Merge tag 'sound-5.7-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "This became a slightly big pull request, as the accumulated ASoC fixes
  are included here. Some highlights:

   - Revert of ASoC DAI startup changes that caused regression on some
     x86 platforms

   - Regression fix in HD-audio power management and driver blacklist

   - A collection of ASoC DAPM and topology fixes

   - Continued USB-audio fixes and quirks

   - Lots of small device-specific fixes

   - Rockchip S/PDIF DT stuff update for validation issues"

* tag 'sound-5.7-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (51 commits)
  ALSA: hda: Always use jackpoll helper for jack update after resume
  ALSA: hda/realtek - Add new codec supported for ALC245
  ALSA: usb-audio: Fix usb audio refcnt leak when getting spdif
  ALSA: usb-audio: Add connector notifier delegation
  ALSA: usb-audio: Apply async workaround for Scarlett 2i4 2nd gen
  ASoC: wm8960: Fix wrong clock after suspend & resume
  ALSA: usx2y: Fix potential NULL dereference
  ALSA: usb-audio: Add quirk for Focusrite Scarlett 2i2
  ASoC: wm89xx: Add missing dependency
  ASoC: dapm: fixup dapm kcontrol widget
  ASoC: rsnd: Fix "status check failed" spam for multi-SSI
  ASoC: rsnd: Don't treat master SSI in multi SSI setup as parent
  ASoC: meson: gx-card: fix codec-to-codec link setup
  ASoC: meson: axg-card: fix codec-to-codec link setup
  ALSA: usb-audio: Add static mapping table for ALC1220-VB-based mobos
  ALSA: hda: Remove ASUS ROG Zenith from the blacklist
  ALSA: hda/realtek - Fix unexpected init_amp override
  ALSA: usb-audio: Filter out unsupported sample rates on Focusrite devices
  ASoC: SOF: Intel: add min/max channels for SSP on Baytrail/Broadwell
  ASoC: stm32: sai: fix sai probe
  ...

4 files changed, 36 insertions, 30 deletions

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 86a632bf4d50..7e3ae4534df9 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -641,8 +641,18 @@ static void hda_jackpoll_work(struct work_struct *work)
 	struct hda_codec *codec =
 		container_of(work, struct hda_codec, jackpoll_work.work);
 
-	snd_hda_jack_set_dirty_all(codec);
-	snd_hda_jack_poll_all(codec);
+	/* for non-polling trigger: we need nothing if already powered on */
+	if (!codec->jackpoll_interval && snd_hdac_is_power_on(&codec->core))
+		return;
+
+	/* the power-up/down sequence triggers the runtime resume */
+	snd_hda_power_up_pm(codec);
+	/* update jacks manually if polling is required, too */
+	if (codec->jackpoll_interval) {
+		snd_hda_jack_set_dirty_all(codec);
+		snd_hda_jack_poll_all(codec);
+	}
+	snd_hda_power_down_pm(codec);
 
 	if (!codec->jackpoll_interval)
 		return;
@@ -2951,18 +2961,14 @@ static int hda_codec_runtime_resume(struct device *dev)
 static int hda_codec_force_resume(struct device *dev)
 {
 	struct hda_codec *codec = dev_to_hda_codec(dev);
-	bool forced_resume = hda_codec_need_resume(codec);
 	int ret;
 
-	/* The get/put pair below enforces the runtime resume even if the
-	 * device hasn't been used at suspend time.  This trick is needed to
-	 * update the jack state change during the sleep.
-	 */
-	if (forced_resume)
-		pm_runtime_get_noresume(dev);
 	ret = pm_runtime_force_resume(dev);
-	if (forced_resume)
-		pm_runtime_put(dev);
+	/* schedule jackpoll work for jack detection update */
+	if (codec->jackpoll_interval ||
+	    (pm_runtime_suspended(dev) && hda_codec_need_resume(codec)))
+		schedule_delayed_work(&codec->jackpoll_work,
+				      codec->jackpoll_interval);
 	return ret;
 }
 
