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authorLinus Torvalds <torvalds@linux-foundation.org>2022-07-14 21:34:16 +0300
committerLinus Torvalds <torvalds@linux-foundation.org>2022-07-14 21:34:16 +0300
commitc4634a3c7dcabed7321304efc00b5a81559adeca (patch)
tree004245c547057e25a346ff96322ef32f761adfe1
parentd11219ad53dcf61ced53ca60fe0c4a8d34393e6c (diff)
parent9b043a8f386485c74c0f8eea2c287d5bdbdf3279 (diff)
downloadlinux-c4634a3c7dcabed7321304efc00b5a81559adeca.tar.xz
Merge tag 'sound-5.19-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai: "Hopefully the last one for 5.19. This became bigger than wished, but all changes are pretty device-specific small fixes, which look less worrisome. The majority of changes are about various ASoC fixes, while the usual HD-audio quirks are included as well" * tag 'sound-5.19-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (28 commits) ALSA: hda/realtek - Enable the headset-mic on a Xiaomi's laptop ALSA: hda/realtek - Fix headset mic problem for a HP machine with alc221 ALSA: hda/realtek: fix mute/micmute LEDs for HP machines ALSA: hda/realtek - Fix headset mic problem for a HP machine with alc671 ALSA: hda - Add fixup for Dell Latitidue E5430 ALSA: hda/conexant: Apply quirk for another HP ProDesk 600 G3 model ALSA: hda/realtek: Fix headset mic for Acer SF313-51 ASoC: Intel: Skylake: Correct the handling of fmt_config flexible array ASoC: Intel: Skylake: Correct the ssp rate discovery in skl_get_ssp_clks() ASoC: rt5640: Fix the wrong state of JD1 and JD2 ASoC: Intel: sof_rt5682: fix out-of-bounds array access ASoC: qdsp6: fix potential memory leak in q6apm_get_audioreach_graph() ASoC: tas2764: Fix amp gain register offset & default ASoC: tas2764: Correct playback volume range ASoC: tas2764: Fix and extend FSYNC polarity handling ASoC: tas2764: Add post reset delays ASoC: dt-bindings: Fix description for msm8916 ASoC: doc: Capitalize RESET line name ASoC: arizona: Update arizona_aif_cfg_changed to use RX_BCLK_RATE ASoC: cs47l92: Fix event generation for OUT1 demux ...
-rw-r--r--Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml8
-rw-r--r--Documentation/sound/soc/dai.rst2
-rw-r--r--sound/pci/hda/patch_conexant.c1
-rw-r--r--sound/pci/hda/patch_realtek.c20
-rw-r--r--sound/soc/codecs/arizona.c4
-rw-r--r--sound/soc/codecs/cs47l92.c8
-rw-r--r--sound/soc/codecs/max98396.c10
-rw-r--r--sound/soc/codecs/rt5640.c30
-rw-r--r--sound/soc/codecs/sgtl5000.c9
-rw-r--r--sound/soc/codecs/sgtl5000.h1
-rw-r--r--sound/soc/codecs/tas2764.c46
-rw-r--r--sound/soc/codecs/tas2764.h6
-rw-r--r--sound/soc/codecs/tlv320adcx140.c13
-rw-r--r--sound/soc/codecs/wcd9335.c17
-rw-r--r--sound/soc/codecs/wm5102.c21
-rw-r--r--sound/soc/codecs/wm8998.c21
-rw-r--r--sound/soc/generic/audio-graph-card2.c6
-rw-r--r--sound/soc/intel/boards/sof_rt5682.c10
-rw-r--r--sound/soc/intel/skylake/skl-nhlt.c40
-rw-r--r--sound/soc/qcom/qdsp6/q6apm.c1
-rw-r--r--sound/soc/ti/omap-mcbsp-priv.h2
-rw-r--r--sound/soc/ti/omap-mcbsp-st.c14
-rw-r--r--sound/soc/ti/omap-mcbsp.c19
23 files changed, 189 insertions, 120 deletions
diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml
index e9a533080b32..ef18a572a1ff 100644
--- a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml
+++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml
@@ -25,12 +25,12 @@ properties:
- qcom,sc7280-lpass-cpu
reg:
- minItems: 2
+ minItems: 1
maxItems: 6
description: LPAIF core registers
reg-names:
- minItems: 2
+ minItems: 1
maxItems: 6
clocks:
@@ -42,12 +42,12 @@ properties:
maxItems: 10
interrupts:
- minItems: 2
+ minItems: 1
maxItems: 4
description: LPAIF DMA buffer interrupt
interrupt-names:
- minItems: 2
+ minItems: 1
maxItems: 4
qcom,adsp:
diff --git a/Documentation/sound/soc/dai.rst b/Documentation/sound/soc/dai.rst
index 009b07e5a0f3..bf8431386d26 100644
--- a/Documentation/sound/soc/dai.rst
+++ b/Documentation/sound/soc/dai.rst
@@ -10,7 +10,7 @@ AC97
====
AC97 is a five wire interface commonly found on many PC sound cards. It is
-now also popular in many portable devices. This DAI has a reset line and time
+now also popular in many portable devices. This DAI has a RESET line and time
multiplexes its data on its SDATA_OUT (playback) and SDATA_IN (capture) lines.
