diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
index ee936d1aa724c13adc48d0d91487004c2e792da8..c2930d65728ed23c6eb90bb1dc604c3ea924b006 100644
--- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
+++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
@@ -114,7 +114,7 @@ properties:
ports:
$ref: /schemas/graph.yaml#/properties/ports
- properties:
+ patternProperties:
port(@[0-9a-f]+)?:
$ref: audio-graph-port.yaml#
unevaluatedProperties: false
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 675849d07284af0edab250b7d88dd87d0266d150..8e6dd8a257c567d1de5c48c6cd963800094d1b30 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -712,6 +712,12 @@ struct snd_soc_dai_link {
/* Do not create a PCM for this DAI link (Backend link) */
unsigned int ignore:1;
+ /* This flag will reorder stop sequence. By enabling this flag
+ * DMA controller stop sequence will be invoked first followed by
+ * CPU DAI driver stop sequence
+ */
+ unsigned int stop_dma_first:1;
+
#ifdef CONFIG_SND_SOC_TOPOLOGY
struct snd_soc_dobj dobj; /* For topology */
#endif
diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c
index 3a78fdad1ab4132f785a1f90a4b5f54cc2a0f1a2..da5c8be84a82841a43e524386769b85bb9c1f740 100644
--- a/sound/core/memalloc.c
+++ b/sound/core/memalloc.c
@@ -261,7 +261,7 @@ static int snd_dma_continuous_mmap(struct snd_dma_buffer *dmab,
struct vm_area_struct *area)
{
return remap_pfn_range(area, area->vm_start,
- dmab->addr >> PAGE_SHIFT,
+ page_to_pfn(virt_to_page(dmab->area)),
area->vm_end - area->vm_start,
area->vm_page_prot);
}
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index c88c4316c417d9eb0fbe7900e4d0c5b45e6ee9f3..09c0e2a6489c4b4700050cf0ed710a456445ccfb 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -246,12 +246,18 @@ static bool hw_support_mmap(struct snd_pcm_substream *substream)
if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_MMAP))
return false;
- if (substream->ops->mmap ||
- (substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV &&
- substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV_UC))
+ if (substream->ops->mmap || substream->ops->page)
return true;
- return dma_can_mmap(substream->dma_buffer.dev.dev);
+ switch (substream->dma_buffer.dev.type) {
+ case SNDRV_DMA_TYPE_UNKNOWN:
+ return false;
+ case SNDRV_DMA_TYPE_CONTINUOUS:
+ case SNDRV_DMA_TYPE_VMALLOC:
+ return true;
+ default:
+ return dma_can_mmap(substream->dma_buffer.dev.dev);
+ }
}
static int constrain_mask_params(struct snd_pcm_substream *substream,
@@ -3669,6 +3675,8 @@ static vm_fault_t snd_pcm_mmap_data_fault(struct vm_fault *vmf)
return VM_FAULT_SIGBUS;
if (substream->ops->page)
page = substream->ops->page(substream, offset);
+ else if (!snd_pcm_get_dma_buf(substream))
+ page = virt_to_page(runtime->dma_area + offset);
else
page = snd_sgbuf_get_page(snd_pcm_get_dma_buf(substream), offset);
if (!page)
diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c
index d8be146793eee2d513a53539a0c1197904bdedd0..c9d0ba353463bda2109d45474bc1c88155e17bdf 100644
--- a/sound/hda/intel-dsp-config.c
+++ b/sound/hda/intel-dsp-config.c
@@ -319,6 +319,10 @@ static const struct config_entry config_table[] = {
.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC,
.device = 0x4b55,
},
+ {
+ .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC,
+ .device = 0x4b58,
+ },
#endif
/* Alder Lake */
diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c
index 5bbe6695689d753cc16bcc6448da3fe745953e7e..7ad8c5f7b664b45038e117a53581ca48c2f2263c 100644
--- a/sound/isa/sb/sb16_csp.c
+++ b/sound/isa/sb/sb16_csp.c
@@ -816,6 +816,7 @@ static int snd_sb_csp_start(struct snd_sb_csp * p, int sample_width, int channel
mixR = snd_sbmixer_read(p->chip, SB_DSP4_PCM_DEV + 1);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL & 0x7);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR & 0x7);
+ spin_unlock_irqrestore(&p->chip->mixer_lock, flags);
spin_lock(&p->chip->reg_lock);
set_mode_register(p->chip, 0xc0); /* c0 = STOP */
@@ -855,6 +856,7 @@ static int snd_sb_csp_start(struct snd_sb_csp * p, int sample_width, int channel
spin_unlock(&p->chip->reg_lock);
/* restore PCM volume */
+ spin_lock_irqsave(&p->chip->mixer_lock, flags);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR);
spin_unlock_irqrestore(&p->chip->mixer_lock, flags);
@@ -880,6 +882,7 @@ static int snd_sb_csp_stop(struct snd_sb_csp * p)
