diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h
index a8bebac1e4b28dd3d0195894dc96e18f74184992..957295364a5e3c1aa3bc8a9108a7da02e7b6ee44 100644
--- a/include/sound/hdaudio_ext.h
+++ b/include/sound/hdaudio_ext.h
@@ -56,6 +56,9 @@ struct hdac_ext_stream {
u32 pphcldpl;
u32 pphcldpu;
+ u32 pplcllpl;
+ u32 pplcllpu;
+
bool decoupled:1;
bool link_locked:1;
bool link_prepared;
diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h
index 4038dd421150a3f03182be5d1f0a503812029c15..1dc59005d241fbe683fb98d62d86e8c04f3708eb 100644
--- a/include/sound/tas2781-tlv.h
+++ b/include/sound/tas2781-tlv.h
@@ -15,7 +15,7 @@
#ifndef __TAS2781_TLV_H__
#define __TAS2781_TLV_H__
-static const DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0);
+static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0);
static const DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0);
#endif
diff --git a/sound/oss/dmasound/dmasound_paula.c b/sound/oss/dmasound/dmasound_paula.c
index 0ba8f0c4cd99a27aaff1d3b43edd306daf0ad857..3a593da09280dca9e5b59dc96c6c2cb27cac6b6b 100644
--- a/sound/oss/dmasound/dmasound_paula.c
+++ b/sound/oss/dmasound/dmasound_paula.c
@@ -725,7 +725,13 @@ static void __exit amiga_audio_remove(struct platform_device *pdev)
dmasound_deinit();
}
-static struct platform_driver amiga_audio_driver = {
+/*
+ * amiga_audio_remove() lives in .exit.text. For drivers registered via
+ * module_platform_driver_probe() this is ok because they cannot get unbound at
+ * runtime. So mark the driver struct with __refdata to prevent modpost
+ * triggering a section mismatch warning.
+ */
+static struct platform_driver amiga_audio_driver __refdata = {
.remove_new = __exit_p(amiga_audio_remove),
.driver = {
.name = "amiga-audio",
diff --git a/sound/pci/emu10k1/emu10k1_callback.c b/sound/pci/emu10k1/emu10k1_callback.c
index d36234b88fb4219fec68e3f68103d01ee4c224ab..941bfbf812ed305bbfb368771d66134703ba8bea 100644
--- a/sound/pci/emu10k1/emu10k1_callback.c
+++ b/sound/pci/emu10k1/emu10k1_callback.c
@@ -255,7 +255,7 @@ lookup_voices(struct snd_emux *emu, struct snd_emu10k1 *hw,
/* check if sample is finished playing (non-looping only) */
if (bp != best + V_OFF && bp != best + V_FREE &&
(vp->reg.sample_mode & SNDRV_SFNT_SAMPLE_SINGLESHOT)) {
- val = snd_emu10k1_ptr_read(hw, CCCA_CURRADDR, vp->ch) - 64;
+ val = snd_emu10k1_ptr_read(hw, CCCA_CURRADDR, vp->ch);
if (val >= vp->reg.loopstart)
bp = best + V_OFF;
}
@@ -362,7 +362,7 @@ start_voice(struct snd_emux_voice *vp)
map = (hw->silent_page.addr << hw->address_mode) | (hw->address_mode ? MAP_PTI_MASK1 : MAP_PTI_MASK0);
- addr = vp->reg.start + 64;
+ addr = vp->reg.start;
temp = vp->reg.parm.filterQ;
ccca = (temp << 28) | addr;
if (vp->apitch < 0xe400)
@@ -430,9 +430,6 @@ start_voice(struct snd_emux_voice *vp)
/* Q & current address (Q 4bit value, MSB) */
CCCA, ccca,
- /* cache */
- CCR, REG_VAL_PUT(CCR_CACHEINVALIDSIZE, 64),
-
/* reset volume */
VTFT, vtarget | vp->ftarget,
CVCF, vtarget | CVCF_CURRENTFILTER_MASK,
diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c
index 72ec872afb8d27de1d2b23288988bf4a5e4f4b88..8fb688e4141485cdf1d29ba73ed25ddc17a2a936 100644
--- a/sound/pci/hda/cs35l41_hda_property.c
+++ b/sound/pci/hda/cs35l41_hda_property.c
@@ -108,7 +108,10 @@ static const struct cs35l41_config cs35l41_config_table[] = {
{ "10431F12", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 },
{ "10431F1F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 },
{ "10431F62", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 },
+ { "10433A60", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 },
{ "17AA386F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 },
+ { "17AA3877", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 },
+ { "17AA3878", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 },
{ "17AA38A9", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 2, -1, 0, 0, 0 },
{ "17AA38AB", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 2, -1, 0, 0, 0 },
{ "17AA38B4", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 },
@@ -496,7 +499,10 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = {
{ "CSC3551", "10431F12", generic_dsd_config },
{ "CSC3551", "10431F1F", generic_dsd_config },
{ "CSC3551", "10431F62", generic_dsd_config },
+ { "CSC3551", "10433A60", generic_dsd_config },
{ "CSC3551", "17AA386F", generic_dsd_config },
+ { "CSC3551", "17AA3877", generic_dsd_config },
+ { "CSC3551", "17AA3878", generic_dsd_config },
{ "CSC3551", "17AA38A9", generic_dsd_config },
{ "CSC3551", "17AA38AB", generic_dsd_config },
{ "CSC3551", "17AA38B4", generic_dsd_config },
diff --git a/sound/pci/hda/cs35l56_hda_i2c.c b/sound/pci/hda/cs35l56_hda_i2c.c
index 13beee807308f1763145cc1f9c1590e427236dc6..40f2f97944d54c916d58279a94f84e6be69b5bba 100644
--- a/sound/pci/hda/cs35l56_hda_i2c.c
+++ b/sound/pci/hda/cs35l56_hda_i2c.c
@@ -56,10 +56,19 @@ static const struct i2c_device_id cs35l56_hda_i2c_id[] = {
{}
};
+static const struct acpi_device_id cs35l56_acpi_hda_match[] = {
+ { "CSC3554", 0 },
+ { "CSC3556", 0 },
+ { "CSC3557", 0 },
+ {}
+};
+MODULE_DEVICE_TABLE(acpi, cs35l56_acpi_hda_match);
+
static struct i2c_driver cs35l56_hda_i2c_driver = {
.driver = {
- .name = "cs35l56-hda",
- .pm = &cs35l56_hda_pm_ops,
+ .name = "cs35l56-hda",
+ .acpi_match_table = cs35l56_acpi_hda_match,
+ .pm = &cs35l56_hda_pm_ops,
},
.id_table = cs35l56_hda_i2c_id,
.probe = cs35l56_hda_i2c_probe,
diff --git a/sound/pci/hda/cs35l56_hda_spi.c b/sound/pci/hda/cs35l56_hda_spi.c
index a3b2fa76663d3685cf404e59785cdc00c502e493..7f02155fe61e3cd529e4c688bf6d734848baf45d 100644
--- a/sound/pci/hda/cs35l56_hda_spi.c
+++ b/sound/pci/hda/cs35l56_hda_spi.c
@@ -56,10 +56,19 @@ static const struct spi_device_id cs35l56_hda_spi_id[] = {
{}
};
+static const struct acpi_device_id cs35l56_acpi_hda_match[] = {
+ { "CSC3554", 0 },
+ { "CSC3556", 0 },
+ { "CSC3557", 0 },
+ {}
+};
+MODULE_DEVICE_TABLE(acpi, cs35l56_acpi_hda_match);
+
static struct spi_driver cs35l56_hda_spi_driver = {
.driver = {
- .name = "cs35l56-hda",
- .pm = &cs35l56_hda_pm_ops,
+ .name = "cs35l56-hda",
+ .acpi_match_table = cs35l56_acpi_hda_match,
+ .pm = &cs35l56_hda_pm_ops,
},
.id_table = cs35l56_hda_spi_id,
.probe = cs35l56_hda_spi_probe,
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a17c36a36aa5375fd8295911a2ffc707cb14263e..cdcb28aa9d7bf028d429aeea0016cec7c6bc0c22 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6875,11 +6875,38 @@ static void alc287_fixup_legion_16ithg6_speakers(struct hda_codec *cdc, const st
comp_generic_fixup(cdc, action, "i2c", "CLSA0101", "-%s:00-cs35l41-hda.%d", 2);
}
+static void cs35l56_fixup_i2c_two(struct hda_codec *cdc, const struct hda_fixup *fix, int action)
+{
+ comp_generic_fixup(cdc, action, "i2c", "CSC3556", "-%s:00-cs35l56-hda.%d", 2);
+}
+
+static void cs35l56_fixup_i2c_four(struct hda_codec *cdc, const struct hda_fixup *fix, int action)
+{
+ comp_generic_fixup(cdc, action, "i2c", "CSC3556", "-%s:00-cs35l56-hda.%d", 4);
+}
+
+static void cs35l56_fixup_spi_two(struct hda_codec *cdc, const struct hda_fixup *fix, int action)
+{
+ comp_generic_fixup(cdc, action, "spi", "CSC3556", "-%s:00-cs35l56-hda.%d", 2);
+}
+
static void cs35l56_fixup_spi_four(struct hda_codec *cdc, const struct hda_fixup *fix, int action)
{
comp_generic_fixup(cdc, action, "spi", "CSC3556", "-%s:00-cs35l56-hda.%d", 4);
}
+static void alc285_fixup_asus_ga403u(struct hda_codec *cdc, const struct hda_fixup *fix, int action)
+{
+ /*
+ * The same SSID has been re-used in different hardware, they have
+ * different codecs and the newer GA403U has a ALC285.
+ */
+ if (cdc->core.vendor_id == 0x10ec0285)
+ cs35l56_fixup_i2c_two(cdc, fix, action);
+ else
+ alc_fixup_inv_dmic(cdc, fix, action);
+}
+
static void tas2781_fixup_i2c(struct hda_codec *cdc,
const struct hda_fixup *fix, int action)
{
@@ -7436,6 +7463,10 @@ enum {
ALC256_FIXUP_ACER_SFG16_MICMUTE_LED,
ALC256_FIXUP_HEADPHONE_AMP_VOL,
ALC245_FIXUP_HP_SPECTRE_X360_EU0XXX,
+ ALC285_FIXUP_CS35L56_SPI_2,
+ ALC285_FIXUP_CS35L56_I2C_2,
+ ALC285_FIXUP_CS35L56_I2C_4,
+ ALC285_FIXUP_ASUS_GA403U,
};
/* A special fixup for Lenovo C940 and Yoga Duet 7;
@@ -9643,6 +9674,22 @@ static const struct hda_fixup alc269_fixups[] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc245_fixup_hp_spectre_x360_eu0xxx,
},
+ [ALC285_FIXUP_CS35L56_SPI_2] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = cs35l56_fixup_spi_two,
+ },
+ [ALC285_FIXUP_CS35L56_I2C_2] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = cs35l56_fixup_i2c_two,
+ },
+ [ALC285_FIXUP_CS35L56_I2C_4] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = cs35l56_fixup_i2c_four,
+ },
+ [ALC285_FIXUP_ASUS_GA403U] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc285_fixup_asus_ga403u,
+ },
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -10096,7 +10143,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1a83, "ASUS UM5302LA", ALC294_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x1043, 0x1a8f, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2),
SND_PCI_QUIRK(0x1043, 0x1b11, "ASUS UX431DA", ALC294_FIXUP_ASUS_COEF_1B),
- SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x1b13, "ASUS U41SV/GA403U", ALC285_FIXUP_ASUS_GA403U),
SND_PCI_QUIRK(0x1043, 0x1b93, "ASUS G614JVR/JIR", ALC245_FIXUP_CS35L41_SPI_2),
SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1043, 0x1c03, "ASUS UM3406HA", ALC287_FIXUP_CS35L41_I2C_2),
@@ -10104,6 +10151,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1c33, "ASUS UX5304MA", ALC245_FIXUP_CS35L41_SPI_2),
SND_PCI_QUIRK(0x1043, 0x1c43, "ASUS UX8406MA", ALC245_FIXUP_CS35L41_SPI_2),
SND_PCI_QUIRK(0x1043, 0x1c62, "ASUS GU603", ALC289_FIXUP_ASUS_GA401),
+ SND_PCI_QUIRK(0x1043, 0x1c63, "ASUS GU605M", ALC285_FIXUP_CS35L56_SPI_2),
SND_PCI_QUIRK(0x1043, 0x1c92, "ASUS ROG Strix G15", ALC285_FIXUP_ASUS_G533Z_PINS),
SND_PCI_QUIRK(0x1043, 0x1c9f, "ASUS G614JU/JV/JI", ALC285_FIXUP_ASUS_HEADSET_MIC),
SND_PCI_QUIRK(0x1043, 0x1caf, "ASUS G634JY/JZ/JI/JG", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS),
@@ -10115,11 +10163,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401),
SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE),
SND_PCI_QUIRK(0x1043, 0x1da2, "ASUS UP6502ZA/ZD", ALC245_FIXUP_CS35L41_SPI_2),
+ SND_PCI_QUIRK(0x1043, 0x1df3, "ASUS UM5606", ALC285_FIXUP_CS35L56_I2C_4),