The bit clock (BCLK) is always driven by the CODEC (usually 12.288MHz) and the
frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 3e541a4c0423..83ae21a01bbf 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -944,6 +944,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK),
SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x103c, 0x82b4, "HP ProDesk 600 G3", CXT_FIXUP_HP_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x836e, "HP ProBook 455 G5", CXT_FIXUP_MUTE_LED_GPIO),
SND_PCI_QUIRK(0x103c, 0x837f, "HP ProBook 470 G5", CXT_FIXUP_MUTE_LED_GPIO),
SND_PCI_QUIRK(0x103c, 0x83b2, "HP EliteBook 840 G5", CXT_FIXUP_HP_DOCK),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 007dd8b5e1f2..2f55bc43bfa9 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6901,6 +6901,7 @@ enum {
ALC298_FIXUP_LENOVO_SPK_VOLUME,
ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER,
ALC269_FIXUP_ATIV_BOOK_8,
+ ALC221_FIXUP_HP_288PRO_MIC_NO_PRESENCE,
ALC221_FIXUP_HP_MIC_NO_PRESENCE,
ALC256_FIXUP_ASUS_HEADSET_MODE,
ALC256_FIXUP_ASUS_MIC,
@@ -7837,6 +7838,16 @@ static const struct hda_fixup alc269_fixups[] = {
.chained = true,
.chain_id = ALC269_FIXUP_NO_SHUTUP
},
+ [ALC221_FIXUP_HP_288PRO_MIC_NO_PRESENCE] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */
+ { 0x1a, 0x01813030 }, /* use as headphone mic, without its own jack detect */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_HEADSET_MODE
+ },
[ALC221_FIXUP_HP_MIC_NO_PRESENCE] = {
.type = HDA_FIXUP_PINS,
.v.pins = (const struct hda_pintbl[]) {
@@ -8886,6 +8897,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
SND_PCI_QUIRK(0x1025, 0x129c, "Acer SWIFT SF314-55", ALC256_FIXUP_ACER_HEADSET_MIC),
+ SND_PCI_QUIRK(0x1025, 0x129d, "Acer SWIFT SF313-51", ALC256_FIXUP_ACER_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1025, 0x1300, "Acer SWIFT SF314-56", ALC256_FIXUP_ACER_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC),
@@ -8895,6 +8907,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1025, 0x1466, "Acer Aspire A515-56", ALC255_FIXUP_ACER_HEADPHONE_AND_MIC),
SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
+ SND_PCI_QUIRK(0x1028, 0x053c, "Dell Latitude E5430", ALC292_FIXUP_DELL_E7X),
SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS),
SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X),
SND_PCI_QUIRK(0x1028, 0x05be, "Dell Latitude E6540", ALC292_FIXUP_DELL_E7X),
@@ -9010,6 +9023,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x2335, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+ SND_PCI_QUIRK(0x103c, 0x2b5e, "HP 288 Pro G2 MT", ALC221_FIXUP_HP_288PRO_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x802e, "HP Z240 SFF", ALC221_FIXUP_HP_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x802f, "HP Z240", ALC221_FIXUP_HP_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x8077, "HP", ALC256_FIXUP_HP_HEADSET_MIC),
@@ -9096,6 +9110,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x89c6, "Zbook Fury 17 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x89ca, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
SND_PCI_QUIRK(0x103c, 0x8a78, "HP Dev One", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST),
+ SND_PCI_QUIRK(0x103c, 0x8aa0, "HP ProBook 440 G9 (MB 8A9E)", ALC236_FIXUP_HP_GPIO_LED),
+ SND_PCI_QUIRK(0x103c, 0x8aa3, "HP ProBook 450 G9 (MB 8AA1)", ALC236_FIXUP_HP_GPIO_LED),
+ SND_PCI_QUIRK(0x103c, 0x8aa8, "HP EliteBook 640 G9 (MB 8AA6)", ALC236_FIXUP_HP_GPIO_LED),
+ SND_PCI_QUIRK(0x103c, 0x8aab, "HP EliteBook 650 G9 (MB 8AA9)", ALC236_FIXUP_HP_GPIO_LED),
SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
@@ -9355,6 +9373,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
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),
+ SND_PCI_QUIRK(0x1d72, 0x1945, "Redmi G", ALC256_FIXUP_ASUS_HEADSET_MIC),
SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC),
SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC),
SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED),
@@ -11217,6 +11236,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800),
SND_PCI_QUIRK(0x103c, 0x8719, "HP", ALC897_FIXUP_HP_HSMIC_VERB),
SND_PCI_QUIRK(0x103c, 0x873e, "HP", ALC671_FIXUP_HP_HEADSET_MIC2),
+ SND_PCI_QUIRK(0x103c, 0x877e, "HP 288 Pro G6", ALC671_FIXUP_HP_HEADSET_MIC2),