mixR = snd_sbmixer_read(p->chip, SB_DSP4_PCM_DEV + 1);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL & 0x7);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR & 0x7);
+ spin_unlock_irqrestore(&p->chip->mixer_lock, flags);
spin_lock(&p->chip->reg_lock);
if (p->running & SNDRV_SB_CSP_ST_QSOUND) {
@@ -894,6 +897,7 @@ static int snd_sb_csp_stop(struct snd_sb_csp * p)
spin_unlock(&p->chip->reg_lock);
/* restore PCM volume */
+ spin_lock_irqsave(&p->chip->mixer_lock, flags);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR);
spin_unlock_irqrestore(&p->chip->mixer_lock, flags);
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 672fd28e2449bd6328cbd128baff7cf5ea25ccd3..65d2c55399195a4c6b89afe3f62f1debbe8fbeed 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1944,6 +1944,8 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid)
static const struct snd_pci_quirk force_connect_list[] = {
SND_PCI_QUIRK(0x103c, 0x870f, "HP", 1),
SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1),
+ SND_PCI_QUIRK(0x1462, 0xec94, "MS-7C94", 1),
+ SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", 1),
{}
};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 1389cfd5e0dbb6b423ae4ef8798abeb548176abd..21c521596c9d02c8f5c91429e59d2d4a1f06300b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8274,9 +8274,11 @@ 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, 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),
SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC),
+ SND_PCI_QUIRK(0x1025, 0x142b, "Acer Swift SF314-42", ALC255_FIXUP_ACER_MIC_NO_PRESENCE),
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),
@@ -8626,6 +8628,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x3151, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
SND_PCI_QUIRK(0x17aa, 0x3176, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
+ SND_PCI_QUIRK(0x17aa, 0x31af, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340),
SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME),
SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF),
SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP),
diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c
index 84e3906abd4f33050b0adaf4535d06fcdffd2593..9449fb40a956bd064462b4f4386a8878e0b1be7e 100644
--- a/sound/soc/amd/acp-da7219-max98357a.c
+++ b/sound/soc/amd/acp-da7219-max98357a.c
@@ -576,6 +576,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = {
| SND_SOC_DAIFMT_CBM_CFM,
.init = cz_rt5682_init,
.dpcm_playback = 1,
+ .stop_dma_first = 1,
.ops = &cz_rt5682_play_ops,
SND_SOC_DAILINK_REG(designware1, rt5682, platform),
},
@@ -585,6 +586,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = {
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBM_CFM,
.dpcm_capture = 1,
+ .stop_dma_first = 1,
.ops = &cz_rt5682_cap_ops,
SND_SOC_DAILINK_REG(designware2, rt5682, platform),
},
@@ -594,6 +596,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = {
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBM_CFM,
.dpcm_playback = 1,
+ .stop_dma_first = 1,
.ops = &cz_rt5682_max_play_ops,
SND_SOC_DAILINK_REG(designware3, mx, platform),
},
@@ -604,6 +607,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = {
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBM_CFM,
.dpcm_capture = 1,
+ .stop_dma_first = 1,
.ops = &cz_rt5682_dmic0_cap_ops,
SND_SOC_DAILINK_REG(designware3, adau, platform),
},
@@ -614,6 +618,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = {
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBM_CFM,
.dpcm_capture = 1,
+ .stop_dma_first = 1,
.ops = &cz_rt5682_dmic1_cap_ops,
SND_SOC_DAILINK_REG(designware2, adau, platform),
},
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 7ebae3f09435c0c83ddb67de36b65d716f32bde3..a3b784ed4f70a28c6c158942da927c5ed5674be6 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -1325,7 +1325,7 @@ config SND_SOC_SSM2305
high-efficiency mono Class-D audio power amplifiers.
config SND_SOC_SSM2518
- tristate
+ tristate "Analog Devices SSM2518 Class-D Amplifier"
depends on I2C
config SND_SOC_SSM2602
@@ -1557,6 +1557,7 @@ config SND_SOC_WCD934X
Qualcomm SoCs like SDM845.
config SND_SOC_WCD938X
+ depends on SND_SOC_WCD938X_SDW
tristate
config SND_SOC_WCD938X_SDW
@@ -1813,11 +1814,6 @@ config SND_SOC_ZL38060
which consists of a Digital Signal Processor (DSP), several Digital
Audio Interfaces (DAIs), analog outputs, and a block of 14 GPIOs.