SND_PCI_QUIRK(0x1043, 0x1e02, "ASUS UX3402ZA", ALC245_FIXUP_CS35L41_SPI_2),
SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502),
SND_PCI_QUIRK(0x1043, 0x1e12, "ASUS UM3402", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS),
SND_PCI_QUIRK(0x1043, 0x1e5e, "ASUS ROG Strix G513", ALC294_FIXUP_ASUS_G513_PINS),
+ SND_PCI_QUIRK(0x1043, 0x1e63, "ASUS H7606W", ALC285_FIXUP_CS35L56_I2C_2),
+ SND_PCI_QUIRK(0x1043, 0x1e83, "ASUS GA605W", ALC285_FIXUP_CS35L56_I2C_2),
SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401),
SND_PCI_QUIRK(0x1043, 0x1ee2, "ASUS UM6702RA/RC", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x1043, 0x1c52, "ASUS Zephyrus G15 2022", ALC289_FIXUP_ASUS_GA401),
@@ -10133,7 +10184,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x3a30, "ASUS G814JVR/JIR", ALC245_FIXUP_CS35L41_SPI_2),
SND_PCI_QUIRK(0x1043, 0x3a40, "ASUS G814JZR", ALC245_FIXUP_CS35L41_SPI_2),
SND_PCI_QUIRK(0x1043, 0x3a50, "ASUS G834JYR/JZR", ALC245_FIXUP_CS35L41_SPI_2),
- SND_PCI_QUIRK(0x1043, 0x3a60, "ASUS G634JYR/JZR", ALC245_FIXUP_CS35L41_SPI_2),
+ SND_PCI_QUIRK(0x1043, 0x3a60, "ASUS G634JYR/JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS),
SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
@@ -10159,7 +10210,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x10ec, 0x1254, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK),
SND_PCI_QUIRK(0x10ec, 0x12cc, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK),
SND_PCI_QUIRK(0x10ec, 0x12f6, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK),
- SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE),
+ SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_ASPIRE_HEADSET_MIC),
SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC),
SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_AMP),
SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_AMP),
@@ -10333,6 +10384,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN),
SND_PCI_QUIRK(0x17aa, 0x386f, "Legion 7i 16IAX7", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x17aa, 0x3870, "Lenovo Yoga 7 14ARB7", ALC287_FIXUP_YOGA7_14ARB7_I2C),
+ SND_PCI_QUIRK(0x17aa, 0x3877, "Lenovo Legion 7 Slim 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2),
+ SND_PCI_QUIRK(0x17aa, 0x3878, "Lenovo Legion 7 Slim 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2),
SND_PCI_QUIRK(0x17aa, 0x387d, "Yoga S780-16 pro Quad AAC", ALC287_FIXUP_TAS2781_I2C),
SND_PCI_QUIRK(0x17aa, 0x387e, "Yoga S780-16 pro Quad YC", ALC287_FIXUP_TAS2781_I2C),
SND_PCI_QUIRK(0x17aa, 0x3881, "YB9 dual power mode2 YC", ALC287_FIXUP_TAS2781_I2C),
@@ -10403,6 +10456,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1d05, 0x1147, "TongFang GMxTGxx", ALC269_FIXUP_NO_SHUTUP),
SND_PCI_QUIRK(0x1d05, 0x115c, "TongFang GMxTGxx", ALC269_FIXUP_NO_SHUTUP),
SND_PCI_QUIRK(0x1d05, 0x121b, "TongFang GMxAGxx", ALC269_FIXUP_NO_SHUTUP),
+ SND_PCI_QUIRK(0x1d05, 0x1387, "TongFang GMxIXxx", ALC2XX_FIXUP_HEADSET_MIC),
SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC),
SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC),
diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c
index 8c8b1dcac6281c1d7448a529ba316d5346fd9030..5f35b90eab8d3f1aa46e6d4ea6bdd8d6d49638a7 100644
--- a/sound/soc/amd/acp/acp-pci.c
+++ b/sound/soc/amd/acp/acp-pci.c
@@ -115,7 +115,10 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id
goto unregister_dmic_dev;
}
- acp_init(chip);
+ ret = acp_init(chip);
+ if (ret)
+ goto unregister_dmic_dev;
+
res = devm_kcalloc(&pci->dev, num_res, sizeof(struct resource), GFP_KERNEL);
if (!res) {
ret = -ENOMEM;
@@ -133,11 +136,9 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id
}
}
- if (flag == FLAG_AMD_LEGACY_ONLY_DMIC) {
- ret = check_acp_pdm(pci, chip);
- if (ret < 0)
- goto skip_pdev_creation;
- }
+ ret = check_acp_pdm(pci, chip);
+ if (ret < 0)
+ goto skip_pdev_creation;
chip->flag = flag;
memset(&pdevinfo, 0, sizeof(pdevinfo));
diff --git a/sound/soc/codecs/cs-amp-lib.c b/sound/soc/codecs/cs-amp-lib.c
index 01ef4db5407da52cc04c6813ba07970c24d91541..287ac01a387357beffe2a8f76ab5f3ec15dc7b3d 100644
--- a/sound/soc/codecs/cs-amp-lib.c
+++ b/sound/soc/codecs/cs-amp-lib.c
@@ -56,6 +56,11 @@ static int _cs_amp_write_cal_coeffs(struct cs_dsp *dsp,
dev_dbg(dsp->dev, "Calibration: Ambient=%#x, Status=%#x, CalR=%d\n",
data->calAmbient, data->calStatus, data->calR);
+ if (list_empty(&dsp->ctl_list)) {
+ dev_info(dsp->dev, "Calibration disabled due to missing firmware controls\n");
+ return -ENOENT;
+ }
+
ret = cs_amp_write_cal_coeff(dsp, controls, controls->ambient, data->calAmbient);
if (ret)
return ret;
diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c
index 860d5cda67bffe83c5c4934497cdf8c1c510922b..94685449f0f48c9b7bd534b1947da07ab26fad53 100644
--- a/sound/soc/codecs/cs42l43.c
+++ b/sound/soc/codecs/cs42l43.c
@@ -2364,7 +2364,8 @@ static int cs42l43_codec_runtime_resume(struct device *dev)
static int cs42l43_codec_suspend(struct device *dev)
{
- struct cs42l43 *cs42l43 = dev_get_drvdata(dev);
+ struct cs42l43_codec *priv = dev_get_drvdata(dev);
+ struct cs42l43 *cs42l43 = priv->core;
disable_irq(cs42l43->irq);
@@ -2373,7 +2374,8 @@ static int cs42l43_codec_suspend(struct device *dev)
static int cs42l43_codec_suspend_noirq(struct device *dev)
{
- struct cs42l43 *cs42l43 = dev_get_drvdata(dev);
+ struct cs42l43_codec *priv = dev_get_drvdata(dev);
+ struct cs42l43 *cs42l43 = priv->core;
enable_irq(cs42l43->irq);
@@ -2382,7 +2384,8 @@ static int cs42l43_codec_suspend_noirq(struct device *dev)
static int cs42l43_codec_resume(struct device *dev)
{
- struct cs42l43 *cs42l43 = dev_get_drvdata(dev);
+ struct cs42l43_codec *priv = dev_get_drvdata(dev);
+ struct cs42l43 *cs42l43 = priv->core;
enable_irq(cs42l43->irq);
@@ -2391,7 +2394,8 @@ static int cs42l43_codec_resume(struct device *dev)
static int cs42l43_codec_resume_noirq(struct device *dev)
{
- struct cs42l43 *cs42l43 = dev_get_drvdata(dev);
+ struct cs42l43_codec *priv = dev_get_drvdata(dev);
+ struct cs42l43 *cs42l43 = priv->core;
disable_irq(cs42l43->irq);
diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c
index 15289dadafea091d2693149e600d72e0cbb975c0..17bd6b5160772e01d8597767868a9d2472cae276 100644
--- a/sound/soc/codecs/es8326.c
+++ b/sound/soc/codecs/es8326.c
@@ -412,9 +412,9 @@ static const struct _coeff_div coeff_div_v3[] = {
{125, 48000, 6000000, 0x04, 0x04, 0x1F, 0x2D, 0x8A, 0x0A, 0x27, 0x27},
{128, 8000, 1024000, 0x60, 0x00, 0x05, 0x75, 0x8A, 0x1B, 0x1F, 0x7F},
- {128, 16000, 2048000, 0x20, 0x00, 0x31, 0x35, 0x8A, 0x1B, 0x1F, 0x3F},
- {128, 44100, 5644800, 0xE0, 0x00, 0x01, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F},
- {128, 48000, 6144000, 0xE0, 0x00, 0x01, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F},
+ {128, 16000, 2048000, 0x20, 0x00, 0x31, 0x35, 0x08, 0x19, 0x1F, 0x3F},
+ {128, 44100, 5644800, 0xE0, 0x00, 0x01, 0x2D, 0x48, 0x08, 0x1F, 0x1F},
+ {128, 48000, 6144000, 0xE0, 0x00, 0x01, 0x2D, 0x48, 0x08, 0x1F, 0x1F},
{144, 8000, 1152000, 0x20, 0x00, 0x03, 0x35, 0x8A, 0x1B, 0x23, 0x47},
{144, 16000, 2304000, 0x20, 0x00, 0x11, 0x35, 0x8A, 0x1B, 0x23, 0x47},
{192, 8000, 1536000, 0x60, 0x02, 0x0D, 0x75, 0x8A, 0x1B, 0x1F, 0x7F},
@@ -423,10 +423,10 @@ static const struct _coeff_div coeff_div_v3[] = {
{200, 48000, 9600000, 0x04, 0x04, 0x0F, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F},
{250, 48000, 12000000, 0x04, 0x04, 0x0F, 0x2D, 0xCA, 0x0A, 0x27, 0x27},
- {256, 8000, 2048000, 0x60, 0x00, 0x31, 0x35, 0x8A, 0x1B, 0x1F, 0x7F},
- {256, 16000, 4096000, 0x20, 0x00, 0x01, 0x35, 0x8A, 0x1B, 0x1F, 0x3F},
- {256, 44100, 11289600, 0xE0, 0x00, 0x30, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F},
- {256, 48000, 12288000, 0xE0, 0x00, 0x30, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F},
+ {256, 8000, 2048000, 0x60, 0x00, 0x31, 0x35, 0x08, 0x19, 0x1F, 0x7F},
+ {256, 16000, 4096000, 0x20, 0x00, 0x01, 0x35, 0x08, 0x19, 0x1F, 0x3F},
+ {256, 44100, 11289600, 0xE0, 0x01, 0x01, 0x2D, 0x48, 0x08, 0x1F, 0x1F},
+ {256, 48000, 12288000, 0xE0, 0x01, 0x01, 0x2D, 0x48, 0x08, 0x1F, 0x1F},
{288, 8000, 2304000, 0x20, 0x00, 0x01, 0x35, 0x8A, 0x1B, 0x23, 0x47},
{384, 8000, 3072000, 0x60, 0x02, 0x05, 0x75, 0x8A, 0x1B, 0x1F, 0x7F},
{384, 16000, 6144000, 0x20, 0x02, 0x03, 0x35, 0x8A, 0x1B, 0x1F, 0x3F},
@@ -435,10 +435,10 @@ static const struct _coeff_div coeff_div_v3[] = {
{400, 48000, 19200000, 0xE4, 0x04, 0x35, 0x6d, 0xCA, 0x0A, 0x1F, 0x1F},
{500, 48000, 24000000, 0xF8, 0x04, 0x3F, 0x6D, 0xCA, 0x0A, 0x1F, 0x1F},
- {512, 8000, 4096000, 0x60, 0x00, 0x01, 0x35, 0x8A, 0x1B, 0x1F, 0x7F},
- {512, 16000, 8192000, 0x20, 0x00, 0x30, 0x35, 0x8A, 0x1B, 0x1F, 0x3F},
- {512, 44100, 22579200, 0xE0, 0x00, 0x00, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F},
- {512, 48000, 24576000, 0xE0, 0x00, 0x00, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F},
+ {512, 8000, 4096000, 0x60, 0x00, 0x01, 0x08, 0x19, 0x1B, 0x1F, 0x7F},
+ {512, 16000, 8192000, 0x20, 0x00, 0x30, 0x35, 0x08, 0x19, 0x1F, 0x3F},
+ {512, 44100, 22579200, 0xE0, 0x00, 0x00, 0x2D, 0x48, 0x08, 0x1F, 0x1F},
+ {512, 48000, 24576000, 0xE0, 0x00, 0x00, 0x2D, 0x48, 0x08, 0x1F, 0x1F},
{768, 8000, 6144000, 0x60, 0x02, 0x11, 0x35, 0x8A, 0x1B, 0x1F, 0x7F},
{768, 16000, 12288000, 0x20, 0x02, 0x01, 0x35, 0x8A, 0x1B, 0x1F, 0x3F},
{768, 32000, 24576000, 0xE0, 0x02, 0x30, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F},
@@ -835,7 +835,6 @@ static void es8326_jack_detect_handler(struct work_struct *work)
dev_dbg(comp->dev, "Report hp remove event\n");
snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET);
/* mute adc when mic path switch */
- regmap_write(es8326->regmap, ES8326_ADC_SCALE, 0x33);
regmap_write(es8326->regmap, ES8326_ADC1_SRC, 0x44);
regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66);
es8326->hp = 0;
@@ -843,6 +842,7 @@ static void es8326_jack_detect_handler(struct work_struct *work)
regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x01);
regmap_write(es8326->regmap, ES8326_SYS_BIAS, 0x0a);
regmap_update_bits(es8326->regmap, ES8326_HP_DRIVER_REF, 0x0f, 0x03);
+ regmap_write(es8326->regmap, ES8326_INT_SOURCE, ES8326_INT_SRC_PIN9);
/*
* Inverted HPJACK_POL bit to trigger one IRQ to double check HP Removal event
*/
@@ -865,6 +865,8 @@ static void es8326_jack_detect_handler(struct work_struct *work)
* set auto-check mode, then restart jack_detect_work after 400ms.
* Don't report jack status.
