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authorLinus Torvalds <torvalds@linux-foundation.org>2023-01-13 08:20:29 -0600
committerLinus Torvalds <torvalds@linux-foundation.org>2023-01-13 08:20:29 -0600
commit689968db7b6145b2e4beb8b472d31162ffa5ad7d (patch)
treefe155e661187ff3f79c1ce2d1a053f4e3146a5b0 /sound
parentd863f0539b525ba714f85c15ea961b225a15dd21 (diff)
parent56b88b50565cd8b946a2d00b0c83927b7ebb055e (diff)
Merge tag 'sound-6.2-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai: "This became a slightly big update, but it's more or less expected, as the first batch after holidays. All changes (but for the last two last-minute fixes) have been stewed in linux-next long enough, so it's fairly safe to take: - PCM UAF fix in 32bit compat layer - ASoC board-specific fixes for Intel, AMD, Medathek, Qualcomm - SOF power management fixes - ASoC Intel link failure fixes - A series of fixes for USB-audio regressions - CS35L41 HD-audio codec regression fixes - HD-audio device-specific fixes / quirks Note that one SPI patch has been taken in ASoC subtree mistakenly, and the same fix is found in spi tree, but it should be OK to apply" * tag 'sound-6.2-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (39 commits) ALSA: pcm: Move rwsem lock inside snd_ctl_elem_read to prevent UAF ALSA: usb-audio: Fix possible NULL pointer dereference in snd_usb_pcm_has_fixed_rate() ALSA: hda/realtek: Enable mute/micmute LEDs on HP Spectre x360 13-aw0xxx ASoC: fsl-asoc-card: Fix naming of AC'97 CODEC widgets ASoC: fsl_ssi: Rename AC'97 streams to avoid collisions with AC'97 CODEC ALSA: hda/hdmi: Add a HP device 0x8715 to force connect list ALSA: control-led: use strscpy in set_led_id() ALSA: usb-audio: Always initialize fixed_rate in snd_usb_find_implicit_fb_sync_format() ASoC: dt-bindings: qcom,lpass-tx-macro: correct clocks on SC7280 ASoC: dt-bindings: qcom,lpass-wsa-macro: correct clocks on SM8250 ASoC: qcom: Fix building APQ8016 machine driver without SOUNDWIRE ALSA: hda: cs35l41: Check runtime suspend capability at runtime_idle ALSA: hda: cs35l41: Don't return -EINVAL from system suspend/resume ASoC: fsl_micfil: Correct the number of steps on SX controls ALSA: hda/realtek: fix mute/micmute LEDs don't work for a HP platform Revert "ALSA: usb-audio: Drop superfluous interface setup at parsing" ALSA: usb-audio: More refactoring of hw constraint rules ALSA: usb-audio: Relax hw constraints for implicit fb sync ALSA: usb-audio: Make sure to stop endpoints before closing EPs ALSA: hda - Enable headset mic on another Dell laptop with ALC3254 ...
Diffstat (limited to 'sound')
-rw-r--r--sound/core/control.c24
-rw-r--r--sound/core/control_led.c5
-rw-r--r--sound/pci/hda/cs35l41_hda.c20
-rw-r--r--sound/pci/hda/patch_hdmi.c1
-rw-r--r--sound/pci/hda/patch_realtek.c55
-rw-r--r--sound/soc/amd/yc/acp6x-mach.c14
-rw-r--r--sound/soc/codecs/rt9120.c12
-rw-r--r--sound/soc/codecs/wm8904.c7
-rw-r--r--sound/soc/fsl/fsl-asoc-card.c8
-rw-r--r--sound/soc/fsl/fsl_micfil.c16
-rw-r--r--sound/soc/fsl/fsl_ssi.c4
-rw-r--r--sound/soc/intel/boards/Kconfig2
-rw-r--r--sound/soc/intel/boards/sof_nau8825.c31
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-adl-match.c20
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-rpl-match.c50
-rw-r--r--sound/soc/mediatek/Kconfig4
-rw-r--r--sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c22
-rw-r--r--sound/soc/qcom/Kconfig21
-rw-r--r--sound/soc/qcom/Makefile2
-rw-r--r--sound/soc/qcom/common.c114
-rw-r--r--sound/soc/qcom/common.h10
-rw-r--r--sound/soc/qcom/lpass-cpu.c5
-rw-r--r--sound/soc/qcom/sc8280xp.c1
-rw-r--r--sound/soc/qcom/sdw.c123
-rw-r--r--sound/soc/qcom/sdw.h18
-rw-r--r--sound/soc/qcom/sm8250.c1
-rw-r--r--sound/soc/sof/debug.c4
-rw-r--r--sound/soc/sof/pm.c9
-rw-r--r--sound/usb/implicit.c3
-rw-r--r--sound/usb/pcm.c222
-rw-r--r--sound/usb/stream.c6
31 files changed, 553 insertions, 281 deletions
diff --git a/sound/core/control.c b/sound/core/control.c
index 50e7ba66f187..82aa1af1d1d8 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -1203,14 +1203,19 @@ static int snd_ctl_elem_read(struct snd_card *card,
const u32 pattern = 0xdeadbeef;
int ret;
+ down_read(&card->controls_rwsem);
kctl = snd_ctl_find_id(card, &control->id);
- if (kctl == NULL)
- return -ENOENT;
+ if (kctl == NULL) {
+ ret = -ENOENT;
+ goto unlock;
+ }
index_offset = snd_ctl_get_ioff(kctl, &control->id);
vd = &kctl->vd[index_offset];
- if (!(vd->access & SNDRV_CTL_ELEM_ACCESS_READ) || kctl->get == NULL)
- return -EPERM;
+ if (!(vd->access & SNDRV_CTL_ELEM_ACCESS_READ) || kctl->get == NULL) {
+ ret = -EPERM;
+ goto unlock;
+ }
snd_ctl_build_ioff(&control->id, kctl, index_offset);
@@ -1220,7 +1225,7 @@ static int snd_ctl_elem_read(struct snd_card *card,
info.id = control->id;
ret = __snd_ctl_elem_info(card, kctl, &info, NULL);
if (ret < 0)
- return ret;
+ goto unlock;
#endif
if (!snd_ctl_skip_validation(&info))
@@ -1230,7 +1235,7 @@ static int snd_ctl_elem_read(struct snd_card *card,
ret = kctl->get(kctl, control);
snd_power_unref(card);
if (ret < 0)
- return ret;
+ goto unlock;
if (!snd_ctl_skip_validation(&info) &&
sanity_check_elem_value(card, control, &info, pattern) < 0) {
dev_err(card->dev,
@@ -1238,8 +1243,11 @@ static int snd_ctl_elem_read(struct snd_card *card,
control->id.iface, control->id.device,
control->id.subdevice, control->id.name,
control->id.index);
- return -EINVAL;
+ ret = -EINVAL;
+ goto unlock;
}
+unlock:
+ up_read(&card->controls_rwsem);
return ret;
}
@@ -1253,9 +1261,7 @@ static int snd_ctl_elem_read_user(struct snd_card *card,
if (IS_ERR(control))
return PTR_ERR(control);
- down_read(&card->controls_rwsem);
result = snd_ctl_elem_read(card, control);
- up_read(&card->controls_rwsem);
if (result < 0)
goto error;
diff --git a/sound/core/control_led.c b/sound/core/control_led.c
index f975cc85772b..3cadd40100f3 100644
--- a/sound/core/control_led.c
+++ b/sound/core/control_led.c
@@ -530,12 +530,11 @@ static ssize_t set_led_id(struct snd_ctl_led_card *led_card, const char *buf, si
bool attach)
{
char buf2[256], *s, *os;
- size_t len = max(sizeof(s) - 1, count);
struct snd_ctl_elem_id id;
int err;
- strncpy(buf2, buf, len);
- buf2[len] = '\0';
+ if (strscpy(buf2, buf, sizeof(buf2)) < 0)
+ return -E2BIG;
memset(&id, 0, sizeof(id));
id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
s = buf2;
diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c
index 91842c0c8c74..f7815ee24f83 100644
--- a/sound/pci/hda/cs35l41_hda.c
+++ b/sound/pci/hda/cs35l41_hda.c
@@ -598,8 +598,8 @@ static int cs35l41_system_suspend(struct device *dev)
dev_dbg(cs35l41->dev, "System Suspend\n");
if (cs35l41->hw_cfg.bst_type == CS35L41_EXT_BOOST_NO_VSPK_SWITCH) {
- dev_err(cs35l41->dev, "System Suspend not supported\n");
- return -EINVAL;