-config SND_SOC_ZX_AUD96P22
- tristate "ZTE ZX AUD96P22 CODEC"
- depends on I2C
- select REGMAP_I2C
-
# Amp
config SND_SOC_LM4857
tristate
diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index 3000bc128b5bcbed1967cd3765b4fa45393c244e..38356ea2bd6ef363f05bb962d4041536c0645c05 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -1695,6 +1695,8 @@ static const struct regmap_config rt5631_regmap_config = {
.reg_defaults = rt5631_reg,
.num_reg_defaults = ARRAY_SIZE(rt5631_reg),
.cache_type = REGCACHE_RBTREE,
+ .use_single_read = true,
+ .use_single_write = true,
};
static int rt5631_i2c_probe(struct i2c_client *i2c,
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index e4c91571abaefba346fa095c37a982585b145c86..abcd6f48378880651f7e4e761ede1a934589277d 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -973,10 +973,14 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert)
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_dapm_get_pin_status(dapm, "PLL1") &&
+ !snd_soc_dapm_get_pin_status(dapm, "PLL2B"))
snd_soc_component_update_bits(component,
RT5682_PWR_ANLG_1, RT5682_PWR_MB, 0);
- if (!snd_soc_dapm_get_pin_status(dapm, "Vref2"))
+ if (!snd_soc_dapm_get_pin_status(dapm, "Vref2") &&
+ !snd_soc_dapm_get_pin_status(dapm, "PLL1") &&
+ !snd_soc_dapm_get_pin_status(dapm, "PLL2B"))
snd_soc_component_update_bits(component,
RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0);
snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3,
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index 51870d50f4195e348e600905602ee3cc7620e5ff..b504d63385b38b3e3c074b87dc11d3d48e07bc18 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -1604,6 +1604,8 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c,
ret);
return ret;
}
+ regcache_cache_only(aic31xx->regmap, true);
+
aic31xx->dev = &i2c->dev;
aic31xx->irq = i2c->irq;
diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h
index 81952984613d2f3f984b184c5019507f1437f21a..2513922a0292314881d6bb2300ee592bd30595e0 100644
--- a/sound/soc/codecs/tlv320aic31xx.h
+++ b/sound/soc/codecs/tlv320aic31xx.h
@@ -151,8 +151,8 @@ struct aic31xx_pdata {
#define AIC31XX_WORD_LEN_24BITS 0x02
#define AIC31XX_WORD_LEN_32BITS 0x03
#define AIC31XX_IFACE1_MASTER_MASK GENMASK(3, 2)
-#define AIC31XX_BCLK_MASTER BIT(2)
-#define AIC31XX_WCLK_MASTER BIT(3)
+#define AIC31XX_BCLK_MASTER BIT(3)
+#define AIC31XX_WCLK_MASTER BIT(2)
/* AIC31XX_DATA_OFFSET */
#define AIC31XX_DATA_OFFSET_MASK GENMASK(7, 0)
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index c63b717040ed25c57a5d68f52e9fe7e0eb3996ad..dcd8aeb45cb317af4199a2f9084870df43897b5e 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -250,8 +250,8 @@ static DECLARE_TLV_DB_SCALE(tlv_pcm, -6350, 50, 0);
static DECLARE_TLV_DB_SCALE(tlv_driver_gain, -600, 100, 0);
/* -12dB min, 0.5dB steps */
static DECLARE_TLV_DB_SCALE(tlv_adc_vol, -1200, 50, 0);
-
-static DECLARE_TLV_DB_LINEAR(tlv_spk_vol, TLV_DB_GAIN_MUTE, 0);
+/* -6dB min, 1dB steps */
+static DECLARE_TLV_DB_SCALE(tlv_tas_driver_gain, -5850, 50, 0);
static DECLARE_TLV_DB_SCALE(tlv_amp_vol, 0, 600, 1);
static const char * const lo_cm_text[] = {
@@ -1063,21 +1063,20 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4 = {
};
static const struct snd_kcontrol_new aic32x4_tas2505_snd_controls[] = {
- SOC_DOUBLE_R_S_TLV("PCM Playback Volume", AIC32X4_LDACVOL,
- AIC32X4_LDACVOL, 0, -0x7f, 0x30, 7, 0, tlv_pcm),
+ SOC_SINGLE_S8_TLV("PCM Playback Volume",
+ AIC32X4_LDACVOL, -0x7f, 0x30, tlv_pcm),
SOC_ENUM("DAC Playback PowerTune Switch", l_ptm_enum),
- SOC_DOUBLE_R_S_TLV("HP Driver Playback Volume", AIC32X4_HPLGAIN,
- AIC32X4_HPLGAIN, 0, -0x6, 0x1d, 5, 0,
- tlv_driver_gain),
- SOC_DOUBLE_R("HP DAC Playback Switch", AIC32X4_HPLGAIN,
- AIC32X4_HPLGAIN, 6, 0x01, 1),
- SOC_SINGLE("Auto-mute Switch", AIC32X4_DACMUTE, 4, 7, 0),
+ SOC_SINGLE_TLV("HP Driver Gain Volume",
