Merge tag 'v5.4.59' into 5.4-2.1.x-imx

This is the 5.4.59 stable release

Conflicts (manual resolve):

drivers/gpu/drm/imx/dw_hdmi-imx.c:
drivers/gpu/drm/imx/imx-ldb.c:
drivers/gpu/drm/imx/ipuv3/ipuv3-crtc.c:
Port changes from upstream commit [1a279871012d3], which extends
component lifetime by moving drm structures allocation/free from
bind() to probe().

sound/soc/fsl/fsl_sai.c:
Apply patch [b8ae2bf5ccc66] from upstream, which uses FIFO watermark
mask macro.

Signed-off-by: Andrey Zhizhikin <andrey.z@gmail.com>

17 files changed, 121 insertions, 24 deletions

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ea25b8d0350d..88629906f314 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4391,6 +4391,7 @@ static void alc233_fixup_lenovo_line2_mic_hotkey(struct hda_codec *codec,
 {
 	struct alc_spec *spec = codec->spec;
 
+	spec->micmute_led_polarity = 1;
 	alc_fixup_hp_gpio_led(codec, action, 0, 0x04);
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
 		spec->init_amp = ALC_INIT_DEFAULT;
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 6508e2d2bf05..74e63f2e8d6a 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -997,10 +997,10 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai)
 	unsigned char offset = sai->soc->reg_offset;
 
 	regmap_update_bits(sai->regmap, FSL_SAI_TCR1(offset),
-				sai->soc->fifo_depth - 1,
+				FSL_SAI_CR1_RFW_MASK(sai->soc->fifo_depth),
 				sai->soc->fifo_depth - FSL_SAI_MAXBURST_TX);
 	regmap_update_bits(sai->regmap, FSL_SAI_RCR1(offset),
-				sai->soc->fifo_depth - 1,
+				FSL_SAI_CR1_RFW_MASK(sai->soc->fifo_depth),
 				FSL_SAI_MAXBURST_RX - 1);
 
 	snd_soc_dai_init_dma_data(cpu_dai, &sai->dma_params_tx,
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index 91e153e88ae2..e59ba6c9c01f 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -110,7 +110,7 @@
 #define FSL_SAI_CSR_FRDE	BIT(0)
 
 /* SAI Transmit and Receive Configuration 1 Register */
-#define FSL_SAI_CR1_RFW_MASK	0x1f
+#define FSL_SAI_CR1_RFW_MASK(x)	((x) - 1)
 
 /* SAI Transmit and Receive Configuration 2 Register */
 #define FSL_SAI_CR2_SYNC	BIT(30)
diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c
index adf416a49b48..60fb87495050 100644
--- a/sound/soc/intel/boards/bxt_rt298.c
+++ b/sound/soc/intel/boards/bxt_rt298.c
@@ -556,6 +556,7 @@ static int bxt_card_late_probe(struct snd_soc_card *card)
 /* broxton audio machine driver for SPT + RT298S */
 static struct snd_soc_card broxton_rt298 = {
 	.name = "broxton-rt298",
+	.owner = THIS_MODULE,
 	.dai_link = broxton_rt298_dais,
 	.num_links = ARRAY_SIZE(broxton_rt298_dais),
 	.controls = broxton_controls,
@@ -571,6 +572,7 @@ static struct snd_soc_card broxton_rt298 = {
 
 static struct snd_soc_card geminilake_rt298 = {
 	.name = "geminilake-rt298",
+	.owner = THIS_MODULE,
 	.dai_link = broxton_rt298_dais,
 	.num_links = ARRAY_SIZE(broxton_rt298_dais),
 	.controls = broxton_controls,
diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c
index 1f698adde506..7126344017fa 100644
--- a/sound/soc/meson/axg-card.c
+++ b/sound/soc/meson/axg-card.c
@@ -266,7 +266,7 @@ static int axg_card_add_tdm_loopback(struct snd_soc_card *card,
 
 	lb = &card->dai_link[*index + 1];
 
