Merge tag 'sound-5.8-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound into master
Pull sound fixes from Takashi Iwai:
"This became fairly large, containing mostly the collection of ASoC
fixes that slipped from the previous request, so I sent now a bit
earlier than usual. But all changes look small and mostly
device-specific, hence nothing to worry too much.
Majority of changes are for x86 based platforms and their CODEC
drivers, in order to address some issues hit by their recent tests and
fuzzing. The rest are other ASoC device-specific fixes (imx, qcom,
wm8974, amd, rockchip) as well as a trivial fix for a kernel WARNING
hit by syzkaller"
* tag 'sound-5.8-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (28 commits)
ALSA: hda/realtek: Fixed ALC298 sound bug by adding quirk for Samsung Notebook Pen S
ALSA: info: Drop WARN_ON() from buffer NULL sanity check
ASoC: rt5682: Report the button event in the headset type only
ASoC: Intel: bytcht_es8316: Add missed put_device()
ASoC: rt5682: Enable Vref2 under using PLL2
ASoC: rt286: fix unexpected interrupt happens
ASoC: wm8974: remove unsupported clock mode
ASoC: wm8974: fix Boost Mixer Aux Switch
ASoC: SOF: core: fix null-ptr-deref bug during device removal
ASoc: codecs: max98373: remove Idle_bias_on to let codec suspend
ASoC: codecs: max98373: Removed superfluous volume control from chip default
ASoC: topology: fix tlvs in error handling for widget_dmixer
ASoC: topology: fix kernel oops on route addition error
ASoC: SOF: imx: add min/max channels for SAI/ESAI on i.MX8/i.MX8M
ASoC: Intel: bdw-rt5677: fix non BE conversion
ASoC: soc-dai: set dai_link dpcm_ flags with a helper
MAINTAINERS: Add Shengjiu to reviewer list of sound/soc/fsl
ASoC: core: Remove only the registered component in devm functions
MAINTAINERS: Change Maintainer for some at91 drivers
ASoC: dt-bindings: simple-card: Fix 'make dt_binding_check' warnings
...
diff --git a/Documentation/devicetree/bindings/sound/simple-card.yaml b/Documentation/devicetree/bindings/sound/simple-card.yaml
index 8132d0c..35e6690 100644
--- a/Documentation/devicetree/bindings/sound/simple-card.yaml
+++ b/Documentation/devicetree/bindings/sound/simple-card.yaml
@@ -378,6 +378,8 @@
- |
sound {
compatible = "simple-audio-card";
+ #address-cells = <1>;
+ #size-cells = <0>;
simple-audio-card,name = "rsnd-ak4643";
simple-audio-card,format = "left_j";
@@ -391,10 +393,12 @@
"ak4642 Playback", "DAI1 Playback";
dpcmcpu: simple-audio-card,cpu@0 {
+ reg = <0>;
sound-dai = <&rcar_sound 0>;
};
simple-audio-card,cpu@1 {
+ reg = <1>;
sound-dai = <&rcar_sound 1>;
};
@@ -418,6 +422,8 @@
- |
sound {
compatible = "simple-audio-card";
+ #address-cells = <1>;
+ #size-cells = <0>;
simple-audio-card,routing =
"pcm3168a Playback", "DAI1 Playback",
@@ -426,6 +432,7 @@
"pcm3168a Playback", "DAI4 Playback";
simple-audio-card,dai-link@0 {
+ reg = <0>;
format = "left_j";
bitclock-master = <&sndcpu0>;
frame-master = <&sndcpu0>;
@@ -439,22 +446,23 @@
};
simple-audio-card,dai-link@1 {
+ reg = <1>;
format = "i2s";
bitclock-master = <&sndcpu1>;
frame-master = <&sndcpu1>;
convert-channels = <8>; /* TDM Split */
- sndcpu1: cpu@0 {
+ sndcpu1: cpu0 {
sound-dai = <&rcar_sound 1>;