+ AIC32X4_HPLGAIN, 0, 0x74, 1, tlv_tas_driver_gain),
+ SOC_SINGLE("HP DAC Playback Switch", AIC32X4_HPLGAIN, 6, 1, 1),
- SOC_SINGLE_RANGE_TLV("Speaker Driver Playback Volume", TAS2505_SPKVOL1,
- 0, 0, 117, 1, tlv_spk_vol),
- SOC_SINGLE_TLV("Speaker Amplifier Playback Volume", TAS2505_SPKVOL2,
- 4, 5, 0, tlv_amp_vol),
+ SOC_SINGLE_TLV("Speaker Driver Playback Volume",
+ TAS2505_SPKVOL1, 0, 0x74, 1, tlv_tas_driver_gain),
+ SOC_SINGLE_TLV("Speaker Amplifier Playback Volume",
+ TAS2505_SPKVOL2, 4, 5, 0, tlv_amp_vol),
+
+ SOC_SINGLE("Auto-mute Switch", AIC32X4_DACMUTE, 4, 7, 0),
};
static const struct snd_kcontrol_new hp_output_mixer_controls[] = {
diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c
index 78b76eceff8fa2fcadb5d455104275fe4594695a..2fcc97370be2bba8284d0dd42642f9cfe11d28c5 100644
--- a/sound/soc/codecs/wcd938x.c
+++ b/sound/soc/codecs/wcd938x.c
@@ -3317,13 +3317,6 @@ static int wcd938x_soc_codec_probe(struct snd_soc_component *component)
(WCD938X_DIGITAL_INTR_LEVEL_0 + i), 0);
}
- ret = wcd938x_irq_init(wcd938x, component->dev);
- if (ret) {
- dev_err(component->dev, "%s: IRQ init failed: %d\n",
- __func__, ret);
- return ret;
- }
-
wcd938x->hphr_pdm_wd_int = regmap_irq_get_virq(wcd938x->irq_chip,
WCD938X_IRQ_HPHR_PDM_WD_INT);
wcd938x->hphl_pdm_wd_int = regmap_irq_get_virq(wcd938x->irq_chip,
@@ -3553,7 +3546,6 @@ static int wcd938x_bind(struct device *dev)
}
wcd938x->sdw_priv[AIF1_PB] = dev_get_drvdata(wcd938x->rxdev);
wcd938x->sdw_priv[AIF1_PB]->wcd938x = wcd938x;
- wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq;
wcd938x->txdev = wcd938x_sdw_device_get(wcd938x->txnode);
if (!wcd938x->txdev) {
@@ -3562,7 +3554,6 @@ static int wcd938x_bind(struct device *dev)
}
wcd938x->sdw_priv[AIF1_CAP] = dev_get_drvdata(wcd938x->txdev);
wcd938x->sdw_priv[AIF1_CAP]->wcd938x = wcd938x;
- wcd938x->sdw_priv[AIF1_CAP]->slave_irq = wcd938x->virq;
wcd938x->tx_sdw_dev = dev_to_sdw_dev(wcd938x->txdev);
if (!wcd938x->tx_sdw_dev) {
dev_err(dev, "could not get txslave with matching of dev\n");
@@ -3595,6 +3586,15 @@ static int wcd938x_bind(struct device *dev)
return PTR_ERR(wcd938x->regmap);
}
+ ret = wcd938x_irq_init(wcd938x, dev);
+ if (ret) {
+ dev_err(dev, "%s: IRQ init failed: %d\n", __func__, ret);
+ return ret;
+ }
+
+ wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq;
+ wcd938x->sdw_priv[AIF1_CAP]->slave_irq = wcd938x->virq;
+
ret = wcd938x_set_micbias_data(wcd938x);
if (ret < 0) {
dev_err(dev, "%s: bad micbias pdata\n", __func__);
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 37aa020f23f631c3635c0a925de1124db07b2177..549d98241daec1ce6e6c22e19772ac6209b57613 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -282,6 +282,7 @@
/*
* HALO_CCM_CORE_CONTROL
*/
+#define HALO_CORE_RESET 0x00000200
#define HALO_CORE_EN 0x00000001
/*
@@ -1213,7 +1214,7 @@ static int wm_coeff_tlv_get(struct snd_kcontrol *kctl,
mutex_lock(&ctl->dsp->pwr_lock);
- ret = wm_coeff_read_ctrl_raw(ctl, ctl->cache, size);
+ ret = wm_coeff_read_ctrl(ctl, ctl->cache, size);
if (!ret && copy_to_user(bytes, ctl->cache, size))
ret = -EFAULT;
@@ -3333,7 +3334,8 @@ static int wm_halo_start_core(struct wm_adsp *dsp)
{
return regmap_update_bits(dsp->regmap,
dsp->base + HALO_CCM_CORE_CONTROL,
- HALO_CORE_EN, HALO_CORE_EN);
+ HALO_CORE_RESET | HALO_CORE_EN,
+ HALO_CORE_RESET | HALO_CORE_EN);
}
static void wm_halo_stop_core(struct wm_adsp *dsp)
diff --git a/sound/soc/intel/boards/sof_sdw_max98373.c b/sound/soc/intel/boards/sof_sdw_max98373.c
index 0e7ed906b34177715caa17c855b31ed7a11a99df..25daef910aee184fea5abbb3b68c385f8f613005 100644
--- a/sound/soc/intel/boards/sof_sdw_max98373.c
+++ b/sound/soc/intel/boards/sof_sdw_max98373.c
@@ -55,43 +55,68 @@ static int spk_init(struct snd_soc_pcm_runtime *rtd)
return ret;
}
-static int max98373_sdw_trigger(struct snd_pcm_substream *substream, int cmd)
+static int mx8373_enable_spk_pin(struct snd_pcm_substream *substream, bool enable)
