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authorLinus Torvalds <torvalds@linux-foundation.org>2019-10-24 06:09:42 -0400
committerLinus Torvalds <torvalds@linux-foundation.org>2019-10-24 06:09:42 -0400
commitf632bfaa33ed147ba9eade15f137fc32a13bd4b8 (patch)
treef1289b360a22724363c1220f563070efd9fd6229
parentfa8a74de0622e97c7d0c9dff8bf6718548bd6e45 (diff)
parent4750c212174892d26645cdf5ad73fb0e9d594ed3 (diff)
download96b-common-f632bfaa33ed147ba9eade15f137fc32a13bd4b8.tar.gz
Merge tag 'sound-5.4-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai: "This is a usual small bump in the middle, we've got a set of ASoC fixes in this week as shown in diffstat. The only change in the core stuff is about (somewhat minor) PCM debugfs error handling. The major changes are rather for Intel SOF and topology coverage, as well as other platform (rockchip, samsung, stm) and codec fixes. As non-ASoC changes, a couple of new HD-audio chip fixes and a typo correction of USB-audio driver validation code are found" * tag 'sound-5.4-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (29 commits) ALSA: hda: Add Tigerlake/Jasperlake PCI ID ALSA: usb-audio: Fix copy&paste error in the validator ALSA: hda/realtek - Add support for ALC711 ASoC: SOF: control: return true when kcontrol values change ASoC: stm32: sai: fix sysclk management on shutdown ASoC: Intel: sof-rt5682: add a check for devm_clk_get ASoC: rsnd: Reinitialize bit clock inversion flag for every format setting ASoC: simple_card_utils.h: Fix potential multiple redefinition error ASoC: msm8916-wcd-digital: add missing MIX2 path for RX1/2 ASoC: core: Fix pcm code debugfs error ASoc: rockchip: i2s: Fix RPM imbalance ASoC: wm_adsp: Don't generate kcontrols without READ flags ASoC: intel: bytcr_rt5651: add null check to support_button_press ASoC: intel: sof_rt5682: add remove function to disable jack ASoC: rt5682: add NULL handler to set_jack function ASoC: intel: sof_rt5682: use separate route map for dmic ASoC: SOF: Intel: hda: Disable DMI L1 entry during capture ASoC: SOF: Intel: initialise and verify FW crash dump data. ASoC: SOF: Intel: hda: fix warnings during FW load ASoC: SOF: pcm: harden PCM STOP sequence ...
-rw-r--r--include/sound/simple_card_utils.h8
-rw-r--r--sound/pci/hda/hda_intel.c6
-rw-r--r--sound/pci/hda/patch_realtek.c3
-rw-r--r--sound/soc/codecs/max98373.c20
-rw-r--r--sound/soc/codecs/msm8916-wcd-digital.c22
-rw-r--r--sound/soc/codecs/rt5651.c3
-rw-r--r--sound/soc/codecs/rt5682.c12
-rw-r--r--sound/soc/codecs/wm8994.c43
-rw-r--r--sound/soc/codecs/wm_adsp.c10
-rw-r--r--sound/soc/intel/boards/sof_rt5682.c60
-rw-r--r--sound/soc/rockchip/rockchip_i2s.c2
-rw-r--r--sound/soc/samsung/arndale_rt5631.c34
-rw-r--r--sound/soc/sh/rcar/core.c1
-rw-r--r--sound/soc/soc-pcm.c17
-rw-r--r--sound/soc/soc-topology.c2
-rw-r--r--sound/soc/sof/control.c26
-rw-r--r--sound/soc/sof/intel/Kconfig10
-rw-r--r--sound/soc/sof/intel/bdw.c7
-rw-r--r--sound/soc/sof/intel/byt.c6
-rw-r--r--sound/soc/sof/intel/hda-ctrl.c12
-rw-r--r--sound/soc/sof/intel/hda-loader.c1
-rw-r--r--sound/soc/sof/intel/hda-stream.c45
-rw-r--r--sound/soc/sof/intel/hda.c7
-rw-r--r--sound/soc/sof/intel/hda.h5
-rw-r--r--sound/soc/sof/loader.c4
-rw-r--r--sound/soc/sof/pcm.c35
-rw-r--r--sound/soc/sof/topology.c4
-rw-r--r--sound/soc/stm/stm32_sai_sub.c21
-rw-r--r--sound/usb/validate.c2
29 files changed, 330 insertions, 98 deletions
diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h
