Merge tag 'sound-fix-6.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "A few more small fixes for 6.18-rc1.

  Most of changes are about ASoC Intel and SOF drivers, while a few
  other device-specific fixes are found for HD-audio, USB-audio, ASoC
  RT722VB and Meson"

* tag 'sound-fix-6.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ASoC: rt722: add settings for rt722VB
  ASoC: meson: aiu-encoder-i2s: fix bit clock polarity
  ALSA: usb: fpc: replace kmalloc_array followed by copy_from_user with memdup_array_user
  ALSA: hda/tas2781: Enable init_profile_id for device initialization
  ALSA: emu10k1: Fix typo in docs
  ALSA: hda/realtek: Add quirk for ASUS ROG Zephyrus Duo
  ASoC: SOF: Intel: Read the LLP via the associated Link DMA channel
  ASoC: SOF: ipc4-pcm: do not report invalid delay values
  ASoC: SOF: sof-audio: add dev_dbg_ratelimited wrapper
  ASoC: SOF: Intel: hda-pcm: Place the constraint on period time instead of buffer time
  ASoC: SOF: ipc4-topology: Account for different ChainDMA host buffer size
  ASoC: SOF: ipc4-topology: Correct the minimum host DMA buffer size
  ASoC: SOF: ipc4-pcm: fix start offset calculation for chain DMA
  ASoC: SOF: ipc4-pcm: fix delay calculation when DSP resamples
  ASoC: SOF: ipc3-topology: Fix multi-core and static pipelines tear down
  ALSA: hda/hdmi: Add pin fix for HP ProDesk model

Author: torvalds

Documentation/sound/cards/emu-mixer.rst

@@ -66,7 +66,7 @@ FX-bus
 
 name='Clock Source',index=0
 ---------------------------
-This control allows switching the word clock between interally generated
+This control allows switching the word clock between internally generated
 44.1 or 48 kHz, or a number of external sources.
 
 Note: the sources for the 1616 CardBus card are unclear. Please report your

sound/hda/codecs/hdmi/hdmi.c

@@ -1549,6 +1549,7 @@ static const struct snd_pci_quirk force_connect_list[] = {
 	SND_PCI_QUIRK(0x103c, 0x83e2, "HP EliteDesk 800 G4", 1),
 	SND_PCI_QUIRK(0x103c, 0x83ef, "HP MP9 G4 Retail System AMS", 1),
 	SND_PCI_QUIRK(0x103c, 0x845a, "HP EliteDesk 800 G4 DM 65W", 1),
+	SND_PCI_QUIRK(0x103c, 0x83f3, "HP ProDesk 400", 1),
 	SND_PCI_QUIRK(0x103c, 0x870f, "HP", 1),
 	SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1),
 	SND_PCI_QUIRK(0x103c, 0x8711, "HP", 1),

sound/hda/codecs/realtek/alc269.c

@@ -3735,6 +3735,7 @@ enum {
 	ALC285_FIXUP_ASUS_GA605K_HEADSET_MIC,
 	ALC285_FIXUP_ASUS_GA605K_I2C_SPEAKER2_TO_DAC1,
 	ALC269_FIXUP_POSITIVO_P15X_HEADSET_MIC,
+	ALC289_FIXUP_ASUS_ZEPHYRUS_DUAL_SPK,
 };
 
 /* A special fixup for Lenovo C940 and Yoga Duet 7;
@@ -6164,6 +6165,14 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE,
 	},
+	[ALC289_FIXUP_ASUS_ZEPHYRUS_DUAL_SPK] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x17, 0x90170151 }, /* Internal Speaker LFE */
+			{ 0x1e, 0x90170150 }, /* Internal Speaker */
+			{ }
+		},
+	}
 };
 