{
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+ struct snd_soc_dai *codec_dai;
+ struct snd_soc_dai *cpu_dai;
int ret;
+ int j;
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- case SNDRV_PCM_TRIGGER_RESUME:
- case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- /* enable max98373 first */
- ret = max_98373_trigger(substream, cmd);
- if (ret < 0)
- break;
-
- ret = sdw_trigger(substream, cmd);
- break;
- case SNDRV_PCM_TRIGGER_STOP:
- case SNDRV_PCM_TRIGGER_SUSPEND:
- case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- ret = sdw_trigger(substream, cmd);
- if (ret < 0)
- break;
-
- ret = max_98373_trigger(substream, cmd);
- break;
- default:
- ret = -EINVAL;
- break;
+ /* set spk pin by playback only */
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ return 0;
+
+ cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ for_each_rtd_codec_dais(rtd, j, codec_dai) {
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_component_get_dapm(cpu_dai->component);
+ char pin_name[16];
+
+ snprintf(pin_name, ARRAY_SIZE(pin_name), "%s Spk",
+ codec_dai->component->name_prefix);
+
+ if (enable)
+ ret = snd_soc_dapm_enable_pin(dapm, pin_name);
+ else
+ ret = snd_soc_dapm_disable_pin(dapm, pin_name);
+
+ if (!ret)
+ snd_soc_dapm_sync(dapm);
}
- return ret;
+ return 0;
+}
+
+static int mx8373_sdw_prepare(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ /* according to soc_pcm_prepare dai link prepare is called first */
+ ret = sdw_prepare(substream);
+ if (ret < 0)
+ return ret;
+
+ return mx8373_enable_spk_pin(substream, true);
+}
+
+static int mx8373_sdw_hw_free(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ /* according to soc_pcm_hw_free dai link free is called first */
+ ret = sdw_hw_free(substream);
+ if (ret < 0)
+ return ret;
+
+ return mx8373_enable_spk_pin(substream, false);
}
static const struct snd_soc_ops max_98373_sdw_ops = {
.startup = sdw_startup,
- .prepare = sdw_prepare,
- .trigger = max98373_sdw_trigger,
- .hw_free = sdw_hw_free,
+ .prepare = mx8373_sdw_prepare,
+ .trigger = sdw_trigger,
+ .hw_free = mx8373_sdw_hw_free,
.shutdown = sdw_shutdown,
};
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 46513bb97904473cf6f4baeddc04a1c9c802f0f9..d1c570ca21ea781f84ff8c5bc6db6d8a9ea0ddc5 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1015,6 +1015,7 @@ out:
static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
{
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
int ret = -EINVAL, _ret = 0;
int rollback = 0;
@@ -1055,14 +1056,23 @@ start_err:
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback);
- if (ret < 0)
- break;
+ if (rtd->dai_link->stop_dma_first) {
+ ret = snd_soc_pcm_component_trigger(substream, cmd, rollback);
+ if (ret < 0)
+ break;
- ret = snd_soc_pcm_component_trigger(substream, cmd, rollback);
- if (ret < 0)
- break;
+ ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback);
+ if (ret < 0)
+ break;
+ } else {
+ ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback);
+ if (ret < 0)
+ break;
+ ret = snd_soc_pcm_component_trigger(substream, cmd, rollback);
+ if (ret < 0)
+ break;
+ }
ret = snd_soc_link_trigger(substream, cmd, rollback);
break;
}
diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c
index a00262184efab4da818ac0b45a38c3b9956fbbd7..d04ce84fe7cc2a4ead4540c651595b8fe75ad1b5 100644
--- a/sound/soc/sof/intel/pci-tgl.c
+++ b/sound/soc/sof/intel/pci-tgl.c
@@ -89,6 +89,7 @@ static const struct sof_dev_desc adls_desc = {
static const struct sof_dev_desc adl_desc = {
.machines = snd_soc_acpi_intel_adl_machines,
.alt_machines = snd_soc_acpi_intel_adl_sdw_machines,
+ .use_acpi_target_states = true,
.resindex_lpe_base = 0,
.resindex_pcicfg_base = -1,
.resindex_imr_base = -1,
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index 573374b89b100a9ae296a437e27bc5e12a30f4e1..d3276b4595affb18b33083ccf1156fb717fc06bf 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -213,19 +213,19 @@ snd_pcm_uframes_t tegra_pcm_pointer(struct snd_soc_component *component,