-	lb->name = kasprintf(GFP_KERNEL, "%s-lb", pad->name);
+	lb->name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-lb", pad->name);
 	if (!lb->name)
 		return -ENOMEM;
 
diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c
index 358c8c0d861c..f7e8e9da68a0 100644
--- a/sound/soc/meson/axg-tdm-formatter.c
+++ b/sound/soc/meson/axg-tdm-formatter.c
@@ -70,7 +70,7 @@ EXPORT_SYMBOL_GPL(axg_tdm_formatter_set_channel_masks);
 static int axg_tdm_formatter_enable(struct axg_tdm_formatter *formatter)
 {
 	struct axg_tdm_stream *ts = formatter->stream;
-	bool invert = formatter->drv->quirks->invert_sclk;
+	bool invert;
 	int ret;
 
 	/* Do nothing if the formatter is already enabled */
@@ -96,11 +96,12 @@ static int axg_tdm_formatter_enable(struct axg_tdm_formatter *formatter)
 		return ret;
 
 	/*
-	 * If sclk is inverted, invert it back and provide the inversion
-	 * required by the formatter
+	 * If sclk is inverted, it means the bit should latched on the
+	 * rising edge which is what our HW expects. If not, we need to
+	 * invert it before the formatter.
 	 */
-	invert ^= axg_tdm_sclk_invert(ts->iface->fmt);
-	ret = clk_set_phase(formatter->sclk, invert ? 180 : 0);
+	invert = axg_tdm_sclk_invert(ts->iface->fmt);
+	ret = clk_set_phase(formatter->sclk, invert ? 0 : 180);
 	if (ret)
 		return ret;
 
diff --git a/sound/soc/meson/axg-tdm-formatter.h b/sound/soc/meson/axg-tdm-formatter.h
index 9ef98e955cb2..a1f0dcc0ff13 100644
--- a/sound/soc/meson/axg-tdm-formatter.h
+++ b/sound/soc/meson/axg-tdm-formatter.h
@@ -16,7 +16,6 @@ struct snd_kcontrol;
 
 struct axg_tdm_formatter_hw {
 	unsigned int skew_offset;
-	bool invert_sclk;
 };
 
 struct axg_tdm_formatter_ops {
diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c
index d51f3344be7c..e25336f73912 100644
--- a/sound/soc/meson/axg-tdm-interface.c
+++ b/sound/soc/meson/axg-tdm-interface.c
@@ -119,18 +119,25 @@ static int axg_tdm_iface_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
 	struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
 
-	/* These modes are not supported */
-	if (fmt & (SND_SOC_DAIFMT_CBS_CFM | SND_SOC_DAIFMT_CBM_CFS)) {
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		if (!iface->mclk) {
+			dev_err(dai->dev, "cpu clock master: mclk missing\n");
+			return -ENODEV;
+		}
+		break;
+
+	case SND_SOC_DAIFMT_CBM_CFM:
+		break;
+
+	case SND_SOC_DAIFMT_CBS_CFM:
+	case SND_SOC_DAIFMT_CBM_CFS:
 		dev_err(dai->dev, "only CBS_CFS and CBM_CFM are supported\n");
+		/* Fall-through */
+	default:
 		return -EINVAL;
 	}
 
-	/* If the TDM interface is the clock master, it requires mclk */
-	if (!iface->mclk && (fmt & SND_SOC_DAIFMT_CBS_CFS)) {
-		dev_err(dai->dev, "cpu clock master: mclk missing\n");
-		return -ENODEV;
-	}
-
 	iface->fmt = fmt;
 	return 0;
 }
@@ -319,7 +326,8 @@ static int axg_tdm_iface_hw_params(struct snd_pcm_substream *substream,
 	if (ret)
 		return ret;
 
-	if (iface->fmt & SND_SOC_DAIFMT_CBS_CFS) {
+	if ((iface->fmt & SND_SOC_DAIFMT_MASTER_MASK) ==
+	    SND_SOC_DAIFMT_CBS_CFS) {
 		ret = axg_tdm_iface_set_sclk(dai, params);
 		if (ret)
 			return ret;
diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c
index 973d4c02ef8d..88ed95ae886b 100644
--- a/sound/soc/meson/axg-tdmin.c
+++ b/sound/soc/meson/axg-tdmin.c
@@ -228,15 +228,29 @@ static const struct axg_tdm_formatter_driver axg_tdmin_drv = {
 	.regmap_cfg	= &axg_tdmin_regmap_cfg,
 	.ops		= &axg_tdmin_ops,
 	.quirks		= &(const struct axg_tdm_formatter_hw) {
-		.invert_sclk	= false,
 		.skew_offset	= 2,
 	},
 };
 