+ dev_err_once(cs35l41->dev, "System Suspend not supported\n");
+ return 0; /* don't block the whole system suspend */
}
ret = pm_runtime_force_suspend(dev);
@@ -624,8 +624,8 @@ static int cs35l41_system_resume(struct device *dev)
dev_dbg(cs35l41->dev, "System Resume\n");
if (cs35l41->hw_cfg.bst_type == CS35L41_EXT_BOOST_NO_VSPK_SWITCH) {
- dev_err(cs35l41->dev, "System Resume not supported\n");
- return -EINVAL;
+ dev_err_once(cs35l41->dev, "System Resume not supported\n");
+ return 0; /* don't block the whole system resume */
}
if (cs35l41->reset_gpio) {
@@ -647,6 +647,15 @@ static int cs35l41_system_resume(struct device *dev)
return ret;
}
+static int cs35l41_runtime_idle(struct device *dev)
+{
+ struct cs35l41_hda *cs35l41 = dev_get_drvdata(dev);
+
+ if (cs35l41->hw_cfg.bst_type == CS35L41_EXT_BOOST_NO_VSPK_SWITCH)
+ return -EBUSY; /* suspend not supported yet on this model */
+ return 0;
+}
+
static int cs35l41_runtime_suspend(struct device *dev)
{
struct cs35l41_hda *cs35l41 = dev_get_drvdata(dev);
@@ -1536,7 +1545,8 @@ void cs35l41_hda_remove(struct device *dev)
EXPORT_SYMBOL_NS_GPL(cs35l41_hda_remove, SND_HDA_SCODEC_CS35L41);
const struct dev_pm_ops cs35l41_hda_pm_ops = {
- RUNTIME_PM_OPS(cs35l41_runtime_suspend, cs35l41_runtime_resume, NULL)
+ RUNTIME_PM_OPS(cs35l41_runtime_suspend, cs35l41_runtime_resume,
+ cs35l41_runtime_idle)
SYSTEM_SLEEP_PM_OPS(cs35l41_system_suspend, cs35l41_system_resume)
};
EXPORT_SYMBOL_NS_GPL(cs35l41_hda_pm_ops, SND_HDA_SCODEC_CS35L41);
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 386dd9d9143f..9ea633fe9339 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1981,6 +1981,7 @@ 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(0x103c, 0x8711, "HP", 1),
+ SND_PCI_QUIRK(0x103c, 0x8715, "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 3794b522c222..6fab7c8fc19a 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3564,6 +3564,15 @@ static void alc256_init(struct hda_codec *codec)
hda_nid_t hp_pin = alc_get_hp_pin(spec);
bool hp_pin_sense;
+ if (spec->ultra_low_power) {
+ alc_update_coef_idx(codec, 0x03, 1<<1, 1<<1);
+ alc_update_coef_idx(codec, 0x08, 3<<2, 3<<2);
+ alc_update_coef_idx(codec, 0x08, 7<<4, 0);
+ alc_update_coef_idx(codec, 0x3b, 1<<15, 0);
+ alc_update_coef_idx(codec, 0x0e, 7<<6, 7<<6);
+ msleep(30);
+ }
+
if (!hp_pin)
hp_pin = 0x21;
@@ -3575,14 +3584,6 @@ static void alc256_init(struct hda_codec *codec)
msleep(2);
alc_update_coefex_idx(codec, 0x57, 0x04, 0x0007, 0x1); /* Low power */
- if (spec->ultra_low_power) {
- alc_update_coef_idx(codec, 0x03, 1<<1, 1<<1);
- alc_update_coef_idx(codec, 0x08, 3<<2, 3<<2);
- alc_update_coef_idx(codec, 0x08, 7<<4, 0);
- alc_update_coef_idx(codec, 0x3b, 1<<15, 0);
- alc_update_coef_idx(codec, 0x0e, 7<<6, 7<<6);
- msleep(30);
- }
snd_hda_codec_write(codec, hp_pin, 0,
AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
@@ -3713,6 +3714,13 @@ static void alc225_init(struct hda_codec *codec)
hda_nid_t hp_pin = alc_get_hp_pin(spec);
bool hp1_pin_sense, hp2_pin_sense;
+ if (spec->ultra_low_power) {
+ alc_update_coef_idx(codec, 0x08, 0x0f << 2, 3<<2);
+ alc_update_coef_idx(codec, 0x0e, 7<<6, 7<<6);
+ alc_update_coef_idx(codec, 0x33, 1<<11, 0);
+ msleep(30);
+ }
+
if (spec->codec_variant != ALC269_TYPE_ALC287 &&
spec->codec_variant != ALC269_TYPE_ALC245)
/* required only at boot or S3 and S4 resume time */
@@ -3734,12 +3742,6 @@ static void alc225_init(struct hda_codec *codec)
msleep(2);
alc_update_coefex_idx(codec, 0x57, 0x04, 0x0007, 0x1); /* Low power */
- if (spec->ultra_low_power) {
- alc_update_coef_idx(codec, 0x08, 0x0f << 2, 3<<2);
- alc_update_coef_idx(codec, 0x0e, 7<<6, 7<<6);
- alc_update_coef_idx(codec, 0x33, 1<<11, 0);
- msleep(30);
- }
if (hp1_pin_sense || spec->ultra_low_power)
snd_hda_codec_write(codec, hp_pin, 0,
@@ -4644,6 +4646,16 @@ static void alc285_fixup_hp_coef_micmute_led(struct hda_codec *codec,
}
}
+static void alc285_fixup_hp_gpio_micmute_led(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+
+ if (action == HDA_FIXUP_ACT_PRE_PROBE)
+ spec->micmute_led_polarity = 1;
+ alc_fixup_hp_gpio_led(codec, action, 0, 0x04);
+}
+
static void alc236_fixup_hp_coef_micmute_led(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
{
@@ -4665,6 +4677,13 @@ static void alc285_fixup_hp_mute_led(struct hda_codec *codec,
alc285_fixup_hp_coef_micmute_led(codec, fix, action);
}
+static void alc285_fixup_hp_spectre_x360_mute_led(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ alc285_fixup_hp_mute_led_coefbit(codec, fix, action);
+ alc285_fixup_hp_gpio_micmute_led(codec, fix, action);
+}
+
static void alc236_fixup_hp_mute_led(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
{
@@ -7106,6 +7125,7 @@ enum {
ALC285_FIXUP_ASUS_G533Z_PINS,
ALC285_FIXUP_HP_GPIO_LED,
ALC285_FIXUP_HP_MUTE_LED,
+ ALC285_FIXUP_HP_SPECTRE_X360_MUTE_LED,
ALC236_FIXUP_HP_GPIO_LED,
ALC236_FIXUP_HP_MUTE_LED,
ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF,
@@ -8486,6 +8506,10 @@ static const struct hda_fixup alc269_fixups[] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc285_fixup_hp_mute_led,
},
+ [ALC285_FIXUP_HP_SPECTRE_X360_MUTE_LED] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc285_fixup_hp_spectre_x360_mute_led,
+ },
[ALC236_FIXUP_HP_GPIO_LED] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc236_fixup_hp_gpio_led,
@@ -9239,6 +9263,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x0b1a, "Dell Precision 5570", ALC289_FIXUP_DUAL_SPK),
SND_PCI_QUIRK(0x1028, 0x0b37, "Dell Inspiron 16 Plus 7620 2-in-1", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS),
SND_PCI_QUIRK(0x1028, 0x0b71, "Dell Inspiron 16 Plus 7620", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS),
+ SND_PCI_QUIRK(0x1028, 0x0c03, "Dell Precision 5340", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x0c19, "Dell Precision 3340", ALC236_FIXUP_DELL_DUAL_CODECS),
SND_PCI_QUIRK(0x1028, 0x0c1a, "Dell Precision 3340", ALC236_FIXUP_DELL_DUAL_CODECS),
SND_PCI_QUIRK(0x1028, 0x0c1b, "Dell Precision 3440", ALC236_FIXUP_DELL_DUAL_CODECS),
@@ -9327,6 +9352,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x86c7, "HP Envy AiO 32", ALC274_FIXUP_HP_ENVY_GPIO),
SND_PCI_QUIRK(0x103c, 0x86e7, "HP Spectre x360 15-eb0xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1),
SND_PCI_QUIRK(0x103c, 0x86e8, "HP Spectre x360 15-eb0xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1),
+ SND_PCI_QUIRK(0x103c, 0x86f9, "HP Spectre x360 13-aw0xxx", ALC285_FIXUP_HP_SPECTRE_X360_MUTE_LED),
SND_PCI_QUIRK(0x103c, 0x8716, "HP Elite Dragonfly G2 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT),
SND_PCI_QUIRK(0x103c, 0x8720, "HP EliteBook x360 1040 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT),
SND_PCI_QUIRK(0x103c, 0x8724, "HP EliteBook 850 G7", ALC285_FIXUP_HP_GPIO_LED),