};
- cpu@1 {
+ cpu1 {
sound-dai = <&rcar_sound 2>;
};
- cpu@2 {
+ cpu2 {
sound-dai = <&rcar_sound 3>;
};
- cpu@3 {
+ cpu3 {
sound-dai = <&rcar_sound 4>;
};
codec {
@@ -466,6 +474,7 @@
};
simple-audio-card,dai-link@2 {
+ reg = <2>;
format = "i2s";
bitclock-master = <&sndcpu2>;
frame-master = <&sndcpu2>;
diff --git a/MAINTAINERS b/MAINTAINERS
index d53db30..0887816 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -6956,6 +6956,7 @@
M: Nicolin Chen <nicoleotsuka@gmail.com>
M: Xiubo Li <Xiubo.Lee@gmail.com>
R: Fabio Estevam <festevam@gmail.com>
+R: Shengjiu Wang <shengjiu.wang@gmail.com>
L: alsa-devel@alsa-project.org (moderated for non-subscribers)
L: linuxppc-dev@lists.ozlabs.org
S: Maintained
@@ -11333,17 +11334,17 @@
F: include/dt-bindings/iio/adc/at91-sama5d2_adc.h
MICROCHIP SAMA5D2-COMPATIBLE SHUTDOWN CONTROLLER
-M: Nicolas Ferre <nicolas.ferre@microchip.com>
+M: Claudiu Beznea <claudiu.beznea@microchip.com>
S: Supported
F: drivers/power/reset/at91-sama5d2_shdwc.c
MICROCHIP SPI DRIVER
-M: Nicolas Ferre <nicolas.ferre@microchip.com>
+M: Tudor Ambarus <tudor.ambarus@microchip.com>
S: Supported
F: drivers/spi/spi-atmel.*
MICROCHIP SSC DRIVER
-M: Nicolas Ferre <nicolas.ferre@microchip.com>
+M: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
L: linux-arm-kernel@lists.infradead.org (moderated for non-subscribers)
S: Supported
F: drivers/misc/atmel-ssc.c
diff --git a/include/sound/rt5670.h b/include/sound/rt5670.h
index f9024c7..02e1d77 100644
--- a/include/sound/rt5670.h
+++ b/include/sound/rt5670.h
@@ -12,6 +12,7 @@
int jd_mode;
bool in2_diff;
bool dev_gpio;
+ bool gpio1_is_ext_spk_en;
bool dmic_en;
unsigned int dmic1_data_pin;
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 212257e..71e178c 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -161,6 +161,7 @@
int snd_soc_dai_compress_new(struct snd_soc_dai *dai,
struct snd_soc_pcm_runtime *rtd, int num);
bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream);
+void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link);
void snd_soc_dai_action(struct snd_soc_dai *dai,
int stream, int action);
static inline void snd_soc_dai_activate(struct snd_soc_dai *dai,
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 2756f9b..3ce7f0f 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -444,6 +444,8 @@
const struct snd_soc_component_driver *component_driver,
struct snd_soc_dai_driver *dai_drv, int num_dai);
void snd_soc_unregister_component(struct device *dev);
+void snd_soc_unregister_component_by_driver(struct device *dev,
+ const struct snd_soc_component_driver *component_driver);
struct snd_soc_component *snd_soc_lookup_component_nolocked(struct device *dev,
const char *driver_name);
struct snd_soc_component *snd_soc_lookup_component(struct device *dev,
diff --git a/sound/core/info.c b/sound/core/info.c
index 8c6bc52..9fec307 100644
--- a/sound/core/info.c
+++ b/sound/core/info.c
@@ -606,7 +606,9 @@
{
int c;
- if (snd_BUG_ON(!buffer || !buffer->buffer))
+ if (snd_BUG_ON(!buffer))
+ return 1;
+ if (!buffer->buffer)
return 1;
if (len <= 0 || buffer->stop || buffer->error)
return 1;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 1b06c42..1b2d8e5 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -7587,6 +7587,7 @@
SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8),
+ SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC),
SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC),
SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC),
diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c
index f25ce50..ebf4388 100644
--- a/sound/soc/amd/raven/pci-acp3x.c
+++ b/sound/soc/amd/raven/pci-acp3x.c
@@ -232,9 +232,7 @@
}
pm_runtime_set_autosuspend_delay(&pci->dev, 2000);
pm_runtime_use_autosuspend(&pci->dev);
- pm_runtime_set_active(&pci->dev);
pm_runtime_put_noidle(&pci->dev);
- pm_runtime_enable(&pci->dev);
pm_runtime_allow(&pci->dev);
return 0;
@@ -303,7 +301,7 @@
ret = acp3x_deinit(adata->acp3x_base);
if (ret)
dev_err(&pci->dev, "ACP de-init failed\n");
- pm_runtime_disable(&pci->dev);
+ pm_runtime_forbid(&pci->dev);
pm_runtime_get_noresume(&pci->dev);
pci_disable_msi(pci);
pci_release_regions(pci);
diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index 96718e3..d87402a 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -779,13 +779,6 @@
regmap_write(max98373->regmap,
MAX98373_R202A_PCM_TO_SPK_MONO_MIX_2,
0x1);
- /* Set inital volume (0dB) */
- regmap_write(max98373->regmap,
- MAX98373_R203D_AMP_DIG_VOL_CTRL,
- 0x00);
- regmap_write(max98373->regmap,
- MAX98373_R203E_AMP_PATH_GAIN,
- 0x00);
/* Enable DC blocker */
regmap_write(max98373->regmap,
MAX98373_R203F_AMP_DSP_CFG,
@@ -869,7 +862,6 @@
.num_dapm_widgets = ARRAY_SIZE(max98373_dapm_widgets),
.dapm_routes = max98373_audio_map,
.num_dapm_routes = ARRAY_SIZE(max98373_audio_map),
- .idle_bias_on = 1,
.use_pmdown_time = 1,
.endianness = 1,
.non_legacy_dai_naming = 1,
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index 9593a9a..e8d14ee 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -272,13 +272,13 @@
regmap_read(rt286->regmap, RT286_GET_MIC1_SENSE, &buf);
*mic = buf & 0x80000000;
}
- if (!*mic) {
+
+ if (!*hp) {
snd_soc_dapm_disable_pin(dapm, "HV");
snd_soc_dapm_disable_pin(dapm, "VREF");
- }
- if (!*hp)
snd_soc_dapm_disable_pin(dapm, "LDO1");
- snd_soc_dapm_sync(dapm);
+ snd_soc_dapm_sync(dapm);
+ }
return 0;
}
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index 70fee68..dfbc0ca 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -31,18 +31,19 @@
#include "rt5670.h"
#include "rt5670-dsp.h"
-#define RT5670_DEV_GPIO BIT(0)
-#define RT5670_IN2_DIFF BIT(1)
-#define RT5670_DMIC_EN BIT(2)
-#define RT5670_DMIC1_IN2P BIT(3)
-#define RT5670_DMIC1_GPIO6 BIT(4)
-#define RT5670_DMIC1_GPIO7 BIT(5)
-#define RT5670_DMIC2_INR BIT(6)
-#define RT5670_DMIC2_GPIO8 BIT(7)
-#define RT5670_DMIC3_GPIO5 BIT(8)
-#define RT5670_JD_MODE1 BIT(9)
-#define RT5670_JD_MODE2 BIT(10)
-#define RT5670_JD_MODE3 BIT(11)
+#define RT5670_DEV_GPIO BIT(0)
+#define RT5670_IN2_DIFF BIT(1)
+#define RT5670_DMIC_EN BIT(2)
+#define RT5670_DMIC1_IN2P BIT(3)
+#define RT5670_DMIC1_GPIO6 BIT(4)
+#define RT5670_DMIC1_GPIO7 BIT(5)
+#define RT5670_DMIC2_INR BIT(6)
+#define RT5670_DMIC2_GPIO8 BIT(7)
+#define RT5670_DMIC3_GPIO5 BIT(8)
+#define RT5670_JD_MODE1 BIT(9)
+#define RT5670_JD_MODE2 BIT(10)
+#define RT5670_JD_MODE3 BIT(11)
+#define RT5670_GPIO1_IS_EXT_SPK_EN BIT(12)