*/
+ regmap_write(es8326->regmap, ES8326_INT_SOURCE,
+ (ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON));
regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x01);
es8326_enable_micbias(es8326->component);
usleep_range(50000, 70000);
@@ -891,7 +893,6 @@ static void es8326_jack_detect_handler(struct work_struct *work)
snd_soc_jack_report(es8326->jack,
SND_JACK_HEADSET, SND_JACK_HEADSET);
- regmap_write(es8326->regmap, ES8326_ADC_SCALE, 0x33);
regmap_update_bits(es8326->regmap, ES8326_PGA_PDN,
0x08, 0x08);
regmap_update_bits(es8326->regmap, ES8326_PGAGAIN,
@@ -987,7 +988,7 @@ static int es8326_resume(struct snd_soc_component *component)
regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x0E);
regmap_write(es8326->regmap, ES8326_ANA_LP, 0xf0);
usleep_range(10000, 15000);
- regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0xe9);
+ regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0xd9);
regmap_write(es8326->regmap, ES8326_ANA_MICBIAS, 0xcb);
/* set headphone default type and detect pin */
regmap_write(es8326->regmap, ES8326_HPDET_TYPE, 0x83);
@@ -1038,8 +1039,7 @@ static int es8326_resume(struct snd_soc_component *component)
es8326_enable_micbias(es8326->component);
usleep_range(50000, 70000);
regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x00);
- regmap_write(es8326->regmap, ES8326_INT_SOURCE,
- (ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON));
+ regmap_write(es8326->regmap, ES8326_INT_SOURCE, ES8326_INT_SRC_PIN9);
regmap_write(es8326->regmap, ES8326_INTOUT_IO,
es8326->interrupt_clk);
regmap_write(es8326->regmap, ES8326_SDINOUT1_IO,
@@ -1060,6 +1060,8 @@ static int es8326_resume(struct snd_soc_component *component)
es8326->hp = 0;
es8326->hpl_vol = 0x03;
es8326->hpr_vol = 0x03;
+
+ es8326_irq(es8326->irq, es8326);
return 0;
}
@@ -1070,6 +1072,9 @@ static int es8326_suspend(struct snd_soc_component *component)
cancel_delayed_work_sync(&es8326->jack_detect_work);
es8326_disable_micbias(component);
es8326->calibrated = false;
+ regmap_write(es8326->regmap, ES8326_CLK_MUX, 0x2d);
+ regmap_write(es8326->regmap, ES8326_DAC2HPMIX, 0x00);
+ regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x3b);
regmap_write(es8326->regmap, ES8326_CLK_CTL, ES8326_CLK_OFF);
regcache_cache_only(es8326->regmap, true);
regcache_mark_dirty(es8326->regmap);
diff --git a/sound/soc/codecs/es8326.h b/sound/soc/codecs/es8326.h
index ee12caef810532380cdf1b5d6b0b204afef78e63..c3e52e7bdef57de0377cb7b467bf6fd8fd62b8c9 100644
--- a/sound/soc/codecs/es8326.h
+++ b/sound/soc/codecs/es8326.h
@@ -104,7 +104,7 @@
#define ES8326_MUTE (3 << 0)
/* ES8326_CLK_CTL */
-#define ES8326_CLK_ON (0x7e << 0)
+#define ES8326_CLK_ON (0x7f << 0)
#define ES8326_CLK_OFF (0 << 0)
/* ES8326_CLK_INV */
diff --git a/sound/soc/codecs/rt1316-sdw.c b/sound/soc/codecs/rt1316-sdw.c
index 47511f70119ae3b1d810ce8561d6026ccbbd98da..0b3bf920bcab2307c0107387e0ad728552bb6b9c 100644
--- a/sound/soc/codecs/rt1316-sdw.c
+++ b/sound/soc/codecs/rt1316-sdw.c
@@ -537,7 +537,7 @@ static int rt1316_sdw_hw_params(struct snd_pcm_substream *substream,
retval = sdw_stream_add_slave(rt1316->sdw_slave, &stream_config,
&port_config, 1, sdw_stream);
if (retval) {
- dev_err(dai->dev, "Unable to configure port\n");
+ dev_err(dai->dev, "%s: Unable to configure port\n", __func__);
return retval;
}
@@ -577,12 +577,12 @@ static int rt1316_sdw_parse_dt(struct rt1316_sdw_priv *rt1316, struct device *de
if (rt1316->bq_params_cnt) {
rt1316->bq_params = devm_kzalloc(dev, rt1316->bq_params_cnt, GFP_KERNEL);
if (!rt1316->bq_params) {
- dev_err(dev, "Could not allocate bq_params memory\n");
+ dev_err(dev, "%s: Could not allocate bq_params memory\n", __func__);
ret = -ENOMEM;
} else {
ret = device_property_read_u8_array(dev, "realtek,bq-params", rt1316->bq_params, rt1316->bq_params_cnt);
if (ret < 0)
- dev_err(dev, "Could not read list of realtek,bq-params\n");
+ dev_err(dev, "%s: Could not read list of realtek,bq-params\n", __func__);
}
}
@@ -759,7 +759,7 @@ static int __maybe_unused rt1316_dev_resume(struct device *dev)
time = wait_for_completion_timeout(&slave->initialization_complete,
msecs_to_jiffies(RT1316_PROBE_TIMEOUT));
if (!time) {
- dev_err(&slave->dev, "Initialization not complete, timed out\n");
+ dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__);
sdw_show_ping_status(slave->bus, true);
return -ETIMEDOUT;
diff --git a/sound/soc/codecs/rt1318-sdw.c b/sound/soc/codecs/rt1318-sdw.c
index ff364bde4a084943d78da479dc876f7b328eb02b..462c9a4b1be5ddb27c078b4c49e9c2ee3737e467 100644
--- a/sound/soc/codecs/rt1318-sdw.c
+++ b/sound/soc/codecs/rt1318-sdw.c
@@ -606,7 +606,7 @@ static int rt1318_sdw_hw_params(struct snd_pcm_substream *substream,
retval = sdw_stream_add_slave(rt1318->sdw_slave, &stream_config,
&port_config, 1, sdw_stream);
if (retval) {
- dev_err(dai->dev, "Unable to configure port\n");
+ dev_err(dai->dev, "%s: Unable to configure port\n", __func__);
return retval;
}
@@ -631,8 +631,8 @@ static int rt1318_sdw_hw_params(struct snd_pcm_substream *substream,
sampling_rate = RT1318_SDCA_RATE_192000HZ;
break;
default:
- dev_err(component->dev, "Rate %d is not supported\n",
- params_rate(params));
+ dev_err(component->dev, "%s: Rate %d is not supported\n",
+ __func__, params_rate(params));
return -EINVAL;
}
@@ -835,7 +835,7 @@ static int __maybe_unused rt1318_dev_resume(struct device *dev)
time = wait_for_completion_timeout(&slave->initialization_complete,
msecs_to_jiffies(RT1318_PROBE_TIMEOUT));
if (!time) {
- dev_err(&slave->dev, "Initialization not complete, timed out\n");
+ dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__);
return -ETIMEDOUT;
}
diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c
index e67c2e19cb1a7291170ada3b69cbdda4aadb8b6c..f9ee42c13dbac34afd0f79ff5299050106d82357 100644
--- a/sound/soc/codecs/rt5682-sdw.c
+++ b/sound/soc/codecs/rt5682-sdw.c
@@ -132,7 +132,7 @@ static int rt5682_sdw_hw_params(struct snd_pcm_substream *substream,
retval = sdw_stream_add_slave(rt5682->slave, &stream_config,
&port_config, 1, sdw_stream);
if (retval) {
- dev_err(dai->dev, "Unable to configure port\n");
+ dev_err(dai->dev, "%s: Unable to configure port\n", __func__);
return retval;
}
@@ -315,8 +315,8 @@ static int rt5682_sdw_init(struct device *dev, struct regmap *regmap,
&rt5682_sdw_indirect_regmap);
if (IS_ERR(rt5682->regmap)) {
ret = PTR_ERR(rt5682->regmap);
- dev_err(dev, "Failed to allocate register map: %d\n",
- ret);
+ dev_err(dev, "%s: Failed to allocate register map: %d\n",
+ __func__, ret);
return ret;
}
@@ -400,7 +400,7 @@ static int rt5682_io_init(struct device *dev, struct sdw_slave *slave)
}
if (val != DEVICE_ID) {
- dev_err(dev, "Device with ID register %x is not rt5682\n", val);
+ dev_err(dev, "%s: Device with ID register %x is not rt5682\n", __func__, val);
ret = -ENODEV;
goto err_nodev;
}
@@ -648,7 +648,7 @@ static int rt5682_bus_config(struct sdw_slave *slave,
ret = rt5682_clock_config(&slave->dev);
if (ret < 0)
- dev_err(&slave->dev, "Invalid clk config");
+ dev_err(&slave->dev, "%s: Invalid clk config", __func__);
return ret;
}
@@ -763,19 +763,19 @@ static int __maybe_unused rt5682_dev_resume(struct device *dev)
return 0;
if (!slave->unattach_request) {
+ mutex_lock(&rt5682->disable_irq_lock);
if (rt5682->disable_irq == true) {
- mutex_lock(&rt5682->disable_irq_lock);
sdw_write_no_pm(slave, SDW_SCP_INTMASK1, SDW_SCP_INT1_IMPL_DEF);
rt5682->disable_irq = false;
- mutex_unlock(&rt5682->disable_irq_lock);
}
+ mutex_unlock(&rt5682->disable_irq_lock);
goto regmap_sync;
}
time = wait_for_completion_timeout(&slave->initialization_complete,
msecs_to_jiffies(RT5682_PROBE_TIMEOUT));
if (!time) {
- dev_err(&slave->dev, "Initialization not complete, timed out\n");
+ dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__);
sdw_show_ping_status(slave->bus, true);
return -ETIMEDOUT;
diff --git a/sound/soc/codecs/rt700.c b/sound/soc/codecs/rt700.c
index 0ebf344a1b6094a38a4c38c6817b8cf0c9242f48..434b926f96c8376c1c5b8b73c37b01b2d64641fe 100644
--- a/sound/soc/codecs/rt700.c
+++ b/sound/soc/codecs/rt700.c
@@ -37,8 +37,8 @@ static int rt700_index_write(struct regmap *regmap,
ret = regmap_write(regmap, addr, value);
if (ret < 0)
- pr_err("Failed to set private value: %06x <= %04x ret=%d\n",
- addr, value, ret);
+ pr_err("%s: Failed to set private value: %06x <= %04x ret=%d\n",
+ __func__, addr, value, ret);
return ret;
}
@@ -52,8 +52,8 @@ static int rt700_index_read(struct regmap *regmap,
*value = 0;
ret = regmap_read(regmap, addr, value);
if (ret < 0)
- pr_err("Failed to get private value: %06x => %04x ret=%d\n",
- addr, *value, ret);
+ pr_err("%s: Failed to get private value: %06x => %04x ret=%d\n",
+ __func__, addr, *value, ret);
return ret;
}
@@ -930,14 +930,14 @@ static int rt700_pcm_hw_params(struct snd_pcm_substream *substream,
port_config.num += 2;
break;
default:
- dev_err(component->dev, "Invalid DAI id %d\n", dai->id);
+ dev_err(component->dev, "%s: Invalid DAI id %d\n", __func__, dai->id);
return -EINVAL;
}
retval = sdw_stream_add_slave(rt700->slave, &stream_config,
&port_config, 1, sdw_stream);
if (retval) {
- dev_err(dai->dev, "Unable to configure port\n");
+ dev_err(dai->dev, "%s: Unable to configure port\n", __func__);
return retval;
}
@@ -945,8 +945,8 @@ static int rt700_pcm_hw_params(struct snd_pcm_substream *substream,
/* bit 3:0 Number of Channel */
val |= (params_channels(params) - 1);
} else {
- dev_err(component->dev, "Unsupported channels %d\n",
- params_channels(params));
+ dev_err(component->dev, "%s: Unsupported channels %d\n",
+ __func__, params_channels(params));
return -EINVAL;
}
diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c
index 935e597022d3242187b378107e302a36bedd17f5..2636c2eea4bc8be6af732d3c2d5f03b8d78be22f 100644
--- a/sound/soc/codecs/rt711-sdca-sdw.c
+++ b/sound/soc/codecs/rt711-sdca-sdw.c
@@ -438,20 +438,20 @@ static int __maybe_unused rt711_sdca_dev_resume(struct device *dev)
return 0;
if (!slave->unattach_request) {
+ mutex_lock(&rt711->disable_irq_lock);
if (rt711->disable_irq == true) {
- mutex_lock(&rt711->disable_irq_lock);
sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0);
sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8);
rt711->disable_irq = false;
- mutex_unlock(&rt711->disable_irq_lock);
}
+ mutex_unlock(&rt711->disable_irq_lock);
goto regmap_sync;
}
time = wait_for_completion_timeout(&slave->initialization_complete,
msecs_to_jiffies(RT711_PROBE_TIMEOUT));
if (!time) {
- dev_err(&slave->dev, "Initialization not complete, timed out\n");
+ dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__);
sdw_show_ping_status(slave->bus, true);
return -ETIMEDOUT;
diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c
index 447154cb60104d31bb66ef268d7b60c513e273e3..1e8dbfc3ecd969be3a87cb5f00aed853a56e2a41 100644
--- a/sound/soc/codecs/rt711-sdca.c
+++ b/sound/soc/codecs/rt711-sdca.c
@@ -36,8 +36,8 @@ static int rt711_sdca_index_write(struct rt711_sdca_priv *rt711,
ret = regmap_write(regmap, addr, value);
if (ret < 0)
dev_err(&rt711->slave->dev,
- "Failed to set private value: %06x <= %04x ret=%d\n",
- addr, value, ret);
+ "%s: Failed to set private value: %06x <= %04x ret=%d\n",
+ __func__, addr, value, ret);
return ret;
}
@@ -52,8 +52,8 @@ static int rt711_sdca_index_read(struct rt711_sdca_priv *rt711,
ret = regmap_read(regmap, addr, value);
if (ret < 0)
dev_err(&rt711->slave->dev,
- "Failed to get private value: %06x => %04x ret=%d\n",
- addr, *value, ret);
+ "%s: Failed to get private value: %06x => %04x ret=%d\n",
+ __func__, addr, *value, ret);
return ret;
}
@@ -1293,13 +1293,13 @@ static int rt711_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
retval = sdw_stream_add_slave(rt711->slave, &stream_config,
&port_config, 1, sdw_stream);
if (retval) {
- dev_err(dai->dev, "Unable to configure port\n");
+ dev_err(dai->dev, "%s: Unable to configure port\n", __func__);
return retval;
}
if (params_channels(params) > 16) {
- dev_err(component->dev, "Unsupported channels %d\n",
- params_channels(params));
+ dev_err(component->dev, "%s: Unsupported channels %d\n",
+ __func__, params_channels(params));
return -EINVAL;
}
@@ -1318,8 +1318,8 @@ static int rt711_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
sampling_rate = RT711_SDCA_RATE_192000HZ;
break;
default:
- dev_err(component->dev, "Rate %d is not supported\n",
- params_rate(params));
