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-rw-r--r--sound/pci/hda/hda_controller.c34
-rw-r--r--sound/pci/hda/hda_intel.c3
-rw-r--r--sound/pci/hda/hda_priv.h1
-rw-r--r--sound/pci/hda/patch_hdmi.c4
-rw-r--r--sound/pci/hda/patch_realtek.c2
-rw-r--r--sound/soc/codecs/alc5623.c2
-rw-r--r--sound/soc/codecs/cs42l52.c6
-rw-r--r--sound/soc/codecs/cs42l73.c6
-rw-r--r--sound/soc/codecs/tlv320aic3x.c9
-rw-r--r--sound/soc/fsl/fsl_spdif.h4
-rw-r--r--sound/soc/intel/sst-dsp-priv.h2
-rw-r--r--sound/soc/intel/sst-haswell-ipc.c7
-rw-r--r--sound/soc/jz4740/Makefile2
-rw-r--r--sound/soc/sh/rcar/src.c4
-rw-r--r--sound/soc/sh/rcar/ssi.c4
-rw-r--r--sound/soc/soc-dapm.c1
-rw-r--r--sound/usb/card.c12
-rw-r--r--sound/usb/card.h1
-rw-r--r--sound/usb/endpoint.c15
-rw-r--r--sound/usb/pcm.c5
-rw-r--r--sound/usb/usbaudio.h1
21 files changed, 79 insertions, 46 deletions
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 248b90abb882..480bbddbd801 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -1059,24 +1059,26 @@ static void azx_init_cmd_io(struct azx *chip)
 
 	/* reset the corb hw read pointer */
 	azx_writew(chip, CORBRP, ICH6_CORBRP_RST);
-	for (timeout = 1000; timeout > 0; timeout--) {
-		if ((azx_readw(chip, CORBRP) & ICH6_CORBRP_RST) == ICH6_CORBRP_RST)
-			break;
-		udelay(1);
-	}
-	if (timeout <= 0)
-		dev_err(chip->card->dev, "CORB reset timeout#1, CORBRP = %d\n",
-			azx_readw(chip, CORBRP));
+	if (!(chip->driver_caps & AZX_DCAPS_CORBRP_SELF_CLEAR)) {
+		for (timeout = 1000; timeout > 0; timeout--) {
+			if ((azx_readw(chip, CORBRP) & ICH6_CORBRP_RST) == ICH6_CORBRP_RST)
+				break;
+			udelay(1);
+		}
+		if (timeout <= 0)
+			dev_err(chip->card->dev, "CORB reset timeout#1, CORBRP = %d\n",
+				azx_readw(chip, CORBRP));
 
-	azx_writew(chip, CORBRP, 0);
-	for (timeout = 1000; timeout > 0; timeout--) {
-		if (azx_readw(chip, CORBRP) == 0)
-			break;
-		udelay(1);
+		azx_writew(chip, CORBRP, 0);
+		for (timeout = 1000; timeout > 0; timeout--) {
+			if (azx_readw(chip, CORBRP) == 0)
+				break;
+			udelay(1);
+		}
+		if (timeout <= 0)
+			dev_err(chip->card->dev, "CORB reset timeout#2, CORBRP = %d\n",
+				azx_readw(chip, CORBRP));
 	}
-	if (timeout <= 0)
-		dev_err(chip->card->dev, "CORB reset timeout#2, CORBRP = %d\n",
-			azx_readw(chip, CORBRP));
 
 	/* enable corb dma */
 	azx_writeb(chip, CORBCTL, ICH6_CORBCTL_RUN);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index d6bca62ef387..b540ad71eb0d 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -249,7 +249,8 @@ enum {
 /* quirks for Nvidia */
 #define AZX_DCAPS_PRESET_NVIDIA \
 	(AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI |\
-	 AZX_DCAPS_ALIGN_BUFSIZE | AZX_DCAPS_NO_64BIT)
+	 AZX_DCAPS_ALIGN_BUFSIZE | AZX_DCAPS_NO_64BIT |\
+	 AZX_DCAPS_CORBRP_SELF_CLEAR)
 
 #define AZX_DCAPS_PRESET_CTHDA \
 	(AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB | AZX_DCAPS_4K_BDLE_BOUNDARY)
diff --git a/sound/pci/hda/hda_priv.h b/sound/pci/hda/hda_priv.h
index ba38b819f984..4a7cb01fa912 100644
--- a/sound/pci/hda/hda_priv.h
+++ b/sound/pci/hda/hda_priv.h
@@ -189,6 +189,7 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
 #define AZX_DCAPS_COUNT_LPIB_DELAY  (1 << 25)	/* Take LPIB as delay */
 #define AZX_DCAPS_PM_RUNTIME	(1 << 26)	/* runtime PM support */
 #define AZX_DCAPS_I915_POWERWELL (1 << 27)	/* HSW i915 powerwell support */
+#define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28)	/* CORBRP clears itself after reset */
 