static unsigned long rt5670_quirk;
static unsigned int quirk_override;
@@ -602,9 +603,9 @@
EXPORT_SYMBOL_GPL(rt5670_set_jack_detect);
static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0);
-static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0);
+static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0);
static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0);
-static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0);
+static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 3000);
static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0);
/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */
@@ -1447,6 +1448,33 @@
return 0;
}
+static int rt5670_spk_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
+ struct rt5670_priv *rt5670 = snd_soc_component_get_drvdata(component);
+
+ if (!rt5670->pdata.gpio1_is_ext_spk_en)
+ return 0;
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2,
+ RT5670_GP1_OUT_MASK, RT5670_GP1_OUT_HI);
+ break;
+
+ case SND_SOC_DAPM_PRE_PMD:
+ regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2,
+ RT5670_GP1_OUT_MASK, RT5670_GP1_OUT_LO);
+ break;
+
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
static int rt5670_bst1_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
@@ -1860,7 +1888,9 @@
};
static const struct snd_soc_dapm_widget rt5672_specific_dapm_widgets[] = {
- SND_SOC_DAPM_PGA("SPO Amp", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA_E("SPO Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+ rt5670_spk_event, SND_SOC_DAPM_PRE_PMD |
+ SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_OUTPUT("SPOLP"),
SND_SOC_DAPM_OUTPUT("SPOLN"),
SND_SOC_DAPM_OUTPUT("SPORP"),
@@ -2857,14 +2887,14 @@
},
{
.callback = rt5670_quirk_cb,
- .ident = "Lenovo Thinkpad Tablet 10",
+ .ident = "Lenovo Miix 2 10",
.matches = {
DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
DMI_MATCH(DMI_PRODUCT_VERSION, "Lenovo Miix 2 10"),
},
.driver_data = (unsigned long *)(RT5670_DMIC_EN |
RT5670_DMIC1_IN2P |
- RT5670_DEV_GPIO |
+ RT5670_GPIO1_IS_EXT_SPK_EN |
RT5670_JD_MODE2),
},
{
@@ -2924,6 +2954,10 @@
rt5670->pdata.dev_gpio = true;
dev_info(&i2c->dev, "quirk dev_gpio\n");
}
+ if (rt5670_quirk & RT5670_GPIO1_IS_EXT_SPK_EN) {
+ rt5670->pdata.gpio1_is_ext_spk_en = true;
+ dev_info(&i2c->dev, "quirk GPIO1 is external speaker enable\n");
+ }
if (rt5670_quirk & RT5670_IN2_DIFF) {
rt5670->pdata.in2_diff = true;
dev_info(&i2c->dev, "quirk IN2_DIFF\n");
@@ -3023,6 +3057,13 @@
RT5670_GP1_PF_MASK, RT5670_GP1_PF_OUT);
}
+ if (rt5670->pdata.gpio1_is_ext_spk_en) {
+ regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1,
+ RT5670_GP1_PIN_MASK, RT5670_GP1_PIN_GPIO1);
+ regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2,
+ RT5670_GP1_PF_MASK, RT5670_GP1_PF_OUT);
+ }
+
if (rt5670->pdata.jd_mode) {
regmap_update_bits(rt5670->regmap, RT5670_GLB_CLK,
RT5670_SCLK_SRC_MASK, RT5670_SCLK_SRC_RCCLK);
diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h
index a8c3e44..de02033 100644
--- a/sound/soc/codecs/rt5670.h
+++ b/sound/soc/codecs/rt5670.h
@@ -757,7 +757,7 @@
#define RT5670_PWR_VREF2_BIT 4
#define RT5670_PWR_FV2 (0x1 << 3)
#define RT5670_PWR_FV2_BIT 3
-#define RT5670_LDO_SEL_MASK (0x3)