SND_PCI_QUIRK(0x103c, 0x885f, "HP 288 Pro G8", ALC671_FIXUP_HP_HEADSET_MIC2),
SND_PCI_QUIRK(0x1043, 0x1080, "Asus UX501VW", ALC668_FIXUP_HEADSET_MODE),
SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_ASUS_Nx50),
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index e32871b3f68a..7434aeeda292 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -1760,8 +1760,8 @@ static bool arizona_aif_cfg_changed(struct snd_soc_component *component,
if (bclk != (val & ARIZONA_AIF1_BCLK_FREQ_MASK))
return true;
- val = snd_soc_component_read(component, base + ARIZONA_AIF_TX_BCLK_RATE);
- if (lrclk != (val & ARIZONA_AIF1TX_BCPF_MASK))
+ val = snd_soc_component_read(component, base + ARIZONA_AIF_RX_BCLK_RATE);
+ if (lrclk != (val & ARIZONA_AIF1RX_BCPF_MASK))
return true;
val = snd_soc_component_read(component, base + ARIZONA_AIF_FRAME_CTRL_1);
diff --git a/sound/soc/codecs/cs47l92.c b/sound/soc/codecs/cs47l92.c
index a1b8dcdb9f7b..444026b7d54b 100644
--- a/sound/soc/codecs/cs47l92.c
+++ b/sound/soc/codecs/cs47l92.c
@@ -119,7 +119,13 @@ static int cs47l92_put_demux(struct snd_kcontrol *kcontrol,
end:
snd_soc_dapm_mutex_unlock(dapm);
- return snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL);
+ ret = snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL);
+ if (ret < 0) {
+ dev_err(madera->dev, "Failed to update demux power state: %d\n", ret);
+ return ret;
+ }
+
+ return change;
}
static SOC_ENUM_SINGLE_DECL(cs47l92_outdemux_enum,
diff --git a/sound/soc/codecs/max98396.c b/sound/soc/codecs/max98396.c
index 56eb62bb041f..34db38812807 100644
--- a/sound/soc/codecs/max98396.c
+++ b/sound/soc/codecs/max98396.c
@@ -342,12 +342,15 @@ static int max98396_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
{
struct snd_soc_component *component = codec_dai->component;
struct max98396_priv *max98396 = snd_soc_component_get_drvdata(component);
- unsigned int format = 0;
+ unsigned int format_mask, format = 0;
unsigned int bclk_pol = 0;
int ret, status;
int reg;
bool update = false;
+ format_mask = MAX98396_PCM_MODE_CFG_FORMAT_MASK |
+ MAX98396_PCM_MODE_CFG_LRCLKEDGE;
+
dev_dbg(component->dev, "%s: fmt 0x%08X\n", __func__, fmt);
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
@@ -395,7 +398,7 @@ static int max98396_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
ret = regmap_read(max98396->regmap, MAX98396_R2041_PCM_MODE_CFG, &reg);
if (ret < 0)
return -EINVAL;
- if (format != (reg & MAX98396_PCM_BCLKEDGE_BSEL_MASK)) {
+ if (format != (reg & format_mask)) {
update = true;
} else {
ret = regmap_read(max98396->regmap,
@@ -412,8 +415,7 @@ static int max98396_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
regmap_update_bits(max98396->regmap,
MAX98396_R2041_PCM_MODE_CFG,
- MAX98396_PCM_BCLKEDGE_BSEL_MASK,
- format);
+ format_mask, format);
regmap_update_bits(max98396->regmap,
MAX98396_R2042_PCM_CLK_SETUP,
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 69c80d80ed9d..18b3da9211e3 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -1984,7 +1984,12 @@ static int rt5640_set_bias_level(struct snd_soc_component *component,
snd_soc_component_write(component, RT5640_PWR_DIG2, 0x0000);
snd_soc_component_write(component, RT5640_PWR_VOL, 0x0000);
snd_soc_component_write(component, RT5640_PWR_MIXER, 0x0000);
- snd_soc_component_write(component, RT5640_PWR_ANLG1, 0x0000);
+ if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER)
+ snd_soc_component_write(component, RT5640_PWR_ANLG1,
+ 0x0018);
+ else
+ snd_soc_component_write(component, RT5640_PWR_ANLG1,
+ 0x0000);
snd_soc_component_write(component, RT5640_PWR_ANLG2, 0x0000);
break;
@@ -2393,9 +2398,15 @@ static void rt5640_jack_work(struct work_struct *work)
static irqreturn_t rt5640_irq(int irq, void *data)
{
struct rt5640_priv *rt5640 = data;
+ int delay = 0;
+
+ if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) {
+ cancel_delayed_work_sync(&rt5640->jack_work);
+ delay = 100;
+ }
if (rt5640->jack)
- queue_delayed_work(system_long_wq, &rt5640->jack_work, 0);
+ queue_delayed_work(system_long_wq, &rt5640->jack_work, delay);
return IRQ_HANDLED;
}
@@ -2580,6 +2591,12 @@ static void rt5640_enable_hda_jack_detect(
snd_soc_component_update_bits(component, RT5640_DUMMY1, 0x400, 0x0);
+ snd_soc_component_update_bits(component, RT5640_PWR_ANLG1,