 /* position fix mode */
 enum {
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 0cb5b89cd0c8..1edbb9c47c2d 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1127,8 +1127,10 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec,
 					    AMP_OUT_UNMUTE);
 
 	eld = &per_pin->sink_eld;
-	if (!eld->monitor_present)
+	if (!eld->monitor_present) {
+		hdmi_set_channel_count(codec, per_pin->cvt_nid, channels);
 		return;
+	}
 
 	if (!non_pcm && per_pin->chmap_set)
 		ca = hdmi_manual_channel_allocation(channels, per_pin->chmap);
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c643dfc0a826..5f7c765391f1 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4621,6 +4621,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0667, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0668, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0669, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0674, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x067e, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x067f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index f500905e9373..2acf82f4a08a 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -1018,13 +1018,13 @@ static int alc5623_i2c_probe(struct i2c_client *client,
 		dev_err(&client->dev, "failed to read vendor ID1: %d\n", ret);
 		return ret;
 	}
-	vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8);
 
 	ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID2, &vid2);
 	if (ret < 0) {
 		dev_err(&client->dev, "failed to read vendor ID2: %d\n", ret);
 		return ret;
 	}
+	vid2 >>= 8;
 
 	if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) {
 		dev_err(&client->dev, "unknown or wrong codec\n");
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 460d35547a68..2213a037c893 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -1229,8 +1229,10 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client,
 	}
 
 	if (cs42l52->pdata.reset_gpio) {
-		ret = gpio_request_one(cs42l52->pdata.reset_gpio,
-				       GPIOF_OUT_INIT_HIGH, "CS42L52 /RST");
+		ret = devm_gpio_request_one(&i2c_client->dev,
+					    cs42l52->pdata.reset_gpio,
+					    GPIOF_OUT_INIT_HIGH,
+					    "CS42L52 /RST");
 		if (ret < 0) {
 			dev_err(&i2c_client->dev, "Failed to request /RST %d: %d\n",
 				cs42l52->pdata.reset_gpio, ret);
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 0ee60a19a263..ae3717992d56 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -1443,8 +1443,10 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client,
 	i2c_set_clientdata(i2c_client, cs42l73);
 
 	if (cs42l73->pdata.reset_gpio) {
-		ret = gpio_request_one(cs42l73->pdata.reset_gpio,
-				       GPIOF_OUT_INIT_HIGH, "CS42L73 /RST");
+		ret = devm_gpio_request_one(&i2c_client->dev,
+					    cs42l73->pdata.reset_gpio,
+					    GPIOF_OUT_INIT_HIGH,
+					    "CS42L73 /RST");
 		if (ret < 0) {
 			dev_err(&i2c_client->dev, "Failed to request /RST %d: %d\n",
 				cs42l73->pdata.reset_gpio, ret);
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index b1835103e9b4..d7349bc89ad3 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1399,7 +1399,6 @@ static int aic3x_probe(struct snd_soc_codec *codec)
 	}
 
 	aic3x_add_widgets(codec);
-	list_add(&aic3x->list, &reset_list);
 
 	return 0;
 
@@ -1569,7 +1568,13 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
 
 	ret = snd_soc_register_codec(&i2c->dev,
 			&soc_codec_dev_aic3x, &aic3x_dai, 1);
-	return ret;
+
+	if (ret != 0)
+		goto err_gpio;
+
+	list_add(&aic3x->list, &reset_list);
+
+	return 0;
 
 err_gpio:
 	if (gpio_is_valid(aic3x->gpio_reset) &&
diff --git a/sound/soc/fsl/fsl_spdif.h b/sound/soc/fsl/fsl_spdif.h
index b1266790d117..605a10b2112b 100644
--- a/sound/soc/fsl/fsl_spdif.h
+++ b/sound/soc/fsl/fsl_spdif.h
@@ -144,8 +144,8 @@ enum spdif_gainsel {
 
 /* SPDIF Clock register */
 #define STC_SYSCLK_DIV_OFFSET		11
-#define STC_SYSCLK_DIV_MASK		(0x1ff << STC_TXCLK_SRC_OFFSET)
-#define STC_SYSCLK_DIV(x)		((((x) - 1) << STC_TXCLK_DIV_OFFSET) & STC_SYSCLK_DIV_MASK)
+#define STC_SYSCLK_DIV_MASK		(0x1ff << STC_SYSCLK_DIV_OFFSET)
+#define STC_SYSCLK_DIV(x)		((((x) - 1) << STC_SYSCLK_DIV_OFFSET) & STC_SYSCLK_DIV_MASK)
 #define STC_TXCLK_SRC_OFFSET		8
 #define STC_TXCLK_SRC_MASK		(0x7 << STC_TXCLK_SRC_OFFSET)
 #define STC_TXCLK_SRC_SET(x)		((x << STC_TXCLK_SRC_OFFSET) & STC_TXCLK_SRC_MASK)
diff --git a/sound/soc/intel/sst-dsp-priv.h b/sound/soc/intel/sst-dsp-priv.h
index fe8e81aad646..30ca14a6a835 100644
--- a/sound/soc/intel/sst-dsp-priv.h
+++ b/sound/soc/intel/sst-dsp-priv.h
@@ -136,7 +136,7 @@ struct sst_module_data {
 	enum sst_data_type data_type;	/* type of module data */
 