+#define RT5670_LDO_SEL_MASK (0x7)
#define RT5670_LDO_SEL_SFT 0
/* Power Management for Analog 2 (0x64) */
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index 7d6670a..d503b5b 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -967,13 +967,12 @@
rt5682_enable_push_button_irq(component, false);
snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1,
RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW);
- if (snd_soc_dapm_get_pin_status(dapm, "MICBIAS"))
+ if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS"))
+ snd_soc_component_update_bits(component,
+ RT5682_PWR_ANLG_1, RT5682_PWR_MB, 0);
+ if (!snd_soc_dapm_get_pin_status(dapm, "Vref2"))
snd_soc_component_update_bits(component,
RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0);
- else
- snd_soc_component_update_bits(component,
- RT5682_PWR_ANLG_1,
- RT5682_PWR_VREF2 | RT5682_PWR_MB, 0);
snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3,
RT5682_PWR_CBJ, 0);
@@ -992,16 +991,17 @@
rt5682->hs_jack = hs_jack;
- if (!rt5682->is_sdw) {
- if (!hs_jack) {
- regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
- RT5682_JD1_EN_MASK, RT5682_JD1_DIS);
- regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
- RT5682_POW_JDH | RT5682_POW_JDL, 0);
- cancel_delayed_work_sync(&rt5682->jack_detect_work);
- return 0;
- }
+ if (!hs_jack) {
+ regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
+ RT5682_JD1_EN_MASK, RT5682_JD1_DIS);
+ regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
+ RT5682_POW_JDH | RT5682_POW_JDL, 0);
+ cancel_delayed_work_sync(&rt5682->jack_detect_work);
+ return 0;
+ }
+
+ if (!rt5682->is_sdw) {
switch (rt5682->pdata.jd_src) {
case RT5682_JD1:
snd_soc_component_update_bits(component,
@@ -1082,7 +1082,8 @@
/* jack was out, report jack type */
rt5682->jack_type =
rt5682_headset_detect(rt5682->component, 1);
- } else {
+ } else if ((rt5682->jack_type & SND_JACK_HEADSET) ==
+ SND_JACK_HEADSET) {
/* jack is already in, report button event */
rt5682->jack_type = SND_JACK_HEADSET;
btn_type = rt5682_button_detect(rt5682->component);
@@ -1608,8 +1609,7 @@
0, set_filter_clk, SND_SOC_DAPM_PRE_PMU),
SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0,
rt5682_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
- SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0,
- NULL, 0),
+ SND_SOC_DAPM_SUPPLY("Vref2", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, NULL, 0),
/* ASRC */
@@ -2492,6 +2492,15 @@
snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS");
snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
RT5682_PWR_MB, RT5682_PWR_MB);
+
+ snd_soc_dapm_force_enable_pin_unlocked(dapm, "Vref2");
+ snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
+ RT5682_PWR_VREF2 | RT5682_PWR_FV2,
+ RT5682_PWR_VREF2);
+ usleep_range(55000, 60000);
+ snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
+ RT5682_PWR_FV2, RT5682_PWR_FV2);
+
snd_soc_dapm_force_enable_pin_unlocked(dapm, "I2S1");
snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2F");
snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2B");
@@ -2517,9 +2526,12 @@
snd_soc_dapm_mutex_lock(dapm);
snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Vref2");