+ dev_err(component->dev, "%s: Rate %d is not supported\n",
+ __func__, params_rate(params));
return -EINVAL;
}
diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c
index 3f5773310ae8cc3b5d94f76aa481724ac35bad0a..0d3b43dd22e63d2343b0ce882166ca7bedc7b67c 100644
--- a/sound/soc/codecs/rt711-sdw.c
+++ b/sound/soc/codecs/rt711-sdw.c
@@ -408,7 +408,7 @@ static int rt711_bus_config(struct sdw_slave *slave,
ret = rt711_clock_config(&slave->dev);
if (ret < 0)
- dev_err(&slave->dev, "Invalid clk config");
+ dev_err(&slave->dev, "%s: Invalid clk config", __func__);
return ret;
}
@@ -536,19 +536,19 @@ static int __maybe_unused rt711_dev_resume(struct device *dev)
return 0;
if (!slave->unattach_request) {
+ mutex_lock(&rt711->disable_irq_lock);
if (rt711->disable_irq == true) {
- mutex_lock(&rt711->disable_irq_lock);
sdw_write_no_pm(slave, SDW_SCP_INTMASK1, SDW_SCP_INT1_IMPL_DEF);
rt711->disable_irq = false;
- mutex_unlock(&rt711->disable_irq_lock);
}
+ mutex_unlock(&rt711->disable_irq_lock);
goto regmap_sync;
}
time = wait_for_completion_timeout(&slave->initialization_complete,
msecs_to_jiffies(RT711_PROBE_TIMEOUT));
if (!time) {
- dev_err(&slave->dev, "Initialization not complete, timed out\n");
+ dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__);
return -ETIMEDOUT;
}
diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c
index 66eaed13b0d6a06ff1a649be6924e16850e65997..5446f9506a16722e8a43571631d109fa27c9fe65 100644
--- a/sound/soc/codecs/rt711.c
+++ b/sound/soc/codecs/rt711.c
@@ -37,8 +37,8 @@ static int rt711_index_write(struct regmap *regmap,
ret = regmap_write(regmap, addr, value);
if (ret < 0)
- pr_err("Failed to set private value: %06x <= %04x ret=%d\n",
- addr, value, ret);
+ pr_err("%s: Failed to set private value: %06x <= %04x ret=%d\n",
+ __func__, addr, value, ret);
return ret;
}
@@ -52,8 +52,8 @@ static int rt711_index_read(struct regmap *regmap,
*value = 0;
ret = regmap_read(regmap, addr, value);
if (ret < 0)
- pr_err("Failed to get private value: %06x => %04x ret=%d\n",
- addr, *value, ret);
+ pr_err("%s: Failed to get private value: %06x => %04x ret=%d\n",
+ __func__, addr, *value, ret);
return ret;
}
@@ -428,7 +428,7 @@ static void rt711_jack_init(struct rt711_priv *rt711)
RT711_HP_JD_FINAL_RESULT_CTL_JD12);
break;
default:
- dev_warn(rt711->component->dev, "Wrong JD source\n");
+ dev_warn(rt711->component->dev, "%s: Wrong JD source\n", __func__);
break;
}
@@ -1020,7 +1020,7 @@ static int rt711_pcm_hw_params(struct snd_pcm_substream *substream,
retval = sdw_stream_add_slave(rt711->slave, &stream_config,
&port_config, 1, sdw_stream);
if (retval) {
- dev_err(dai->dev, "Unable to configure port\n");
+ dev_err(dai->dev, "%s: Unable to configure port\n", __func__);
return retval;
}
@@ -1028,8 +1028,8 @@ static int rt711_pcm_hw_params(struct snd_pcm_substream *substream,
/* bit 3:0 Number of Channel */
val |= (params_channels(params) - 1);
} else {
- dev_err(component->dev, "Unsupported channels %d\n",
- params_channels(params));
+ dev_err(component->dev, "%s: Unsupported channels %d\n",
+ __func__, params_channels(params));
return -EINVAL;
}
diff --git a/sound/soc/codecs/rt712-sdca-dmic.c b/sound/soc/codecs/rt712-sdca-dmic.c
index 0926b26619bd45b69f5b0b4b5508d677ca207766..012b79e72cf6b64e1e5c2837aaf034237cddcaa6 100644
--- a/sound/soc/codecs/rt712-sdca-dmic.c
+++ b/sound/soc/codecs/rt712-sdca-dmic.c
@@ -139,8 +139,8 @@ static int rt712_sdca_dmic_index_write(struct rt712_sdca_dmic_priv *rt712,
ret = regmap_write(regmap, addr, value);
if (ret < 0)
dev_err(&rt712->slave->dev,
- "Failed to set private value: %06x <= %04x ret=%d\n",
- addr, value, ret);
+ "%s: Failed to set private value: %06x <= %04x ret=%d\n",
+ __func__, addr, value, ret);
return ret;
}
@@ -155,8 +155,8 @@ static int rt712_sdca_dmic_index_read(struct rt712_sdca_dmic_priv *rt712,
ret = regmap_read(regmap, addr, value);
if (ret < 0)
dev_err(&rt712->slave->dev,
- "Failed to get private value: %06x => %04x ret=%d\n",
- addr, *value, ret);
+ "%s: Failed to get private value: %06x => %04x ret=%d\n",
+ __func__, addr, *value, ret);
return ret;
}
@@ -317,7 +317,8 @@ static int rt712_sdca_dmic_set_gain_put(struct snd_kcontrol *kcontrol,
for (i = 0; i < p->count; i++) {
err = regmap_write(rt712->mbq_regmap, p->reg_base + i, gain_val[i]);
if (err < 0)
- dev_err(&rt712->slave->dev, "0x%08x can't be set\n", p->reg_base + i);
+ dev_err(&rt712->slave->dev, "%s: 0x%08x can't be set\n",
+ __func__, p->reg_base + i);
}
return changed;
@@ -667,13 +668,13 @@ static int rt712_sdca_dmic_hw_params(struct snd_pcm_substream *substream,
retval = sdw_stream_add_slave(rt712->slave, &stream_config,
&port_config, 1, sdw_stream);
if (retval) {
- dev_err(dai->dev, "Unable to configure port\n");
+ dev_err(dai->dev, "%s: Unable to configure port\n", __func__);
return retval;
}
if (params_channels(params) > 4) {
- dev_err(component->dev, "Unsupported channels %d\n",
- params_channels(params));
+ dev_err(component->dev, "%s: Unsupported channels %d\n",
+ __func__, params_channels(params));
return -EINVAL;
}
@@ -698,8 +699,8 @@ static int rt712_sdca_dmic_hw_params(struct snd_pcm_substream *substream,
sampling_rate = RT712_SDCA_RATE_192000HZ;
break;
default:
- dev_err(component->dev, "Rate %d is not supported\n",
- params_rate(params));
+ dev_err(component->dev, "%s: Rate %d is not supported\n",
+ __func__, params_rate(params));
return -EINVAL;
}
@@ -923,7 +924,8 @@ static int __maybe_unused rt712_sdca_dmic_dev_resume(struct device *dev)
time = wait_for_completion_timeout(&slave->initialization_complete,
msecs_to_jiffies(RT712_PROBE_TIMEOUT));
if (!time) {
- dev_err(&slave->dev, "Initialization not complete, timed out\n");
+ dev_err(&slave->dev, "%s: Initialization not complete, timed out\n",
+ __func__);
sdw_show_ping_status(slave->bus, true);
return -ETIMEDOUT;
diff --git a/sound/soc/codecs/rt712-sdca-sdw.c b/sound/soc/codecs/rt712-sdca-sdw.c
index 01ac555cd79b84a0b1aabe57899b1fedc214a2b5..4e9ab3ef135b34946d37d6280a9afb568cceef51 100644
--- a/sound/soc/codecs/rt712-sdca-sdw.c
+++ b/sound/soc/codecs/rt712-sdca-sdw.c
@@ -438,20 +438,21 @@ static int __maybe_unused rt712_sdca_dev_resume(struct device *dev)
return 0;
if (!slave->unattach_request) {
+ mutex_lock(&rt712->disable_irq_lock);
if (rt712->disable_irq == true) {
- mutex_lock(&rt712->disable_irq_lock);
+
sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0);
sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8);
rt712->disable_irq = false;
- mutex_unlock(&rt712->disable_irq_lock);
}
+ mutex_unlock(&rt712->disable_irq_lock);
goto regmap_sync;
}
time = wait_for_completion_timeout(&slave->initialization_complete,
msecs_to_jiffies(RT712_PROBE_TIMEOUT));
if (!time) {
- dev_err(&slave->dev, "Initialization not complete, timed out\n");
+ dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__);
sdw_show_ping_status(slave->bus, true);
return -ETIMEDOUT;
diff --git a/sound/soc/codecs/rt712-sdca.c b/sound/soc/codecs/rt712-sdca.c
index 6954fbe7ec5f3bb79f8693c23f302a7a1003e11e..b503de9fda80e71cbe78e8916a6a7f41286ac5b2 100644
--- a/sound/soc/codecs/rt712-sdca.c
+++ b/sound/soc/codecs/rt712-sdca.c
@@ -34,8 +34,8 @@ static int rt712_sdca_index_write(struct rt712_sdca_priv *rt712,
ret = regmap_write(regmap, addr, value);
if (ret < 0)
dev_err(&rt712->slave->dev,
- "Failed to set private value: %06x <= %04x ret=%d\n",
- addr, value, ret);
+ "%s: Failed to set private value: %06x <= %04x ret=%d\n",
+ __func__, addr, value, ret);
return ret;
}
@@ -50,8 +50,8 @@ static int rt712_sdca_index_read(struct rt712_sdca_priv *rt712,
ret = regmap_read(regmap, addr, value);
if (ret < 0)
dev_err(&rt712->slave->dev,
- "Failed to get private value: %06x => %04x ret=%d\n",
- addr, *value, ret);
+ "%s: Failed to get private value: %06x => %04x ret=%d\n",
+ __func__, addr, *value, ret);
return ret;
}
@@ -1060,13 +1060,13 @@ static int rt712_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
retval = sdw_stream_add_slave(rt712->slave, &stream_config,
&port_config, 1, sdw_stream);
if (retval) {
- dev_err(dai->dev, "Unable to configure port\n");
+ dev_err(dai->dev, "%s: Unable to configure port\n", __func__);
return retval;
}
if (params_channels(params) > 16) {
- dev_err(component->dev, "Unsupported channels %d\n",
- params_channels(params));
+ dev_err(component->dev, "%s: Unsupported channels %d\n",
+ __func__, params_channels(params));
return -EINVAL;
}
@@ -1085,8 +1085,8 @@ static int rt712_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
sampling_rate = RT712_SDCA_RATE_192000HZ;
break;
default:
- dev_err(component->dev, "Rate %d is not supported\n",
- params_rate(params));
+ dev_err(component->dev, "%s: Rate %d is not supported\n",
+ __func__, params_rate(params));
return -EINVAL;
}
@@ -1106,7 +1106,7 @@ static int rt712_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
sampling_rate);
break;
default:
- dev_err(component->dev, "Wrong DAI id\n");
+ dev_err(component->dev, "%s: Wrong DAI id\n", __func__);
return -EINVAL;
}
diff --git a/sound/soc/codecs/rt715-sdca-sdw.c b/sound/soc/codecs/rt715-sdca-sdw.c
index ab54a67a27ebbfc8fbe19837fa07ed7d084bd429..ee450126106f969588ab52b83434309f8cfb8036 100644
--- a/sound/soc/codecs/rt715-sdca-sdw.c
+++ b/sound/soc/codecs/rt715-sdca-sdw.c
@@ -237,7 +237,7 @@ static int __maybe_unused rt715_dev_resume(struct device *dev)
time = wait_for_completion_timeout(&slave->enumeration_complete,
msecs_to_jiffies(RT715_PROBE_TIMEOUT));
if (!time) {
- dev_err(&slave->dev, "Enumeration not complete, timed out\n");
+ dev_err(&slave->dev, "%s: Enumeration not complete, timed out\n", __func__);
sdw_show_ping_status(slave->bus, true);
return -ETIMEDOUT;
diff --git a/sound/soc/codecs/rt715-sdca.c b/sound/soc/codecs/rt715-sdca.c
index 4533eedd7e189f3b48e36175eb5494f20a6f1be0..3fb7b9adb61de628705d784fbe64e259bf031089 100644
--- a/sound/soc/codecs/rt715-sdca.c
+++ b/sound/soc/codecs/rt715-sdca.c
@@ -41,8 +41,8 @@ static int rt715_sdca_index_write(struct rt715_sdca_priv *rt715,
ret = regmap_write(regmap, addr, value);
if (ret < 0)
dev_err(&rt715->slave->dev,
- "Failed to set private value: %08x <= %04x %d\n",
- addr, value, ret);
+ "%s: Failed to set private value: %08x <= %04x %d\n",
+ __func__, addr, value, ret);
return ret;
}
@@ -59,8 +59,8 @@ static int rt715_sdca_index_read(struct rt715_sdca_priv *rt715,
ret = regmap_read(regmap, addr, value);
if (ret < 0)
dev_err(&rt715->slave->dev,
- "Failed to get private value: %06x => %04x ret=%d\n",
- addr, *value, ret);
+ "%s: Failed to get private value: %06x => %04x ret=%d\n",
+ __func__, addr, *value, ret);
return ret;
}
@@ -152,8 +152,8 @@ static int rt715_sdca_set_amp_gain_put(struct snd_kcontrol *kcontrol,
mc->shift);
ret = regmap_write(rt715->mbq_regmap, mc->reg + i, gain_val);
if (ret != 0) {
- dev_err(component->dev, "Failed to write 0x%x=0x%x\n",
- mc->reg + i, gain_val);
+ dev_err(component->dev, "%s: Failed to write 0x%x=0x%x\n",
+ __func__, mc->reg + i, gain_val);
return ret;
}
}
@@ -188,8 +188,8 @@ static int rt715_sdca_set_amp_gain_4ch_put(struct snd_kcontrol *kcontrol,
ret = regmap_write(rt715->mbq_regmap, reg_base + i,
gain_val);
if (ret != 0) {
- dev_err(component->dev, "Failed to write 0x%x=0x%x\n",
- reg_base + i, gain_val);
+ dev_err(component->dev, "%s: Failed to write 0x%x=0x%x\n",
+ __func__, reg_base + i, gain_val);
return ret;
}
}
@@ -224,8 +224,8 @@ static int rt715_sdca_set_amp_gain_8ch_put(struct snd_kcontrol *kcontrol,
reg = i < 7 ? reg_base + i : (reg_base - 1) | BIT(15);
ret = regmap_write(rt715->mbq_regmap, reg, gain_val);
if (ret != 0) {
- dev_err(component->dev, "Failed to write 0x%x=0x%x\n",
- reg, gain_val);
+ dev_err(component->dev, "%s: Failed to write 0x%x=0x%x\n",
+ __func__, reg, gain_val);
return ret;
}
}
@@ -246,8 +246,8 @@ static int rt715_sdca_set_amp_gain_get(struct snd_kcontrol *kcontrol,
for (i = 0; i < 2; i++) {
ret = regmap_read(rt715->mbq_regmap, mc->reg + i, &val);
if (ret < 0) {
- dev_err(component->dev, "Failed to read 0x%x, ret=%d\n",
- mc->reg + i, ret);
+ dev_err(component->dev, "%s: Failed to read 0x%x, ret=%d\n",
+ __func__, mc->reg + i, ret);
return ret;
}
ucontrol->value.integer.value[i] = rt715_sdca_get_gain(val, mc->shift);
@@ -271,8 +271,8 @@ static int rt715_sdca_set_amp_gain_4ch_get(struct snd_kcontrol *kcontrol,
for (i = 0; i < 4; i++) {
ret = regmap_read(rt715->mbq_regmap, reg_base + i, &val);