 	u32 size;		/* size in bytes */
-	u32 offset;		/* offset in FW file */
+	int32_t offset;		/* offset in FW file */
 	u32 data_offset;	/* offset in ADSP memory space */
 	void *data;		/* module data */
 };
diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c
index f46bb4ddde6f..50e4246d4b57 100644
--- a/sound/soc/intel/sst-haswell-ipc.c
+++ b/sound/soc/intel/sst-haswell-ipc.c
@@ -617,7 +617,7 @@ static void hsw_notification_work(struct work_struct *work)
 	case IPC_POSITION_CHANGED:
 		trace_ipc_notification("DSP stream position changed for",
 			stream->reply.stream_hw_id);
-		sst_dsp_inbox_read(hsw->dsp, pos, sizeof(pos));
+		sst_dsp_inbox_read(hsw->dsp, pos, sizeof(*pos));
 
 		if (stream->notify_position)
 			stream->notify_position(stream, stream->pdata);
@@ -991,7 +991,8 @@ int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream
 		return -EINVAL;
 
 	sst_dsp_read(hsw->dsp, volume,
-		stream->reply.volume_register_address[channel], sizeof(volume));
+		stream->reply.volume_register_address[channel],
+		sizeof(*volume));
 
 	return 0;
 }
@@ -1609,7 +1610,7 @@ int sst_hsw_dx_set_state(struct sst_hsw *hsw,
 	trace_ipc_request("PM enter Dx state", state);
 
 	ret = ipc_tx_message_wait(hsw, header, &state_, sizeof(state_),
-		dx, sizeof(dx));
+		dx, sizeof(*dx));
 	if (ret < 0) {
 		dev_err(hsw->dev, "ipc: error set dx state %d failed\n", state);
 		return ret;
diff --git a/sound/soc/jz4740/Makefile b/sound/soc/jz4740/Makefile
index be873c1b0c20..d32c540555c4 100644
--- a/sound/soc/jz4740/Makefile
+++ b/sound/soc/jz4740/Makefile
@@ -1,10 +1,8 @@
 #
 # Jz4740 Platform Support
 #
-snd-soc-jz4740-objs := jz4740-pcm.o
 snd-soc-jz4740-i2s-objs := jz4740-i2s.o
 
-obj-$(CONFIG_SND_JZ4740_SOC) += snd-soc-jz4740.o
 obj-$(CONFIG_SND_JZ4740_SOC_I2S) += snd-soc-jz4740-i2s.o
 
 # Jz4740 Machine Support
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index 6232b7d307aa..4d0720ed5a90 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -258,7 +258,7 @@ static int rsnd_src_init(struct rsnd_mod *mod,
 {
 	struct rsnd_src *src = rsnd_mod_to_src(mod);
 
-	clk_enable(src->clk);
+	clk_prepare_enable(src->clk);
 
 	return 0;
 }
@@ -269,7 +269,7 @@ static int rsnd_src_quit(struct rsnd_mod *mod,
 {
 	struct rsnd_src *src = rsnd_mod_to_src(mod);
 
-	clk_disable(src->clk);
+	clk_disable_unprepare(src->clk);
 
 	return 0;
 }
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 4b7e20603dd7..1d8387c25bd8 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -171,7 +171,7 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi,
 	u32 cr;
 
 	if (0 == ssi->usrcnt) {
-		clk_enable(ssi->clk);
+		clk_prepare_enable(ssi->clk);
 
 		if (rsnd_dai_is_clk_master(rdai)) {
 			if (rsnd_ssi_clk_from_parent(ssi))
@@ -230,7 +230,7 @@ static void rsnd_ssi_hw_stop(struct rsnd_ssi *ssi,
 				rsnd_ssi_master_clk_stop(ssi);
 		}
 