+ RT5640_PWR_VREF2, RT5640_PWR_VREF2);
+ usleep_range(10000, 15000);
+ snd_soc_component_update_bits(component, RT5640_PWR_ANLG1,
+ RT5640_PWR_FV2, RT5640_PWR_FV2);
+
rt5640->jack = jack;
ret = request_irq(rt5640->irq, rt5640_irq,
@@ -2696,16 +2713,13 @@ static int rt5640_probe(struct snd_soc_component *component)
if (device_property_read_u32(component->dev,
"realtek,jack-detect-source", &val) == 0) {
- if (val <= RT5640_JD_SRC_GPIO4) {
+ if (val <= RT5640_JD_SRC_GPIO4)
rt5640->jd_src = val << RT5640_JD_SFT;
- } else if (val == RT5640_JD_SRC_HDA_HEADER) {
+ else if (val == RT5640_JD_SRC_HDA_HEADER)
rt5640->jd_src = RT5640_JD_SRC_HDA_HEADER;
- snd_soc_component_update_bits(component, RT5640_DUMMY1,
- 0x0300, 0x0);
- } else {
+ else
dev_warn(component->dev, "Warning: Invalid jack-detect-source value: %d, leaving jack-detect disabled\n",
val);
- }
}
if (!device_property_read_bool(component->dev, "realtek,jack-detect-not-inverted"))
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 2aa48aef6a97..3363d1696ad7 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1795,6 +1795,9 @@ static int sgtl5000_i2c_remove(struct i2c_client *client)
{
struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client);
+ regmap_write(sgtl5000->regmap, SGTL5000_CHIP_DIG_POWER, SGTL5000_DIG_POWER_DEFAULT);
+ regmap_write(sgtl5000->regmap, SGTL5000_CHIP_ANA_POWER, SGTL5000_ANA_POWER_DEFAULT);
+
clk_disable_unprepare(sgtl5000->mclk);
regulator_bulk_disable(sgtl5000->num_supplies, sgtl5000->supplies);
regulator_bulk_free(sgtl5000->num_supplies, sgtl5000->supplies);
@@ -1802,6 +1805,11 @@ static int sgtl5000_i2c_remove(struct i2c_client *client)
return 0;
}
+static void sgtl5000_i2c_shutdown(struct i2c_client *client)
+{
+ sgtl5000_i2c_remove(client);
+}
+
static const struct i2c_device_id sgtl5000_id[] = {
{"sgtl5000", 0},
{},
@@ -1822,6 +1830,7 @@ static struct i2c_driver sgtl5000_i2c_driver = {
},
.probe_new = sgtl5000_i2c_probe,
.remove = sgtl5000_i2c_remove,
+ .shutdown = sgtl5000_i2c_shutdown,
.id_table = sgtl5000_id,
};
diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h
index 56ec5863f250..3a808c762299 100644
--- a/sound/soc/codecs/sgtl5000.h
+++ b/sound/soc/codecs/sgtl5000.h
@@ -80,6 +80,7 @@
/*
* SGTL5000_CHIP_DIG_POWER
*/
+#define SGTL5000_DIG_POWER_DEFAULT 0x0000
#define SGTL5000_ADC_EN 0x0040
#define SGTL5000_DAC_EN 0x0020
#define SGTL5000_DAP_POWERUP 0x0010
diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c
index d395feffb30b..4cb788f3e5f7 100644
--- a/sound/soc/codecs/tas2764.c
+++ b/sound/soc/codecs/tas2764.c
@@ -42,10 +42,12 @@ static void tas2764_reset(struct tas2764_priv *tas2764)
gpiod_set_value_cansleep(tas2764->reset_gpio, 0);
msleep(20);
gpiod_set_value_cansleep(tas2764->reset_gpio, 1);
+ usleep_range(1000, 2000);
}
snd_soc_component_write(tas2764->component, TAS2764_SW_RST,
TAS2764_RST);
+ usleep_range(1000, 2000);
}
static int tas2764_set_bias_level(struct snd_soc_component *component,
@@ -107,8 +109,10 @@ static int tas2764_codec_resume(struct snd_soc_component *component)
struct tas2764_priv *tas2764 = snd_soc_component_get_drvdata(component);
int ret;
- if (tas2764->sdz_gpio)
+ if (tas2764->sdz_gpio) {
gpiod_set_value_cansleep(tas2764->sdz_gpio, 1);
+ usleep_range(1000, 2000);
+ }
ret = snd_soc_component_update_bits(component, TAS2764_PWR_CTRL,
TAS2764_PWR_CTRL_MASK,
@@ -131,7 +135,8 @@ static const char * const tas2764_ASI1_src[] = {
};
static SOC_ENUM_SINGLE_DECL(
- tas2764_ASI1_src_enum, TAS2764_TDM_CFG2, 4, tas2764_ASI1_src);
+ tas2764_ASI1_src_enum, TAS2764_TDM_CFG2, TAS2764_TDM_CFG2_SCFG_SHIFT,
+ tas2764_ASI1_src);
static const struct snd_kcontrol_new tas2764_asi1_mux =
SOC_DAPM_ENUM("ASI1 Source", tas2764_ASI1_src_enum);
@@ -329,20 +334,22 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
{
struct snd_soc_component *component = dai->component;
struct tas2764_priv *tas2764 = snd_soc_component_get_drvdata(component);
- u8 tdm_rx_start_slot = 0, asi_cfg_1 = 0;
- int iface;
+ u8 tdm_rx_start_slot = 0, asi_cfg_0 = 0, asi_cfg_1 = 0;
int ret;
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_IF:
+ asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START;
+ fallthrough;
case SND_SOC_DAIFMT_NB_NF:
asi_cfg_1 = TAS2764_TDM_CFG1_RX_RISING;
break;