 static const struct hda_quirk alc269_fixup_tbl[] = {
@@ -6718,6 +6727,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A),
 	SND_PCI_QUIRK(0x1043, 0x1533, "ASUS GV302XA/XJ/XQ/XU/XV/XI", ALC287_FIXUP_CS35L41_I2C_2),
 	SND_PCI_QUIRK(0x1043, 0x1573, "ASUS GZ301VV/VQ/VU/VJ/VA/VC/VE/VVC/VQC/VUC/VJC/VEC/VCC", ALC285_FIXUP_ASUS_HEADSET_MIC),
+	SND_PCI_QUIRK(0x1043, 0x1652, "ASUS ROG Zephyrus Do 15 SE", ALC289_FIXUP_ASUS_ZEPHYRUS_DUAL_SPK),
 	SND_PCI_QUIRK(0x1043, 0x1662, "ASUS GV301QH", ALC294_FIXUP_ASUS_DUAL_SPK),
 	SND_PCI_QUIRK(0x1043, 0x1663, "ASUS GU603ZI/ZJ/ZQ/ZU/ZV", ALC285_FIXUP_ASUS_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1043, 0x1683, "ASUS UM3402YAR", ALC287_FIXUP_CS35L41_I2C_2),

sound/hda/codecs/side-codecs/tas2781_hda_i2c.c

@@ -474,6 +474,12 @@ static void tasdevice_dspfw_init(void *context)
 	if (tas_priv->fmw->nr_configurations > 0)
 		tas_priv->cur_conf = 0;
 
+	/* Init common setting for different audio profiles */
+	if (tas_priv->rcabin.init_profile_id >= 0)
+		tasdevice_select_cfg_blk(tas_priv,
+			tas_priv->rcabin.init_profile_id,
+			TASDEVICE_BIN_BLK_PRE_POWER_UP);
+
 	/* If calibrated data occurs error, dsp will still works with default
 	 * calibrated data inside algo.
 	 */
@@ -770,6 +776,12 @@ static int tas2781_system_resume(struct device *dev)
 	tasdevice_reset(tas_hda->priv);
 	tasdevice_prmg_load(tas_hda->priv, tas_hda->priv->cur_prog);
 
+	/* Init common setting for different audio profiles */
+	if (tas_hda->priv->rcabin.init_profile_id >= 0)
+		tasdevice_select_cfg_blk(tas_hda->priv,
+			tas_hda->priv->rcabin.init_profile_id,
+			TASDEVICE_BIN_BLK_PRE_POWER_UP);
+
 	if (tas_hda->priv->playback_started)
 		tasdevice_tuning_switch(tas_hda->priv, 0);
 

sound/soc/codecs/rt722-sdca-sdw.c

@@ -21,7 +21,7 @@ static int rt722_sdca_mbq_size(struct device *dev, unsigned int reg)
 	switch (reg) {
 	case 0x2f01 ... 0x2f0a:
 	case 0x2f35 ... 0x2f36:
-	case 0x2f50:
+	case 0x2f50 ... 0x2f52:
 	case 0x2f54:
 	case 0x2f58 ... 0x2f5d:
 	case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT722_SDCA_ENT0, RT722_SDCA_CTL_FUNC_STATUS, 0):

sound/soc/codecs/rt722-sdca.c

@@ -1378,6 +1378,9 @@ static void rt722_sdca_dmic_preset(struct rt722_sdca_priv *rt722)
 		/* PHYtiming TDZ/TZD control */
 		regmap_write(rt722->regmap, 0x2f03, 0x06);
 
+		if (rt722->hw_vid == RT722_VB)
+			regmap_write(rt722->regmap, 0x2f52, 0x00);
+
 		/* clear flag */
 		regmap_write(rt722->regmap,
 			SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT722_SDCA_ENT0, RT722_SDCA_CTL_FUNC_STATUS, 0),
@@ -1415,6 +1418,9 @@ static void rt722_sdca_amp_preset(struct rt722_sdca_priv *rt722)
 			SDW_SDCA_CTL(FUNC_NUM_AMP, RT722_SDCA_ENT_OT23,
 				RT722_SDCA_CTL_VENDOR_DEF, CH_08), 0x04);
 