-		clk_disable(ssi->clk);
+		clk_disable_unprepare(ssi->clk);
 	}
 
 	dev_dbg(dev, "ssi%d hw stopped\n", rsnd_mod_id(&ssi->mod));
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index c8a780d0d057..7769b0a2bc5a 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -254,7 +254,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
 static void dapm_kcontrol_free(struct snd_kcontrol *kctl)
 {
 	struct dapm_kcontrol_data *data = snd_kcontrol_chip(kctl);
-	kfree(data->widget);
 	kfree(data->wlist);
 	kfree(data);
 }
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 893d5a1afc3c..c3b5b7dca1c3 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -651,7 +651,7 @@ int snd_usb_autoresume(struct snd_usb_audio *chip)
 	int err = -ENODEV;
 
 	down_read(&chip->shutdown_rwsem);
-	if (chip->probing)
+	if (chip->probing && chip->in_pm)
 		err = 0;
 	else if (!chip->shutdown)
 		err = usb_autopm_get_interface(chip->pm_intf);
@@ -663,7 +663,7 @@ int snd_usb_autoresume(struct snd_usb_audio *chip)
 void snd_usb_autosuspend(struct snd_usb_audio *chip)
 {
 	down_read(&chip->shutdown_rwsem);
-	if (!chip->shutdown && !chip->probing)
+	if (!chip->shutdown && !chip->probing && !chip->in_pm)
 		usb_autopm_put_interface(chip->pm_intf);
 	up_read(&chip->shutdown_rwsem);
 }
@@ -695,8 +695,9 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message)
 			chip->autosuspended = 1;
 	}
 
-	list_for_each_entry(mixer, &chip->mixer_list, list)
-		snd_usb_mixer_suspend(mixer);
+	if (chip->num_suspended_intf == 1)
+		list_for_each_entry(mixer, &chip->mixer_list, list)
+			snd_usb_mixer_suspend(mixer);
 
 	return 0;
 }
@@ -711,6 +712,8 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume)
 		return 0;
 	if (--chip->num_suspended_intf)
 		return 0;
+
+	chip->in_pm = 1;
 	/*
 	 * ALSA leaves material resumption to user space
 	 * we just notify and restart the mixers
@@ -726,6 +729,7 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume)
 	chip->autosuspended = 0;
 
 err_out:
+	chip->in_pm = 0;
 	return err;
 }
 
diff --git a/sound/usb/card.h b/sound/usb/card.h
index 9867ab866857..97acb906acc2 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -92,6 +92,7 @@ struct snd_usb_endpoint {
 	unsigned int curframesize;      /* current packet size in frames (for capture) */
 	unsigned int syncmaxsize;	/* sync endpoint packet size */
 	unsigned int fill_max:1;	/* fill max packet size always */
+	unsigned int udh01_fb_quirk:1;	/* corrupted feedback data */
 	unsigned int datainterval;      /* log_2 of data packet interval */
 	unsigned int syncinterval;	/* P for adaptive mode, 0 otherwise */
 	unsigned char silence_value;
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index e70a87e0d9fe..289f582c9130 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -471,6 +471,10 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip,
 			ep->syncinterval = 3;
 
 		ep->syncmaxsize = le16_to_cpu(get_endpoint(alts, 1)->wMaxPacketSize);
+
+		if (chip->usb_id == USB_ID(0x0644, 0x8038) /* TEAC UD-H01 */ &&
+		    ep->syncmaxsize == 4)
+			ep->udh01_fb_quirk = 1;
 	}
 
 	list_add_tail(&ep->list, &chip->ep_list);
@@ -1105,7 +1109,16 @@ void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep,
 	if (f == 0)
 		return;
 
-	if (unlikely(ep->freqshift == INT_MIN)) {
+	if (unlikely(sender->udh01_fb_quirk)) {
+		/*
+		 * The TEAC UD-H01 firmware sometimes changes the feedback value
+		 * by +/- 0x1.0000.
+		 */
+		if (f < ep->freqn - 0x8000)
+			f += 0x10000;
+		else if (f > ep->freqn + 0x8000)
+			f -= 0x10000;
+	} else if (unlikely(ep->freqshift == INT_MIN)) {
 		/*
 		 * The first time we see a feedback value, determine its format
 		 * by shifting it left or right until it matches the nominal
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 131336d40492..c62a1659106d 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -1501,9 +1501,8 @@ static void retire_playback_urb(struct snd_usb_substream *subs,
 	 * The error should be lower than 2ms since the estimate relies
 	 * on two reads of a counter updated every ms.
 	 */
-	if (printk_ratelimit() &&
-	    abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2)
-		dev_dbg(&subs->dev->dev,
+	if (abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2)
+		dev_dbg_ratelimited(&subs->dev->dev,
 			"delay: estimated %d, actual %d\n",
 			est_delay, subs->last_delay);
 
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 25c4c7e217de..91d0380431b4 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -40,6 +40,7 @@ struct snd_usb_audio {
 	struct rw_semaphore shutdown_rwsem;
 	unsigned int shutdown:1;
 	unsigned int probing:1;
+	unsigned int in_pm:1;
 	unsigned int autosuspended:1;	
 	unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */
generated by cgit-pink 1.4.1 (git 2.36.1) at 2024-10-10 05:13:09 +0000