if (ret < 0) {
- dev_err(component->dev, "Failed to read 0x%x, ret=%d\n",
- reg_base + i, ret);
+ dev_err(component->dev, "%s: Failed to read 0x%x, ret=%d\n",
+ __func__, reg_base + i, ret);
return ret;
}
ucontrol->value.integer.value[i] = rt715_sdca_get_gain(val, gain_sft);
@@ -297,8 +297,8 @@ static int rt715_sdca_set_amp_gain_8ch_get(struct snd_kcontrol *kcontrol,
for (i = 0; i < 8; i += 2) {
ret = regmap_read(rt715->mbq_regmap, reg_base + i, &val_l);
if (ret < 0) {
- dev_err(component->dev, "Failed to read 0x%x, ret=%d\n",
- reg_base + i, ret);
+ dev_err(component->dev, "%s: Failed to read 0x%x, ret=%d\n",
+ __func__, reg_base + i, ret);
return ret;
}
ucontrol->value.integer.value[i] = (val_l >> gain_sft) / 10;
@@ -306,8 +306,8 @@ static int rt715_sdca_set_amp_gain_8ch_get(struct snd_kcontrol *kcontrol,
reg = (i == 6) ? (reg_base - 1) | BIT(15) : reg_base + 1 + i;
ret = regmap_read(rt715->mbq_regmap, reg, &val_r);
if (ret < 0) {
- dev_err(component->dev, "Failed to read 0x%x, ret=%d\n",
- reg, ret);
+ dev_err(component->dev, "%s: Failed to read 0x%x, ret=%d\n",
+ __func__, reg, ret);
return ret;
}
ucontrol->value.integer.value[i + 1] = (val_r >> gain_sft) / 10;
@@ -834,15 +834,15 @@ static int rt715_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
0xaf00);
break;
default:
- dev_err(component->dev, "Invalid DAI id %d\n", dai->id);
+ dev_err(component->dev, "%s: Invalid DAI id %d\n", __func__, dai->id);
return -EINVAL;
}
retval = sdw_stream_add_slave(rt715->slave, &stream_config,
&port_config, 1, sdw_stream);
if (retval) {
- dev_err(component->dev, "Unable to configure port, retval:%d\n",
- retval);
+ dev_err(component->dev, "%s: Unable to configure port, retval:%d\n",
+ __func__, retval);
return retval;
}
@@ -893,8 +893,8 @@ static int rt715_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
val = 0xf;
break;
default:
- dev_err(component->dev, "Unsupported sample rate %d\n",
- params_rate(params));
+ dev_err(component->dev, "%s: Unsupported sample rate %d\n",
+ __func__, params_rate(params));
return -EINVAL;
}
diff --git a/sound/soc/codecs/rt715-sdw.c b/sound/soc/codecs/rt715-sdw.c
index 21f37babd148a487e82568144a791bff07fdf6c0..7e13868ff99f03110c165dcd706cff46a8eeba5d 100644
--- a/sound/soc/codecs/rt715-sdw.c
+++ b/sound/soc/codecs/rt715-sdw.c
@@ -482,7 +482,7 @@ static int rt715_bus_config(struct sdw_slave *slave,
ret = rt715_clock_config(&slave->dev);
if (ret < 0)
- dev_err(&slave->dev, "Invalid clk config");
+ dev_err(&slave->dev, "%s: Invalid clk config", __func__);
return 0;
}
@@ -554,7 +554,7 @@ static int __maybe_unused rt715_dev_resume(struct device *dev)
time = wait_for_completion_timeout(&slave->initialization_complete,
msecs_to_jiffies(RT715_PROBE_TIMEOUT));
if (!time) {
- dev_err(&slave->dev, "Initialization not complete, timed out\n");
+ dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__);
sdw_show_ping_status(slave->bus, true);
return -ETIMEDOUT;
diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c
index 9f732a5abd53f37cd24382522f9dc3ab97ecd7b0..299c9b12377c6ada95a40b4a876a43dd127786be 100644
--- a/sound/soc/codecs/rt715.c
+++ b/sound/soc/codecs/rt715.c
@@ -40,8 +40,8 @@ static int rt715_index_write(struct regmap *regmap, unsigned int reg,
ret = regmap_write(regmap, addr, value);
if (ret < 0) {
- pr_err("Failed to set private value: %08x <= %04x %d\n",
- addr, value, ret);
+ pr_err("%s: Failed to set private value: %08x <= %04x %d\n",
+ __func__, addr, value, ret);
}
return ret;
@@ -55,8 +55,8 @@ static int rt715_index_write_nid(struct regmap *regmap,
ret = regmap_write(regmap, addr, value);
if (ret < 0)
- pr_err("Failed to set private value: %06x <= %04x ret=%d\n",
- addr, value, ret);
+ pr_err("%s: Failed to set private value: %06x <= %04x ret=%d\n",
+ __func__, addr, value, ret);
return ret;
}
@@ -70,8 +70,8 @@ static int rt715_index_read_nid(struct regmap *regmap,
*value = 0;
ret = regmap_read(regmap, addr, value);
if (ret < 0)
- pr_err("Failed to get private value: %06x => %04x ret=%d\n",
- addr, *value, ret);
+ pr_err("%s: Failed to get private value: %06x => %04x ret=%d\n",
+ __func__, addr, *value, ret);
return ret;
}
@@ -862,14 +862,14 @@ static int rt715_pcm_hw_params(struct snd_pcm_substream *substream,
rt715_index_write(rt715->regmap, RT715_SDW_INPUT_SEL, 0xa000);
break;
default:
- dev_err(component->dev, "Invalid DAI id %d\n", dai->id);
+ dev_err(component->dev, "%s: Invalid DAI id %d\n", __func__, dai->id);
return -EINVAL;
}
retval = sdw_stream_add_slave(rt715->slave, &stream_config,
&port_config, 1, sdw_stream);
if (retval) {
- dev_err(dai->dev, "Unable to configure port\n");
+ dev_err(dai->dev, "%s: Unable to configure port\n", __func__);
return retval;
}
@@ -883,8 +883,8 @@ static int rt715_pcm_hw_params(struct snd_pcm_substream *substream,
val |= 0x0 << 8;
break;
default:
- dev_err(component->dev, "Unsupported sample rate %d\n",
- params_rate(params));
+ dev_err(component->dev, "%s: Unsupported sample rate %d\n",
+ __func__, params_rate(params));
return -EINVAL;
}
@@ -892,8 +892,8 @@ static int rt715_pcm_hw_params(struct snd_pcm_substream *substream,
/* bit 3:0 Number of Channel */
val |= (params_channels(params) - 1);
} else {
- dev_err(component->dev, "Unsupported channels %d\n",
- params_channels(params));
+ dev_err(component->dev, "%s: Unsupported channels %d\n",
+ __func__, params_channels(params));
return -EINVAL;
}
diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c
index eb76f4c675b67fd59df00cb41a17955c751bbd44..65d584c1886e819597577ed10551d2fe5d104e53 100644
--- a/sound/soc/codecs/rt722-sdca-sdw.c
+++ b/sound/soc/codecs/rt722-sdca-sdw.c
@@ -467,13 +467,13 @@ static int __maybe_unused rt722_sdca_dev_resume(struct device *dev)
return 0;
if (!slave->unattach_request) {
+ mutex_lock(&rt722->disable_irq_lock);
if (rt722->disable_irq == true) {
- mutex_lock(&rt722->disable_irq_lock);
sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_6);
sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8);
rt722->disable_irq = false;
- mutex_unlock(&rt722->disable_irq_lock);
}
+ mutex_unlock(&rt722->disable_irq_lock);
goto regmap_sync;
}
diff --git a/sound/soc/codecs/rt722-sdca.c b/sound/soc/codecs/rt722-sdca.c
index 0e1c65a20392addb92a6bdbc39319884f4d2f9c9..e0ea3a23f7cc6844691338ff8daae7f2843d2c6e 100644
--- a/sound/soc/codecs/rt722-sdca.c
+++ b/sound/soc/codecs/rt722-sdca.c
@@ -35,8 +35,8 @@ int rt722_sdca_index_write(struct rt722_sdca_priv *rt722,
ret = regmap_write(regmap, addr, value);
if (ret < 0)
dev_err(&rt722->slave->dev,
- "Failed to set private value: %06x <= %04x ret=%d\n",
- addr, value, ret);
+ "%s: Failed to set private value: %06x <= %04x ret=%d\n",
+ __func__, addr, value, ret);
return ret;
}
@@ -51,8 +51,8 @@ int rt722_sdca_index_read(struct rt722_sdca_priv *rt722,
ret = regmap_read(regmap, addr, value);
if (ret < 0)
dev_err(&rt722->slave->dev,
- "Failed to get private value: %06x => %04x ret=%d\n",
- addr, *value, ret);
+ "%s: Failed to get private value: %06x => %04x ret=%d\n",
+ __func__, addr, *value, ret);
return ret;
}
@@ -663,7 +663,8 @@ static int rt722_sdca_dmic_set_gain_put(struct snd_kcontrol *kcontrol,
for (i = 0; i < p->count; i++) {
err = regmap_write(rt722->mbq_regmap, p->reg_base + i, gain_val[i]);
if (err < 0)
- dev_err(&rt722->slave->dev, "%#08x can't be set\n", p->reg_base + i);
+ dev_err(&rt722->slave->dev, "%s: %#08x can't be set\n",
+ __func__, p->reg_base + i);
}
return changed;
@@ -1211,13 +1212,13 @@ static int rt722_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
retval = sdw_stream_add_slave(rt722->slave, &stream_config,
&port_config, 1, sdw_stream);
if (retval) {
- dev_err(dai->dev, "Unable to configure port\n");
+ dev_err(dai->dev, "%s: Unable to configure port\n", __func__);
return retval;
}
if (params_channels(params) > 16) {
- dev_err(component->dev, "Unsupported channels %d\n",
- params_channels(params));
+ dev_err(component->dev, "%s: Unsupported channels %d\n",
+ __func__, params_channels(params));
return -EINVAL;
}
@@ -1236,8 +1237,8 @@ static int rt722_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
sampling_rate = RT722_SDCA_RATE_192000HZ;
break;
default:
- dev_err(component->dev, "Rate %d is not supported\n",
- params_rate(params));
+ dev_err(component->dev, "%s: Rate %d is not supported\n",
+ __func__, params_rate(params));
return -EINVAL;
}
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index e451c009f2d99980bab20dd5d4c55cc26bd73cd5..7d5c096e06cd32b77fc6b73f18002af63bd6c8d5 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -683,11 +683,12 @@ static void wm_adsp_control_remove(struct cs_dsp_coeff_ctl *cs_ctl)
int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type,
unsigned int alg, void *buf, size_t len)
{
- struct cs_dsp_coeff_ctl *cs_ctl = cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg);
+ struct cs_dsp_coeff_ctl *cs_ctl;
struct wm_coeff_ctl *ctl;
int ret;
mutex_lock(&dsp->cs_dsp.pwr_lock);
+ cs_ctl = cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg);
ret = cs_dsp_coeff_write_ctrl(cs_ctl, 0, buf, len);
mutex_unlock(&dsp->cs_dsp.pwr_lock);
diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c
index c018f84fe02529322455035e6ca4fff7ddf2afaf..fc072dc58968cb80d6481cb0c84a4b776bdef150 100644
--- a/sound/soc/intel/avs/boards/da7219.c
+++ b/sound/soc/intel/avs/boards/da7219.c
@@ -296,5 +296,6 @@ static struct platform_driver avs_da7219_driver = {
module_platform_driver(avs_da7219_driver);
+MODULE_DESCRIPTION("Intel da7219 machine driver");
MODULE_AUTHOR("Cezary Rojewski <cezary.rojewski@intel.com>");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/dmic.c b/sound/soc/intel/avs/boards/dmic.c
index ba2bc7f689eb603051870bfcbf28dc5640b7ac66..d9e5e85f523358d26a85218c5050fe8b31876e21 100644
--- a/sound/soc/intel/avs/boards/dmic.c
+++ b/sound/soc/intel/avs/boards/dmic.c
@@ -96,4 +96,5 @@ static struct platform_driver avs_dmic_driver = {
module_platform_driver(avs_dmic_driver);
+MODULE_DESCRIPTION("Intel DMIC machine driver");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/es8336.c b/sound/soc/intel/avs/boards/es8336.c
index 1090082e7d5bfcd47e92ecbd6bed22269fab3678..5c90a60075773409431f957f05f2e3bc03303334 100644
--- a/sound/soc/intel/avs/boards/es8336.c
+++ b/sound/soc/intel/avs/boards/es8336.c
@@ -326,4 +326,5 @@ static struct platform_driver avs_es8336_driver = {
module_platform_driver(avs_es8336_driver);
+MODULE_DESCRIPTION("Intel es8336 machine driver");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/i2s_test.c b/sound/soc/intel/avs/boards/i2s_test.c
index 28f254eb0d03fcfa6f5fc8c4bd0184d73f9c298d..027373d6a16d602c62b07d34b2b0bd984c14ae78 100644
--- a/sound/soc/intel/avs/boards/i2s_test.c
+++ b/sound/soc/intel/avs/boards/i2s_test.c
@@ -204,4 +204,5 @@ static struct platform_driver avs_i2s_test_driver = {
module_platform_driver(avs_i2s_test_driver);
+MODULE_DESCRIPTION("Intel i2s test machine driver");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/max98357a.c b/sound/soc/intel/avs/boards/max98357a.c
index a83b95f25129f90e1a0fd6f5d34e7e6fa799d34f..1ff85e4d8e160b7c61a74fc6dbdf9a32ee410614 100644
--- a/sound/soc/intel/avs/boards/max98357a.c
+++ b/sound/soc/intel/avs/boards/max98357a.c
@@ -154,4 +154,5 @@ static struct platform_driver avs_max98357a_driver = {
module_platform_driver(avs_max98357a_driver)
+MODULE_DESCRIPTION("Intel max98357a machine driver");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/max98373.c b/sound/soc/intel/avs/boards/max98373.c
index 3b980a025e6f697446419f6efec4f071e45495cb..8d31586b73eaec7c10edb319002f1b024e41480b 100644
--- a/sound/soc/intel/avs/boards/max98373.c
+++ b/sound/soc/intel/avs/boards/max98373.c
@@ -211,4 +211,5 @@ static struct platform_driver avs_max98373_driver = {
module_platform_driver(avs_max98373_driver)
+MODULE_DESCRIPTION("Intel max98373 machine driver");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/max98927.c b/sound/soc/intel/avs/boards/max98927.c
index 86dd2b228df3a5ce1a2751221659834cb458c775..572ec58073d06bce6e07bc7dbd05f2b9a3cf5462 100644
--- a/sound/soc/intel/avs/boards/max98927.c
+++ b/sound/soc/intel/avs/boards/max98927.c
@@ -208,4 +208,5 @@ static struct platform_driver avs_max98927_driver = {
module_platform_driver(avs_max98927_driver)
+MODULE_DESCRIPTION("Intel max98927 machine driver");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c