@@ -9406,6 +9432,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x8ad2, "HP EliteBook 860 16 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x8b5d, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
SND_PCI_QUIRK(0x103c, 0x8b5e, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
+ SND_PCI_QUIRK(0x103c, 0x8bf0, "HP", 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),
diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c
index 1f0b5527c594..0d283e41f66d 100644
--- a/sound/soc/amd/yc/acp6x-mach.c
+++ b/sound/soc/amd/yc/acp6x-mach.c
@@ -209,6 +209,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
{
.driver_data = &acp6x_card,
.matches = {
+ DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."),
+ DMI_MATCH(DMI_PRODUCT_NAME, "M5402RA"),
+ }
+ },
+ {
+ .driver_data = &acp6x_card,
+ .matches = {
DMI_MATCH(DMI_BOARD_VENDOR, "Alienware"),
DMI_MATCH(DMI_PRODUCT_NAME, "Alienware m17 R5 AMD"),
}
@@ -220,6 +227,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
DMI_MATCH(DMI_PRODUCT_NAME, "Redmi Book Pro 14 2022"),
}
},
+ {
+ .driver_data = &acp6x_card,
+ .matches = {
+ DMI_MATCH(DMI_BOARD_VENDOR, "Razer"),
+ DMI_MATCH(DMI_PRODUCT_NAME, "Blade 14 (2022) - RZ09-0427"),
+ }
+ },
{}
};
diff --git a/sound/soc/codecs/rt9120.c b/sound/soc/codecs/rt9120.c
index 644300e88b4c..fcf4fbaed3c7 100644
--- a/sound/soc/codecs/rt9120.c
+++ b/sound/soc/codecs/rt9120.c
@@ -177,8 +177,20 @@ static int rt9120_codec_probe(struct snd_soc_component *comp)
return 0;
}
+static int rt9120_codec_suspend(struct snd_soc_component *comp)
+{
+ return pm_runtime_force_suspend(comp->dev);
+}
+
+static int rt9120_codec_resume(struct snd_soc_component *comp)
+{
+ return pm_runtime_force_resume(comp->dev);
+}
+
static const struct snd_soc_component_driver rt9120_component_driver = {
.probe = rt9120_codec_probe,
+ .suspend = rt9120_codec_suspend,
+ .resume = rt9120_codec_resume,
.controls = rt9120_snd_controls,
.num_controls = ARRAY_SIZE(rt9120_snd_controls),
.dapm_widgets = rt9120_dapm_widgets,
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index ca6a01a230af..791d8738d1c0 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -697,6 +697,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w,
int dcs_mask;
int dcs_l, dcs_r;
int dcs_l_reg, dcs_r_reg;
+ int an_out_reg;
int timeout;
int pwr_reg;
@@ -712,6 +713,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w,
dcs_mask = WM8904_DCS_ENA_CHAN_0 | WM8904_DCS_ENA_CHAN_1;
dcs_r_reg = WM8904_DC_SERVO_8;
dcs_l_reg = WM8904_DC_SERVO_9;
+ an_out_reg = WM8904_ANALOGUE_OUT1_LEFT;
dcs_l = 0;
dcs_r = 1;
break;
@@ -720,6 +722,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w,
dcs_mask = WM8904_DCS_ENA_CHAN_2 | WM8904_DCS_ENA_CHAN_3;
dcs_r_reg = WM8904_DC_SERVO_6;
dcs_l_reg = WM8904_DC_SERVO_7;
+ an_out_reg = WM8904_ANALOGUE_OUT2_LEFT;
dcs_l = 2;
dcs_r = 3;
break;
@@ -792,6 +795,10 @@ static int out_pga_event(struct snd_soc_dapm_widget *w,
snd_soc_component_update_bits(component, reg,
WM8904_HPL_ENA_OUTP | WM8904_HPR_ENA_OUTP,
WM8904_HPL_ENA_OUTP | WM8904_HPR_ENA_OUTP);
+
+ /* Update volume, requires PGA to be powered */
+ val = snd_soc_component_read(component, an_out_reg);
+ snd_soc_component_write(component, an_out_reg, val);
break;
case SND_SOC_DAPM_POST_PMU:
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index c836848ef0a6..8d14b5593658 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -121,11 +121,11 @@ static const struct snd_soc_dapm_route audio_map[] = {
static const struct snd_soc_dapm_route audio_map_ac97[] = {
/* 1st half -- Normal DAPM routes */
- {"Playback", NULL, "AC97 Playback"},
- {"AC97 Capture", NULL, "Capture"},
+ {"AC97 Playback", NULL, "CPU AC97 Playback"},
+ {"CPU AC97 Capture", NULL, "AC97 Capture"},
/* 2nd half -- ASRC DAPM routes */
- {"AC97 Playback", NULL, "ASRC-Playback"},
- {"ASRC-Capture", NULL, "AC97 Capture"},
+ {"CPU AC97 Playback", NULL, "ASRC-Playback"},
+ {"ASRC-Capture", NULL, "CPU AC97 Capture"},
};
static const struct snd_soc_dapm_route audio_map_tx[] = {
diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c
index 7b17f152bbf3..94341e4352b3 100644
--- a/sound/soc/fsl/fsl_micfil.c
+++ b/sound/soc/fsl/fsl_micfil.c
@@ -315,21 +315,21 @@ static int hwvad_detected(struct snd_kcontrol *kcontrol,
static const struct snd_kcontrol_new fsl_micfil_snd_controls[] = {
SOC_SINGLE_SX_TLV("CH0 Volume", REG_MICFIL_OUT_CTRL,
- MICFIL_OUTGAIN_CHX_SHIFT(0), 0xF, 0x7, gain_tlv),
+ MICFIL_OUTGAIN_CHX_SHIFT(0), 0x8, 0xF, gain_tlv),
SOC_SINGLE_SX_TLV("CH1 Volume", REG_MICFIL_OUT_CTRL,
- MICFIL_OUTGAIN_CHX_SHIFT(1), 0xF, 0x7, gain_tlv),
+ MICFIL_OUTGAIN_CHX_SHIFT(1), 0x8, 0xF, gain_tlv),
SOC_SINGLE_SX_TLV("CH2 Volume", REG_MICFIL_OUT_CTRL,
- MICFIL_OUTGAIN_CHX_SHIFT(2), 0xF, 0x7, gain_tlv),
+ MICFIL_OUTGAIN_CHX_SHIFT(2), 0x8, 0xF, gain_tlv),
SOC_SINGLE_SX_TLV("CH3 Volume", REG_MICFIL_OUT_CTRL,
- MICFIL_OUTGAIN_CHX_SHIFT(3), 0xF, 0x7, gain_tlv),
+ MICFIL_OUTGAIN_CHX_SHIFT(3), 0x8, 0xF, gain_tlv),
SOC_SINGLE_SX_TLV("CH4 Volume", REG_MICFIL_OUT_CTRL,
- MICFIL_OUTGAIN_CHX_SHIFT(4), 0xF, 0x7, gain_tlv),
+ MICFIL_OUTGAIN_CHX_SHIFT(4), 0x8, 0xF, gain_tlv),
SOC_SINGLE_SX_TLV("CH5 Volume", REG_MICFIL_OUT_CTRL,
- MICFIL_OUTGAIN_CHX_SHIFT(5), 0xF, 0x7, gain_tlv),
+ MICFIL_OUTGAIN_CHX_SHIFT(5), 0x8, 0xF, gain_tlv),
SOC_SINGLE_SX_TLV("CH6 Volume", REG_MICFIL_OUT_CTRL,
- MICFIL_OUTGAIN_CHX_SHIFT(6), 0xF, 0x7, gain_tlv),
+ MICFIL_OUTGAIN_CHX_SHIFT(6), 0x8, 0xF, gain_tlv),
SOC_SINGLE_SX_TLV("CH7 Volume", REG_MICFIL_OUT_CTRL,
- MICFIL_OUTGAIN_CHX_SHIFT(7), 0xF, 0x7, gain_tlv),
+ MICFIL_OUTGAIN_CHX_SHIFT(7), 0x8, 0xF, gain_tlv),
SOC_ENUM_EXT("MICFIL Quality Select",
fsl_micfil_quality_enum,
micfil_quality_get, micfil_quality_set),
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index c9e0e31d5b34..46a53551b955 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1189,14 +1189,14 @@ static struct snd_soc_dai_driver fsl_ssi_ac97_dai = {
.symmetric_channels = 1,
.probe = fsl_ssi_dai_probe,
.playback = {
- .stream_name = "AC97 Playback",
+ .stream_name = "CPU AC97 Playback",
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S20,
},
.capture = {
- .stream_name = "AC97 Capture",
+ .stream_name = "CPU AC97 Capture",
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_48000,
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index a472de1909f4..99308ed85277 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -554,10 +554,12 @@ config SND_SOC_INTEL_SOF_NAU8825_MACH
select SND_SOC_RT1015P
select SND_SOC_MAX98373_I2C
select SND_SOC_MAX98357A
+ select SND_SOC_NAU8315
select SND_SOC_DMIC
select SND_SOC_HDAC_HDMI
select SND_SOC_INTEL_HDA_DSP_COMMON
select SND_SOC_INTEL_SOF_MAXIM_COMMON
+ select SND_SOC_INTEL_SOF_REALTEK_COMMON
help
This adds support for ASoC machine driver for SOF platforms
with nau8825 codec.
diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c
index 27880224359d..a800854c2831 100644
--- a/sound/soc/intel/boards/sof_nau8825.c
+++ b/sound/soc/intel/boards/sof_nau8825.c
@@ -48,6 +48,7 @@
#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(15)
#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(16)
#define SOF_RT1015P_SPEAKER_AMP_PRESENT BIT(17)
+#define SOF_NAU8318_SPEAKER_AMP_PRESENT BIT(18)
static unsigned long sof_nau8825_quirk = SOF_NAU8825_SSP_CODEC(0);
@@ -338,6 +339,13 @@ static struct snd_soc_dai_link_component rt1019p_component[] = {
}
};
+static struct snd_soc_dai_link_component nau8318_components[] = {
+ {
+ .name = "NVTN2012:00",
+ .dai_name = "nau8315-hifi",
+ }
+};
+
static struct snd_soc_dai_link_component dummy_component[] = {
{
.name = "snd-soc-dummy",
@@ -486,6 +494,11 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev,
max_98360a_dai_link(&links[id]);
} else if (sof_nau8825_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) {
sof_rt1015p_dai_link(&links[id]);
+ } else if (sof_nau8825_quirk &
+ SOF_NAU8318_SPEAKER_AMP_PRESENT) {
+ links[id].codecs = nau8318_components;
+ links[id].num_codecs = ARRAY_SIZE(nau8318_components);
+ links[id].init = speaker_codec_init;
} else {
goto devm_err;
}
@@ -618,7 +631,7 @@ static const struct platform_device_id board_ids[] = {
},
{
- .name = "adl_rt1019p_nau8825",
+ .name = "adl_rt1019p_8825",
.driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) |
SOF_SPEAKER_AMP_PRESENT |
SOF_RT1019P_SPEAKER_AMP_PRESENT |
@@ -626,7 +639,7 @@ static const struct platform_device_id board_ids[] = {
SOF_NAU8825_NUM_HDMIDEV(4)),
},
{
- .name = "adl_max98373_nau8825",
+ .name = "adl_max98373_8825",
.driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) |
SOF_SPEAKER_AMP_PRESENT |
SOF_MAX98373_SPEAKER_AMP_PRESENT |
@@ -637,7 +650,7 @@ static const struct platform_device_id board_ids[] = {
},
{
/* The limitation of length of char array, shorten the name */
- .name = "adl_mx98360a_nau8825",
+ .name = "adl_mx98360a_8825",
.driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) |
SOF_SPEAKER_AMP_PRESENT |
SOF_MAX98360A_SPEAKER_AMP_PRESENT |
@@ -648,7 +661,7 @@ static const struct platform_device_id board_ids[] = {
},
{
- .name = "adl_rt1015p_nau8825",
+ .name = "adl_rt1015p_8825",
.driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) |
SOF_SPEAKER_AMP_PRESENT |
SOF_RT1015P_SPEAKER_AMP_PRESENT |
@@ -657,6 +670,16 @@ static const struct platform_device_id board_ids[] = {
SOF_BT_OFFLOAD_SSP(2) |
SOF_SSP_BT_OFFLOAD_PRESENT),
},
+ {
+ .name = "adl_nau8318_8825",
+ .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) |
+ SOF_SPEAKER_AMP_PRESENT |
+ SOF_NAU8318_SPEAKER_AMP_PRESENT |
+ SOF_NAU8825_SSP_AMP(1) |
+ SOF_NAU8825_NUM_HDMIDEV(4) |
+ SOF_BT_OFFLOAD_SSP(2) |
+ SOF_SSP_BT_OFFLOAD_PRESENT),
+ },
{ }
};
MODULE_DEVICE_TABLE(platform, board_ids);
diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c
index 60aee56f94bd..56ee5fef66a8 100644
--- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c
@@ -450,6 +450,11 @@ static const struct snd_soc_acpi_codecs adl_lt6911_hdmi = {
.codecs = {"INTC10B0"}
};
+static const struct snd_soc_acpi_codecs adl_nau8318_amp = {
+ .num_codecs = 1,
+ .codecs = {"NVTN2012"}
+};
+
struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = {
{
.comp_ids = &adl_rt5682_rt5682s_hp,
@@ -474,21 +479,21 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = {
},
{
.id = "10508825",
- .drv_name = "adl_rt1019p_nau8825",
+ .drv_name = "adl_rt1019p_8825",
.machine_quirk = snd_soc_acpi_codec_list,
.quirk_data = &adl_rt1019p_amp,
.sof_tplg_filename = "sof-adl-rt1019-nau8825.tplg",
},
{
.id = "10508825",
- .drv_name = "adl_max98373_nau8825",
+ .drv_name = "adl_max98373_8825",
.machine_quirk = snd_soc_acpi_codec_list,
.quirk_data = &adl_max98373_amp,
.sof_tplg_filename = "sof-adl-max98373-nau8825.tplg",
},
{
.id = "10508825",
- .drv_name = "adl_mx98360a_nau8825",
+ .drv_name = "adl_mx98360a_8825",
.machine_quirk = snd_soc_acpi_codec_list,
.quirk_data = &adl_max98360a_amp,
.sof_tplg_filename = "sof-adl-max98360a-nau8825.tplg",
@@ -502,13 +507,20 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = {
},
{
.id = "10508825",
- .drv_name = "adl_rt1015p_nau8825",
+ .drv_name = "adl_rt1015p_8825",
.machine_quirk = snd_soc_acpi_codec_list,
.quirk_data = &adl_rt1015p_amp,
.sof_tplg_filename = "sof-adl-rt1015-nau8825.tplg",
},
{
.id = "10508825",
+ .drv_name = "adl_nau8318_8825",
+ .machine_quirk = snd_soc_acpi_codec_list,
+ .quirk_data = &adl_nau8318_amp,
+ .sof_tplg_filename = "sof-adl-nau8318-nau8825.tplg",
+ },
+ {
+ .id = "10508825",
.drv_name = "sof_nau8825",
.sof_tplg_filename = "sof-adl-nau8825.tplg",
},
diff --git a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
index 31b43116e3d8..07f96a11ea2f 100644
--- a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c
@@ -203,6 +203,25 @@ static const struct snd_soc_acpi_link_adr rpl_sdw_rt711_link2_rt1316_link01_rt71
{}
};
+static const struct snd_soc_acpi_link_adr rpl_sdw_rt711_link2_rt1316_link01[] = {
+ {
+ .mask = BIT(2),
+ .num_adr = ARRAY_SIZE(rt711_sdca_2_adr),
+ .adr_d = rt711_sdca_2_adr,
+ },
+ {
+ .mask = BIT(0),
+ .num_adr = ARRAY_SIZE(rt1316_0_group2_adr),
+ .adr_d = rt1316_0_group2_adr,
+ },
+ {
+ .mask = BIT(1),
+ .num_adr = ARRAY_SIZE(rt1316_1_group2_adr),
+ .adr_d = rt1316_1_group2_adr,
+ },
+ {}
+};
+
static const struct snd_soc_acpi_link_adr rpl_sdw_rt711_link0_rt1318_link12_rt714_link3[] = {
{
.mask = BIT(0),
@@ -227,6 +246,25 @@ static const struct snd_soc_acpi_link_adr rpl_sdw_rt711_link0_rt1318_link12_rt71
{}
};
+static const struct snd_soc_acpi_link_adr rpl_sdw_rt711_link0_rt1318_link12[] = {
+ {
+ .mask = BIT(0),
+ .num_adr = ARRAY_SIZE(rt711_sdca_0_adr),
+ .adr_d = rt711_sdca_0_adr,
+ },
+ {
+ .mask = BIT(1),
+ .num_adr = ARRAY_SIZE(rt1318_1_group1_adr),
+ .adr_d = rt1318_1_group1_adr,
+ },
+ {
+ .mask = BIT(2),
+ .num_adr = ARRAY_SIZE(rt1318_2_group1_adr),
+ .adr_d = rt1318_2_group1_adr,
+ },
+ {}
+};
+
static const struct snd_soc_acpi_link_adr rpl_sdw_rt1316_link12_rt714_link0[] = {
{
.mask = BIT(1),
@@ -272,12 +310,24 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_sdw_machines[] = {
.sof_tplg_filename = "sof-rpl-rt711-l0-rt1318-l12-rt714-l3.tplg",
},
{
+ .link_mask = 0x7, /* rt711 on link0 & two rt1318s on link1 and link2 */
+ .links = rpl_sdw_rt711_link0_rt1318_link12,
+ .drv_name = "sof_sdw",
+ .sof_tplg_filename = "sof-rpl-rt711-l0-rt1318-l12.tplg",
+ },
+ {
.link_mask = 0x7, /* rt714 on link0 & two rt1316s on link1 and link2 */
.links = rpl_sdw_rt1316_link12_rt714_link0,
.drv_name = "sof_sdw",
.sof_tplg_filename = "sof-rpl-rt1316-l12-rt714-l0.tplg",
},
{
+ .link_mask = 0x7, /* rt711 on link2 & two rt1316s on link0 and link1 */
+ .links = rpl_sdw_rt711_link2_rt1316_link01,
+ .drv_name = "sof_sdw",
+ .sof_tplg_filename = "sof-rpl-rt711-l2-rt1316-l01.tplg",
+ },
+ {
.link_mask = 0x1, /* link0 required */
.links = rpl_rvp,
.drv_name = "sof_sdw",
diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig
index 363fa4d47680..b027fba8233d 100644
--- a/sound/soc/mediatek/Kconfig
+++ b/sound/soc/mediatek/Kconfig
@@ -182,10 +182,12 @@ config SND_SOC_MT8186_MT6366_DA7219_MAX98357
If unsure select "N".