+ case SND_SOC_DAIFMT_IB_IF:
+ asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START;
+ fallthrough;
case SND_SOC_DAIFMT_IB_NF:
asi_cfg_1 = TAS2764_TDM_CFG1_RX_FALLING;
break;
- default:
- dev_err(tas2764->dev, "ASI format Inverse is not found\n");
- return -EINVAL;
}
ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG1,
@@ -353,13 +360,13 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
+ asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START;
+ fallthrough;
case SND_SOC_DAIFMT_DSP_A:
- iface = TAS2764_TDM_CFG2_SCFG_I2S;
tdm_rx_start_slot = 1;
break;
case SND_SOC_DAIFMT_DSP_B:
case SND_SOC_DAIFMT_LEFT_J:
- iface = TAS2764_TDM_CFG2_SCFG_LEFT_J;
tdm_rx_start_slot = 0;
break;
default:
@@ -368,14 +375,15 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
return -EINVAL;
}
- ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG1,
- TAS2764_TDM_CFG1_MASK,
- (tdm_rx_start_slot << TAS2764_TDM_CFG1_51_SHIFT));
+ ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG0,
+ TAS2764_TDM_CFG0_FRAME_START,
+ asi_cfg_0);
if (ret < 0)
return ret;
- ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG2,
- TAS2764_TDM_CFG2_SCFG_MASK, iface);
+ ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG1,
+ TAS2764_TDM_CFG1_MASK,
+ (tdm_rx_start_slot << TAS2764_TDM_CFG1_51_SHIFT));
if (ret < 0)
return ret;
@@ -501,8 +509,10 @@ static int tas2764_codec_probe(struct snd_soc_component *component)
tas2764->component = component;
- if (tas2764->sdz_gpio)
+ if (tas2764->sdz_gpio) {
gpiod_set_value_cansleep(tas2764->sdz_gpio, 1);
+ usleep_range(1000, 2000);
+ }
tas2764_reset(tas2764);
@@ -526,12 +536,12 @@ static int tas2764_codec_probe(struct snd_soc_component *component)
}
static DECLARE_TLV_DB_SCALE(tas2764_digital_tlv, 1100, 50, 0);
-static DECLARE_TLV_DB_SCALE(tas2764_playback_volume, -10000, 50, 0);
+static DECLARE_TLV_DB_SCALE(tas2764_playback_volume, -10050, 50, 1);
static const struct snd_kcontrol_new tas2764_snd_controls[] = {
SOC_SINGLE_TLV("Speaker Volume", TAS2764_DVC, 0,
TAS2764_DVC_MAX, 1, tas2764_playback_volume),
- SOC_SINGLE_TLV("Amp Gain Volume", TAS2764_CHNL_0, 0, 0x14, 0,
+ SOC_SINGLE_TLV("Amp Gain Volume", TAS2764_CHNL_0, 1, 0x14, 0,
tas2764_digital_tlv),
};
@@ -556,7 +566,7 @@ static const struct reg_default tas2764_reg_defaults[] = {
{ TAS2764_SW_RST, 0x00 },
{ TAS2764_PWR_CTRL, 0x1a },
{ TAS2764_DVC, 0x00 },
- { TAS2764_CHNL_0, 0x00 },
+ { TAS2764_CHNL_0, 0x28 },
{ TAS2764_TDM_CFG0, 0x09 },
{ TAS2764_TDM_CFG1, 0x02 },
{ TAS2764_TDM_CFG2, 0x0a },
diff --git a/sound/soc/codecs/tas2764.h b/sound/soc/codecs/tas2764.h
index 67d6fd903c42..f015f22a083b 100644
--- a/sound/soc/codecs/tas2764.h
+++ b/sound/soc/codecs/tas2764.h
@@ -47,6 +47,7 @@
#define TAS2764_TDM_CFG0_MASK GENMASK(3, 1)
#define TAS2764_TDM_CFG0_44_1_48KHZ BIT(3)
#define TAS2764_TDM_CFG0_88_2_96KHZ (BIT(3) | BIT(1))
+#define TAS2764_TDM_CFG0_FRAME_START BIT(0)
/* TDM Configuration Reg1 */
#define TAS2764_TDM_CFG1 TAS2764_REG(0X0, 0x09)
@@ -66,10 +67,7 @@
#define TAS2764_TDM_CFG2_RXS_16BITS 0x0
#define TAS2764_TDM_CFG2_RXS_24BITS BIT(0)
#define TAS2764_TDM_CFG2_RXS_32BITS BIT(1)
-#define TAS2764_TDM_CFG2_SCFG_MASK GENMASK(5, 4)
-#define TAS2764_TDM_CFG2_SCFG_I2S 0x0
-#define TAS2764_TDM_CFG2_SCFG_LEFT_J BIT(4)
-#define TAS2764_TDM_CFG2_SCFG_RIGHT_J BIT(5)
+#define TAS2764_TDM_CFG2_SCFG_SHIFT 4
/* TDM Configuration Reg3 */
#define TAS2764_TDM_CFG3 TAS2764_REG(0X0, 0x0c)
diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c
index b55f0b836932..0b729658fde8 100644
--- a/sound/soc/codecs/tlv320adcx140.c
+++ b/sound/soc/codecs/tlv320adcx140.c
@@ -33,7 +33,6 @@ struct adcx140_priv {
bool micbias_vg;
unsigned int dai_fmt;
- unsigned int tdm_delay;
unsigned int slot_width;
};
@@ -792,12 +791,13 @@ static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai,
{
struct snd_soc_component *component = codec_dai->component;
struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component);
- unsigned int lsb;
- /* TDM based on DSP mode requires slots to be adjacent */
- lsb = __ffs(tx_mask);
- if ((lsb + 1) != __fls(tx_mask)) {
- dev_err(component->dev, "Invalid mask, slots must be adjacent\n");
+ /*
+ * The chip itself supports arbitrary masks, but the driver currently
+ * only supports adjacent slots beginning at the first slot.