index 1c1e2083f474df122259c41f774e246dd7223a1f..55db75efae41425684bdd1c67441e76cae4d4062 100644
--- a/sound/soc/intel/avs/boards/nau8825.c
+++ b/sound/soc/intel/avs/boards/nau8825.c
@@ -313,4 +313,5 @@ static struct platform_driver avs_nau8825_driver = {
module_platform_driver(avs_nau8825_driver)
+MODULE_DESCRIPTION("Intel nau8825 machine driver");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/probe.c b/sound/soc/intel/avs/boards/probe.c
index a9469b5ecb402f1af389c52fd6e1e5d9991cc3e9..8be6887bbc6e81cb0f6af16685524fd01b96e36a 100644
--- a/sound/soc/intel/avs/boards/probe.c
+++ b/sound/soc/intel/avs/boards/probe.c
@@ -69,4 +69,5 @@ static struct platform_driver avs_probe_mb_driver = {
module_platform_driver(avs_probe_mb_driver);
+MODULE_DESCRIPTION("Intel probe machine driver");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/rt274.c b/sound/soc/intel/avs/boards/rt274.c
index bfcb8845fd15d06ec39d7360008ac60f73491d3b..1cf52421608753e1cca23d333eab4a7a9d624b63 100644
--- a/sound/soc/intel/avs/boards/rt274.c
+++ b/sound/soc/intel/avs/boards/rt274.c
@@ -276,4 +276,5 @@ static struct platform_driver avs_rt274_driver = {
module_platform_driver(avs_rt274_driver);
+MODULE_DESCRIPTION("Intel rt274 machine driver");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/rt286.c b/sound/soc/intel/avs/boards/rt286.c
index 28d7d86b1cc99dabed8c76a94ee2c5dc3064582e..4740bba1057032128c60594b9339b820f9f7bc70 100644
--- a/sound/soc/intel/avs/boards/rt286.c
+++ b/sound/soc/intel/avs/boards/rt286.c
@@ -247,4 +247,5 @@ static struct platform_driver avs_rt286_driver = {
module_platform_driver(avs_rt286_driver);
+MODULE_DESCRIPTION("Intel rt286 machine driver");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/rt298.c b/sound/soc/intel/avs/boards/rt298.c
index 80f490b9e11842c34859ff8b6cebc8b7cee51e71..6e409e29f6974654a0a5cbe4eba105f78c055164 100644
--- a/sound/soc/intel/avs/boards/rt298.c
+++ b/sound/soc/intel/avs/boards/rt298.c
@@ -266,4 +266,5 @@ static struct platform_driver avs_rt298_driver = {
module_platform_driver(avs_rt298_driver);
+MODULE_DESCRIPTION("Intel rt298 machine driver");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/rt5514.c b/sound/soc/intel/avs/boards/rt5514.c
index 60105f453ae235c6affdcfef2181b95b101a3e1c..097ae5f73241efea14cf187e85bc88d936be1956 100644
--- a/sound/soc/intel/avs/boards/rt5514.c
+++ b/sound/soc/intel/avs/boards/rt5514.c
@@ -192,4 +192,5 @@ static struct platform_driver avs_rt5514_driver = {
module_platform_driver(avs_rt5514_driver);
+MODULE_DESCRIPTION("Intel rt5514 machine driver");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/rt5663.c b/sound/soc/intel/avs/boards/rt5663.c
index b4762c2a7bf2d1a3b0237479145380243861c4d8..1880c315cc4d1f9be4b7f42113382b322bd9c4cd 100644
--- a/sound/soc/intel/avs/boards/rt5663.c
+++ b/sound/soc/intel/avs/boards/rt5663.c
@@ -265,4 +265,5 @@ static struct platform_driver avs_rt5663_driver = {
module_platform_driver(avs_rt5663_driver);
+MODULE_DESCRIPTION("Intel rt5663 machine driver");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/rt5682.c b/sound/soc/intel/avs/boards/rt5682.c
index 243f979fda98a4d6e67b76cef8b5a192dd6a8ff7..594a971ded9eb2ea339ab2e45ae84da8b4b1dd6d 100644
--- a/sound/soc/intel/avs/boards/rt5682.c
+++ b/sound/soc/intel/avs/boards/rt5682.c
@@ -341,5 +341,6 @@ static struct platform_driver avs_rt5682_driver = {
module_platform_driver(avs_rt5682_driver)
+MODULE_DESCRIPTION("Intel rt5682 machine driver");
MODULE_AUTHOR("Cezary Rojewski <cezary.rojewski@intel.com>");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/boards/ssm4567.c b/sound/soc/intel/avs/boards/ssm4567.c
index 4a0e136835ff5d05118b1d802d2884beabb68a95..d6f7f046c24e5d12bd3189fe800bb05b18ee4444 100644
--- a/sound/soc/intel/avs/boards/ssm4567.c
+++ b/sound/soc/intel/avs/boards/ssm4567.c
@@ -200,4 +200,5 @@ static struct platform_driver avs_ssm4567_driver = {
module_platform_driver(avs_ssm4567_driver)
+MODULE_DESCRIPTION("Intel ssm4567 machine driver");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index 2d25748ca70662bf771c6896297ccb6a0fb0798f..b27e89ff6a1673f57db6e253a818d6fbe3d1ab91 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -263,7 +263,7 @@ int snd_soc_get_volsw(struct snd_kcontrol *kcontrol,
int max = mc->max;
int min = mc->min;
int sign_bit = mc->sign_bit;
- unsigned int mask = (1 << fls(max)) - 1;
+ unsigned int mask = (1ULL << fls(max)) - 1;
unsigned int invert = mc->invert;
int val;
int ret;
diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c
index be7dc1e02284ab62f8cbaeffdd70f26a19ff6232..c12c7f820529476de0273474082b8174ab0ae052 100644
--- a/sound/soc/sof/amd/acp.c
+++ b/sound/soc/sof/amd/acp.c
@@ -704,6 +704,10 @@ int amd_sof_acp_probe(struct snd_sof_dev *sdev)
goto unregister_dev;
}
+ ret = acp_init(sdev);
+ if (ret < 0)
+ goto free_smn_dev;
+
sdev->ipc_irq = pci->irq;
ret = request_threaded_irq(sdev->ipc_irq, acp_irq_handler, acp_irq_thread,
IRQF_SHARED, "AudioDSP", sdev);
@@ -713,10 +717,6 @@ int amd_sof_acp_probe(struct snd_sof_dev *sdev)
goto free_smn_dev;
}
- ret = acp_init(sdev);
- if (ret < 0)
- goto free_ipc_irq;
-
/* scan SoundWire capabilities exposed by DSDT */
ret = acp_sof_scan_sdw_devices(sdev, chip->sdw_acpi_dev_addr);
if (ret < 0) {
diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c
index 9b00ede2a486a2ff2d619ab714ed2c1665eb3463..cc84d4c81be9d363d701b1d5c658e26a62079435 100644
--- a/sound/soc/sof/core.c
+++ b/sound/soc/sof/core.c
@@ -339,8 +339,7 @@ static int sof_init_environment(struct snd_sof_dev *sdev)
ret = snd_sof_probe(sdev);
if (ret < 0) {
dev_err(sdev->dev, "failed to probe DSP %d\n", ret);
- sof_ops_free(sdev);
- return ret;
+ goto err_sof_probe;
}
/* check machine info */
@@ -358,15 +357,18 @@ static int sof_init_environment(struct snd_sof_dev *sdev)
ret = validate_sof_ops(sdev);
if (ret < 0) {
snd_sof_remove(sdev);
+ snd_sof_remove_late(sdev);
return ret;
}
}
+ return 0;
+
err_machine_check:
- if (ret) {
- snd_sof_remove(sdev);
- sof_ops_free(sdev);
- }
+ snd_sof_remove(sdev);
+err_sof_probe:
+ snd_sof_remove_late(sdev);
+ sof_ops_free(sdev);
return ret;
}
diff --git a/sound/soc/sof/intel/hda-common-ops.c b/sound/soc/sof/intel/hda-common-ops.c
index 2b385cddc385c5bd59e11acfe8e6bda45704fdd1..d71bb66b9991164cdb8b0ed000e461d9e3a0719c 100644
--- a/sound/soc/sof/intel/hda-common-ops.c
+++ b/sound/soc/sof/intel/hda-common-ops.c
@@ -57,6 +57,9 @@ struct snd_sof_dsp_ops sof_hda_common_ops = {
.pcm_pointer = hda_dsp_pcm_pointer,
.pcm_ack = hda_dsp_pcm_ack,
+ .get_dai_frame_counter = hda_dsp_get_stream_llp,
+ .get_host_byte_counter = hda_dsp_get_stream_ldp,
+
/* firmware loading */
.load_firmware = snd_sof_load_firmware_raw,
diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c
index c50ca9e72d37385816ddb3cd6ef7456ed50a58e9..b073720b4cf432466e18bf8840dd87eb5efac98e 100644
--- a/sound/soc/sof/intel/hda-dai-ops.c
+++ b/sound/soc/sof/intel/hda-dai-ops.c
@@ -7,6 +7,7 @@
#include <sound/pcm_params.h>
#include <sound/hdaudio_ext.h>
+#include <sound/hda_register.h>
#include <sound/hda-mlink.h>
#include <sound/sof/ipc4/header.h>
#include <uapi/sound/sof/header.h>
@@ -362,6 +363,16 @@ static int hda_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *cpu_dai,
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
snd_hdac_ext_stream_clear(hext_stream);
+
+ /*
+ * Save the LLP registers in case the stream is
+ * restarting due PAUSE_RELEASE, or START without a pcm
+ * close/open since in this case the LLP register is not reset
+ * to 0 and the delay calculation will return with invalid
+ * results.
+ */
+ hext_stream->pplcllpl = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPL);
+ hext_stream->pplcllpu = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPU);
break;
default:
dev_err(sdev->dev, "unknown trigger command %d\n", cmd);
diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index 31ffa1a8f2ac04ddd5c31aadec5400c52757dd19..ef5c915db8ffb47a622a8c753f14dd950fb9c45c 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -681,17 +681,27 @@ static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend)
struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata;
const struct sof_intel_dsp_desc *chip = hda->desc;
struct hdac_bus *bus = sof_to_bus(sdev);
+ bool imr_lost = false;
int ret, j;
/*
- * The memory used for IMR boot loses its content in deeper than S3 state
- * We must not try IMR boot on next power up (as it will fail).
- *
+ * The memory used for IMR boot loses its content in deeper than S3
+ * state on CAVS platforms.
+ * On ACE platforms due to the system architecture the IMR content is
+ * lost at S3 state already, they are tailored for s2idle use.
+ * We must not try IMR boot on next power up in these cases as it will
+ * fail.
+ */
+ if (sdev->system_suspend_target > SOF_SUSPEND_S3 ||
+ (chip->hw_ip_version >= SOF_INTEL_ACE_1_0 &&
+ sdev->system_suspend_target == SOF_SUSPEND_S3))
+ imr_lost = true;
+
+ /*
* In case of firmware crash or boot failure set the skip_imr_boot to true
* as well in order to try to re-load the firmware to do a 'cold' boot.
*/
- if (sdev->system_suspend_target > SOF_SUSPEND_S3 ||
- sdev->fw_state == SOF_FW_CRASHED ||
+ if (imr_lost || sdev->fw_state == SOF_FW_CRASHED ||
sdev->fw_state == SOF_FW_BOOT_FAILED)
hda->skip_imr_boot = true;
diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c
index 18f07364d2198425bffd3e111a546e11b536cd63..d7b446f3f973e3d532d8eaef241aac2f3a30a54d 100644
--- a/sound/soc/sof/intel/hda-pcm.c
+++ b/sound/soc/sof/intel/hda-pcm.c
@@ -259,8 +259,37 @@ int hda_dsp_pcm_open(struct snd_sof_dev *sdev,
snd_pcm_hw_constraint_mask64(substream->runtime, SNDRV_PCM_HW_PARAM_FORMAT,
SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S32);
+ /*
+ * The dsp_max_burst_size_in_ms is the length of the maximum burst size
+ * of the host DMA in the ALSA buffer.
+ *
+ * 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.
+ *
+ * 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)
+ 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,
+ UINT_MAX);
+
/* binding pcm substream to hda stream */
substream->runtime->private_data = &dsp_stream->hstream;
+
+ /*
+ * Reset the llp cache values (they are used for LLP compensation in
+ * case the counter is not reset)
+ */
+ dsp_stream->pplcllpl = 0;
+ dsp_stream->pplcllpu = 0;
+
return 0;
}
diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c
index b387b1a69d7ea3ceaed9fe814b174d9040e3eae1..0c189d3b19c1af6448d5d1264802ef493e5c7b14 100644
--- a/sound/soc/sof/intel/hda-stream.c
+++ b/sound/soc/sof/intel/hda-stream.c
@@ -1063,3 +1063,73 @@ snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream,
return pos;
}
+
+#define merge_u64(u32_u, u32_l) (((u64)(u32_u) << 32) | (u32_l))
+
+/**
+ * hda_dsp_get_stream_llp - Retrieve the LLP (Linear Link Position) of the stream
+ * @sdev: SOF device
+ * @component: ASoC component
+ * @substream: PCM substream
+ *
+ * Returns the raw Linear Link Position value
+ */
+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);
+ u32 llp_l, llp_u;
+
+ /*
+ * The pplc_addr have been calculated during probe in
+ * hda_dsp_stream_init():
+ * pplc_addr = sdev->bar[HDA_DSP_PP_BAR] +
+ * SOF_HDA_PPLC_BASE +
+ * SOF_HDA_PPLC_MULTI * total_stream +
+ * SOF_HDA_PPLC_INTERVAL * stream_index
+ *
+ * Use this pre-calculated address to avoid repeated re-calculation.
+ */
+ llp_l = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPL);
+ llp_u = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPU);
+
+ /* Compensate the LLP counter with the saved offset */
+ if (hext_stream->pplcllpl || hext_stream->pplcllpu)
+ return merge_u64(llp_u, llp_l) -
+ merge_u64(hext_stream->pplcllpu, hext_stream->pplcllpl);
+
+ return merge_u64(llp_u, llp_l);
+}
+
+/**
+ * hda_dsp_get_stream_ldp - Retrieve the LDP (Linear DMA Position) of the stream
+ * @sdev: SOF device
+ * @component: ASoC component
+ * @substream: PCM substream
+ *
+ * Returns the raw Linear Link Position value
+ */
+u64 hda_dsp_get_stream_ldp(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);
+ u32 ldp_l, ldp_u;
+
+ /*
+ * The pphc_addr have been calculated during probe in
+ * hda_dsp_stream_init():
+ * pphc_addr = sdev->bar[HDA_DSP_PP_BAR] +
+ * SOF_HDA_PPHC_BASE +
+ * SOF_HDA_PPHC_INTERVAL * stream_index
+ *
+ * Use this pre-calculated address to avoid repeated re-calculation.