index 985a5f583de4..31f76b6abf71 100644
--- a/include/sound/simple_card_utils.h
+++ b/include/sound/simple_card_utils.h
@@ -135,9 +135,9 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv,
struct link_info *li);
#ifdef DEBUG
-inline void asoc_simple_debug_dai(struct asoc_simple_priv *priv,
- char *name,
- struct asoc_simple_dai *dai)
+static inline void asoc_simple_debug_dai(struct asoc_simple_priv *priv,
+ char *name,
+ struct asoc_simple_dai *dai)
{
struct device *dev = simple_priv_to_dev(priv);
@@ -167,7 +167,7 @@ inline void asoc_simple_debug_dai(struct asoc_simple_priv *priv,
dev_dbg(dev, "%s clk %luHz\n", name, clk_get_rate(dai->clk));
}
-inline void asoc_simple_debug_info(struct asoc_simple_priv *priv)
+static inline void asoc_simple_debug_info(struct asoc_simple_priv *priv)
{
struct snd_soc_card *card = simple_priv_to_card(priv);
struct device *dev = simple_priv_to_dev(priv);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 240f4ca76391..a815bc811799 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2399,6 +2399,12 @@ static const struct pci_device_id azx_ids[] = {
/* Icelake */
{ PCI_DEVICE(0x8086, 0x34c8),
.driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+ /* Jasperlake */
+ { PCI_DEVICE(0x8086, 0x38c8),
+ .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+ /* Tigerlake */
+ { PCI_DEVICE(0x8086, 0xa0c8),
+ .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
/* Elkhart Lake */
{ PCI_DEVICE(0x8086, 0x4b55),
.driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ce4f11659765..085a2f95e076 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -393,6 +393,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
case 0x10ec0700:
case 0x10ec0701:
case 0x10ec0703:
+ case 0x10ec0711:
alc_update_coef_idx(codec, 0x10, 1<<15, 0);
break;
case 0x10ec0662:
@@ -8019,6 +8020,7 @@ static int patch_alc269(struct hda_codec *codec)
case 0x10ec0700:
case 0x10ec0701:
case 0x10ec0703:
+ case 0x10ec0711:
spec->codec_variant = ALC269_TYPE_ALC700;
spec->gen.mixer_nid = 0; /* ALC700 does not have any loopback mixer path */
alc_update_coef_idx(codec, 0x4a, 1 << 15, 0); /* Combo jack auto trigger control */
@@ -9233,6 +9235,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = {
HDA_CODEC_ENTRY(0x10ec0700, "ALC700", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0701, "ALC701", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0703, "ALC703", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0711, "ALC711", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0867, "ALC891", patch_alc662),
HDA_CODEC_ENTRY(0x10ec0880, "ALC880", patch_alc880),
HDA_CODEC_ENTRY(0x10ec0882, "ALC882", patch_alc882),
diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index e609abcf3220..eb709d528259 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -901,16 +901,20 @@ static void max98373_slot_config(struct i2c_client *i2c,
max98373->i_slot = value & 0xF;
else
max98373->i_slot = 1;
-
- max98373->reset_gpio = of_get_named_gpio(dev->of_node,
+ if (dev->of_node) {
+ max98373->reset_gpio = of_get_named_gpio(dev->of_node,
"maxim,reset-gpio", 0);
- if (!gpio_is_valid(max98373->reset_gpio)) {
- dev_err(dev, "Looking up %s property in node %s failed %d\n",
- "maxim,reset-gpio", dev->of_node->full_name,
- max98373->reset_gpio);
+ if (!gpio_is_valid(max98373->reset_gpio)) {
+ dev_err(dev, "Looking up %s property in node %s failed %d\n",
+ "maxim,reset-gpio", dev->of_node->full_name,