+ */
+ if (tx_mask != GENMASK(__fls(tx_mask), 0)) {
+ dev_err(component->dev, "Only lower adjacent slots are supported\n");
return -EINVAL;
}
@@ -812,7 +812,6 @@ static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai,
return -EINVAL;
}
- adcx140->tdm_delay = lsb;
adcx140->slot_width = slot_width;
return 0;
diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c
index d9f135200688..3cb7a3eab8c7 100644
--- a/sound/soc/codecs/wcd9335.c
+++ b/sound/soc/codecs/wcd9335.c
@@ -342,7 +342,7 @@ struct wcd9335_codec {
struct regulator_bulk_data supplies[WCD9335_MAX_SUPPLY];
unsigned int rx_port_value[WCD9335_RX_MAX];
- unsigned int tx_port_value;
+ unsigned int tx_port_value[WCD9335_TX_MAX];
int hph_l_gain;
int hph_r_gain;
u32 rx_bias_count;
@@ -1334,8 +1334,13 @@ static int slim_tx_mixer_get(struct snd_kcontrol *kc,
struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kc);
struct wcd9335_codec *wcd = dev_get_drvdata(dapm->dev);
+ struct snd_soc_dapm_widget *widget = snd_soc_dapm_kcontrol_widget(kc);
+ struct soc_mixer_control *mixer =
+ (struct soc_mixer_control *)kc->private_value;
+ int dai_id = widget->shift;
+ int port_id = mixer->shift;
- ucontrol->value.integer.value[0] = wcd->tx_port_value;
+ ucontrol->value.integer.value[0] = wcd->tx_port_value[port_id] == dai_id;
return 0;
}
@@ -1358,12 +1363,12 @@ static int slim_tx_mixer_put(struct snd_kcontrol *kc,
case AIF2_CAP:
case AIF3_CAP:
/* only add to the list if value not set */
- if (enable && !(wcd->tx_port_value & BIT(port_id))) {
- wcd->tx_port_value |= BIT(port_id);
+ if (enable && wcd->tx_port_value[port_id] != dai_id) {
+ wcd->tx_port_value[port_id] = dai_id;
list_add_tail(&wcd->tx_chs[port_id].list,
&wcd->dai[dai_id].slim_ch_list);
- } else if (!enable && (wcd->tx_port_value & BIT(port_id))) {
- wcd->tx_port_value &= ~BIT(port_id);
+ } else if (!enable && wcd->tx_port_value[port_id] == dai_id) {
+ wcd->tx_port_value[port_id] = -1;
list_del_init(&wcd->tx_chs[port_id].list);
}
break;
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index da2f8998df87..b034df47a5ef 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -680,12 +680,17 @@ static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol,
{
struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
struct arizona *arizona = dev_get_drvdata(component->dev->parent);
+ uint16_t dac_comp_coeff = get_unaligned_be16(ucontrol->value.bytes.data);
+ int ret = 0;
mutex_lock(&arizona->dac_comp_lock);
- arizona->dac_comp_coeff = get_unaligned_be16(ucontrol->value.bytes.data);
+ if (arizona->dac_comp_coeff != dac_comp_coeff) {
+ arizona->dac_comp_coeff = dac_comp_coeff;
+ ret = 1;
+ }
mutex_unlock(&arizona->dac_comp_lock);
- return 0;
+ return ret;
}
static int wm5102_out_comp_switch_get(struct snd_kcontrol *kcontrol,
@@ -706,12 +711,20 @@ static int wm5102_out_comp_switch_put(struct snd_kcontrol *kcontrol,
{
struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
struct arizona *arizona = dev_get_drvdata(component->dev->parent);
+ struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value;
+ int ret = 0;
+
+ if (ucontrol->value.integer.value[0] > mc->max)
+ return -EINVAL;
mutex_lock(&arizona->dac_comp_lock);
- arizona->dac_comp_enabled = ucontrol->value.integer.value[0];
+ if (arizona->dac_comp_enabled != ucontrol->value.integer.value[0]) {
+ arizona->dac_comp_enabled = ucontrol->value.integer.value[0];
+ ret = 1;
+ }
mutex_unlock(&arizona->dac_comp_lock);
- return 0;
+ return ret;
}
static const char * const wm5102_osr_text[] = {
diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c
index 00b59fc9b1fe..ab5481187c71 100644
--- a/sound/soc/codecs/wm8998.c
+++ b/sound/soc/codecs/wm8998.c
@@ -108,6 +108,7 @@ static int wm8998_inmux_put(struct snd_kcontrol *kcontrol,
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
unsigned int mode_reg, mode_index;
unsigned int mux, inmode, src_val, mode_val;
+ int change, ret;
mux = ucontrol->value.enumerated.item[0];
if (mux > 1)
@@ -137,14 +138,20 @@ static int wm8998_inmux_put(struct snd_kcontrol *kcontrol,
snd_soc_component_update_bits(component, mode_reg,
ARIZONA_IN1_MODE_MASK, mode_val);
- snd_soc_component_update_bits(component, e->reg,
- ARIZONA_IN1L_SRC_MASK |
- ARIZONA_IN1L_SRC_SE_MASK,
- src_val);
+ change = snd_soc_component_update_bits(component, e->reg,
+ ARIZONA_IN1L_SRC_MASK |
+ ARIZONA_IN1L_SRC_SE_MASK,
+ src_val);
- return snd_soc_dapm_mux_update_power(dapm, kcontrol,
- ucontrol->value.enumerated.item[0],
- e, NULL);
+ ret = snd_soc_dapm_mux_update_power(dapm, kcontrol,
+ ucontrol->value.enumerated.item[0],
+ e, NULL);
+ if (ret < 0) {
+ dev_err(arizona->dev, "Failed to update demux power state: %d\n", ret);
+ return ret;
+ }
+
+ return change;
}
static const char * const wm8998_inmux_texts[] = {
diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c
index 77ac4051b827..d34b29a49268 100644
--- a/sound/soc/generic/audio-graph-card2.c
+++ b/sound/soc/generic/audio-graph-card2.c
@@ -90,12 +90,12 @@ links indicates connection part of CPU side (= A).