}
EXPORT_SYMBOL_GPL(tegra_pcm_pointer);
-static int tegra_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream,
+static int tegra_pcm_preallocate_dma_buffer(struct device *dev, struct snd_pcm *pcm, int stream,
size_t size)
{
struct snd_pcm_substream *substream = pcm->streams[stream].substream;
struct snd_dma_buffer *buf = &substream->dma_buffer;
- buf->area = dma_alloc_wc(pcm->card->dev, size, &buf->addr, GFP_KERNEL);
+ buf->area = dma_alloc_wc(dev, size, &buf->addr, GFP_KERNEL);
if (!buf->area)
return -ENOMEM;
buf->private_data = NULL;
buf->dev.type = SNDRV_DMA_TYPE_DEV;
- buf->dev.dev = pcm->card->dev;
+ buf->dev.dev = dev;
buf->bytes = size;
return 0;
@@ -244,31 +244,28 @@ static void tegra_pcm_deallocate_dma_buffer(struct snd_pcm *pcm, int stream)
if (!buf->area)
return;
- dma_free_wc(pcm->card->dev, buf->bytes, buf->area, buf->addr);
+ dma_free_wc(buf->dev.dev, buf->bytes, buf->area, buf->addr);
buf->area = NULL;
}
-static int tegra_pcm_dma_allocate(struct snd_soc_pcm_runtime *rtd,
+static int tegra_pcm_dma_allocate(struct device *dev, struct snd_soc_pcm_runtime *rtd,
size_t size)
{
- struct snd_card *card = rtd->card->snd_card;
struct snd_pcm *pcm = rtd->pcm;
int ret;
- ret = dma_set_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
+ ret = dma_set_mask_and_coherent(dev, DMA_BIT_MASK(32));
if (ret < 0)
return ret;
if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
- ret = tegra_pcm_preallocate_dma_buffer(pcm,
- SNDRV_PCM_STREAM_PLAYBACK, size);
+ ret = tegra_pcm_preallocate_dma_buffer(dev, pcm, SNDRV_PCM_STREAM_PLAYBACK, size);
if (ret)
goto err;
}
if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
- ret = tegra_pcm_preallocate_dma_buffer(pcm,
- SNDRV_PCM_STREAM_CAPTURE, size);
+ ret = tegra_pcm_preallocate_dma_buffer(dev, pcm, SNDRV_PCM_STREAM_CAPTURE, size);
if (ret)
goto err_free_play;
}
@@ -284,7 +281,16 @@ err:
int tegra_pcm_construct(struct snd_soc_component *component,
struct snd_soc_pcm_runtime *rtd)
{
- return tegra_pcm_dma_allocate(rtd, tegra_pcm_hardware.buffer_bytes_max);
+ struct device *dev = component->dev;
+
+ /*
+ * Fallback for backwards-compatibility with older device trees that
+ * have the iommus property in the virtual, top-level "sound" node.
+ */
+ if (!of_get_property(dev->of_node, "iommus", NULL))
+ dev = rtd->card->snd_card->dev;
+
+ return tegra_pcm_dma_allocate(dev, rtd, tegra_pcm_hardware.buffer_bytes_max);
}
EXPORT_SYMBOL_GPL(tegra_pcm_construct);
diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c
index a7c0484d44ec79dbf74d2a02aa96f2d3958e7da9..265bbc5a2f96a8eba48febe8e7f42c965ba0a49c 100644
--- a/sound/soc/ti/j721e-evm.c
+++ b/sound/soc/ti/j721e-evm.c
@@ -197,7 +197,7 @@ static int j721e_configure_refclk(struct j721e_priv *priv,
return ret;
}
- if (priv->hsdiv_rates[domain->parent_clk_id] != scki) {
+ if (domain->parent_clk_id == -1 || priv->hsdiv_rates[domain->parent_clk_id] != scki) {
dev_dbg(priv->dev,
"%s configuration for %u Hz: %s, %dxFS (SCKI: %u Hz)\n",
audio_domain == J721E_AUDIO_DOMAIN_CPB ? "CPB" : "IVI",
@@ -278,23 +278,29 @@ static int j721e_audio_startup(struct snd_pcm_substream *substream)
j721e_rule_rate, &priv->rate_range,
SNDRV_PCM_HW_PARAM_RATE, -1);
- mutex_unlock(&priv->mutex);
if (ret)
- return ret;
+ goto out;
/* Reset TDM slots to 32 */
ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 32);
if (ret && ret != -ENOTSUPP)
- return ret;
+ goto out;
for_each_rtd_codec_dais(rtd, i, codec_dai) {
ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 2, 32);
if (ret && ret != -ENOTSUPP)
- return ret;
+ goto out;
}
- return 0;
+ if (ret == -ENOTSUPP)
+ ret = 0;
+out:
+ if (ret)
+ domain->active--;
+ mutex_unlock(&priv->mutex);
+
+ return ret;
}
static int j721e_audio_hw_params(struct snd_pcm_substream *substream,
