[PATCH AUTOSEL 6.14 01/20] ASoC: imx-card: Adjust over allocation of memory in imx_card_parse_of()

[PATCH AUTOSEL 6.14 01/20] ASoC: imx-card: Adjust over allocation of memory in imx_card_parse_of()

[Sasha Levin]

From: Chenyuan Yang <chenyuan0y@gmail.com>

[ Upstream commit a9a69c3b38c89d7992fb53db4abb19104b531d32 ]

Incorrect types are used as sizeof() arguments in devm_kcalloc().
It should be sizeof(dai_link_data) for link_data instead of
sizeof(snd_soc_dai_link).

This is found by our static analysis tool.

Signed-off-by: Chenyuan Yang <chenyuan0y@gmail.com>
Link: https://patch.msgid.link/20250406210854.149316-1-chenyuan0y@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/fsl/imx-card.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/fsl/imx-card.c b/sound/soc/fsl/imx-card.c
index 21f617f6f9fa8..566214cb3d60c 100644
--- a/sound/soc/fsl/imx-card.c
+++ b/sound/soc/fsl/imx-card.c
@@ -543,7 +543,7 @@ static int imx_card_parse_of(struct imx_card_data *data)
 	if (!card->dai_link)
 		return -ENOMEM;
 
-	data->link_data = devm_kcalloc(dev, num_links, sizeof(*link), GFP_KERNEL);
+	data->link_data = devm_kcalloc(dev, num_links, sizeof(*link_data), GFP_KERNEL);
 	if (!data->link_data)
 		return -ENOMEM;
 
-- 
2.39.5

[PATCH AUTOSEL 6.14 08/20] ASoC: cs42l43: Disable headphone clamps during type detection

[Sasha Levin]

From: Charles Keepax <ckeepax@opensource.cirrus.com>

[ Upstream commit 70ad2e6bd180f94be030aef56e59693e36d945f3 ]

The headphone clamps cause fairly loud pops during type detect
because they sink current from the detection process itself. Disable
the clamps whilst the type detect runs, to improve the detection
pop performance.

Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20250423090944.1504538-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/codecs/cs42l43-jack.c | 7 +++++++
 1 file changed, 7 insertions(+)

diff --git a/sound/soc/codecs/cs42l43-jack.c b/sound/soc/codecs/cs42l43-jack.c
index 20e6ab6f0d4ad..6165ac16c3a95 100644
--- a/sound/soc/codecs/cs42l43-jack.c
+++ b/sound/soc/codecs/cs42l43-jack.c
@@ -654,6 +654,10 @@ static int cs42l43_run_type_detect(struct cs42l43_codec *priv)
 
 	reinit_completion(&priv->type_detect);
 
+	regmap_update_bits(cs42l43->regmap, CS42L43_STEREO_MIC_CLAMP_CTRL,
+			   CS42L43_SMIC_HPAMP_CLAMP_DIS_FRC_VAL_MASK,
+			   CS42L43_SMIC_HPAMP_CLAMP_DIS_FRC_VAL_MASK);
+
 	cs42l43_start_hs_bias(priv, true);
 	regmap_update_bits(cs42l43->regmap, CS42L43_HS2,
 			   CS42L43_HSDET_MODE_MASK, 0x3 << CS42L43_HSDET_MODE_SHIFT);
@@ -665,6 +669,9 @@ static int cs42l43_run_type_detect(struct cs42l43_codec *priv)
 			   CS42L43_HSDET_MODE_MASK, 0x2 << CS42L43_HSDET_MODE_SHIFT);
 	cs42l43_stop_hs_bias(priv);
 
+	regmap_update_bits(cs42l43->regmap, CS42L43_STEREO_MIC_CLAMP_CTRL,
+			   CS42L43_SMIC_HPAMP_CLAMP_DIS_FRC_VAL_MASK, 0);
+
 	if (!time_left)
 		return -ETIMEDOUT;
 
-- 
2.39.5

[PATCH AUTOSEL 6.14 09/20] ASoC: Intel: bytcr_rt5640: Add DMI quirk for Acer Aspire SW3-013

[Sasha Levin]

From: Takashi Iwai <tiwai@suse.de>

[ Upstream commit a549b927ea3f5e50b1394209b64e6e17e31d4db8 ]

Acer Aspire SW3-013 requires the very same quirk as other Acer Aspire
model for making it working.