+static const struct axg_tdm_formatter_driver g12a_tdmin_drv = {
+	.component_drv	= &axg_tdmin_component_drv,
+	.regmap_cfg	= &axg_tdmin_regmap_cfg,
+	.ops		= &axg_tdmin_ops,
+	.quirks		= &(const struct axg_tdm_formatter_hw) {
+		.skew_offset	= 3,
+	},
+};
+
 static const struct of_device_id axg_tdmin_of_match[] = {
 	{
 		.compatible = "amlogic,axg-tdmin",
 		.data = &axg_tdmin_drv,
+	}, {
+		.compatible = "amlogic,g12a-tdmin",
+		.data = &g12a_tdmin_drv,
+	}, {
+		.compatible = "amlogic,sm1-tdmin",
+		.data = &g12a_tdmin_drv,
 	}, {}
 };
 MODULE_DEVICE_TABLE(of, axg_tdmin_of_match);
diff --git a/sound/soc/meson/axg-tdmout.c b/sound/soc/meson/axg-tdmout.c
index 418ec314b37d..3ceabddae629 100644
--- a/sound/soc/meson/axg-tdmout.c
+++ b/sound/soc/meson/axg-tdmout.c
@@ -238,7 +238,6 @@ static const struct axg_tdm_formatter_driver axg_tdmout_drv = {
 	.regmap_cfg	= &axg_tdmout_regmap_cfg,
 	.ops		= &axg_tdmout_ops,
 	.quirks		= &(const struct axg_tdm_formatter_hw) {
-		.invert_sclk = true,
 		.skew_offset = 1,
 	},
 };
@@ -248,7 +247,6 @@ static const struct axg_tdm_formatter_driver g12a_tdmout_drv = {
 	.regmap_cfg	= &axg_tdmout_regmap_cfg,
 	.ops		= &axg_tdmout_ops,
 	.quirks		= &(const struct axg_tdm_formatter_hw) {
-		.invert_sclk = true,
 		.skew_offset = 2,
 	},
 };
@@ -309,7 +307,6 @@ static const struct axg_tdm_formatter_driver sm1_tdmout_drv = {
 	.regmap_cfg	= &axg_tdmout_regmap_cfg,
 	.ops		= &axg_tdmout_ops,
 	.quirks		= &(const struct axg_tdm_formatter_hw) {
-		.invert_sclk = true,
 		.skew_offset = 2,
 	},
 };
diff --git a/sound/soc/sof/nocodec.c b/sound/soc/sof/nocodec.c
index 71cf5f9db79d..849c3bcdca9e 100644
--- a/sound/soc/sof/nocodec.c
+++ b/sound/soc/sof/nocodec.c
@@ -14,6 +14,7 @@
 
 static struct snd_soc_card sof_nocodec_card = {
 	.name = "nocodec", /* the sof- prefix is added by the core */
+	.owner = THIS_MODULE
 };
 
 static int sof_nocodec_bes_setup(struct device *dev,
diff --git a/sound/usb/card.h b/sound/usb/card.h
index f39f23e3525d..d8ec5caf464d 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -133,6 +133,7 @@ struct snd_usb_substream {
 	unsigned int tx_length_quirk:1;	/* add length specifier to transfers */
 	unsigned int fmt_type;		/* USB audio format type (1-3) */
 	unsigned int pkt_offset_adj;	/* Bytes to drop from beginning of packets (for non-compliant devices) */
+	unsigned int stream_offset_adj;	/* Bytes to drop from beginning of stream (for non-compliant devices) */
 
 	unsigned int running: 1;	/* running status */
 
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index d39bf5b648d1..49f0dc0e3e4d 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -184,6 +184,7 @@ static const struct rc_config {
 	{ USB_ID(0x041e, 0x3042), 0, 1, 1, 1,  1,  0x000d }, /* Usb X-Fi S51 */
 	{ USB_ID(0x041e, 0x30df), 0, 1, 1, 1,  1,  0x000d }, /* Usb X-Fi S51 Pro */
 	{ USB_ID(0x041e, 0x3237), 0, 1, 1, 1,  1,  0x000d }, /* Usb X-Fi S51 Pro */
+	{ USB_ID(0x041e, 0x3263), 0, 1, 1, 1,  1,  0x000d }, /* Usb X-Fi S51 Pro */
 	{ USB_ID(0x041e, 0x3048), 2, 2, 6, 6,  2,  0x6e91 }, /* Toshiba SB0500 */
 };
 