config SND_SOC_MT8186_MT6366_RT1019_RT5682S
- tristate "ASoC Audio driver for MT8186 with RT1019 RT5682S codec"
+ tristate "ASoC Audio driver for MT8186 with RT1019 RT5682S MAX98357A/MAX98360 codec"
depends on I2C && GPIOLIB
depends on SND_SOC_MT8186 && MTK_PMIC_WRAP
+ select SND_SOC_MAX98357A
select SND_SOC_MT6358
+ select SND_SOC_MAX98357A
select SND_SOC_RT1015P
select SND_SOC_RT5682S
select SND_SOC_BT_SCO
diff --git a/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c b/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c
index 8f77a0bc1dc8..af44e331dae8 100644
--- a/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c
+++ b/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c
@@ -1083,6 +1083,21 @@ static struct snd_soc_card mt8186_mt6366_rt1019_rt5682s_soc_card = {
.num_configs = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_codec_conf),
};
+static struct snd_soc_card mt8186_mt6366_rt5682s_max98360_soc_card = {
+ .name = "mt8186_rt5682s_max98360",
+ .owner = THIS_MODULE,
+ .dai_link = mt8186_mt6366_rt1019_rt5682s_dai_links,
+ .num_links = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_dai_links),
+ .controls = mt8186_mt6366_rt1019_rt5682s_controls,
+ .num_controls = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_controls),
+ .dapm_widgets = mt8186_mt6366_rt1019_rt5682s_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_widgets),
+ .dapm_routes = mt8186_mt6366_rt1019_rt5682s_routes,
+ .num_dapm_routes = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_routes),
+ .codec_conf = mt8186_mt6366_rt1019_rt5682s_codec_conf,
+ .num_configs = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_codec_conf),
+};
+
static int mt8186_mt6366_rt1019_rt5682s_dev_probe(struct platform_device *pdev)
{
struct snd_soc_card *card;
@@ -1232,9 +1247,14 @@ err_adsp_node:
#if IS_ENABLED(CONFIG_OF)
static const struct of_device_id mt8186_mt6366_rt1019_rt5682s_dt_match[] = {
- { .compatible = "mediatek,mt8186-mt6366-rt1019-rt5682s-sound",
+ {
+ .compatible = "mediatek,mt8186-mt6366-rt1019-rt5682s-sound",
.data = &mt8186_mt6366_rt1019_rt5682s_soc_card,
},
+ {
+ .compatible = "mediatek,mt8186-mt6366-rt5682s-max98360-sound",
+ .data = &mt8186_mt6366_rt5682s_max98360_soc_card,
+ },
{}
};
MODULE_DEVICE_TABLE(of, mt8186_mt6366_rt1019_rt5682s_dt_match);
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index 96a6d4731e6f..e7b00d1d9e99 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -2,7 +2,6 @@
menuconfig SND_SOC_QCOM
tristate "ASoC support for QCOM platforms"
depends on ARCH_QCOM || COMPILE_TEST
- imply SND_SOC_QCOM_COMMON
help
Say Y or M if you want to add support to use audio devices
in Qualcomm Technologies SOC-based platforms.
@@ -60,14 +59,16 @@ config SND_SOC_STORM
config SND_SOC_APQ8016_SBC
tristate "SoC Audio support for APQ8016 SBC platforms"
select SND_SOC_LPASS_APQ8016
- depends on SND_SOC_QCOM_COMMON
+ select SND_SOC_QCOM_COMMON
help
Support for Qualcomm Technologies LPASS audio block in
APQ8016 SOC-based systems.
Say Y if you want to use audio devices on MI2S.
config SND_SOC_QCOM_COMMON
- depends on SOUNDWIRE
+ tristate
+
+config SND_SOC_QCOM_SDW
tristate
config SND_SOC_QDSP6_COMMON
@@ -144,7 +145,7 @@ config SND_SOC_MSM8996
depends on QCOM_APR
depends on COMMON_CLK
select SND_SOC_QDSP6
- depends on SND_SOC_QCOM_COMMON
+ select SND_SOC_QCOM_COMMON
help
Support for Qualcomm Technologies LPASS audio block in
APQ8096 SoC-based systems.
@@ -155,7 +156,7 @@ config SND_SOC_SDM845
depends on QCOM_APR && I2C && SOUNDWIRE
depends on COMMON_CLK
select SND_SOC_QDSP6
- depends on SND_SOC_QCOM_COMMON
+ select SND_SOC_QCOM_COMMON
select SND_SOC_RT5663
select SND_SOC_MAX98927
imply SND_SOC_CROS_EC_CODEC
@@ -169,7 +170,8 @@ config SND_SOC_SM8250
depends on QCOM_APR && SOUNDWIRE
depends on COMMON_CLK
select SND_SOC_QDSP6
- depends on SND_SOC_QCOM_COMMON
+ select SND_SOC_QCOM_COMMON
+ select SND_SOC_QCOM_SDW
help
To add support for audio on Qualcomm Technologies Inc.
SM8250 SoC-based systems.
@@ -180,7 +182,8 @@ config SND_SOC_SC8280XP
depends on QCOM_APR && SOUNDWIRE
depends on COMMON_CLK
select SND_SOC_QDSP6
- depends on SND_SOC_QCOM_COMMON
+ select SND_SOC_QCOM_COMMON
+ select SND_SOC_QCOM_SDW
help
To add support for audio on Qualcomm Technologies Inc.
SC8280XP SoC-based systems.
@@ -190,7 +193,7 @@ config SND_SOC_SC7180
tristate "SoC Machine driver for SC7180 boards"
depends on I2C && GPIOLIB
depends on SOUNDWIRE || SOUNDWIRE=n
- depends on SND_SOC_QCOM_COMMON
+ select SND_SOC_QCOM_COMMON
select SND_SOC_LPASS_SC7180
select SND_SOC_MAX98357A
select SND_SOC_RT5682_I2C
@@ -204,7 +207,7 @@ config SND_SOC_SC7180
config SND_SOC_SC7280
tristate "SoC Machine driver for SC7280 boards"
depends on I2C && SOUNDWIRE
- depends on SND_SOC_QCOM_COMMON
+ select SND_SOC_QCOM_COMMON
select SND_SOC_LPASS_SC7280
select SND_SOC_MAX98357A
select SND_SOC_WCD938X_SDW
diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile
index 8b97172cf990..254350d9dc06 100644
--- a/sound/soc/qcom/Makefile
+++ b/sound/soc/qcom/Makefile
@@ -28,6 +28,7 @@ snd-soc-sdm845-objs := sdm845.o
snd-soc-sm8250-objs := sm8250.o
snd-soc-sc8280xp-objs := sc8280xp.o
snd-soc-qcom-common-objs := common.o
+snd-soc-qcom-sdw-objs := sdw.o
obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o
obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o
@@ -38,6 +39,7 @@ obj-$(CONFIG_SND_SOC_SC8280XP) += snd-soc-sc8280xp.o
obj-$(CONFIG_SND_SOC_SDM845) += snd-soc-sdm845.o
obj-$(CONFIG_SND_SOC_SM8250) += snd-soc-sm8250.o
obj-$(CONFIG_SND_SOC_QCOM_COMMON) += snd-soc-qcom-common.o
+obj-$(CONFIG_SND_SOC_QCOM_SDW) += snd-soc-qcom-sdw.o
#DSP lib
obj-$(CONFIG_SND_SOC_QDSP6) += qdsp6/
diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c
index 49c74c1662a3..96fe80241fb4 100644
--- a/sound/soc/qcom/common.c
+++ b/sound/soc/qcom/common.c
@@ -180,120 +180,6 @@ err_put_np:
}
EXPORT_SYMBOL_GPL(qcom_snd_parse_of);
-int qcom_snd_sdw_prepare(struct snd_pcm_substream *substream,
- struct sdw_stream_runtime *sruntime,
- bool *stream_prepared)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
- int ret;
-
- if (!sruntime)
- return 0;
-
- switch (cpu_dai->id) {
- case WSA_CODEC_DMA_RX_0:
- case WSA_CODEC_DMA_RX_1:
- case RX_CODEC_DMA_RX_0:
- case RX_CODEC_DMA_RX_1:
- case TX_CODEC_DMA_TX_0:
- case TX_CODEC_DMA_TX_1:
- case TX_CODEC_DMA_TX_2:
- case TX_CODEC_DMA_TX_3:
- break;
- default:
- return 0;
- }
-
- if (*stream_prepared) {
- sdw_disable_stream(sruntime);
- sdw_deprepare_stream(sruntime);
- *stream_prepared = false;
- }
-
- ret = sdw_prepare_stream(sruntime);
- if (ret)
- return ret;
-
- /**
- * NOTE: there is a strict hw requirement about the ordering of port
- * enables and actual WSA881x PA enable. PA enable should only happen
- * after soundwire ports are enabled if not DC on the line is
- * accumulated resulting in Click/Pop Noise
- * PA enable/mute are handled as part of codec DAPM and digital mute.