Link: https://bugzilla.kernel.org/show_bug.cgi?id=220011
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250420085716.12095-1-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/intel/boards/bytcr_rt5640.c | 13 +++++++++++++
 1 file changed, 13 insertions(+)

diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index 6446cda0f8572..0f3b8f44e7011 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -576,6 +576,19 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
 					BYT_RT5640_SSP0_AIF2 |
 					BYT_RT5640_MCLK_EN),
 	},
+	{       /* Acer Aspire SW3-013 */
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "Acer"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "Aspire SW3-013"),
+		},
+		.driver_data = (void *)(BYT_RT5640_DMIC1_MAP |
+					BYT_RT5640_JD_SRC_JD2_IN4N |
+					BYT_RT5640_OVCD_TH_2000UA |
+					BYT_RT5640_OVCD_SF_0P75 |
+					BYT_RT5640_DIFF_MIC |
+					BYT_RT5640_SSP0_AIF1 |
+					BYT_RT5640_MCLK_EN),
+	},
 	{
 		.matches = {
 			DMI_MATCH(DMI_SYS_VENDOR, "Acer"),
-- 
2.39.5

[PATCH AUTOSEL 6.14 10/20] ALSA: hda/realtek: Add quirk for HP Spectre x360 15-df1xxx

[Sasha Levin]

From: Takashi Iwai <tiwai@suse.de>

[ Upstream commit be0c40da888840fe91b45474cb70779e6cbaf7ca ]

HP Spectre x360 15-df1xxx with SSID 13c:863e requires similar
workarounds that were applied to another HP Spectre x360 models;
it has a mute LED only, no micmute LEDs, and needs the speaker GPIO
seup.

Link: https://bugzilla.kernel.org/show_bug.cgi?id=220054
Link: https://patch.msgid.link/20250427081035.11567-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/pci/hda/patch_realtek.c | 42 +++++++++++++++++++++++++++++++++++
 1 file changed, 42 insertions(+)

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 356df48c97309..6e3f0b031a41c 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6962,6 +6962,41 @@ static void alc285_fixup_hp_spectre_x360_eb1(struct hda_codec *codec,
 	}
 }
 