+		if (rt722->hw_vid == RT722_VB)
+			regmap_write(rt722->regmap, 0x2f54, 0x00);
+
 		/* clear flag */
 		regmap_write(rt722->regmap,
 			SDW_SDCA_CTL(FUNC_NUM_AMP, RT722_SDCA_ENT0, RT722_SDCA_CTL_FUNC_STATUS, 0),
@@ -1506,6 +1512,9 @@ static void rt722_sdca_jack_preset(struct rt722_sdca_priv *rt722)
 		rt722_sdca_index_write(rt722, RT722_VENDOR_REG, RT722_DIGITAL_MISC_CTRL4,
 			0x0010);
 
+		if (rt722->hw_vid == RT722_VB)
+			regmap_write(rt722->regmap, 0x2f51, 0x00);
+
 		/* clear flag */
 		regmap_write(rt722->regmap,
 			SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT722_SDCA_ENT0, RT722_SDCA_CTL_FUNC_STATUS, 0),
@@ -1516,6 +1525,7 @@ static void rt722_sdca_jack_preset(struct rt722_sdca_priv *rt722)
 int rt722_sdca_io_init(struct device *dev, struct sdw_slave *slave)
 {
 	struct rt722_sdca_priv *rt722 = dev_get_drvdata(dev);
+	unsigned int val;
 
 	rt722->disable_irq = false;
 
@@ -1545,6 +1555,10 @@ int rt722_sdca_io_init(struct device *dev, struct sdw_slave *slave)
 
 	pm_runtime_get_noresume(&slave->dev);
 
+	rt722_sdca_index_read(rt722, RT722_VENDOR_REG, RT722_JD_PRODUCT_NUM, &val);
+	rt722->hw_vid = (val & 0x0f00) >> 8;
+	dev_dbg(&slave->dev, "%s hw_vid=0x%x\n", __func__, rt722->hw_vid);
+
 	rt722_sdca_dmic_preset(rt722);
 	rt722_sdca_amp_preset(rt722);
 	rt722_sdca_jack_preset(rt722);

sound/soc/codecs/rt722-sdca.h

@@ -39,6 +39,7 @@ struct  rt722_sdca_priv {
 	/* For DMIC */
 	bool fu1e_dapm_mute;
 	bool fu1e_mixer_mute[4];
+	int hw_vid;
 };
 
 struct rt722_sdca_dmic_kctrl_priv {
@@ -233,6 +234,11 @@ enum rt722_sdca_jd_src {
 	RT722_JD1,
 };
 
+enum rt722_sdca_version {
+	RT722_VA,
+	RT722_VB,
+};
+
 int rt722_sdca_io_init(struct device *dev, struct sdw_slave *slave);
 int rt722_sdca_init(struct device *dev, struct regmap *regmap, struct sdw_slave *slave);
 int rt722_sdca_index_write(struct rt722_sdca_priv *rt722,

sound/soc/meson/aiu-encoder-i2s.c

@@ -236,8 +236,12 @@ static int aiu_encoder_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	    inv == SND_SOC_DAIFMT_IB_IF)
 		val |= AIU_CLK_CTRL_LRCLK_INVERT;
 
-	if (inv == SND_SOC_DAIFMT_IB_NF ||
-	    inv == SND_SOC_DAIFMT_IB_IF)
+	/*
+	 * The SoC changes data on the rising edge of the bitclock
+	 * so an inversion of the bitclock is required in normal mode
+	 */
+	if (inv == SND_SOC_DAIFMT_NB_NF ||
+	    inv == SND_SOC_DAIFMT_NB_IF)
 		val |= AIU_CLK_CTRL_AOCLK_INVERT;
 
 	/* Signal skew */
@@ -328,4 +332,3 @@ const struct snd_soc_dai_ops aiu_encoder_i2s_dai_ops = {
 	.startup	= aiu_encoder_i2s_startup,
 	.shutdown	= aiu_encoder_i2s_shutdown,
 };
-

sound/soc/sof/intel/hda-pcm.c

@@ -29,6 +29,8 @@
 #define SDnFMT_BITS(x)	((x) << 4)
 #define SDnFMT_CHAN(x)	((x) << 0)
 