- */
-
- ret = sdw_enable_stream(sruntime);
- if (ret) {
- sdw_deprepare_stream(sruntime);
- return ret;
- }
- *stream_prepared = true;
-
- return ret;
-}
-EXPORT_SYMBOL_GPL(qcom_snd_sdw_prepare);
-
-int qcom_snd_sdw_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params,
- struct sdw_stream_runtime **psruntime)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai;
- struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
- struct sdw_stream_runtime *sruntime;
- int i;
-
- switch (cpu_dai->id) {
- case WSA_CODEC_DMA_RX_0:
- case RX_CODEC_DMA_RX_0:
- case RX_CODEC_DMA_RX_1:
- case TX_CODEC_DMA_TX_0:
- case TX_CODEC_DMA_TX_1:
- case TX_CODEC_DMA_TX_2:
- case TX_CODEC_DMA_TX_3:
- for_each_rtd_codec_dais(rtd, i, codec_dai) {
- sruntime = snd_soc_dai_get_stream(codec_dai, substream->stream);
- if (sruntime != ERR_PTR(-ENOTSUPP))
- *psruntime = sruntime;
- }
- break;
- }
-
- return 0;
-
-}
-EXPORT_SYMBOL_GPL(qcom_snd_sdw_hw_params);
-
-int qcom_snd_sdw_hw_free(struct snd_pcm_substream *substream,
- struct sdw_stream_runtime *sruntime, bool *stream_prepared)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
-
- switch (cpu_dai->id) {
- case WSA_CODEC_DMA_RX_0:
- case WSA_CODEC_DMA_RX_1:
- case RX_CODEC_DMA_RX_0:
- case RX_CODEC_DMA_RX_1:
- case TX_CODEC_DMA_TX_0:
- case TX_CODEC_DMA_TX_1:
- case TX_CODEC_DMA_TX_2:
- case TX_CODEC_DMA_TX_3:
- if (sruntime && *stream_prepared) {
- sdw_disable_stream(sruntime);
- sdw_deprepare_stream(sruntime);
- *stream_prepared = false;
- }
- break;
- default:
- break;
- }
-
- return 0;
-}
-EXPORT_SYMBOL_GPL(qcom_snd_sdw_hw_free);
-
int qcom_snd_wcd_jack_setup(struct snd_soc_pcm_runtime *rtd,
struct snd_soc_jack *jack, bool *jack_setup)
{
diff --git a/sound/soc/qcom/common.h b/sound/soc/qcom/common.h
index 3ef5bb6d12df..d7f80ee5ae26 100644
--- a/sound/soc/qcom/common.h
+++ b/sound/soc/qcom/common.h
@@ -5,19 +5,9 @@
#define __QCOM_SND_COMMON_H__
#include <sound/soc.h>
-#include <linux/soundwire/sdw.h>
int qcom_snd_parse_of(struct snd_soc_card *card);
int qcom_snd_wcd_jack_setup(struct snd_soc_pcm_runtime *rtd,
struct snd_soc_jack *jack, bool *jack_setup);
-int qcom_snd_sdw_prepare(struct snd_pcm_substream *substream,
- struct sdw_stream_runtime *runtime,
- bool *stream_prepared);
-int qcom_snd_sdw_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params,
- struct sdw_stream_runtime **psruntime);
-int qcom_snd_sdw_hw_free(struct snd_pcm_substream *substream,
- struct sdw_stream_runtime *sruntime,
- bool *stream_prepared);
#endif
diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c
index 54353842dc07..dbdaaa85ce48 100644
--- a/sound/soc/qcom/lpass-cpu.c
+++ b/sound/soc/qcom/lpass-cpu.c
@@ -1037,10 +1037,11 @@ static void of_lpass_cpu_parse_dai_data(struct device *dev,
struct lpass_data *data)
{
struct device_node *node;
- int ret, id;
+ int ret, i, id;
/* Allow all channels by default for backwards compatibility */
- for (id = 0; id < data->variant->num_dai; id++) {
+ for (i = 0; i < data->variant->num_dai; i++) {
+ id = data->variant->dai_driver[i].id;
data->mi2s_playback_sd_mode[id] = LPAIF_I2SCTL_MODE_8CH;
data->mi2s_capture_sd_mode[id] = LPAIF_I2SCTL_MODE_8CH;
}
diff --git a/sound/soc/qcom/sc8280xp.c b/sound/soc/qcom/sc8280xp.c
index ade44ad7c585..14d9fea33d16 100644
--- a/sound/soc/qcom/sc8280xp.c
+++ b/sound/soc/qcom/sc8280xp.c
@@ -12,6 +12,7 @@
#include <linux/input-event-codes.h>
#include "qdsp6/q6afe.h"
#include "common.h"
+#include "sdw.h"
#define DRIVER_NAME "sc8280xp"
diff --git a/sound/soc/qcom/sdw.c b/sound/soc/qcom/sdw.c
new file mode 100644
index 000000000000..10249519a39e
--- /dev/null
+++ b/sound/soc/qcom/sdw.c
@@ -0,0 +1,123 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2018, Linaro Limited.
+// Copyright (c) 2018, The Linux Foundation. All rights reserved.
+
+#include <linux/module.h>
+#include <sound/soc.h>
+#include "qdsp6/q6afe.h"
+#include "sdw.h"
+
+int qcom_snd_sdw_prepare(struct snd_pcm_substream *substream,
+ struct sdw_stream_runtime *sruntime,
+ bool *stream_prepared)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ int ret;
+
+ if (!sruntime)
+ return 0;
+
+ switch (cpu_dai->id) {
+ case WSA_CODEC_DMA_RX_0:
+ case WSA_CODEC_DMA_RX_1:
+ case RX_CODEC_DMA_RX_0:
+ case RX_CODEC_DMA_RX_1:
+ case TX_CODEC_DMA_TX_0:
+ case TX_CODEC_DMA_TX_1:
+ case TX_CODEC_DMA_TX_2:
+ case TX_CODEC_DMA_TX_3:
+ break;
+ default:
+ return 0;
+ }
+
+ if (*stream_prepared) {
+ sdw_disable_stream(sruntime);
+ sdw_deprepare_stream(sruntime);
+ *stream_prepared = false;
+ }
+
+ ret = sdw_prepare_stream(sruntime);
+ if (ret)
+ return ret;
+
+ /**
+ * NOTE: there is a strict hw requirement about the ordering of port
+ * enables and actual WSA881x PA enable. PA enable should only happen
+ * after soundwire ports are enabled if not DC on the line is
+ * accumulated resulting in Click/Pop Noise
+ * PA enable/mute are handled as part of codec DAPM and digital mute.