+ max98373->reset_gpio);
+ } else {
+ dev_dbg(dev, "maxim,reset-gpio=%d",
+ max98373->reset_gpio);
+ }
} else {
- dev_dbg(dev, "maxim,reset-gpio=%d",
- max98373->reset_gpio);
+ /* this makes reset_gpio as invalid */
+ max98373->reset_gpio = -1;
}
if (!device_property_read_u32(dev, "maxim,spkfb-slot-no", &value))
diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c
index 9fa5d44fdc79..58b2468fb2a7 100644
--- a/sound/soc/codecs/msm8916-wcd-digital.c
+++ b/sound/soc/codecs/msm8916-wcd-digital.c
@@ -243,6 +243,10 @@ static const char *const rx_mix1_text[] = {
"ZERO", "IIR1", "IIR2", "RX1", "RX2", "RX3"
};
+static const char * const rx_mix2_text[] = {
+ "ZERO", "IIR1", "IIR2"
+};
+
static const char *const dec_mux_text[] = {
"ZERO", "ADC1", "ADC2", "ADC3", "DMIC1", "DMIC2"
};
@@ -270,6 +274,16 @@ static const struct soc_enum rx3_mix1_inp_enum[] = {
SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX3_B2_CTL, 0, 6, rx_mix1_text),
};
+/* RX1 MIX2 */
+static const struct soc_enum rx_mix2_inp1_chain_enum =
+ SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX1_B3_CTL,
+ 0, 3, rx_mix2_text);
+
+/* RX2 MIX2 */
+static const struct soc_enum rx2_mix2_inp1_chain_enum =
+ SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX2_B3_CTL,
+ 0, 3, rx_mix2_text);
+
/* DEC */
static const struct soc_enum dec1_mux_enum = SOC_ENUM_SINGLE(
LPASS_CDC_CONN_TX_B1_CTL, 0, 6, dec_mux_text);
@@ -309,6 +323,10 @@ static const struct snd_kcontrol_new rx3_mix1_inp2_mux = SOC_DAPM_ENUM(
"RX3 MIX1 INP2 Mux", rx3_mix1_inp_enum[1]);
static const struct snd_kcontrol_new rx3_mix1_inp3_mux = SOC_DAPM_ENUM(
"RX3 MIX1 INP3 Mux", rx3_mix1_inp_enum[2]);
+static const struct snd_kcontrol_new rx1_mix2_inp1_mux = SOC_DAPM_ENUM(
+ "RX1 MIX2 INP1 Mux", rx_mix2_inp1_chain_enum);
+static const struct snd_kcontrol_new rx2_mix2_inp1_mux = SOC_DAPM_ENUM(
+ "RX2 MIX2 INP1 Mux", rx2_mix2_inp1_chain_enum);
/* Digital Gain control -38.4 dB to +38.4 dB in 0.3 dB steps */
static const DECLARE_TLV_DB_SCALE(digital_gain, -3840, 30, 0);
@@ -740,6 +758,10 @@ static const struct snd_soc_dapm_widget msm8916_wcd_digital_dapm_widgets[] = {
&rx3_mix1_inp2_mux),
SND_SOC_DAPM_MUX("RX3 MIX1 INP3", SND_SOC_NOPM, 0, 0,
&rx3_mix1_inp3_mux),
+ SND_SOC_DAPM_MUX("RX1 MIX2 INP1", SND_SOC_NOPM, 0, 0,
+ &rx1_mix2_inp1_mux),
+ SND_SOC_DAPM_MUX("RX2 MIX2 INP1", SND_SOC_NOPM, 0, 0,
+ &rx2_mix2_inp1_mux),
SND_SOC_DAPM_MUX("CIC1 MUX", SND_SOC_NOPM, 0, 0, &cic1_mux),
SND_SOC_DAPM_MUX("CIC2 MUX", SND_SOC_NOPM, 0, 0, &cic2_mux),
diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c
index 762595de956c..c506c9305043 100644
--- a/sound/soc/codecs/rt5651.c
+++ b/sound/soc/codecs/rt5651.c
@@ -1770,6 +1770,9 @@ static int rt5651_detect_headset(struct snd_soc_component *component)
static bool rt5651_support_button_press(struct rt5651_priv *rt5651)
{
+ if (!rt5651->hp_jack)
+ return false;
+
/* Button press support only works with internal jack-detection */
return (rt5651->hp_jack->status & SND_JACK_MICROPHONE) &&
rt5651->gpiod_hp_det == NULL;
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index 1ef470700ed5..c50b75ce82e0 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -995,6 +995,16 @@ static int rt5682_set_jack_detect(struct snd_soc_component *component,
{
struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+ rt5682->hs_jack = hs_jack;
+
+ 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);
+ return 0;
+ }
+
switch (rt5682->pdata.jd_src) {