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index d11d00efc574..7b41f9748978 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -1417,6 +1417,12 @@ static void retire_capture_urb(struct snd_usb_substream *subs,
 			// continue;
 		}
 		bytes = urb->iso_frame_desc[i].actual_length;
+		if (subs->stream_offset_adj > 0) {
+			unsigned int adj = min(subs->stream_offset_adj, bytes);
+			cp += adj;
+			bytes -= adj;
+			subs->stream_offset_adj -= adj;
+		}
 		frames = bytes / stride;
 		if (!subs->txfr_quirk)
 			bytes = frames * stride;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 562179492a33..1573229d8cf4 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -3570,6 +3570,62 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 		}
 	}
 },
+{
+	/*
+	 * PIONEER DJ DDJ-RB
+	 * PCM is 4 channels out, 2 dummy channels in @ 44.1 fixed
+	 * The feedback for the output is the dummy input.
+	 */
+	USB_DEVICE_VENDOR_SPEC(0x2b73, 0x000e),
+	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = (const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 0,
+				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
+				.data = &(const struct audioformat) {
+					.formats = SNDRV_PCM_FMTBIT_S24_3LE,
+					.channels = 4,
+					.iface = 0,
+					.altsetting = 1,
+					.altset_idx = 1,
+					.endpoint = 0x01,
+					.ep_attr = USB_ENDPOINT_XFER_ISOC|
+						   USB_ENDPOINT_SYNC_ASYNC,
+					.rates = SNDRV_PCM_RATE_44100,
+					.rate_min = 44100,
+					.rate_max = 44100,
+					.nr_rates = 1,
+					.rate_table = (unsigned int[]) { 44100 }
+				}
+			},
+			{
+				.ifnum = 0,
+				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
+				.data = &(const struct audioformat) {
+					.formats = SNDRV_PCM_FMTBIT_S24_3LE,
+					.channels = 2,
+					.iface = 0,
+					.altsetting = 1,
+					.altset_idx = 1,
+					.endpoint = 0x82,
+					.ep_attr = USB_ENDPOINT_XFER_ISOC|
+						 USB_ENDPOINT_SYNC_ASYNC|
+						 USB_ENDPOINT_USAGE_IMPLICIT_FB,
+					.rates = SNDRV_PCM_RATE_44100,
+					.rate_min = 44100,
+					.rate_max = 44100,
+					.nr_rates = 1,
+					.rate_table = (unsigned int[]) { 44100 }
+				}
+			},
+			{
+				.ifnum = -1
+			}
+		}
+	}
+},
 
 #define ALC1220_VB_DESKTOP(vend, prod) { \
 	USB_DEVICE(vend, prod),	\
@@ -3623,7 +3679,13 @@ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */
  * with.
  */
 {
-	USB_DEVICE(0x534d, 0x2109),
+	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
+		       USB_DEVICE_ID_MATCH_INT_CLASS |
+		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
+	.idVendor = 0x534d,
+	.idProduct = 0x2109,
+	.bInterfaceClass = USB_CLASS_AUDIO,
+	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL,
 	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
 		.vendor_name = "MacroSilicon",
 		.product_name = "MS2109",
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index a8bb953cc468..a756f50d9f07 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1432,6 +1432,9 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs,
 	case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */
 		set_format_emu_quirk(subs, fmt);
 		break;
+	case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */
+		subs->stream_offset_adj = 2;
+		break;
 	}
 }
 
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index 11785f9652ad..d01edd5da6cf 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -94,6 +94,7 @@ static void snd_usb_init_substream(struct snd_usb_stream *as,
 	subs->tx_length_quirk = as->chip->tx_length_quirk;
 	subs->speed = snd_usb_get_speed(subs->dev);
 	subs->pkt_offset_adj = 0;
+	subs->stream_offset_adj = 0;
 
 	snd_usb_set_pcm_ops(as->pcm, stream);