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index a5fab12defde..457a2c065485 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1004,7 +1004,8 @@ static void __azx_runtime_resume(struct azx *chip, bool from_rt)
 
 	if (status && from_rt) {
 		list_for_each_codec(codec, &chip->bus)
-			if (status & (1 << codec->addr))
+			if (!codec->relaxed_resume &&
+			    (status & (1 << codec->addr)))
 				schedule_delayed_work(&codec->jackpoll_work,
 						      codec->jackpoll_interval);
 	}
@@ -1044,9 +1045,7 @@ static int azx_suspend(struct device *dev)
 static int azx_resume(struct device *dev)
 {
 	struct snd_card *card = dev_get_drvdata(dev);
-	struct hda_codec *codec;
 	struct azx *chip;
-	bool forced_resume = false;
 
 	if (!azx_is_pm_ready(card))
 		return 0;
@@ -1058,19 +1057,7 @@ static int azx_resume(struct device *dev)
 	if (azx_acquire_irq(chip, 1) < 0)
 		return -EIO;
 
-	/* check for the forced resume */
-	list_for_each_codec(codec, &chip->bus) {
-		if (hda_codec_need_resume(codec)) {
-			forced_resume = true;
-			break;
-		}
-	}
-
-	if (forced_resume)
-		pm_runtime_get_noresume(dev);
 	pm_runtime_force_resume(dev);
-	if (forced_resume)
-		pm_runtime_put(dev);
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 
 	trace_azx_resume(chip);
@@ -2092,7 +2079,6 @@ static void pcm_mmap_prepare(struct snd_pcm_substream *substream,
  * should be ignored from the beginning.
  */
 static const struct snd_pci_quirk driver_blacklist[] = {
-	SND_PCI_QUIRK(0x1043, 0x874f, "ASUS ROG Zenith II / Strix", 0),
 	SND_PCI_QUIRK(0x1462, 0xcb59, "MSI TRX40 Creator", 0),
 	SND_PCI_QUIRK(0x1462, 0xcb60, "MSI TRX40", 0),
 	{}
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index bb287a916dae..4eff16053bd5 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -38,6 +38,10 @@ static bool static_hdmi_pcm;
 module_param(static_hdmi_pcm, bool, 0644);
 MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info");
 
+static bool enable_acomp = true;
+module_param(enable_acomp, bool, 0444);
+MODULE_PARM_DESC(enable_acomp, "Enable audio component binding (default=yes)");
+
 struct hdmi_spec_per_cvt {
 	hda_nid_t cvt_nid;
 	int assigned;
@@ -2505,6 +2509,11 @@ static void generic_acomp_init(struct hda_codec *codec,
 {
 	struct hdmi_spec *spec = codec->spec;
 
+	if (!enable_acomp) {
+		codec_info(codec, "audio component disabled by module option\n");
+		return;
+	}
+
 	spec->port2pin = port2pin;
 	setup_drm_audio_ops(codec, ops);
 	if (!snd_hdac_acomp_init(&codec->bus->core, &spec->drm_audio_ops,
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index dc5557d79c43..c1a85c8f7b69 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -377,6 +377,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
 	case 0x10ec0233:
 	case 0x10ec0235:
 	case 0x10ec0236:
+	case 0x10ec0245:
 	case 0x10ec0255:
 	case 0x10ec0256:
 	case 0x10ec0257:
@@ -797,9 +798,11 @@ static void alc_ssid_check(struct hda_codec *codec, const hda_nid_t *ports)
 {
 	if (!alc_subsystem_id(codec, ports)) {
 		struct alc_spec *spec = codec->spec;
-		codec_dbg(codec,
-			  "realtek: Enable default setup for auto mode as fallback\n");
-		spec->init_amp = ALC_INIT_DEFAULT;
+		if (spec->init_amp == ALC_INIT_UNDEFINED) {
+			codec_dbg(codec,
+				  "realtek: Enable default setup for auto mode as fallback\n");
+			spec->init_amp = ALC_INIT_DEFAULT;
+		}
 	}
 }
 
@@ -8196,6 +8199,7 @@ static int patch_alc269(struct hda_codec *codec)
 		spec->gen.mixer_nid = 0;
 		break;
 	case 0x10ec0215:
+	case 0x10ec0245:
 	case 0x10ec0285:
 	case 0x10ec0289:
 		spec->codec_variant = ALC269_TYPE_ALC215;
@@ -9457,6 +9461,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = {
 	HDA_CODEC_ENTRY(0x10ec0234, "ALC234", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0235, "ALC233", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0236, "ALC236", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0245, "ALC245", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0255, "ALC255", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0256, "ALC256", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0257, "ALC257", patch_alc269),