+ */
+
+ ret = sdw_enable_stream(sruntime);
+ if (ret) {
+ sdw_deprepare_stream(sruntime);
+ return ret;
+ }
+ *stream_prepared = true;
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(qcom_snd_sdw_prepare);
+
+int qcom_snd_sdw_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct sdw_stream_runtime **psruntime)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct sdw_stream_runtime *sruntime;
+ int i;
+
+ switch (cpu_dai->id) {
+ case WSA_CODEC_DMA_RX_0:
+ case RX_CODEC_DMA_RX_0:
+ case RX_CODEC_DMA_RX_1:
+ case TX_CODEC_DMA_TX_0:
+ case TX_CODEC_DMA_TX_1:
+ case TX_CODEC_DMA_TX_2:
+ case TX_CODEC_DMA_TX_3:
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
+ sruntime = snd_soc_dai_get_stream(codec_dai, substream->stream);
+ if (sruntime != ERR_PTR(-ENOTSUPP))
+ *psruntime = sruntime;
+ }
+ break;
+ }
+
+ return 0;
+
+}
+EXPORT_SYMBOL_GPL(qcom_snd_sdw_hw_params);
+
+int qcom_snd_sdw_hw_free(struct snd_pcm_substream *substream,
+ struct sdw_stream_runtime *sruntime, bool *stream_prepared)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+
+ switch (cpu_dai->id) {
+ case WSA_CODEC_DMA_RX_0:
+ case WSA_CODEC_DMA_RX_1:
+ case RX_CODEC_DMA_RX_0:
+ case RX_CODEC_DMA_RX_1:
+ case TX_CODEC_DMA_TX_0:
+ case TX_CODEC_DMA_TX_1:
+ case TX_CODEC_DMA_TX_2:
+ case TX_CODEC_DMA_TX_3:
+ if (sruntime && *stream_prepared) {
+ sdw_disable_stream(sruntime);
+ sdw_deprepare_stream(sruntime);
+ *stream_prepared = false;
+ }
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(qcom_snd_sdw_hw_free);
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/qcom/sdw.h b/sound/soc/qcom/sdw.h
new file mode 100644
index 000000000000..d74cbb84da13
--- /dev/null
+++ b/sound/soc/qcom/sdw.h
@@ -0,0 +1,18 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+// Copyright (c) 2018, The Linux Foundation. All rights reserved.
+
+#ifndef __QCOM_SND_SDW_H__
+#define __QCOM_SND_SDW_H__
+
+#include <linux/soundwire/sdw.h>
+
+int qcom_snd_sdw_prepare(struct snd_pcm_substream *substream,
+ struct sdw_stream_runtime *runtime,
+ bool *stream_prepared);
+int qcom_snd_sdw_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct sdw_stream_runtime **psruntime);
+int qcom_snd_sdw_hw_free(struct snd_pcm_substream *substream,
+ struct sdw_stream_runtime *sruntime,
+ bool *stream_prepared);
+#endif
diff --git a/sound/soc/qcom/sm8250.c b/sound/soc/qcom/sm8250.c
index 8dbe9ef41b1c..9626a9ef78c2 100644
--- a/sound/soc/qcom/sm8250.c
+++ b/sound/soc/qcom/sm8250.c
@@ -12,6 +12,7 @@
#include <linux/input-event-codes.h>
#include "qdsp6/q6afe.h"
#include "common.h"
+#include "sdw.h"
#define DRIVER_NAME "sm8250"
#define MI2S_BCLK_RATE 1536000
diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c
index d9a3ce7b69e1..ade0507328af 100644
--- a/sound/soc/sof/debug.c
+++ b/sound/soc/sof/debug.c
@@ -353,7 +353,9 @@ int snd_sof_dbg_init(struct snd_sof_dev *sdev)
return err;
}
- return 0;
+ return snd_sof_debugfs_buf_item(sdev, &sdev->fw_state,
+ sizeof(sdev->fw_state),
+ "fw_state", 0444);
}
EXPORT_SYMBOL_GPL(snd_sof_dbg_init);
diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c
index df740be645e8..8722bbd7fd3d 100644
--- a/sound/soc/sof/pm.c
+++ b/sound/soc/sof/pm.c
@@ -182,7 +182,7 @@ static int sof_suspend(struct device *dev, bool runtime_suspend)
const struct sof_ipc_pm_ops *pm_ops = sdev->ipc->ops->pm;
const struct sof_ipc_tplg_ops *tplg_ops = sdev->ipc->ops->tplg;
pm_message_t pm_state;
- u32 target_state = 0;
+ u32 target_state = snd_sof_dsp_power_target(sdev);
int ret;
/* do nothing if dsp suspend callback is not set */
@@ -192,6 +192,9 @@ static int sof_suspend(struct device *dev, bool runtime_suspend)
if (runtime_suspend && !sof_ops(sdev)->runtime_suspend)
return 0;
+ if (tplg_ops && tplg_ops->tear_down_all_pipelines)
+ tplg_ops->tear_down_all_pipelines(sdev, false);
+
if (sdev->fw_state != SOF_FW_BOOT_COMPLETE)
goto suspend;
@@ -206,7 +209,6 @@ static int sof_suspend(struct device *dev, bool runtime_suspend)
}
}
- target_state = snd_sof_dsp_power_target(sdev);
pm_state.event = target_state;
/* Skip to platform-specific suspend if DSP is entering D0 */
@@ -217,9 +219,6 @@ static int sof_suspend(struct device *dev, bool runtime_suspend)
goto suspend;
}
- if (tplg_ops->tear_down_all_pipelines)
- tplg_ops->tear_down_all_pipelines(sdev, false);
-
/* suspend DMA trace */
sof_fw_trace_suspend(sdev, pm_state);
diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c
index 41ac7185b42b..4727043fd745 100644
--- a/sound/usb/implicit.c
+++ b/sound/usb/implicit.c
@@ -471,7 +471,7 @@ snd_usb_find_implicit_fb_sync_format(struct snd_usb_audio *chip,
subs = find_matching_substream(chip, stream, target->sync_ep,
target->fmt_type);
if (!subs)
- return sync_fmt;
+ goto end;
high_score = 0;
list_for_each_entry(fp, &subs->fmt_list, list) {
@@ -485,6 +485,7 @@ snd_usb_find_implicit_fb_sync_format(struct snd_usb_audio *chip,
}
}
+ end:
if (fixed_rate)
*fixed_rate = snd_usb_pcm_has_fixed_rate(subs);
return sync_fmt;
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 99a66d0ef5b2..d959da7a1afb 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -160,9 +160,12 @@ find_substream_format(struct snd_usb_substream *subs,
bool snd_usb_pcm_has_fixed_rate(struct snd_usb_substream *subs)
{
const struct audioformat *fp;
- struct snd_usb_audio *chip = subs->stream->chip;
+ struct snd_usb_audio *chip;
int rate = -1;
+ if (!subs)
+ return false;
+ chip = subs->stream->chip;
if (!(chip->quirk_flags & QUIRK_FLAG_FIXED_RATE))
return false;
list_for_each_entry(fp, &subs->fmt_list, list) {
@@ -525,6 +528,8 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream,
if (snd_usb_endpoint_compatible(chip, subs->data_endpoint,
fmt, hw_params))
goto unlock;
+ if (stop_endpoints(subs, false))
+ sync_pending_stops(subs);
close_endpoints(chip, subs);
}
@@ -787,11 +792,27 @@ static int apply_hw_params_minmax(struct snd_interval *it, unsigned int rmin,
return changed;
}
+/* get the specified endpoint object that is being used by other streams
+ * (i.e. the parameter is locked)
+ */
+static const struct snd_usb_endpoint *
+get_endpoint_in_use(struct snd_usb_audio *chip, int endpoint,
+ const struct snd_usb_endpoint *ref_ep)
+{
+ const struct snd_usb_endpoint *ep;
+
+ ep = snd_usb_get_endpoint(chip, endpoint);
+ if (ep && ep->cur_audiofmt && (ep != ref_ep || ep->opened > 1))
+ return ep;
+ return NULL;
+}
+
static int hw_rule_rate(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
struct snd_usb_substream *subs = rule->private;
struct snd_usb_audio *chip = subs->stream->chip;
+ const struct snd_usb_endpoint *ep;
const struct audioformat *fp;
struct snd_interval *it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
unsigned int rmin, rmax, r;
@@ -803,6 +824,29 @@ static int hw_rule_rate(struct snd_pcm_hw_params *params,
list_for_each_entry(fp, &subs->fmt_list, list) {
if (!hw_check_valid_format(subs, params, fp))
continue;
+
+ ep = get_endpoint_in_use(chip, fp->endpoint,
+ subs->data_endpoint);
+ if (ep) {
+ hwc_debug("rate limit %d for ep#%x\n",
+ ep->cur_rate, fp->endpoint);
+ rmin = min(rmin, ep->cur_rate);
+ rmax = max(rmax, ep->cur_rate);
+ continue;
+ }
+
+ if (fp->implicit_fb) {
+ ep = get_endpoint_in_use(chip, fp->sync_ep,
+ subs->sync_endpoint);
+ if (ep) {
+ hwc_debug("rate limit %d for sync_ep#%x\n",