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 2f6a62416c057466f1c8af82d7621a8e8822714b..a1f8c3a026f57901f4372181a85a1dabbdf6d641 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -907,7 +907,7 @@ static void usb_audio_disconnect(struct usb_interface *intf)
}
}
- if (chip->quirk_type & QUIRK_SETUP_DISABLE_AUTOSUSPEND)
+ if (chip->quirk_type == QUIRK_SETUP_DISABLE_AUTOSUSPEND)
usb_enable_autosuspend(interface_to_usbdev(intf));
chip->num_interfaces--;
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index 52de52288e1051be41fb01d8449f8e1df24d378b..14456f61539e1a1481d4b6cc2c4130fe0fcfd524 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -324,6 +324,12 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip,
sources[ret - 1],
visited, validate);
if (ret > 0) {
+ /*
+ * For Samsung USBC Headset (AKG), setting clock selector again
+ * will result in incorrect default clock setting problems
+ */
+ if (chip->usb_id == USB_ID(0x04e8, 0xa051))
+ return ret;
err = uac_clock_selector_set_val(chip, entity_id, cur);
if (err < 0)
return err;
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 30b3e128e28d86865a39b0333314a0588f8b2d30..9b713b4a5ec4cf4732533eb415550460af0a4a4f 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -1816,6 +1816,15 @@ static void get_connector_control_name(struct usb_mixer_interface *mixer,
strlcat(name, " - Output Jack", name_size);
}
+/* get connector value to "wake up" the USB audio */
+static int connector_mixer_resume(struct usb_mixer_elem_list *list)
+{
+ struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list);
+
+ get_connector_value(cval, NULL, NULL);
+ return 0;
+}
+
/* Build a mixer control for a UAC connector control (jack-detect) */
static void build_connector_control(struct usb_mixer_interface *mixer,
const struct usbmix_name_map *imap,
@@ -1833,6 +1842,10 @@ static void build_connector_control(struct usb_mixer_interface *mixer,
if (!cval)
return;
snd_usb_mixer_elem_init_std(&cval->head, mixer, term->id);
+
+ /* set up a specific resume callback */
+ cval->head.resume = connector_mixer_resume;
+
/*
* UAC2: The first byte from reading the UAC2_TE_CONNECTOR control returns the
* number of channels connected.
@@ -3295,7 +3308,15 @@ static void snd_usb_mixer_dump_cval(struct snd_info_buffer *buffer,
{
struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list);
static const char * const val_types[] = {
- "BOOLEAN", "INV_BOOLEAN", "S8", "U8", "S16", "U16", "S32", "U32",
+ [USB_MIXER_BOOLEAN] = "BOOLEAN",
+ [USB_MIXER_INV_BOOLEAN] = "INV_BOOLEAN",
+ [USB_MIXER_S8] = "S8",
+ [USB_MIXER_U8] = "U8",
+ [USB_MIXER_S16] = "S16",
+ [USB_MIXER_U16] = "U16",
+ [USB_MIXER_S32] = "S32",
+ [USB_MIXER_U32] = "U32",
+ [USB_MIXER_BESPOKEN] = "BESPOKEN",
};
snd_iprintf(buffer, " Info: id=%i, control=%i, cmask=0x%x, "
"channels=%i, type=\"%s\"\n", cval->head.id,
@@ -3634,23 +3655,15 @@ static int restore_mixer_value(struct usb_mixer_elem_list *list)
return 0;
}
-static int default_mixer_resume(struct usb_mixer_elem_list *list)
-{
- struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list);
-
- /* get connector value to "wake up" the USB audio */
- if (cval->val_type == USB_MIXER_BOOLEAN && cval->channels == 1)
- get_connector_value(cval, NULL, NULL);
-
- return 0;
-}
-
static int default_mixer_reset_resume(struct usb_mixer_elem_list *list)
{
- int err = default_mixer_resume(list);
+ int err;
- if (err < 0)
- return err;
+ if (list->resume) {
+ err = list->resume(list);
+ if (err < 0)
+ return err;
+ }
return restore_mixer_value(list);
}
@@ -3689,7 +3702,7 @@ void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list,
list->id = unitid;
list->dump = snd_usb_mixer_dump_cval;
#ifdef CONFIG_PM
- list->resume = default_mixer_resume;
+ list->resume = NULL;
list->reset_resume = default_mixer_reset_resume;
#endif
}
diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c