+ */
+ ldp_l = readl(hext_stream->pphc_addr + AZX_REG_PPHCLDPL);
+ ldp_u = readl(hext_stream->pphc_addr + AZX_REG_PPHCLDPU);
+
+ return ((u64)ldp_u << 32) | ldp_l;
+}
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index b36eb7c7891335a3038d5e1402d6f73ede754b81..81a1d4606d3cde8ecb1b9b2ef859c7b0393555f5 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -662,6 +662,12 @@ bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev);
snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream,
int direction, bool can_sleep);
+u64 hda_dsp_get_stream_llp(struct snd_sof_dev *sdev,
+ struct snd_soc_component *component,
+ struct snd_pcm_substream *substream);
+u64 hda_dsp_get_stream_ldp(struct snd_sof_dev *sdev,
+ struct snd_soc_component *component,
+ struct snd_pcm_substream *substream);
struct hdac_ext_stream *
hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction, u32 flags);
diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c
index 7ae017a00184e52371052c188d75f13fbfc053df..aeb4350cce6bba3b229af876e188c4ead7f2b201 100644
--- a/sound/soc/sof/intel/lnl.c
+++ b/sound/soc/sof/intel/lnl.c
@@ -29,15 +29,17 @@ static const struct snd_sof_debugfs_map lnl_dsp_debugfs[] = {
};
/* this helps allows the DSP to setup DMIC/SSP */
-static int hdac_bus_offload_dmic_ssp(struct hdac_bus *bus)
+static int hdac_bus_offload_dmic_ssp(struct hdac_bus *bus, bool enable)
{
int ret;
- ret = hdac_bus_eml_enable_offload(bus, true, AZX_REG_ML_LEPTR_ID_INTEL_SSP, true);
+ ret = hdac_bus_eml_enable_offload(bus, true,
+ AZX_REG_ML_LEPTR_ID_INTEL_SSP, enable);
if (ret < 0)
return ret;
- ret = hdac_bus_eml_enable_offload(bus, true, AZX_REG_ML_LEPTR_ID_INTEL_DMIC, true);
+ ret = hdac_bus_eml_enable_offload(bus, true,
+ AZX_REG_ML_LEPTR_ID_INTEL_DMIC, enable);
if (ret < 0)
return ret;
@@ -52,7 +54,19 @@ static int lnl_hda_dsp_probe(struct snd_sof_dev *sdev)
if (ret < 0)
return ret;
- return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev));
+ return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev), true);
+}
+
+static void lnl_hda_dsp_remove(struct snd_sof_dev *sdev)
+{
+ int ret;
+
+ ret = hdac_bus_offload_dmic_ssp(sof_to_bus(sdev), false);
+ if (ret < 0)
+ dev_warn(sdev->dev,
+ "Failed to disable offload for DMIC/SSP: %d\n", ret);
+
+ hda_dsp_remove(sdev);
}
static int lnl_hda_dsp_resume(struct snd_sof_dev *sdev)
@@ -63,7 +77,7 @@ static int lnl_hda_dsp_resume(struct snd_sof_dev *sdev)
if (ret < 0)
return ret;
- return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev));
+ return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev), true);
}
static int lnl_hda_dsp_runtime_resume(struct snd_sof_dev *sdev)
@@ -74,7 +88,7 @@ static int lnl_hda_dsp_runtime_resume(struct snd_sof_dev *sdev)
if (ret < 0)
return ret;
- return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev));
+ return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev), true);
}
static int lnl_dsp_post_fw_run(struct snd_sof_dev *sdev)
@@ -97,9 +111,11 @@ int sof_lnl_ops_init(struct snd_sof_dev *sdev)
/* common defaults */
memcpy(&sof_lnl_ops, &sof_hda_common_ops, sizeof(struct snd_sof_dsp_ops));
- /* probe */
- if (!sdev->dspless_mode_selected)
+ /* probe/remove */
+ if (!sdev->dspless_mode_selected) {
sof_lnl_ops.probe = lnl_hda_dsp_probe;
+ sof_lnl_ops.remove = lnl_hda_dsp_remove;
+ }
/* shutdown */
sof_lnl_ops.shutdown = hda_dsp_shutdown;
@@ -134,8 +150,6 @@ int sof_lnl_ops_init(struct snd_sof_dev *sdev)
sof_lnl_ops.runtime_resume = lnl_hda_dsp_runtime_resume;
}
- sof_lnl_ops.get_stream_position = mtl_dsp_get_stream_hda_link_position;
-
/* dsp core get/put */
sof_lnl_ops.core_get = mtl_dsp_core_get;
sof_lnl_ops.core_put = mtl_dsp_core_put;
diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c
index df05dc77b8d5e3bef5f3e55ea7e82837b5a89504..060c34988e90d122caf12cc30fe42ba5f1d0c87d 100644
--- a/sound/soc/sof/intel/mtl.c
+++ b/sound/soc/sof/intel/mtl.c
@@ -626,18 +626,6 @@ static int mtl_dsp_disable_interrupts(struct snd_sof_dev *sdev)
return mtl_enable_interrupts(sdev, false);
}
-u64 mtl_dsp_get_stream_hda_link_position(struct snd_sof_dev *sdev,
- struct snd_soc_component *component,
- struct snd_pcm_substream *substream)
-{
- struct hdac_stream *hstream = substream->runtime->private_data;
- u32 llp_l, llp_u;
-
- llp_l = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, MTL_PPLCLLPL(hstream->index));
- llp_u = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, MTL_PPLCLLPU(hstream->index));
- return ((u64)llp_u << 32) | llp_l;
-}
-
int mtl_dsp_core_get(struct snd_sof_dev *sdev, int core)
{
const struct sof_ipc_pm_ops *pm_ops = sdev->ipc->ops->pm;
@@ -707,8 +695,6 @@ int sof_mtl_ops_init(struct snd_sof_dev *sdev)
sof_mtl_ops.core_get = mtl_dsp_core_get;
sof_mtl_ops.core_put = mtl_dsp_core_put;
- sof_mtl_ops.get_stream_position = mtl_dsp_get_stream_hda_link_position;
-
sdev->private = kzalloc(sizeof(struct sof_ipc4_fw_data), GFP_KERNEL);
if (!sdev->private)
return -ENOMEM;
diff --git a/sound/soc/sof/intel/mtl.h b/sound/soc/sof/intel/mtl.h
index cc5a1f46fd09560e9fefc10d6b4775b82294bfd4..ea8c1b83f7127d58f76bd5db018eeb0f0d9b1d7f 100644
--- a/sound/soc/sof/intel/mtl.h
+++ b/sound/soc/sof/intel/mtl.h
@@ -6,12 +6,6 @@
* Copyright(c) 2020-2022 Intel Corporation. All rights reserved.
*/
-/* HDA Registers */
-#define MTL_PPLCLLPL_BASE 0x948
-#define MTL_PPLCLLPU_STRIDE 0x10
-#define MTL_PPLCLLPL(x) (MTL_PPLCLLPL_BASE + (x) * MTL_PPLCLLPU_STRIDE)
-#define MTL_PPLCLLPU(x) (MTL_PPLCLLPL_BASE + 0x4 + (x) * MTL_PPLCLLPU_STRIDE)
-
/* DSP Registers */
#define MTL_HFDSSCS 0x1000
#define MTL_HFDSSCS_SPA_MASK BIT(16)
@@ -103,9 +97,5 @@ int mtl_dsp_ipc_get_window_offset(struct snd_sof_dev *sdev, u32 id);
void mtl_ipc_dump(struct snd_sof_dev *sdev);
-u64 mtl_dsp_get_stream_hda_link_position(struct snd_sof_dev *sdev,
- struct snd_soc_component *component,
- struct snd_pcm_substream *substream);
-
int mtl_dsp_core_get(struct snd_sof_dev *sdev, int core);
int mtl_dsp_core_put(struct snd_sof_dev *sdev, int core);
diff --git a/sound/soc/sof/ipc4-mtrace.c b/sound/soc/sof/ipc4-mtrace.c
index 9f1e33ee8826123cdc57bd0da78b876e79bf6f27..0e04bea9432ddab2e60b2f61d209689d560b39fb 100644
--- a/sound/soc/sof/ipc4-mtrace.c
+++ b/sound/soc/sof/ipc4-mtrace.c
@@ -4,6 +4,7 @@
#include <linux/debugfs.h>
#include <linux/sched/signal.h>
+#include <linux/sched/clock.h>
#include <sound/sof/ipc4/header.h>
#include "sof-priv.h"
#include "ipc4-priv.h"
@@ -412,7 +413,6 @@ static int ipc4_mtrace_enable(struct snd_sof_dev *sdev)
const struct sof_ipc_ops *iops = sdev->ipc->ops;
struct sof_ipc4_msg msg;
u64 system_time;
- ktime_t kt;
int ret;
if (priv->mtrace_state != SOF_MTRACE_DISABLED)
@@ -424,9 +424,12 @@ static int ipc4_mtrace_enable(struct snd_sof_dev *sdev)
msg.primary |= SOF_IPC4_MOD_INSTANCE(SOF_IPC4_MOD_INIT_BASEFW_INSTANCE_ID);
msg.extension = SOF_IPC4_MOD_EXT_MSG_PARAM_ID(SOF_IPC4_FW_PARAM_SYSTEM_TIME);
- /* The system time is in usec, UTC, epoch is 1601-01-01 00:00:00 */
- kt = ktime_add_us(ktime_get_real(), FW_EPOCH_DELTA * USEC_PER_SEC);
- system_time = ktime_to_us(kt);
+ /*
+ * local_clock() is used to align with dmesg, so both kernel and firmware logs have
+ * the same base and a minor delta due to the IPC. system time is in us format but
+ * local_clock() returns the time in ns, so convert to ns.
+ */
+ system_time = div64_u64(local_clock(), NSEC_PER_USEC);
msg.data_size = sizeof(system_time);
msg.data_ptr = &system_time;
ret = iops->set_get_data(sdev, &msg, msg.data_size, true);
diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c
index 0f332c8cdbe6afe6fc9449b48194ebb749db5d3d..e915f9f87a6c35d74f1cf7096accca70dce688da 100644
--- a/sound/soc/sof/ipc4-pcm.c
+++ b/sound/soc/sof/ipc4-pcm.c
@@ -15,6 +15,28 @@
#include "ipc4-topology.h"
#include "ipc4-fw-reg.h"
+/**
+ * struct sof_ipc4_timestamp_info - IPC4 timestamp info
+ * @host_copier: the host copier of the pcm stream
+ * @dai_copier: the dai copier of the pcm stream
+ * @stream_start_offset: reported by fw in memory window (converted to frames)
+ * @stream_end_offset: reported by fw in memory window (converted to frames)
+ * @llp_offset: llp offset in memory window
+ * @boundary: wrap boundary should be used for the LLP frame counter
+ * @delay: Calculated and stored in pointer callback. The stored value is
+ * returned in the delay callback.
+ */
+struct sof_ipc4_timestamp_info {
+ struct sof_ipc4_copier *host_copier;
+ struct sof_ipc4_copier *dai_copier;
+ u64 stream_start_offset;
+ u64 stream_end_offset;
+ u32 llp_offset;
+
+ u64 boundary;
+ snd_pcm_sframes_t delay;
+};
+
static int sof_ipc4_set_multi_pipeline_state(struct snd_sof_dev *sdev, u32 state,
struct ipc4_pipeline_set_state_data *trigger_list)
{
@@ -423,8 +445,19 @@ static int sof_ipc4_trigger_pipelines(struct snd_soc_component *component,
}
/* return if this is the final state */
- if (state == SOF_IPC4_PIPE_PAUSED)
+ if (state == SOF_IPC4_PIPE_PAUSED) {
+ struct sof_ipc4_timestamp_info *time_info;
+
+ /*
+ * Invalidate the stream_start_offset to make sure that it is
+ * going to be updated if the stream resumes
+ */
+ time_info = spcm->stream[substream->stream].private;
+ if (time_info)
+ time_info->stream_start_offset = SOF_IPC4_INVALID_STREAM_POSITION;
+
goto free;
+ }
skip_pause_transition:
/* else set the RUNNING/RESET state in the DSP */
ret = sof_ipc4_set_multi_pipeline_state(sdev, state, trigger_list);
@@ -464,14 +497,12 @@ static int sof_ipc4_pcm_trigger(struct snd_soc_component *component,
/* determine the pipeline state */
switch (cmd) {
- case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- state = SOF_IPC4_PIPE_PAUSED;
- break;
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_START:
state = SOF_IPC4_PIPE_RUNNING;
break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_STOP:
state = SOF_IPC4_PIPE_PAUSED;
@@ -703,6 +734,10 @@ static int sof_ipc4_pcm_setup(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm
if (abi_version < SOF_IPC4_FW_REGS_ABI_VER)
support_info = false;
+ /* For delay reporting the get_host_byte_counter callback is needed */
+ if (!sof_ops(sdev) || !sof_ops(sdev)->get_host_byte_counter)
+ support_info = false;
+
for_each_pcm_streams(stream) {
pipeline_list = &spcm->stream[stream].pipeline_list;
@@ -835,7 +870,6 @@ static int sof_ipc4_get_stream_start_offset(struct snd_sof_dev *sdev,
struct sof_ipc4_copier *host_copier = time_info->host_copier;
struct sof_ipc4_copier *dai_copier = time_info->dai_copier;
struct sof_ipc4_pipeline_registers ppl_reg;
- u64 stream_start_position;
u32 dai_sample_size;
u32 ch, node_index;
u32 offset;
@@ -852,38 +886,51 @@ static int sof_ipc4_get_stream_start_offset(struct snd_sof_dev *sdev,
if (ppl_reg.stream_start_offset == SOF_IPC4_INVALID_STREAM_POSITION)
return -EINVAL;
- stream_start_position = ppl_reg.stream_start_offset;
ch = dai_copier->data.out_format.fmt_cfg;
ch = SOF_IPC4_AUDIO_FORMAT_CFG_CHANNELS_COUNT(ch);
dai_sample_size = (dai_copier->data.out_format.bit_depth >> 3) * ch;
- /* convert offset to sample count */
- do_div(stream_start_position, dai_sample_size);
- time_info->stream_start_offset = stream_start_position;
+
+ /* convert offsets to frame count */
+ time_info->stream_start_offset = ppl_reg.stream_start_offset;
+ do_div(time_info->stream_start_offset, dai_sample_size);
+ time_info->stream_end_offset = ppl_reg.stream_end_offset;
+ do_div(time_info->stream_end_offset, dai_sample_size);
+
+ /*
+ * Calculate the wrap boundary need to be used for delay calculation
+ * The host counter is in bytes, it will wrap earlier than the frames
+ * based link counter.
+ */
+ time_info->boundary = div64_u64(~((u64)0),
+ frames_to_bytes(substream->runtime, 1));
+ /* Initialize the delay value to 0 (no delay) */
+ time_info->delay = 0;
return 0;
}
-static snd_pcm_sframes_t sof_ipc4_pcm_delay(struct snd_soc_component *component,
- struct snd_pcm_substream *substream)
+static int sof_ipc4_pcm_pointer(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream,
+ snd_pcm_uframes_t *pointer)
{
struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component);
struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct sof_ipc4_timestamp_info *time_info;
struct sof_ipc4_llp_reading_slot llp;
- snd_pcm_uframes_t head_ptr, tail_ptr;
+ snd_pcm_uframes_t head_cnt, tail_cnt;
struct snd_sof_pcm_stream *stream;
+ u64 dai_cnt, host_cnt, host_ptr;
struct snd_sof_pcm *spcm;
- u64 tmp_ptr;
int ret;
spcm = snd_sof_find_spcm_dai(component, rtd);
if (!spcm)
- return 0;
+ return -EOPNOTSUPP;
stream = &spcm->stream[substream->stream];
time_info = stream->private;
if (!time_info)
- return 0;
+ return -EOPNOTSUPP;
/*
* stream_start_offset is updated to memory window by FW based on
@@ -893,45 +940,116 @@ static snd_pcm_sframes_t sof_ipc4_pcm_delay(struct snd_soc_component *component,
if (time_info->stream_start_offset == SOF_IPC4_INVALID_STREAM_POSITION) {
ret = sof_ipc4_get_stream_start_offset(sdev, substream, stream, time_info);
if (ret < 0)
- return 0;
+ return -EOPNOTSUPP;
}
+ /* For delay calculation we need the host counter */
+ host_cnt = snd_sof_pcm_get_host_byte_counter(sdev, component, substream);
+ host_ptr = host_cnt;
+
+ /* convert the host_cnt to frames */
+ host_cnt = div64_u64(host_cnt, frames_to_bytes(substream->runtime, 1));
+
/*
- * HDaudio links don't support the LLP counter reported by firmware
- * the link position is read directly from hardware registers.
+ * If the LLP counter is not reported by firmware in the SRAM window
+ * then read the dai (link) counter via host accessible means if
+ * available.
*/
if (!time_info->llp_offset) {
- tmp_ptr = snd_sof_pcm_get_stream_position(sdev, component, substream);
- if (!tmp_ptr)
- return 0;
+ dai_cnt = snd_sof_pcm_get_dai_frame_counter(sdev, component, substream);
+ if (!dai_cnt)
+ return -EOPNOTSUPP;
} else {
sof_mailbox_read(sdev, time_info->llp_offset, &llp, sizeof(llp));
- tmp_ptr = ((u64)llp.reading.llp_u << 32) | llp.reading.llp_l;
+ dai_cnt = ((u64)llp.reading.llp_u << 32) | llp.reading.llp_l;
}
+ dai_cnt += time_info->stream_end_offset;
- /* In two cases dai dma position is not accurate
+ /* In two cases dai dma counter is not accurate
* (1) dai pipeline is started before host pipeline
- * (2) multiple streams mixed into one. Each stream has the same dai dma position
+ * (2) multiple streams mixed into one. Each stream has the same dai dma
+ * counter
*
- * Firmware calculates correct stream_start_offset for all cases including above two.