if (!rt5682->jack_type)
snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
+ RT5682_PWR_VREF2 | RT5682_PWR_FV2 |
RT5682_PWR_MB, 0);
+
snd_soc_dapm_disable_pin_unlocked(dapm, "I2S1");
snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2F");
snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2B");
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 06ba365..7cfc896 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -186,7 +186,7 @@
/* Boost mixer */
static const struct snd_kcontrol_new wm8974_boost_mixer[] = {
-SOC_DAPM_SINGLE("Aux Switch", WM8974_INPPGA, 6, 1, 0),
+SOC_DAPM_SINGLE("Aux Switch", WM8974_INPPGA, 6, 1, 1),
};
/* Input PGA */
@@ -474,6 +474,10 @@
iface |= 0x0008;
break;
case SND_SOC_DAIFMT_DSP_A:
+ if ((fmt & SND_SOC_DAIFMT_INV_MASK) == SND_SOC_DAIFMT_IB_IF ||
+ (fmt & SND_SOC_DAIFMT_INV_MASK) == SND_SOC_DAIFMT_NB_IF) {
+ return -EINVAL;
+ }
iface |= 0x00018;
break;
default:
diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c
index 9ad35d9..97b4f54 100644
--- a/sound/soc/generic/audio-graph-card.c
+++ b/sound/soc/generic/audio-graph-card.c
@@ -317,8 +317,8 @@
if (ret < 0)
goto out_put_node;
- dai_link->dpcm_playback = 1;
- dai_link->dpcm_capture = 1;
+ snd_soc_dai_link_set_capabilities(dai_link);
+
dai_link->ops = &graph_ops;
dai_link->init = asoc_simple_dai_init;
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 55e9f88..04d4d28 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -231,8 +231,8 @@
if (ret < 0)
goto out_put_node;
- dai_link->dpcm_playback = 1;
- dai_link->dpcm_capture = 1;
+ snd_soc_dai_link_set_capabilities(dai_link);
+
dai_link->ops = &simple_ops;
dai_link->init = asoc_simple_dai_init;
diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c
index 5f96d7a..bed4d5f 100644
--- a/sound/soc/intel/boards/bdw-rt5677.c
+++ b/sound/soc/intel/boards/bdw-rt5677.c
@@ -354,6 +354,7 @@
{
.name = "Codec DSP",
.stream_name = "Wake on Voice",
+ .capture_only = 1,
.ops = &bdw_rt5677_dsp_ops,
SND_SOC_DAILINK_REG(dsp),
},
diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c
index 9e5fc94..ecbc58e 100644
--- a/sound/soc/intel/boards/bytcht_es8316.c
+++ b/sound/soc/intel/boards/bytcht_es8316.c
@@ -543,8 +543,10 @@
if (cnt) {
ret = device_add_properties(codec_dev, props);
- if (ret)
+ if (ret) {
+ put_device(codec_dev);
return ret;
+ }
}
devm_acpi_dev_add_driver_gpios(codec_dev, byt_cht_es8316_gpios);
diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c
index 7a43c70..22e4327 100644
--- a/sound/soc/intel/boards/cht_bsw_rt5672.c
+++ b/sound/soc/intel/boards/cht_bsw_rt5672.c
@@ -253,21 +253,20 @@
params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
/*
- * Default mode for SSP configuration is TDM 4 slot
+ * Default mode for SSP configuration is TDM 4 slot. One board/design,
+ * the Lenovo Miix 2 10 uses not 1 but 2 codecs connected to SSP2. The
+ * second piggy-backed, output-only codec is inside the keyboard-dock
+ * (which has extra speakers). Unlike the main rt5672 codec, we cannot
+ * configure this codec, it is hard coded to use 2 channel 24 bit I2S.
+ * Since we only support 2 channels anyways, there is no need for TDM
+ * on any cht-bsw-rt5672 designs. So we simply use I2S 2ch everywhere.