case RT5682_JD1:
snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_2,
@@ -1032,8 +1042,6 @@ static int rt5682_set_jack_detect(struct snd_soc_component *component,
break;
}
- rt5682->hs_jack = hs_jack;
-
return 0;
}
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index c3d06e8bc54f..d5fb7f5dd551 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -533,13 +533,10 @@ static SOC_ENUM_SINGLE_DECL(dac_osr,
static SOC_ENUM_SINGLE_DECL(adc_osr,
WM8994_OVERSAMPLING, 1, osr_text);
-static const struct snd_kcontrol_new wm8994_snd_controls[] = {
+static const struct snd_kcontrol_new wm8994_common_snd_controls[] = {
SOC_DOUBLE_R_TLV("AIF1ADC1 Volume", WM8994_AIF1_ADC1_LEFT_VOLUME,
WM8994_AIF1_ADC1_RIGHT_VOLUME,
1, 119, 0, digital_tlv),
-SOC_DOUBLE_R_TLV("AIF1ADC2 Volume", WM8994_AIF1_ADC2_LEFT_VOLUME,
- WM8994_AIF1_ADC2_RIGHT_VOLUME,
- 1, 119, 0, digital_tlv),
SOC_DOUBLE_R_TLV("AIF2ADC Volume", WM8994_AIF2_ADC_LEFT_VOLUME,
WM8994_AIF2_ADC_RIGHT_VOLUME,
1, 119, 0, digital_tlv),
@@ -556,8 +553,6 @@ SOC_ENUM("AIF2DACR Source", aif2dacr_src),
SOC_DOUBLE_R_TLV("AIF1DAC1 Volume", WM8994_AIF1_DAC1_LEFT_VOLUME,
WM8994_AIF1_DAC1_RIGHT_VOLUME, 1, 96, 0, digital_tlv),
-SOC_DOUBLE_R_TLV("AIF1DAC2 Volume", WM8994_AIF1_DAC2_LEFT_VOLUME,
- WM8994_AIF1_DAC2_RIGHT_VOLUME, 1, 96, 0, digital_tlv),
SOC_DOUBLE_R_TLV("AIF2DAC Volume", WM8994_AIF2_DAC_LEFT_VOLUME,
WM8994_AIF2_DAC_RIGHT_VOLUME, 1, 96, 0, digital_tlv),
@@ -565,17 +560,12 @@ SOC_SINGLE_TLV("AIF1 Boost Volume", WM8994_AIF1_CONTROL_2, 10, 3, 0, aif_tlv),
SOC_SINGLE_TLV("AIF2 Boost Volume", WM8994_AIF2_CONTROL_2, 10, 3, 0, aif_tlv),
SOC_SINGLE("AIF1DAC1 EQ Switch", WM8994_AIF1_DAC1_EQ_GAINS_1, 0, 1, 0),
-SOC_SINGLE("AIF1DAC2 EQ Switch", WM8994_AIF1_DAC2_EQ_GAINS_1, 0, 1, 0),
SOC_SINGLE("AIF2 EQ Switch", WM8994_AIF2_EQ_GAINS_1, 0, 1, 0),
WM8994_DRC_SWITCH("AIF1DAC1 DRC Switch", WM8994_AIF1_DRC1_1, 2),
WM8994_DRC_SWITCH("AIF1ADC1L DRC Switch", WM8994_AIF1_DRC1_1, 1),
WM8994_DRC_SWITCH("AIF1ADC1R DRC Switch", WM8994_AIF1_DRC1_1, 0),
-WM8994_DRC_SWITCH("AIF1DAC2 DRC Switch", WM8994_AIF1_DRC2_1, 2),
-WM8994_DRC_SWITCH("AIF1ADC2L DRC Switch", WM8994_AIF1_DRC2_1, 1),
-WM8994_DRC_SWITCH("AIF1ADC2R DRC Switch", WM8994_AIF1_DRC2_1, 0),
-
WM8994_DRC_SWITCH("AIF2DAC DRC Switch", WM8994_AIF2_DRC_1, 2),
WM8994_DRC_SWITCH("AIF2ADCL DRC Switch", WM8994_AIF2_DRC_1, 1),
WM8994_DRC_SWITCH("AIF2ADCR DRC Switch", WM8994_AIF2_DRC_1, 0),
@@ -594,9 +584,6 @@ SOC_SINGLE("Sidetone HPF Switch", WM8994_SIDETONE, 6, 1, 0),
SOC_ENUM("AIF1ADC1 HPF Mode", aif1adc1_hpf),
SOC_DOUBLE("AIF1ADC1 HPF Switch", WM8994_AIF1_ADC1_FILTERS, 12, 11, 1, 0),
-SOC_ENUM("AIF1ADC2 HPF Mode", aif1adc2_hpf),
-SOC_DOUBLE("AIF1ADC2 HPF Switch", WM8994_AIF1_ADC2_FILTERS, 12, 11, 1, 0),
-
SOC_ENUM("AIF2ADC HPF Mode", aif2adc_hpf),
SOC_DOUBLE("AIF2ADC HPF Switch", WM8994_AIF2_ADC_FILTERS, 12, 11, 1, 0),
@@ -637,6 +624,24 @@ SOC_SINGLE("AIF2DAC 3D Stereo Switch", WM8994_AIF2_DAC_FILTERS_2,
8, 1, 0),
};
+/* Controls not available on WM1811 */
+static const struct snd_kcontrol_new wm8994_snd_controls[] = {
+SOC_DOUBLE_R_TLV("AIF1ADC2 Volume", WM8994_AIF1_ADC2_LEFT_VOLUME,
+ WM8994_AIF1_ADC2_RIGHT_VOLUME,
+ 1, 119, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("AIF1DAC2 Volume", WM8994_AIF1_DAC2_LEFT_VOLUME,
+ WM8994_AIF1_DAC2_RIGHT_VOLUME, 1, 96, 0, digital_tlv),
+
+SOC_SINGLE("AIF1DAC2 EQ Switch", WM8994_AIF1_DAC2_EQ_GAINS_1, 0, 1, 0),
+
+WM8994_DRC_SWITCH("AIF1DAC2 DRC Switch", WM8994_AIF1_DRC2_1, 2),