- * Driver subtracts stream_start_offset from dai dma position to get accurate one
+ * Firmware calculates correct stream_start_offset for all cases
+ * including above two.
+ * Driver subtracts stream_start_offset from dai dma counter to get
+ * accurate one
*/
- tmp_ptr -= time_info->stream_start_offset;
- /* Calculate the delay taking into account that both pointer can wrap */
- div64_u64_rem(tmp_ptr, substream->runtime->boundary, &tmp_ptr);
+ /*
+ * On stream start the dai counter might not yet have reached the
+ * stream_start_offset value which means that no frames have left the
+ * DSP yet from the audio stream (on playback, capture streams have
+ * offset of 0 as we start capturing right away).
+ * In this case we need to adjust the distance between the counters by
+ * increasing the host counter by (offset - dai_counter).
+ * Otherwise the dai_counter needs to be adjusted to reflect the number
+ * of valid frames passed on the DAI side.
+ *
+ * The delay is the difference between the counters on the two
+ * sides of the DSP.
+ */
+ if (dai_cnt < time_info->stream_start_offset) {
+ host_cnt += time_info->stream_start_offset - dai_cnt;
+ dai_cnt = 0;
+ } else {
+ dai_cnt -= time_info->stream_start_offset;
+ }
+
+ /* Wrap the dai counter at the boundary where the host counter wraps */
+ div64_u64_rem(dai_cnt, time_info->boundary, &dai_cnt);
+
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- head_ptr = substream->runtime->status->hw_ptr;
- tail_ptr = tmp_ptr;
+ head_cnt = host_cnt;
+ tail_cnt = dai_cnt;
} else {
- head_ptr = tmp_ptr;
- tail_ptr = substream->runtime->status->hw_ptr;
+ head_cnt = dai_cnt;
+ tail_cnt = host_cnt;
+ }
+
+ if (head_cnt < tail_cnt) {
+ time_info->delay = time_info->boundary - tail_cnt + head_cnt;
+ goto out;
}
- if (head_ptr < tail_ptr)
- return substream->runtime->boundary - tail_ptr + head_ptr;
+ time_info->delay = head_cnt - tail_cnt;
+
+out:
+ /*
+ * Convert the host byte counter to PCM pointer which wraps in buffer
+ * and it is in frames
+ */
+ div64_u64_rem(host_ptr, snd_pcm_lib_buffer_bytes(substream), &host_ptr);
+ *pointer = bytes_to_frames(substream->runtime, host_ptr);
+
+ return 0;
+}
+
+static snd_pcm_sframes_t sof_ipc4_pcm_delay(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
+ struct sof_ipc4_timestamp_info *time_info;
+ struct snd_sof_pcm_stream *stream;
+ struct snd_sof_pcm *spcm;
+
+ spcm = snd_sof_find_spcm_dai(component, rtd);
+ if (!spcm)
+ return 0;
+
+ stream = &spcm->stream[substream->stream];
+ time_info = stream->private;
+ /*
+ * Report the stored delay value calculated in the pointer callback.
+ * In the unlikely event that the calculation was skipped/aborted, the
+ * default 0 delay returned.
+ */
+ if (time_info)
+ return time_info->delay;
+
+ /* No delay information available, report 0 as delay */
+ return 0;
- return head_ptr - tail_ptr;
}
const struct sof_ipc_pcm_ops ipc4_pcm_ops = {
@@ -941,6 +1059,7 @@ const struct sof_ipc_pcm_ops ipc4_pcm_ops = {
.dai_link_fixup = sof_ipc4_pcm_dai_link_fixup,
.pcm_setup = sof_ipc4_pcm_setup,
.pcm_free = sof_ipc4_pcm_free,
+ .pointer = sof_ipc4_pcm_pointer,
.delay = sof_ipc4_pcm_delay,
.ipc_first_on_start = true,
.platform_stop_during_hw_free = true,
diff --git a/sound/soc/sof/ipc4-priv.h b/sound/soc/sof/ipc4-priv.h
index f3b908b093f9562ddeb6932b4f1743a27c2b3c09..afed618a15f061a8588466490ee38ea19a80bc3d 100644
--- a/sound/soc/sof/ipc4-priv.h
+++ b/sound/soc/sof/ipc4-priv.h
@@ -92,20 +92,6 @@ struct sof_ipc4_fw_data {
struct mutex pipeline_state_mutex; /* protect pipeline triggers, ref counts and states */
};
-/**
- * struct sof_ipc4_timestamp_info - IPC4 timestamp info
- * @host_copier: the host copier of the pcm stream
- * @dai_copier: the dai copier of the pcm stream
- * @stream_start_offset: reported by fw in memory window
- * @llp_offset: llp offset in memory window
- */
-struct sof_ipc4_timestamp_info {
- struct sof_ipc4_copier *host_copier;
- struct sof_ipc4_copier *dai_copier;
- u64 stream_start_offset;
- u32 llp_offset;
-};
-
extern const struct sof_ipc_fw_loader_ops ipc4_loader_ops;
extern const struct sof_ipc_tplg_ops ipc4_tplg_ops;
extern const struct sof_ipc_tplg_control_ops tplg_ipc4_control_ops;
diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index f28edd9830c1b3e25961add70dff87a489cfa119..5cca058421260978dd18e992b09dfff58b44bbdb 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -412,8 +412,9 @@ static int sof_ipc4_widget_setup_pcm(struct snd_sof_widget *swidget)
struct sof_ipc4_available_audio_format *available_fmt;
struct snd_soc_component *scomp = swidget->scomp;
struct sof_ipc4_copier *ipc4_copier;
+ struct snd_sof_pcm *spcm;
int node_type = 0;
- int ret;
+ int ret, dir;
ipc4_copier = kzalloc(sizeof(*ipc4_copier), GFP_KERNEL);
if (!ipc4_copier)
@@ -447,6 +448,25 @@ static int sof_ipc4_widget_setup_pcm(struct snd_sof_widget *swidget)
}
dev_dbg(scomp->dev, "host copier '%s' node_type %u\n", swidget->widget->name, node_type);
+ spcm = snd_sof_find_spcm_comp(scomp, swidget->comp_id, &dir);
+ if (!spcm)
+ goto skip_gtw_cfg;
+
+ if (dir == SNDRV_PCM_STREAM_PLAYBACK) {
+ struct snd_sof_pcm_stream *sps = &spcm->stream[dir];
+
+ sof_update_ipc_object(scomp, &sps->dsp_max_burst_size_in_ms,
+ SOF_COPIER_DEEP_BUFFER_TOKENS,
+ swidget->tuples,
+ swidget->num_tuples, sizeof(u32), 1);
+ /* Set default DMA buffer size if it is not specified in topology */
+ if (!sps->dsp_max_burst_size_in_ms)
+ sps->dsp_max_burst_size_in_ms = SOF_IPC4_MIN_DMA_BUFFER_SIZE;
+ } else {
+ /* Capture data is copied from DSP to host in 1ms bursts */
+ spcm->stream[dir].dsp_max_burst_size_in_ms = 1;
+ }
+
skip_gtw_cfg:
ipc4_copier->gtw_attr = kzalloc(sizeof(*ipc4_copier->gtw_attr), GFP_KERNEL);
if (!ipc4_copier->gtw_attr) {
diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h
index 6cf21e829e07272ccf4c7005f74f9ae61403d39b..3cd748e13460916517d9533c48d2172d556fc344 100644
--- a/sound/soc/sof/ops.h
+++ b/sound/soc/sof/ops.h
@@ -523,12 +523,26 @@ static inline int snd_sof_pcm_platform_ack(struct snd_sof_dev *sdev,
return 0;
}
-static inline u64 snd_sof_pcm_get_stream_position(struct snd_sof_dev *sdev,
- struct snd_soc_component *component,
- struct snd_pcm_substream *substream)
+static inline u64
+snd_sof_pcm_get_dai_frame_counter(struct snd_sof_dev *sdev,
+ struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
{
- if (sof_ops(sdev) && sof_ops(sdev)->get_stream_position)
- return sof_ops(sdev)->get_stream_position(sdev, component, substream);
+ if (sof_ops(sdev) && sof_ops(sdev)->get_dai_frame_counter)
+ return sof_ops(sdev)->get_dai_frame_counter(sdev, component,
+ substream);
+
+ return 0;
+}
+
+static inline u64
+snd_sof_pcm_get_host_byte_counter(struct snd_sof_dev *sdev,
+ struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
+{
+ if (sof_ops(sdev) && sof_ops(sdev)->get_host_byte_counter)
+ return sof_ops(sdev)->get_host_byte_counter(sdev, component,
+ substream);
return 0;
}
diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c
index 33d576b1764783ab3468591703e0c97106893e9e..f03cee94bce62642e3c419d4f956a2011ea4dd3f 100644
--- a/sound/soc/sof/pcm.c
+++ b/sound/soc/sof/pcm.c
@@ -388,13 +388,21 @@ static snd_pcm_uframes_t sof_pcm_pointer(struct snd_soc_component *component,
{
struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component);
+ const struct sof_ipc_pcm_ops *pcm_ops = sof_ipc_get_ops(sdev, pcm);
struct snd_sof_pcm *spcm;
snd_pcm_uframes_t host, dai;
+ int ret = -EOPNOTSUPP;
/* nothing to do for BE */
if (rtd->dai_link->no_pcm)
return 0;
+ if (pcm_ops && pcm_ops->pointer)
+ ret = pcm_ops->pointer(component, substream, &host);
+
+ if (ret != -EOPNOTSUPP)
+ return ret ? ret : host;
+
/* use dsp ops pointer callback directly if set */
if (sof_ops(sdev)->pcm_pointer)
return sof_ops(sdev)->pcm_pointer(sdev, substream);
diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h
index 9ea2ac5adac79ee322f82060b908ce529cd9c43b..86bbb531e142c72be1ca5d710c466d16c9058734 100644
--- a/sound/soc/sof/sof-audio.h
+++ b/sound/soc/sof/sof-audio.h
@@ -103,7 +103,10 @@ struct snd_sof_dai_config_data {
* additional memory in the SOF PCM stream structure
* @pcm_free: Function pointer for PCM free that can be used for freeing any
* additional memory in the SOF PCM stream structure
- * @delay: Function pointer for pcm delay calculation
+ * @pointer: Function pointer for pcm pointer
+ * Note: the @pointer callback may return -EOPNOTSUPP which should be
+ * handled in a same way as if the callback is not provided
+ * @delay: Function pointer for pcm delay reporting
* @reset_hw_params_during_stop: Flag indicating whether the hw_params should be reset during the
* STOP pcm trigger
* @ipc_first_on_start: Send IPC before invoking platform trigger during
@@ -124,6 +127,9 @@ struct sof_ipc_pcm_ops {
int (*dai_link_fixup)(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params);
int (*pcm_setup)(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm);
void (*pcm_free)(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm);
+ int (*pointer)(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream,
+ snd_pcm_uframes_t *pointer);
snd_pcm_sframes_t (*delay)(struct snd_soc_component *component,
struct snd_pcm_substream *substream);
bool reset_hw_params_during_stop;
@@ -322,6 +328,7 @@ struct snd_sof_pcm_stream {
struct work_struct period_elapsed_work;
struct snd_soc_dapm_widget_list *list; /* list of connected DAPM widgets */
bool d0i3_compatible; /* DSP can be in D0I3 when this pcm is opened */
+ unsigned int dsp_max_burst_size_in_ms; /* The maximum size of the host DMA burst in ms */
/*
* flag to indicate that the DSP pipelines should be kept
* active or not while suspending the stream
diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h
index d453a4ce3b219d601813310c22cbf11029a08a77..d3c436f826046bca9f385b429d6d7e1639600f63 100644
--- a/sound/soc/sof/sof-priv.h
+++ b/sound/soc/sof/sof-priv.h
@@ -262,13 +262,25 @@ struct snd_sof_dsp_ops {
int (*pcm_ack)(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream); /* optional */
/*
- * optional callback to retrieve the link DMA position for the substream
- * when the position is not reported in the shared SRAM windows but
- * instead from a host-accessible hardware counter.
+ * optional callback to retrieve the number of frames left/arrived from/to
+ * the DSP on the DAI side (link/codec/DMIC/etc).
+ *
+ * The callback is used when the firmware does not provide this information
+ * via the shared SRAM window and it can be retrieved by host.
*/
- u64 (*get_stream_position)(struct snd_sof_dev *sdev,
- struct snd_soc_component *component,
- struct snd_pcm_substream *substream); /* optional */
+ u64 (*get_dai_frame_counter)(struct snd_sof_dev *sdev,
+ struct snd_soc_component *component,
+ struct snd_pcm_substream *substream); /* optional */
+
+ /*
+ * Optional callback to retrieve the number of bytes left/arrived from/to
+ * the DSP on the host side (bytes between host ALSA buffer and DSP).
+ *
+ * The callback is needed for ALSA delay reporting.
+ */
+ u64 (*get_host_byte_counter)(struct snd_sof_dev *sdev,
+ struct snd_soc_component *component,
+ struct snd_pcm_substream *substream); /* optional */
/* host read DSP stream data */
int (*ipc_msg_data)(struct snd_sof_dev *sdev,
diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c
index b67617b68e509d2c86d78058f7796a64aab00f41..f4437015d43a7500b809a303f175b211662d500f 100644
--- a/sound/usb/line6/driver.c
+++ b/sound/usb/line6/driver.c
@@ -202,7 +202,7 @@ int line6_send_raw_message_async(struct usb_line6 *line6, const char *buffer,
struct urb *urb;
/* create message: */
- msg = kmalloc(sizeof(struct message), GFP_ATOMIC);
+ msg = kzalloc(sizeof(struct message), GFP_ATOMIC);
if (msg == NULL)
return -ENOMEM;
@@ -688,7 +688,7 @@ static int line6_init_cap_control(struct usb_line6 *line6)
int ret;
/* initialize USB buffers: */
- line6->buffer_listen = kmalloc(LINE6_BUFSIZE_LISTEN, GFP_KERNEL);
+ line6->buffer_listen = kzalloc(LINE6_BUFSIZE_LISTEN, GFP_KERNEL);
if (!line6->buffer_listen)
return -ENOMEM;
@@ -697,7 +697,7 @@ static int line6_init_cap_control(struct usb_line6 *line6)
return -ENOMEM;
if (line6->properties->capabilities & LINE6_CAP_CONTROL_MIDI) {
- line6->buffer_message = kmalloc(LINE6_MIDI_MESSAGE_MAXLEN, GFP_KERNEL);
+ line6->buffer_message = kzalloc(LINE6_MIDI_MESSAGE_MAXLEN, GFP_KERNEL);
if (!line6->buffer_message)
return -ENOMEM;