+ ep->cur_rate, fp->sync_ep);
+ rmin = min(rmin, ep->cur_rate);
+ rmax = max(rmax, ep->cur_rate);
+ continue;
+ }
+ }
+
r = snd_usb_endpoint_get_clock_rate(chip, fp->clock);
if (r > 0) {
if (!snd_interval_test(it, r))
@@ -872,6 +916,8 @@ static int hw_rule_format(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
struct snd_usb_substream *subs = rule->private;
+ struct snd_usb_audio *chip = subs->stream->chip;
+ const struct snd_usb_endpoint *ep;
const struct audioformat *fp;
struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
u64 fbits;
@@ -881,6 +927,27 @@ static int hw_rule_format(struct snd_pcm_hw_params *params,
list_for_each_entry(fp, &subs->fmt_list, list) {
if (!hw_check_valid_format(subs, params, fp))
continue;
+
+ ep = get_endpoint_in_use(chip, fp->endpoint,
+ subs->data_endpoint);
+ if (ep) {
+ hwc_debug("format limit %d for ep#%x\n",
+ ep->cur_format, fp->endpoint);
+ fbits |= pcm_format_to_bits(ep->cur_format);
+ continue;
+ }
+
+ if (fp->implicit_fb) {
+ ep = get_endpoint_in_use(chip, fp->sync_ep,
+ subs->sync_endpoint);
+ if (ep) {
+ hwc_debug("format limit %d for sync_ep#%x\n",
+ ep->cur_format, fp->sync_ep);
+ fbits |= pcm_format_to_bits(ep->cur_format);
+ continue;
+ }
+ }
+
fbits |= fp->formats;
}
return apply_hw_params_format_bits(fmt, fbits);
@@ -913,98 +980,95 @@ static int hw_rule_period_time(struct snd_pcm_hw_params *params,
return apply_hw_params_minmax(it, pmin, UINT_MAX);
}
-/* get the EP or the sync EP for implicit fb when it's already set up */
-static const struct snd_usb_endpoint *
-get_sync_ep_from_substream(struct snd_usb_substream *subs)
-{
- struct snd_usb_audio *chip = subs->stream->chip;
- const struct audioformat *fp;
- const struct snd_usb_endpoint *ep;
-
- list_for_each_entry(fp, &subs->fmt_list, list) {
- ep = snd_usb_get_endpoint(chip, fp->endpoint);
- if (ep && ep->cur_audiofmt) {
- /* if EP is already opened solely for this substream,
- * we still allow us to change the parameter; otherwise
- * this substream has to follow the existing parameter
- */
- if (ep->cur_audiofmt != subs->cur_audiofmt || ep->opened > 1)
- return ep;
- }
- if (!fp->implicit_fb)
- continue;
- /* for the implicit fb, check the sync ep as well */
- ep = snd_usb_get_endpoint(chip, fp->sync_ep);
- if (ep && ep->cur_audiofmt)
- return ep;
- }
- return NULL;
-}
-
/* additional hw constraints for implicit feedback mode */
-static int hw_rule_format_implicit_fb(struct snd_pcm_hw_params *params,
- struct snd_pcm_hw_rule *rule)
-{
- struct snd_usb_substream *subs = rule->private;
- const struct snd_usb_endpoint *ep;
- struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
-
- ep = get_sync_ep_from_substream(subs);
- if (!ep)
- return 0;
-
- hwc_debug("applying %s\n", __func__);
- return apply_hw_params_format_bits(fmt, pcm_format_to_bits(ep->cur_format));
-}
-
-static int hw_rule_rate_implicit_fb(struct snd_pcm_hw_params *params,
- struct snd_pcm_hw_rule *rule)
-{
- struct snd_usb_substream *subs = rule->private;
- const struct snd_usb_endpoint *ep;
- struct snd_interval *it;
-
- ep = get_sync_ep_from_substream(subs);
- if (!ep)
- return 0;
-
- hwc_debug("applying %s\n", __func__);
- it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
- return apply_hw_params_minmax(it, ep->cur_rate, ep->cur_rate);
-}
-
static int hw_rule_period_size_implicit_fb(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
struct snd_usb_substream *subs = rule->private;
+ struct snd_usb_audio *chip = subs->stream->chip;
+ const struct audioformat *fp;
const struct snd_usb_endpoint *ep;
struct snd_interval *it;
+ unsigned int rmin, rmax;
- ep = get_sync_ep_from_substream(subs);
- if (!ep)
- return 0;
-
- hwc_debug("applying %s\n", __func__);
it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
- return apply_hw_params_minmax(it, ep->cur_period_frames,
- ep->cur_period_frames);
+ hwc_debug("hw_rule_period_size: (%u,%u)\n", it->min, it->max);
+ rmin = UINT_MAX;
+ rmax = 0;
+ list_for_each_entry(fp, &subs->fmt_list, list) {
+ if (!hw_check_valid_format(subs, params, fp))
+ continue;
+ ep = get_endpoint_in_use(chip, fp->endpoint,
+ subs->data_endpoint);
+ if (ep) {
+ hwc_debug("period size limit %d for ep#%x\n",
+ ep->cur_period_frames, fp->endpoint);
+ rmin = min(rmin, ep->cur_period_frames);
+ rmax = max(rmax, ep->cur_period_frames);
+ continue;
+ }
+
+ if (fp->implicit_fb) {
+ ep = get_endpoint_in_use(chip, fp->sync_ep,
+ subs->sync_endpoint);
+ if (ep) {
+ hwc_debug("period size limit %d for sync_ep#%x\n",
+ ep->cur_period_frames, fp->sync_ep);
+ rmin = min(rmin, ep->cur_period_frames);
+ rmax = max(rmax, ep->cur_period_frames);
+ continue;
+ }
+ }
+ }
+
+ if (!rmax)
+ return 0; /* no limit by implicit fb */
+ return apply_hw_params_minmax(it, rmin, rmax);
}
static int hw_rule_periods_implicit_fb(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
struct snd_usb_substream *subs = rule->private;
+ struct snd_usb_audio *chip = subs->stream->chip;
+ const struct audioformat *fp;
const struct snd_usb_endpoint *ep;
struct snd_interval *it;
+ unsigned int rmin, rmax;
- ep = get_sync_ep_from_substream(subs);
- if (!ep)
- return 0;
-
- hwc_debug("applying %s\n", __func__);
it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIODS);
- return apply_hw_params_minmax(it, ep->cur_buffer_periods,
- ep->cur_buffer_periods);
+ hwc_debug("hw_rule_periods: (%u,%u)\n", it->min, it->max);
+ rmin = UINT_MAX;
+ rmax = 0;
+ list_for_each_entry(fp, &subs->fmt_list, list) {
+ if (!hw_check_valid_format(subs, params, fp))
+ continue;
+ ep = get_endpoint_in_use(chip, fp->endpoint,
+ subs->data_endpoint);
+ if (ep) {
+ hwc_debug("periods limit %d for ep#%x\n",
+ ep->cur_buffer_periods, fp->endpoint);
+ rmin = min(rmin, ep->cur_buffer_periods);
+ rmax = max(rmax, ep->cur_buffer_periods);
+ continue;
+ }
+
+ if (fp->implicit_fb) {
+ ep = get_endpoint_in_use(chip, fp->sync_ep,
+ subs->sync_endpoint);
+ if (ep) {
+ hwc_debug("periods limit %d for sync_ep#%x\n",
+ ep->cur_buffer_periods, fp->sync_ep);
+ rmin = min(rmin, ep->cur_buffer_periods);
+ rmax = max(rmax, ep->cur_buffer_periods);
+ continue;
+ }
+ }
+ }
+
+ if (!rmax)
+ return 0; /* no limit by implicit fb */
+ return apply_hw_params_minmax(it, rmin, rmax);
}
/*
@@ -1113,16 +1177,6 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre
return err;
/* additional hw constraints for implicit fb */
- err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT,
- hw_rule_format_implicit_fb, subs,
- SNDRV_PCM_HW_PARAM_FORMAT, -1);
- if (err < 0)
- return err;
- err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
- hw_rule_rate_implicit_fb, subs,
- SNDRV_PCM_HW_PARAM_RATE, -1);
- if (err < 0)
- return err;
err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
hw_rule_period_size_implicit_fb, subs,
SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1);
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index f75601ca2d52..f10f4e6d3fb8 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -1222,6 +1222,12 @@ static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip,
if (err < 0)
return err;
}
+
+ /* try to set the interface... */
+ usb_set_interface(chip->dev, iface_no, 0);
+ snd_usb_init_pitch(chip, fp);
+ snd_usb_init_sample_rate(chip, fp, fp->rate_max);
+ usb_set_interface(chip->dev, iface_no, altno);
}
return 0;
}