+WM8994_DRC_SWITCH("AIF1ADC2L DRC Switch", WM8994_AIF1_DRC2_1, 1),
+WM8994_DRC_SWITCH("AIF1ADC2R DRC Switch", WM8994_AIF1_DRC2_1, 0),
+
+SOC_ENUM("AIF1ADC2 HPF Mode", aif1adc2_hpf),
+SOC_DOUBLE("AIF1ADC2 HPF Switch", WM8994_AIF1_ADC2_FILTERS, 12, 11, 1, 0),
+};
+
static const struct snd_kcontrol_new wm8994_eq_controls[] = {
SOC_SINGLE_TLV("AIF1DAC1 EQ1 Volume", WM8994_AIF1_DAC1_EQ_GAINS_1, 11, 31, 0,
eq_tlv),
@@ -4258,13 +4263,15 @@ static int wm8994_component_probe(struct snd_soc_component *component)
wm8994_handle_pdata(wm8994);
wm_hubs_add_analogue_controls(component);
- snd_soc_add_component_controls(component, wm8994_snd_controls,
- ARRAY_SIZE(wm8994_snd_controls));
+ snd_soc_add_component_controls(component, wm8994_common_snd_controls,
+ ARRAY_SIZE(wm8994_common_snd_controls));
snd_soc_dapm_new_controls(dapm, wm8994_dapm_widgets,
ARRAY_SIZE(wm8994_dapm_widgets));
switch (control->type) {
case WM8994:
+ snd_soc_add_component_controls(component, wm8994_snd_controls,
+ ARRAY_SIZE(wm8994_snd_controls));
snd_soc_dapm_new_controls(dapm, wm8994_specific_dapm_widgets,
ARRAY_SIZE(wm8994_specific_dapm_widgets));
if (control->revision < 4) {
@@ -4284,8 +4291,10 @@ static int wm8994_component_probe(struct snd_soc_component *component)
}
break;
case WM8958:
+ snd_soc_add_component_controls(component, wm8994_snd_controls,
+ ARRAY_SIZE(wm8994_snd_controls));
snd_soc_add_component_controls(component, wm8958_snd_controls,
- ARRAY_SIZE(wm8958_snd_controls));
+ ARRAY_SIZE(wm8958_snd_controls));
snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets,
ARRAY_SIZE(wm8958_dapm_widgets));
if (control->revision < 1) {
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index ae28d9907c30..9b8bb7bbe945 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -1259,8 +1259,7 @@ static unsigned int wmfw_convert_flags(unsigned int in, unsigned int len)
}
if (in) {
- if (in & WMFW_CTL_FLAG_READABLE)
- out |= rd;
+ out |= rd;
if (in & WMFW_CTL_FLAG_WRITEABLE)
out |= wr;
if (in & WMFW_CTL_FLAG_VOLATILE)
@@ -3697,11 +3696,16 @@ static int wm_adsp_buffer_parse_legacy(struct wm_adsp *dsp)
u32 xmalg, addr, magic;
int i, ret;
+ alg_region = wm_adsp_find_alg_region(dsp, WMFW_ADSP2_XM, dsp->fw_id);
+ if (!alg_region) {
+ adsp_err(dsp, "No algorithm region found\n");
+ return -EINVAL;
+ }
+
buf = wm_adsp_buffer_alloc(dsp);
if (!buf)
return -ENOMEM;
- alg_region = wm_adsp_find_alg_region(dsp, WMFW_ADSP2_XM, dsp->fw_id);
xmalg = dsp->ops->sys_config_size / sizeof(__be32);
addr = alg_region->base + xmalg + ALG_XM_FIELD(magic);
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index a437567b8cee..4f6e58c3954a 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -308,6 +308,9 @@ static const struct snd_soc_dapm_widget sof_widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
SND_SOC_DAPM_SPK("Spk", NULL),
+};
+
+static const struct snd_soc_dapm_widget dmic_widgets[] = {
SND_SOC_DAPM_MIC("SoC DMIC", NULL),
};
@@ -318,10 +321,6 @@ static const struct snd_soc_dapm_route sof_map[] = {
/* other jacks */
{ "IN1P", NULL, "Headset Mic" },
-
- /* digital mics */
- {"DMic", NULL, "SoC DMIC"},
-
};
static const struct snd_soc_dapm_route speaker_map[] = {
@@ -329,6 +328,11 @@ static const struct snd_soc_dapm_route speaker_map[] = {
{ "Spk", NULL, "Speaker" },
};
+static const struct snd_soc_dapm_route dmic_map[] = {
+ /* digital mics */
+ {"DMic", NULL, "SoC DMIC"},
+};
+