+#define HDA_MAX_PERIOD_TIME_HEADROOM	10
+
 static bool hda_always_enable_dmi_l1;
 module_param_named(always_enable_dmi_l1, hda_always_enable_dmi_l1, bool, 0444);
 MODULE_PARM_DESC(always_enable_dmi_l1, "SOF HDA always enable DMI l1");
@@ -291,19 +293,30 @@ int hda_dsp_pcm_open(struct snd_sof_dev *sdev,
 	 * On playback start the DMA will transfer dsp_max_burst_size_in_ms
 	 * amount of data in one initial burst to fill up the host DMA buffer.
 	 * Consequent DMA burst sizes are shorter and their length can vary.
-	 * To make sure that userspace allocate large enough ALSA buffer we need
-	 * to place a constraint on the buffer time.
+	 * To avoid immediate xrun by the initial burst we need to place
+	 * constraint on the period size (via PERIOD_TIME) to cover the size of
+	 * the host buffer.
+	 * We need to add headroom of max 10ms as the firmware needs time to
+	 * settle to the 1ms pacing and initially it can run faster for few
+	 * internal periods.
 	 *
 	 * On capture the DMA will transfer 1ms chunks.
-	 *
-	 * Exact dsp_max_burst_size_in_ms constraint is racy, so set the
-	 * constraint to a minimum of 2x dsp_max_burst_size_in_ms.
 	 */
-	if (spcm->stream[direction].dsp_max_burst_size_in_ms)
+	if (spcm->stream[direction].dsp_max_burst_size_in_ms) {
+		unsigned int period_time = spcm->stream[direction].dsp_max_burst_size_in_ms;
+
+		/*
+		 * add headroom over the maximum burst size to cover the time
+		 * needed for the DMA pace to settle.
+		 * Limit the headroom time to HDA_MAX_PERIOD_TIME_HEADROOM
+		 */
+		period_time += min(period_time, HDA_MAX_PERIOD_TIME_HEADROOM);
+
 		snd_pcm_hw_constraint_minmax(substream->runtime,
-			SNDRV_PCM_HW_PARAM_BUFFER_TIME,
-			spcm->stream[direction].dsp_max_burst_size_in_ms * USEC_PER_MSEC * 2,
+			SNDRV_PCM_HW_PARAM_PERIOD_TIME,
+			period_time * USEC_PER_MSEC,
 			UINT_MAX);
+	}
 
 	/* binding pcm substream to hda stream */
 	substream->runtime->private_data = &dsp_stream->hstream;

sound/soc/sof/intel/hda-stream.c

@@ -1129,10 +1129,35 @@ u64 hda_dsp_get_stream_llp(struct snd_sof_dev *sdev,
 			   struct snd_soc_component *component,
 			   struct snd_pcm_substream *substream)
 {
-	struct hdac_stream *hstream = substream->runtime->private_data;
-	struct hdac_ext_stream *hext_stream = stream_to_hdac_ext_stream(hstream);
+	struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
+	struct snd_soc_pcm_runtime *be_rtd = NULL;
+	struct hdac_ext_stream *hext_stream;
+	struct snd_soc_dai *cpu_dai;
+	struct snd_soc_dpcm *dpcm;
 	u32 llp_l, llp_u;
 
+	/*
+	 * The LLP needs to be read from the Link DMA used for this FE as it is
+	 * allowed to use any combination of Link and Host channels
+	 */
+	for_each_dpcm_be(rtd, substream->stream, dpcm) {
+		if (dpcm->fe != rtd)
+			continue;
+
+		be_rtd = dpcm->be;
+	}
+
+	if (!be_rtd)
+		return 0;
+
+	cpu_dai = snd_soc_rtd_to_cpu(be_rtd, 0);
+	if (!cpu_dai)
+		return 0;
+
+	hext_stream = snd_soc_dai_get_dma_data(cpu_dai, substream);
+	if (!hext_stream)
+		return 0;
+
 	/*
 	 * The pplc_addr have been calculated during probe in
 	 * hda_dsp_stream_init():