ports@0 {
(X) (A) mcpu: port@0 { mcpu0_ep: endpoint { remote-endpoint = <&mcodec0_ep>; }; };
(y) port@1 { mcpu1_ep: endpoint { remote-endpoint = <&cpu1_ep>; }; };
-(y) port@1 { mcpu2_ep: endpoint { remote-endpoint = <&cpu2_ep>; }; };
+(y) port@2 { mcpu2_ep: endpoint { remote-endpoint = <&cpu2_ep>; }; };
};
ports@1 {
(X) port@0 { mcodec0_ep: endpoint { remote-endpoint = <&mcpu0_ep>; }; };
-(y) port@0 { mcodec1_ep: endpoint { remote-endpoint = <&codec1_ep>; }; };
-(y) port@1 { mcodec2_ep: endpoint { remote-endpoint = <&codec2_ep>; }; };
+(y) port@1 { mcodec1_ep: endpoint { remote-endpoint = <&codec1_ep>; }; };
+(y) port@2 { mcodec2_ep: endpoint { remote-endpoint = <&codec2_ep>; }; };
};
};
};
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index 5d67a2c87a1d..4a90a0a5d831 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -69,11 +69,10 @@ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN |
static int is_legacy_cpu;
-static struct snd_soc_jack sof_hdmi[3];
-
struct sof_hdmi_pcm {
struct list_head head;
struct snd_soc_dai *codec_dai;
+ struct snd_soc_jack hdmi_jack;
int device;
};
@@ -434,7 +433,6 @@ static int sof_card_late_probe(struct snd_soc_card *card)
char jack_name[NAME_SIZE];
struct sof_hdmi_pcm *pcm;
int err;
- int i = 0;
/* HDMI is not supported by SOF on Baytrail/CherryTrail */
if (is_legacy_cpu || !ctx->idisp_codec)
@@ -455,17 +453,15 @@ static int sof_card_late_probe(struct snd_soc_card *card)
snprintf(jack_name, sizeof(jack_name),
"HDMI/DP, pcm=%d Jack", pcm->device);
err = snd_soc_card_jack_new(card, jack_name,
- SND_JACK_AVOUT, &sof_hdmi[i]);
+ SND_JACK_AVOUT, &pcm->hdmi_jack);
if (err)
return err;
err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device,
- &sof_hdmi[i]);
+ &pcm->hdmi_jack);
if (err < 0)
return err;
-
- i++;
}
if (sof_rt5682_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) {
diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c
index 2439a574ac2f..deb7b820325e 100644
--- a/sound/soc/intel/skylake/skl-nhlt.c
+++ b/sound/soc/intel/skylake/skl-nhlt.c
@@ -99,7 +99,6 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
struct nhlt_fmt_cfg *fmt_cfg;
struct wav_fmt_ext *wav_fmt;
unsigned long rate;
- bool present = false;
int rate_index = 0;
u16 channels, bps;
u8 clk_src;
@@ -112,9 +111,12 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
if (fmt->fmt_count == 0)
return;
+ fmt_cfg = (struct nhlt_fmt_cfg *)fmt->fmt_config;
for (i = 0; i < fmt->fmt_count; i++) {
- fmt_cfg = &fmt->fmt_config[i];
- wav_fmt = &fmt_cfg->fmt_ext;
+ struct nhlt_fmt_cfg *saved_fmt_cfg = fmt_cfg;
+ bool present = false;
+
+ wav_fmt = &saved_fmt_cfg->fmt_ext;
channels = wav_fmt->fmt.channels;
bps = wav_fmt->fmt.bits_per_sample;
@@ -132,12 +134,18 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
* derive the rate.
*/
for (j = i; j < fmt->fmt_count; j++) {
- fmt_cfg = &fmt->fmt_config[j];
- wav_fmt = &fmt_cfg->fmt_ext;
+ struct nhlt_fmt_cfg *tmp_fmt_cfg = fmt_cfg;
+
+ wav_fmt = &tmp_fmt_cfg->fmt_ext;
if ((fs == wav_fmt->fmt.samples_per_sec) &&
- (bps == wav_fmt->fmt.bits_per_sample))
+ (bps == wav_fmt->fmt.bits_per_sample)) {
channels = max_t(u16, channels,
wav_fmt->fmt.channels);
+ saved_fmt_cfg = tmp_fmt_cfg;
+ }
+ /* Move to the next nhlt_fmt_cfg */
+ tmp_fmt_cfg = (struct nhlt_fmt_cfg *)(tmp_fmt_cfg->config.caps +
+ tmp_fmt_cfg->config.size);
}
rate = channels * bps * fs;
@@ -153,8 +161,11 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
/* Fill rate and parent for sclk/sclkfs */
if (!present) {
+ struct nhlt_fmt_cfg *first_fmt_cfg;
+
+ first_fmt_cfg = (struct nhlt_fmt_cfg *)fmt->fmt_config;
i2s_config_ext = (struct skl_i2s_config_blob_ext *)
- fmt->fmt_config[0].config.caps;
+ first_fmt_cfg->config.caps;
/* MCLK Divider Source Select */
if (is_legacy_blob(i2s_config_ext->hdr.sig)) {
@@ -168,6 +179,9 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
parent = skl_get_parent_clk(clk_src);
+ /* Move to the next nhlt_fmt_cfg */
+ fmt_cfg = (struct nhlt_fmt_cfg *)(fmt_cfg->config.caps +
+ fmt_cfg->config.size);
/*
* Do not copy the config data if there is no parent
* clock available for this clock source select
@@ -176,9 +190,9 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
continue;
sclk[id].rate_cfg[rate_index].rate = rate;
- sclk[id].rate_cfg[rate_index].config = fmt_cfg;
+ sclk[id].rate_cfg[rate_index].config = saved_fmt_cfg;
sclkfs[id].rate_cfg[rate_index].rate = rate;
- sclkfs[id].rate_cfg[rate_index].config = fmt_cfg;