static int speaker_codec_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_card *card = rtd->card;
@@ -342,6 +346,28 @@ static int speaker_codec_init(struct snd_soc_pcm_runtime *rtd)
return ret;
}
+static int dmic_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_card *card = rtd->card;
+ int ret;
+
+ ret = snd_soc_dapm_new_controls(&card->dapm, dmic_widgets,
+ ARRAY_SIZE(dmic_widgets));
+ if (ret) {
+ dev_err(card->dev, "DMic widget addition failed: %d\n", ret);
+ /* Don't need to add routes if widget addition failed */
+ return ret;
+ }
+
+ ret = snd_soc_dapm_add_routes(&card->dapm, dmic_map,
+ ARRAY_SIZE(dmic_map));
+
+ if (ret)
+ dev_err(card->dev, "DMic map addition failed: %d\n", ret);
+
+ return ret;
+}
+
/* sof audio machine driver for rt5682 codec */
static struct snd_soc_card sof_audio_card_rt5682 = {
.name = "sof_rt5682",
@@ -445,6 +471,7 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev,
links[id].name = "dmic01";
links[id].cpus = &cpus[id];
links[id].cpus->dai_name = "DMIC01 Pin";
+ links[id].init = dmic_init;
if (dmic_be_num > 1) {
/* set up 2 BE links at most */
links[id + 1].name = "dmic16k";
@@ -576,6 +603,15 @@ static int sof_audio_probe(struct platform_device *pdev)
/* need to get main clock from pmc */
if (sof_rt5682_quirk & SOF_RT5682_MCLK_BYTCHT_EN) {
ctx->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3");
+ if (IS_ERR(ctx->mclk)) {
+ ret = PTR_ERR(ctx->mclk);
+
+ dev_err(&pdev->dev,
+ "Failed to get MCLK from pmc_plt_clk_3: %d\n",
+ ret);
+ return ret;
+ }
+
ret = clk_prepare_enable(ctx->mclk);
if (ret < 0) {
dev_err(&pdev->dev,
@@ -621,8 +657,24 @@ static int sof_audio_probe(struct platform_device *pdev)
&sof_audio_card_rt5682);
}
+static int sof_rt5682_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+ struct snd_soc_component *component = NULL;
+
+ for_each_card_components(card, component) {
+ if (!strcmp(component->name, rt5682_component[0].name)) {
+ snd_soc_component_set_jack(component, NULL, NULL);
+ break;
+ }
+ }
+
+ return 0;
+}
+
static struct platform_driver sof_audio = {
.probe = sof_audio_probe,
+ .remove = sof_rt5682_remove,
.driver = {
.name = "sof_rt5682",
.pm = &snd_soc_pm_ops,
diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index af2d5a6124c8..61c984f10d8e 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -677,7 +677,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev)
ret = rockchip_pcm_platform_register(&pdev->dev);
if (ret) {
dev_err(&pdev->dev, "Could not register PCM\n");
- return ret;
+ goto err_suspend;
}
return 0;
diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c
index c213913eb984..fd8c6642fb0d 100644
--- a/sound/soc/samsung/arndale_rt5631.c
+++ b/sound/soc/samsung/arndale_rt5631.c
@@ -5,6 +5,7 @@
// Author: Claude <claude@insginal.co.kr>
#include <linux/module.h>
+#include <linux/of_device.h>
#include <linux/platform_device.h>
#include <linux/clk.h>
@@ -74,6 +75,17 @@ static struct snd_soc_card arndale_rt5631 = {
.num_links = ARRAY_SIZE(arndale_rt5631_dai),
};
+static void arndale_put_of_nodes(struct snd_soc_card *card)
+{
+ struct snd_soc_dai_link *dai_link;
+ int i;
+
+ for_each_card_prelinks(card, i, dai_link) {
+ of_node_put(dai_link->cpus->of_node);
+ of_node_put(dai_link->codecs->of_node);
+ }
+}
+
static int arndale_audio_probe(struct platform_device *pdev)
{
int n, ret;
@@ -103,18 +115,31 @@ static int arndale_audio_probe(struct platform_device *pdev)
if (!arndale_rt5631_dai[0].codecs->of_node) {