*/
- ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0),
- SND_SOC_DAIFMT_DSP_B |
- SND_SOC_DAIFMT_IB_NF |
+ ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0),
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0) {
- dev_err(rtd->dev, "can't set format to TDM %d\n", ret);
- return ret;
- }
-
- /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
- ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 0), 0xF, 0xF, 4, 24);
- if (ret < 0) {
- dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret);
+ dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret);
return ret;
}
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index f51b28d..92f51d0 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -72,7 +72,7 @@
config SND_SOC_QDSP6
tristate "SoC ALSA audio driver for QDSP6"
- depends on QCOM_APR && HAS_DMA
+ depends on QCOM_APR
select SND_SOC_QDSP6_COMMON
select SND_SOC_QDSP6_CORE
select SND_SOC_QDSP6_AFE
diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c
index f45e5aa..9539b0d 100644
--- a/sound/soc/rockchip/rk3399_gru_sound.c
+++ b/sound/soc/rockchip/rk3399_gru_sound.c
@@ -219,19 +219,32 @@
return 0;
}
+static int rockchip_sound_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE;
+ return snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE,
+ 8000, 96000);
+}
+
static const struct snd_soc_ops rockchip_sound_max98357a_ops = {
+ .startup = rockchip_sound_startup,
.hw_params = rockchip_sound_max98357a_hw_params,
};
static const struct snd_soc_ops rockchip_sound_rt5514_ops = {
+ .startup = rockchip_sound_startup,
.hw_params = rockchip_sound_rt5514_hw_params,
};
static const struct snd_soc_ops rockchip_sound_da7219_ops = {
+ .startup = rockchip_sound_startup,
.hw_params = rockchip_sound_da7219_hw_params,
};
static const struct snd_soc_ops rockchip_sound_dmic_ops = {
+ .startup = rockchip_sound_startup,
.hw_params = rockchip_sound_dmic_hw_params,
};
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 0f30f5a..2b8abf8 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2573,6 +2573,33 @@
EXPORT_SYMBOL_GPL(snd_soc_register_component);
/**
+ * snd_soc_unregister_component_by_driver - Unregister component using a given driver
+ * from the ASoC core
+ *
+ * @dev: The device to unregister
+ * @component_driver: The component driver to unregister
+ */
+void snd_soc_unregister_component_by_driver(struct device *dev,
+ const struct snd_soc_component_driver *component_driver)
+{
+ struct snd_soc_component *component;
+
+ if (!component_driver)
+ return;
+
+ mutex_lock(&client_mutex);
+ component = snd_soc_lookup_component_nolocked(dev, component_driver->name);
+ if (!component)
+ goto out;
+
+ snd_soc_del_component_unlocked(component);
+
+out:
+ mutex_unlock(&client_mutex);
+}
+EXPORT_SYMBOL_GPL(snd_soc_unregister_component_by_driver);
+
+/**
* snd_soc_unregister_component - Unregister all related component
* from the ASoC core
*
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index b05e18b..4571599 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -391,6 +391,44 @@
return stream->channels_min;
}
+/*
+ * snd_soc_dai_link_set_capabilities() - set dai_link properties based on its DAIs
+ */
+void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link)
+{
+ struct snd_soc_dai_link_component *cpu;
+ struct snd_soc_dai_link_component *codec;
+ struct snd_soc_dai *dai;
+ bool supported[SNDRV_PCM_STREAM_LAST + 1];
+ int direction;
+ int i;
+
+ for_each_pcm_streams(direction) {
+ supported[direction] = true;
+
+ for_each_link_cpus(dai_link, i, cpu) {
+ dai = snd_soc_find_dai(cpu);
+ if (!dai || !snd_soc_dai_stream_valid(dai, direction)) {
+ supported[direction] = false;
+ break;
+ }
+ }
+ if (!supported[direction])
+ continue;
+ for_each_link_codecs(dai_link, i, codec) {
+ dai = snd_soc_find_dai(codec);
+ if (!dai || !snd_soc_dai_stream_valid(dai, direction)) {
+ supported[direction] = false;
+ break;
+ }
+ }
+ }
+
+ dai_link->dpcm_playback = supported[SNDRV_PCM_STREAM_PLAYBACK];
+ dai_link->dpcm_capture = supported[SNDRV_PCM_STREAM_CAPTURE];