+/* GPIO1 = amplifier on/off */
+static void alc285_fixup_hp_spectre_x360_df1(struct hda_codec *codec,
+					     const struct hda_fixup *fix,
+					     int action)
+{
+	struct alc_spec *spec = codec->spec;
+	static const hda_nid_t conn[] = { 0x02 };
+	static const struct hda_pintbl pincfgs[] = {
+		{ 0x14, 0x90170110 },  /* front/high speakers */
+		{ 0x17, 0x90170130 },  /* back/bass speakers */
+		{ }
+	};
+
+	// enable mute led
+	alc285_fixup_hp_mute_led_coefbit(codec, fix, action);
+
+	switch (action) {
+	case HDA_FIXUP_ACT_PRE_PROBE:
+		/* needed for amp of back speakers */
+		spec->gpio_mask |= 0x01;
+		spec->gpio_dir |= 0x01;
+		snd_hda_apply_pincfgs(codec, pincfgs);
+		/* share DAC to have unified volume control */
+		snd_hda_override_conn_list(codec, 0x14, ARRAY_SIZE(conn), conn);
+		snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn);
+		break;
+	case HDA_FIXUP_ACT_INIT:
+		/* need to toggle GPIO to enable the amp of back speakers */
+		alc_update_gpio_data(codec, 0x01, true);
+		msleep(100);
+		alc_update_gpio_data(codec, 0x01, false);
+		break;
+	}
+}
+
 static void alc285_fixup_hp_spectre_x360(struct hda_codec *codec,
 					  const struct hda_fixup *fix, int action)
 {
@@ -7718,6 +7753,7 @@ enum {
 	ALC280_FIXUP_HP_9480M,
 	ALC245_FIXUP_HP_X360_AMP,
 	ALC285_FIXUP_HP_SPECTRE_X360_EB1,
+	ALC285_FIXUP_HP_SPECTRE_X360_DF1,
 	ALC285_FIXUP_HP_ENVY_X360,
 	ALC288_FIXUP_DELL_HEADSET_MODE,
 	ALC288_FIXUP_DELL1_MIC_NO_PRESENCE,
@@ -9786,6 +9822,10 @@ static const struct hda_fixup alc269_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc285_fixup_hp_spectre_x360_eb1
 	},
+	[ALC285_FIXUP_HP_SPECTRE_X360_DF1] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc285_fixup_hp_spectre_x360_df1
+	},
 	[ALC285_FIXUP_HP_ENVY_X360] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc285_fixup_hp_envy_x360,
@@ -10509,6 +10549,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x86c1, "HP Laptop 15-da3001TU", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2),
 	SND_PCI_QUIRK(0x103c, 0x86c7, "HP Envy AiO 32", ALC274_FIXUP_HP_ENVY_GPIO),
 	SND_PCI_QUIRK(0x103c, 0x86e7, "HP Spectre x360 15-eb0xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1),
+	SND_PCI_QUIRK(0x103c, 0x863e, "HP Spectre x360 15-df1xxx", ALC285_FIXUP_HP_SPECTRE_X360_DF1),
 	SND_PCI_QUIRK(0x103c, 0x86e8, "HP Spectre x360 15-eb0xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1),
 	SND_PCI_QUIRK(0x103c, 0x86f9, "HP Spectre x360 13-aw0xxx", ALC285_FIXUP_HP_SPECTRE_X360_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x8716, "HP Elite Dragonfly G2 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT),
@@ -11422,6 +11463,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{.id = ALC295_FIXUP_HP_OMEN, .name = "alc295-hp-omen"},
 	{.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"},
 	{.id = ALC285_FIXUP_HP_SPECTRE_X360_EB1, .name = "alc285-hp-spectre-x360-eb1"},
+	{.id = ALC285_FIXUP_HP_SPECTRE_X360_DF1, .name = "alc285-hp-spectre-x360-df1"},
 	{.id = ALC285_FIXUP_HP_ENVY_X360, .name = "alc285-hp-envy-x360"},
 	{.id = ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, .name = "alc287-ideapad-bass-spk-amp"},
 	{.id = ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN, .name = "alc287-yoga9-bass-spk-pin"},
-- 
2.39.5

[PATCH AUTOSEL 6.14 14/20] ALSA: usb-audio: Fix duplicated name in MIDI substream names

[Sasha Levin]

From: Takashi Iwai <tiwai@suse.de>

[ Upstream commit 0759e77a6d9bd34a874da73721ce4a7dc6665023 ]

The MIDI substream name string is constructed from the combination of
the card shortname (which is taken from USB iProduct) and the USB
iJack.  The problem is that some devices put the product name to the
iJack field, too.  For example, aplaymidi -l output on the Lanchkey MK
49 are like:

  % aplaymidi -l
  Port    Client name            Port name
  44:0    Launchkey MK4 49       Launchkey MK4 49 Launchkey MK4
  44:1    Launchkey MK4 49       Launchkey MK4 49 Launchkey MK4

where the actual iJack name can't be seen because it's truncated due
to the doubly words.

For resolving those situations, this patch compares the iJack string
with the card shortname, and drops if both start with the same words.
Then the result becomes like:

  % aplaymidi -l
  Port    Client name            Port name
  40:0    Launchkey MK4 49       Launchkey MK4 49 MIDI In
  40:1    Launchkey MK4 49       Launchkey MK4 49 DAW In

A caveat is that there are some pre-defined names for certain
devices in the driver code, and this workaround shouldn't be applied
to them.  Similarly, when the iJack isn't specified, we should skip
this check, too.  The patch added those checks in addition to the
string comparison.