dev_err(&pdev->dev,
"Property 'samsung,audio-codec' missing or invalid\n");
- return -EINVAL;
+ ret = -EINVAL;
+ goto err_put_of_nodes;
}
}
ret = devm_snd_soc_register_card(card->dev, card);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret);
+ goto err_put_of_nodes;
+ }
+ return 0;
- if (ret)
- dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret);
-
+err_put_of_nodes:
+ arndale_put_of_nodes(card);
return ret;
}
+static int arndale_audio_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ arndale_put_of_nodes(card);
+ return 0;
+}
+
static const struct of_device_id samsung_arndale_rt5631_of_match[] __maybe_unused = {
{ .compatible = "samsung,arndale-rt5631", },
{ .compatible = "samsung,arndale-alc5631", },
@@ -129,6 +154,7 @@ static struct platform_driver arndale_audio_driver = {
.of_match_table = of_match_ptr(samsung_arndale_rt5631_of_match),
},
.probe = arndale_audio_probe,
+ .remove = arndale_audio_remove,
};
module_platform_driver(arndale_audio_driver);
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index bda5b958d0dc..e9596c2096cd 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -761,6 +761,7 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
}
/* set format */
+ rdai->bit_clk_inv = 0;
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
rdai->sys_delay = 0;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index e163dde5eab1..b600d3eaaf5c 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1070,7 +1070,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
return ret;
}
- snd_soc_dai_trigger(cpu_dai, substream, cmd);
+ ret = snd_soc_dai_trigger(cpu_dai, substream, cmd);
if (ret < 0)
return ret;
@@ -1097,7 +1097,7 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream,
return ret;
}
- snd_soc_dai_bespoke_trigger(cpu_dai, substream, cmd);
+ ret = snd_soc_dai_bespoke_trigger(cpu_dai, substream, cmd);
if (ret < 0)
return ret;
@@ -1146,6 +1146,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe,
{
struct snd_soc_dpcm *dpcm;
unsigned long flags;
+ char *name;
/* only add new dpcms */
for_each_dpcm_be(fe, stream, dpcm) {
@@ -1171,9 +1172,15 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe,
stream ? "<-" : "->", be->dai_link->name);
#ifdef CONFIG_DEBUG_FS
- dpcm->debugfs_state = debugfs_create_dir(be->dai_link->name,
- fe->debugfs_dpcm_root);
- debugfs_create_u32("state", 0644, dpcm->debugfs_state, &dpcm->state);
+ name = kasprintf(GFP_KERNEL, "%s:%s", be->dai_link->name,
+ stream ? "capture" : "playback");
+ if (name) {
+ dpcm->debugfs_state = debugfs_create_dir(name,
+ fe->debugfs_dpcm_root);
+ debugfs_create_u32("state", 0644, dpcm->debugfs_state,
+ &dpcm->state);
+ kfree(name);
+ }
#endif
return 1;
}
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index aa9a1fca46fa..0fd032914a31 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1582,7 +1582,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg,
/* map user to kernel widget ID */
template.id = get_widget_id(le32_to_cpu(w->id));
- if (template.id < 0)
+ if ((int)template.id < 0)
return template.id;
/* strings are allocated here, but used and freed by the widget */
diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c
index a4983f90ff5b..2b8711eda362 100644
--- a/sound/soc/sof/control.c
+++ b/sound/soc/sof/control.c
@@ -60,13 +60,16 @@ int snd_sof_volume_put(struct snd_kcontrol *kcontrol,
struct snd_sof_dev *sdev = scontrol->sdev;
struct sof_ipc_ctrl_data *cdata = scontrol->control_data;