+ sclkfs[id].rate_cfg[rate_index].config = saved_fmt_cfg;
sclk[id].parent_name = parent->name;
sclkfs[id].parent_name = parent->name;
@@ -192,13 +206,13 @@ static void skl_get_mclk(struct skl_dev *skl, struct skl_ssp_clk *mclk,
{
struct skl_i2s_config_blob_ext *i2s_config_ext;
struct skl_i2s_config_blob_legacy *i2s_config;
- struct nhlt_specific_cfg *fmt_cfg;
+ struct nhlt_fmt_cfg *fmt_cfg;
struct skl_clk_parent_src *parent;
u32 clkdiv, div_ratio;
u8 clk_src;
- fmt_cfg = &fmt->fmt_config[0].config;
- i2s_config_ext = (struct skl_i2s_config_blob_ext *)fmt_cfg->caps;
+ fmt_cfg = (struct nhlt_fmt_cfg *)fmt->fmt_config;
+ i2s_config_ext = (struct skl_i2s_config_blob_ext *)fmt_cfg->config.caps;
/* MCLK Divider Source Select and divider */
if (is_legacy_blob(i2s_config_ext->hdr.sig)) {
@@ -227,7 +241,7 @@ static void skl_get_mclk(struct skl_dev *skl, struct skl_ssp_clk *mclk,
return;
mclk[id].rate_cfg[0].rate = parent->rate/div_ratio;
- mclk[id].rate_cfg[0].config = &fmt->fmt_config[0];
+ mclk[id].rate_cfg[0].config = fmt_cfg;
mclk[id].parent_name = parent->name;
}
diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c
index f424d7aa389a..794019286c70 100644
--- a/sound/soc/qcom/qdsp6/q6apm.c
+++ b/sound/soc/qcom/qdsp6/q6apm.c
@@ -75,6 +75,7 @@ static struct audioreach_graph *q6apm_get_audioreach_graph(struct q6apm *apm, ui
id = idr_alloc(&apm->graph_idr, graph, graph_id, graph_id + 1, GFP_KERNEL);
if (id < 0) {
dev_err(apm->dev, "Unable to allocate graph id (%d)\n", graph_id);
+ kfree(graph->graph);
kfree(graph);
mutex_unlock(&apm->lock);
return ERR_PTR(id);
diff --git a/sound/soc/ti/omap-mcbsp-priv.h b/sound/soc/ti/omap-mcbsp-priv.h
index 7865cda4bf0a..da519ea1f303 100644
--- a/sound/soc/ti/omap-mcbsp-priv.h
+++ b/sound/soc/ti/omap-mcbsp-priv.h
@@ -316,8 +316,6 @@ static inline int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg,
/* Sidetone specific API */
int omap_mcbsp_st_init(struct platform_device *pdev);
-void omap_mcbsp_st_cleanup(struct platform_device *pdev);
-
int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp);
int omap_mcbsp_st_stop(struct omap_mcbsp *mcbsp);
diff --git a/sound/soc/ti/omap-mcbsp-st.c b/sound/soc/ti/omap-mcbsp-st.c
index 0bc7d26c660a..7e8179cae92e 100644
--- a/sound/soc/ti/omap-mcbsp-st.c
+++ b/sound/soc/ti/omap-mcbsp-st.c
@@ -347,7 +347,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev)
if (!st_data)
return -ENOMEM;
- st_data->mcbsp_iclk = clk_get(mcbsp->dev, "ick");
+ st_data->mcbsp_iclk = devm_clk_get(mcbsp->dev, "ick");
if (IS_ERR(st_data->mcbsp_iclk)) {
dev_warn(mcbsp->dev,
"Failed to get ick, sidetone might be broken\n");
@@ -359,7 +359,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev)
if (!st_data->io_base_st)
return -ENOMEM;
- ret = sysfs_create_group(&mcbsp->dev->kobj, &sidetone_attr_group);
+ ret = devm_device_add_group(mcbsp->dev, &sidetone_attr_group);
if (ret)
return ret;
@@ -368,16 +368,6 @@ int omap_mcbsp_st_init(struct platform_device *pdev)
return 0;
}
-void omap_mcbsp_st_cleanup(struct platform_device *pdev)
-{
- struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev);
-
- if (mcbsp->st_data) {
- sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group);
- clk_put(mcbsp->st_data->mcbsp_iclk);
- }
-}
-
static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c
index 4479d74f0a45..9933b33c80ca 100644
--- a/sound/soc/ti/omap-mcbsp.c
+++ b/sound/soc/ti/omap-mcbsp.c
@@ -702,8 +702,7 @@ static int omap_mcbsp_init(struct platform_device *pdev)
mcbsp->max_tx_thres = max_thres(mcbsp) - 0x10;
mcbsp->max_rx_thres = max_thres(mcbsp) - 0x10;
- ret = sysfs_create_group(&mcbsp->dev->kobj,
- &additional_attr_group);
+ ret = devm_device_add_group(mcbsp->dev, &additional_attr_group);
if (ret) {
dev_err(mcbsp->dev,
"Unable to create additional controls\n");
@@ -711,16 +710,7 @@ static int omap_mcbsp_init(struct platform_device *pdev)
}
}
- ret = omap_mcbsp_st_init(pdev);
- if (ret)
- goto err_st;
-
- return 0;
-
-err_st:
- if (mcbsp->pdata->buffer_size)
- sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group);
- return ret;
+ return omap_mcbsp_st_init(pdev);
}
/*
@@ -1431,11 +1421,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev)
if (cpu_latency_qos_request_active(&mcbsp->pm_qos_req))
cpu_latency_qos_remove_request(&mcbsp->pm_qos_req);
- if (mcbsp->pdata->buffer_size)
- sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group);
-
- omap_mcbsp_st_cleanup(pdev);
-
return 0;
}