Suggested-by: Paul Davis <paul@linuxaudiosystems.com>
Tested-by: Paul Davis <paul@linuxaudiosystems.com>
Link: https://lore.kernel.org/CAFa_cKmEDQWcJatbYWi6A58Zg4Ma9_6Nr3k5LhqwyxC-P_kXtw@mail.gmail.com
Link: https://patch.msgid.link/20250429183626.20773-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/usb/midi.c | 16 ++++++++++++----
 1 file changed, 12 insertions(+), 4 deletions(-)

diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 826ac870f2469..a792ada18863a 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -1885,10 +1885,18 @@ static void snd_usbmidi_init_substream(struct snd_usb_midi *umidi,
 	}
 
 	port_info = find_port_info(umidi, number);
-	name_format = port_info ? port_info->name :
-		(jack_name != default_jack_name  ? "%s %s" : "%s %s %d");
-	snprintf(substream->name, sizeof(substream->name),
-		 name_format, umidi->card->shortname, jack_name, number + 1);
+	if (port_info || jack_name == default_jack_name ||
+	    strncmp(umidi->card->shortname, jack_name, strlen(umidi->card->shortname)) != 0) {
+		name_format = port_info ? port_info->name :
+			(jack_name != default_jack_name  ? "%s %s" : "%s %s %d");
+		snprintf(substream->name, sizeof(substream->name),
+			 name_format, umidi->card->shortname, jack_name, number + 1);
+	} else {
+		/* The manufacturer included the iProduct name in the jack
+		 * name, do not use both
+		 */
+		strscpy(substream->name, jack_name);
+	}
 
 	*rsubstream = substream;
 }
-- 
2.39.5

[PATCH AUTOSEL 6.14 19/20] ASoC: intel/sdw_utils: Add volume limit to cs42l43 speakers

[Sasha Levin]

From: Stefan Binding <sbinding@opensource.cirrus.com>

[ Upstream commit 02b44a2b2bdcee03cbb92484d31e9ca1b91b2a38 ]

The volume control for cs42l43 speakers has a maximum gain of +31.5 dB.
However, for many use cases, this can cause distorted audio, depending
various factors, such as other signal-processing elements in the chain,
for example if the audio passes through a gain control before reaching
the codec or the signal path has been tuned for a particular maximum
gain in the codec.

In the case of systems which use the soc_sdw_cs42l43 driver, audio will
likely be distorted in all cases above 0 dB, therefore add a volume
limit of 128, which is 0 dB maximum volume inside this driver.

Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20250430103134.24579-2-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/sdw_utils/soc_sdw_cs42l43.c | 10 ++++++++++
 1 file changed, 10 insertions(+)

diff --git a/sound/soc/sdw_utils/soc_sdw_cs42l43.c b/sound/soc/sdw_utils/soc_sdw_cs42l43.c
index 668c9d28a1c12..b415d45d520d0 100644
--- a/sound/soc/sdw_utils/soc_sdw_cs42l43.c
+++ b/sound/soc/sdw_utils/soc_sdw_cs42l43.c
@@ -20,6 +20,8 @@
 #include <sound/soc-dapm.h>
 #include <sound/soc_sdw_utils.h>
 
+#define CS42L43_SPK_VOLUME_0DB	128 /* 0dB Max */
+
 static const struct snd_soc_dapm_route cs42l43_hs_map[] = {
 	{ "Headphone", NULL, "cs42l43 AMP3_OUT" },
 	{ "Headphone", NULL, "cs42l43 AMP4_OUT" },
@@ -117,6 +119,14 @@ int asoc_sdw_cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_so
 			return -ENOMEM;
 	}
 
+	ret = snd_soc_limit_volume(card, "cs42l43 Speaker Digital Volume",
+				   CS42L43_SPK_VOLUME_0DB);
+	if (ret)
+		dev_err(card->dev, "cs42l43 speaker volume limit failed: %d\n", ret);
+	else
+		dev_info(card->dev, "Setting CS42L43 Speaker volume limit to %d\n",
+			 CS42L43_SPK_VOLUME_0DB);
+
 	ret = snd_soc_dapm_add_routes(&card->dapm, cs42l43_spk_map,
 				      ARRAY_SIZE(cs42l43_spk_map));
 	if (ret)
-- 
2.39.5

[PATCH AUTOSEL 6.14 20/20] ASoC: intel/sdw_utils: Add volume limit to cs35l56 speakers

[Sasha Levin]

From: Stefan Binding <sbinding@opensource.cirrus.com>

[ Upstream commit d5463e531c128ff1b141fdba2e13345cd50028a4 ]

The volume control for cs35l56 speakers has a maximum gain of +12 dB.
However, for many use cases, this can cause distorted audio, depending
various factors, such as other signal-processing elements in the chain,
for example if the audio passes through a gain control before reaching
the amp or the signal path has been tuned for a particular maximum
gain in the amp.

