+#define usb_audio_resume       NULL
 #endif         /* CONFIG_PM */
 
 static const struct usb_device_id usb_audio_ids [] = {
@@ -1135,7 +1125,7 @@ static struct usb_driver usb_audio_driver = {
        .disconnect =   usb_audio_disconnect,
        .suspend =      usb_audio_suspend,
        .resume =       usb_audio_resume,
-       .reset_resume = usb_audio_reset_resume,
+       .reset_resume = usb_audio_resume,
        .id_table =     usb_audio_ids,
        .supports_autosuspend = 1,
 };

diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 43bc595..a2ce535 100644 (file)
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -3653,33 +3653,16 @@ static int restore_mixer_value(struct usb_mixer_elem_list *list)
        return 0;
 }
 
-static int default_mixer_reset_resume(struct usb_mixer_elem_list *list)
-{
-       int err;
-
-       if (list->resume) {
-               err = list->resume(list);
-               if (err < 0)
-                       return err;
-       }
-       return restore_mixer_value(list);
-}
-
-int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume)
+int snd_usb_mixer_resume(struct usb_mixer_interface *mixer)
 {
        struct usb_mixer_elem_list *list;
-       usb_mixer_elem_resume_func_t f;
        int id, err;
 
        /* restore cached mixer values */
        for (id = 0; id < MAX_ID_ELEMS; id++) {
                for_each_mixer_elem(list, mixer, id) {
-                       if (reset_resume)
-                               f = list->reset_resume;
-                       else
-                               f = list->resume;
-                       if (f) {
-                               err = f(list);
+                       if (list->resume) {
+                               err = list->resume(list);
                                if (err < 0)
                                        return err;
                        }
@@ -3700,7 +3683,6 @@ void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list,
        list->id = unitid;
        list->dump = snd_usb_mixer_dump_cval;
 #ifdef CONFIG_PM
-       list->resume = NULL;
-       list->reset_resume = default_mixer_reset_resume;
+       list->resume = restore_mixer_value;
 #endif
 }

diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h
index 876bbc9..98ea24d 100644 (file)
--- a/sound/usb/mixer.h
+++ b/sound/usb/mixer.h
@@ -70,7 +70,6 @@ struct usb_mixer_elem_list {
        bool is_std_info;
        usb_mixer_elem_dump_func_t dump;
        usb_mixer_elem_resume_func_t resume;
-       usb_mixer_elem_resume_func_t reset_resume;
 };
 
 /* iterate over mixer element list of the given unit id */
@@ -121,7 +120,7 @@ int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag,
 
 #ifdef CONFIG_PM
 int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer);
-int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume);
+int snd_usb_mixer_resume(struct usb_mixer_interface *mixer);
 #endif
 
 int snd_usb_set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel,

diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index a66ce03..46082dc 100644 (file)
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -151,7 +151,7 @@ static int add_single_ctl_with_resume(struct usb_mixer_interface *mixer,
                *listp = list;
        list->mixer = mixer;
        list->id = id;
-       list->reset_resume = resume;
+       list->resume = resume;
        kctl = snd_ctl_new1(knew, list);
        if (!kctl) {
                kfree(list);