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_link_set_capabilities);
+
void snd_soc_dai_action(struct snd_soc_dai *dai,
int stream, int action)
{
diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c
index 11e5d79..4534a1c 100644
--- a/sound/soc/soc-devres.c
+++ b/sound/soc/soc-devres.c
@@ -48,7 +48,9 @@
static void devm_component_release(struct device *dev, void *res)
{
- snd_soc_unregister_component(*(struct device **)res);
+ const struct snd_soc_component_driver **cmpnt_drv = res;
+
+ snd_soc_unregister_component_by_driver(dev, *cmpnt_drv);
}
/**
@@ -65,7 +67,7 @@
const struct snd_soc_component_driver *cmpnt_drv,
struct snd_soc_dai_driver *dai_drv, int num_dai)
{
- struct device **ptr;
+ const struct snd_soc_component_driver **ptr;
int ret;
ptr = devres_alloc(devm_component_release, sizeof(*ptr), GFP_KERNEL);
@@ -74,7 +76,7 @@
ret = snd_soc_register_component(dev, cmpnt_drv, dai_drv, num_dai);
if (ret == 0) {
- *ptr = dev;
+ *ptr = cmpnt_drv;
devres_add(dev, ptr);
} else {
devres_free(ptr);
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 80a4e71..6184440 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -478,7 +478,7 @@
pcm = soc_component_to_pcm(component);
- snd_soc_unregister_component(dev);
+ snd_soc_unregister_component_by_driver(dev, component->driver);
dmaengine_pcm_release_chan(pcm);
kfree(pcm);
}
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 43e5745..6eaa00c 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1261,17 +1261,29 @@
list_add(&routes[i]->dobj.list, &tplg->comp->dobj_list);
ret = soc_tplg_add_route(tplg, routes[i]);
- if (ret < 0)
+ if (ret < 0) {
+ /*
+ * this route was added to the list, it will
+ * be freed in remove_route() so increment the
+ * counter to skip it in the error handling
+ * below.
+ */
+ i++;
break;
+ }
/* add route, but keep going if some fail */
snd_soc_dapm_add_routes(dapm, routes[i], 1);
}
- /* free memory allocated for all dapm routes in case of error */
- if (ret < 0)
- for (i = 0; i < count ; i++)
- kfree(routes[i]);
+ /*
+ * free memory allocated for all dapm routes not added to the
+ * list in case of error
+ */
+ if (ret < 0) {
+ while (i < count)
+ kfree(routes[i++]);
+ }
/*
* free pointer to array of dapm routes as this is no longer needed.
@@ -1359,7 +1371,6 @@
if (err < 0) {
dev_err(tplg->dev, "ASoC: failed to init %s\n",
mc->hdr.name);
- soc_tplg_free_tlv(tplg, &kc[i]);
goto err_sm;
}
}
@@ -1367,6 +1378,7 @@
err_sm:
for (; i >= 0; i--) {
+ soc_tplg_free_tlv(tplg, &kc[i]);
sm = (struct soc_mixer_control *)kc[i].private_value;
kfree(sm);
kfree(kc[i].name);
diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c
index 339c493..adc7c37 100644
--- a/sound/soc/sof/core.c
+++ b/sound/soc/sof/core.c
@@ -345,15 +345,15 @@
struct snd_sof_pdata *pdata = sdev->pdata;
int ret;
- ret = snd_sof_dsp_power_down_notify(sdev);
- if (ret < 0)
- dev_warn(dev, "error: %d failed to prepare DSP for device removal",
- ret);
-
if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE))
cancel_work_sync(&sdev->probe_work);
if (sdev->fw_state > SOF_FW_BOOT_NOT_STARTED) {
+ ret = snd_sof_dsp_power_down_notify(sdev);
+ if (ret < 0)
+ dev_warn(dev, "error: %d failed to prepare DSP for device removal",
+ ret);
+
snd_sof_fw_unload(sdev);
snd_sof_ipc_free(sdev);
snd_sof_free_debug(sdev);
diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c
index 63f9c20..a4fa845 100644
--- a/sound/soc/sof/imx/imx8.c
+++ b/sound/soc/sof/imx/imx8.c
@@ -375,6 +375,14 @@
static struct snd_soc_dai_driver imx8_dai[] = {
{
.name = "esai-port",
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 8,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 8,
+ },
},
};
diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c
index fa86a9e..287114a3 100644
--- a/sound/soc/sof/imx/imx8m.c
+++ b/sound/soc/sof/imx/imx8m.c
@@ -240,6 +240,14 @@
static struct snd_soc_dai_driver imx8m_dai[] = {
{
.name = "sai-port",
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 32,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 32,
+ },
},
};