In the case of systems which use the soc_sdw_* driver, audio will
likely be distorted in all cases above 0 dB, therefore add a volume
limit of 400, which is 0 dB maximum volume inside this driver.

The volume limit should be applied to both soundwire and soundwire
bridge configurations.

Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://patch.msgid.link/20250430103134.24579-3-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 include/sound/soc_sdw_utils.h                |  1 +
 sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c |  4 ++++
 sound/soc/sdw_utils/soc_sdw_cs_amp.c         | 24 ++++++++++++++++++++
 3 files changed, 29 insertions(+)

diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h
index 36a4a1e1d8ca2..d8bd5d37131aa 100644
--- a/include/sound/soc_sdw_utils.h
+++ b/include/sound/soc_sdw_utils.h
@@ -226,6 +226,7 @@ int asoc_sdw_cs_amp_init(struct snd_soc_card *card,
 			 bool playback);
 int asoc_sdw_cs_spk_feedback_rtd_init(struct snd_soc_pcm_runtime *rtd,
 				      struct snd_soc_dai *dai);
+int asoc_sdw_cs35l56_volume_limit(struct snd_soc_card *card, const char *name_prefix);
 
 /* MAXIM codec support */
 int asoc_sdw_maxim_init(struct snd_soc_card *card,
diff --git a/sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c b/sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c
index 246e5c2e0af55..c7e55f4433514 100644
--- a/sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c
+++ b/sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c
@@ -60,6 +60,10 @@ static int asoc_sdw_bridge_cs35l56_asp_init(struct snd_soc_pcm_runtime *rtd)
 
 	/* 4 x 16-bit sample slots and FSYNC=48000, BCLK=3.072 MHz */
 	for_each_rtd_codec_dais(rtd, i, codec_dai) {
+		ret = asoc_sdw_cs35l56_volume_limit(card, codec_dai->component->name_prefix);
+		if (ret)
+			return ret;
+
 		ret = snd_soc_dai_set_tdm_slot(codec_dai, tx_mask, rx_mask, 4, 16);
 		if (ret < 0)
 			return ret;
diff --git a/sound/soc/sdw_utils/soc_sdw_cs_amp.c b/sound/soc/sdw_utils/soc_sdw_cs_amp.c
index 4b6181cf29716..35b550bcd4ded 100644
--- a/sound/soc/sdw_utils/soc_sdw_cs_amp.c
+++ b/sound/soc/sdw_utils/soc_sdw_cs_amp.c
@@ -16,6 +16,25 @@
 
 #define CODEC_NAME_SIZE	8
 #define CS_AMP_CHANNELS_PER_AMP	4
+#define CS35L56_SPK_VOLUME_0DB 400 /* 0dB Max */
+
+int asoc_sdw_cs35l56_volume_limit(struct snd_soc_card *card, const char *name_prefix)
+{
+	char *volume_ctl_name;
+	int ret;
+
+	volume_ctl_name = kasprintf(GFP_KERNEL, "%s Speaker Volume", name_prefix);
+	if (!volume_ctl_name)
+		return -ENOMEM;
+
+	ret = snd_soc_limit_volume(card, volume_ctl_name, CS35L56_SPK_VOLUME_0DB);
+	if (ret)
+		dev_err(card->dev, "%s limit set failed: %d\n", volume_ctl_name, ret);
+
+	kfree(volume_ctl_name);
+	return ret;
+}
+EXPORT_SYMBOL_NS(asoc_sdw_cs35l56_volume_limit, "SND_SOC_SDW_UTILS");
 
 int asoc_sdw_cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai)
 {
@@ -40,6 +59,11 @@ int asoc_sdw_cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai
 
 		snprintf(widget_name, sizeof(widget_name), "%s SPK",
 			 codec_dai->component->name_prefix);
+
+		ret = asoc_sdw_cs35l56_volume_limit(card, codec_dai->component->name_prefix);
+		if (ret)
+			return ret;
+
 		ret = snd_soc_dapm_add_routes(&card->dapm, &route, 1);
 		if (ret)
 			return ret;
-- 
2.39.5