index f9d698a371539d716754bad57cbe35a86a9f9ce9..3d5848d5481be93156792793f92575f3bba850c8 100644
--- a/sound/usb/mixer_scarlett_gen2.c
+++ b/sound/usb/mixer_scarlett_gen2.c
@@ -228,7 +228,7 @@ enum {
};
static const char *const scarlett2_dim_mute_names[SCARLETT2_DIM_MUTE_COUNT] = {
- "Mute", "Dim"
+ "Mute Playback Switch", "Dim Playback Switch"
};
/* Description of each hardware port type:
@@ -1856,9 +1856,15 @@ static int scarlett2_mute_ctl_get(struct snd_kcontrol *kctl,
struct snd_ctl_elem_value *ucontrol)
{
struct usb_mixer_elem_info *elem = kctl->private_data;
- struct scarlett2_data *private = elem->head.mixer->private_data;
+ struct usb_mixer_interface *mixer = elem->head.mixer;
+ struct scarlett2_data *private = mixer->private_data;
int index = line_out_remap(private, elem->control);
+ mutex_lock(&private->data_mutex);
+ if (private->vol_updated)
+ scarlett2_update_volumes(mixer);
+ mutex_unlock(&private->data_mutex);
+
ucontrol->value.integer.value[0] = private->mute_switch[index];
return 0;
}
@@ -1955,10 +1961,12 @@ static void scarlett2_vol_ctl_set_writable(struct usb_mixer_interface *mixer,
~SNDRV_CTL_ELEM_ACCESS_WRITE;
}
- /* Notify of write bit change */
- snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO,
+ /* Notify of write bit and possible value change */
+ snd_ctl_notify(card,
+ SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO,
&private->vol_ctls[index]->id);
- snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO,
+ snd_ctl_notify(card,
+ SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO,
&private->mute_ctls[index]->id);
}
@@ -2530,14 +2538,18 @@ static int scarlett2_add_direct_monitor_ctl(struct usb_mixer_interface *mixer)
{
struct scarlett2_data *private = mixer->private_data;
const struct scarlett2_device_info *info = private->info;
+ const char *s;
if (!info->direct_monitor)
return 0;
+ s = info->direct_monitor == 1
+ ? "Direct Monitor Playback Switch"
+ : "Direct Monitor Playback Enum";
+
return scarlett2_add_new_ctl(
mixer, &scarlett2_direct_monitor_ctl[info->direct_monitor - 1],
- 0, 1, "Direct Monitor Playback Switch",
- &private->direct_monitor_ctl);
+ 0, 1, s, &private->direct_monitor_ctl);
}
/*** Speaker Switching Control ***/
@@ -2589,7 +2601,9 @@ static int scarlett2_speaker_switch_enable(struct usb_mixer_interface *mixer)
/* disable the line out SW/HW switch */
scarlett2_sw_hw_ctl_ro(private, i);
- snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO,
+ snd_ctl_notify(card,
+ SNDRV_CTL_EVENT_MASK_VALUE |
+ SNDRV_CTL_EVENT_MASK_INFO,
&private->sw_hw_ctls[i]->id);
}
@@ -2913,7 +2927,7 @@ static int scarlett2_dim_mute_ctl_put(struct snd_kcontrol *kctl,
if (private->vol_sw_hw_switch[line_index]) {
private->mute_switch[line_index] = val;
snd_ctl_notify(mixer->chip->card,
- SNDRV_CTL_EVENT_MASK_INFO,
+ SNDRV_CTL_EVENT_MASK_VALUE,
&private->mute_ctls[i]->id);
}
}
@@ -3455,7 +3469,7 @@ static int scarlett2_add_msd_ctl(struct usb_mixer_interface *mixer)
/* Add MSD control */
return scarlett2_add_new_ctl(mixer, &scarlett2_msd_ctl,
- 0, 1, "MSD Mode", NULL);
+ 0, 1, "MSD Mode Switch", NULL);
}
/*** Cleanup/Suspend Callbacks ***/
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 8b8bee3c3dd635aa50ec87df83c54fe628d95acb..326d1b0ea5e6928c6c324d5a1a221e0368b2793f 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1897,6 +1897,10 @@ static const struct registration_quirk registration_quirks[] = {
REG_QUIRK_ENTRY(0x0951, 0x16d8, 2), /* Kingston HyperX AMP */
REG_QUIRK_ENTRY(0x0951, 0x16ed, 2), /* Kingston HyperX Cloud Alpha S */
REG_QUIRK_ENTRY(0x0951, 0x16ea, 2), /* Kingston HyperX Cloud Flight S */
+ REG_QUIRK_ENTRY(0x0ecb, 0x1f46, 2), /* JBL Quantum 600 */
+ REG_QUIRK_ENTRY(0x0ecb, 0x2039, 2), /* JBL Quantum 400 */
+ REG_QUIRK_ENTRY(0x0ecb, 0x203c, 2), /* JBL Quantum 600 */
+ REG_QUIRK_ENTRY(0x0ecb, 0x203e, 2), /* JBL Quantum 800 */
{ 0 } /* terminator */
};