unsigned int i, channels = scontrol->num_channels;
+ bool change = false;
+ u32 value;
/* update each channel */
for (i = 0; i < channels; i++) {
- cdata->chanv[i].value =
- mixer_to_ipc(ucontrol->value.integer.value[i],
+ value = mixer_to_ipc(ucontrol->value.integer.value[i],
scontrol->volume_table, sm->max + 1);
+ change = change || (value != cdata->chanv[i].value);
cdata->chanv[i].channel = i;
+ cdata->chanv[i].value = value;
}
/* notify DSP of mixer updates */
@@ -76,8 +79,7 @@ int snd_sof_volume_put(struct snd_kcontrol *kcontrol,
SOF_CTRL_TYPE_VALUE_CHAN_GET,
SOF_CTRL_CMD_VOLUME,
true);
-
- return 0;
+ return change;
}
int snd_sof_switch_get(struct snd_kcontrol *kcontrol,
@@ -105,11 +107,15 @@ int snd_sof_switch_put(struct snd_kcontrol *kcontrol,
struct snd_sof_dev *sdev = scontrol->sdev;
struct sof_ipc_ctrl_data *cdata = scontrol->control_data;
unsigned int i, channels = scontrol->num_channels;
+ bool change = false;
+ u32 value;
/* update each channel */
for (i = 0; i < channels; i++) {
- cdata->chanv[i].value = ucontrol->value.integer.value[i];
+ value = ucontrol->value.integer.value[i];
+ change = change || (value != cdata->chanv[i].value);
cdata->chanv[i].channel = i;
+ cdata->chanv[i].value = value;
}
/* notify DSP of mixer updates */
@@ -120,7 +126,7 @@ int snd_sof_switch_put(struct snd_kcontrol *kcontrol,
SOF_CTRL_CMD_SWITCH,
true);
- return 0;
+ return change;
}
int snd_sof_enum_get(struct snd_kcontrol *kcontrol,
@@ -148,11 +154,15 @@ int snd_sof_enum_put(struct snd_kcontrol *kcontrol,
struct snd_sof_dev *sdev = scontrol->sdev;
struct sof_ipc_ctrl_data *cdata = scontrol->control_data;
unsigned int i, channels = scontrol->num_channels;
+ bool change = false;
+ u32 value;
/* update each channel */
for (i = 0; i < channels; i++) {
- cdata->chanv[i].value = ucontrol->value.enumerated.item[i];
+ value = ucontrol->value.enumerated.item[i];
+ change = change || (value != cdata->chanv[i].value);
cdata->chanv[i].channel = i;
+ cdata->chanv[i].value = value;
}
/* notify DSP of enum updates */
@@ -163,7 +173,7 @@ int snd_sof_enum_put(struct snd_kcontrol *kcontrol,
SOF_CTRL_CMD_ENUM,
true);
- return 0;
+ return change;
}
int snd_sof_bytes_get(struct snd_kcontrol *kcontrol,
diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig
index 479ba249e219..d62f51d33be1 100644
--- a/sound/soc/sof/intel/Kconfig
+++ b/sound/soc/sof/intel/Kconfig
@@ -273,6 +273,16 @@ config SND_SOC_SOF_HDA_AUDIO_CODEC
Say Y if you want to enable HDAudio codecs with SOF.
If unsure select "N".
+config SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1
+ bool "SOF enable DMI Link L1"
+ help
+ This option enables DMI L1 for both playback and capture
+ and disables known workarounds for specific HDaudio platforms.
+ Only use to look into power optimizations on platforms not
+ affected by DMI L1 issues. This option is not recommended.
+ Say Y if you want to enable DMI Link L1
+ If unsure, select "N".
+
endif ## SND_SOC_SOF_HDA_COMMON
config SND_SOC_SOF_HDA_LINK_BASELINE
diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c
index e282179263e8..80e2826fb447 100644
--- a/sound/soc/sof/intel/bdw.c
+++ b/sound/soc/sof/intel/bdw.c
@@ -37,6 +37,7 @@
#define MBOX_SIZE 0x1000
#define MBOX_DUMP_SIZE 0x30
#define EXCEPT_OFFSET 0x800
+#define EXCEPT_MAX_HDR_SIZE 0x400
/* DSP peripherals */
#define DMAC0_OFFSET 0xFE000
@@ -228,6 +229,11 @@ static void bdw_get_registers(struct snd_sof_dev *sdev,
/* note: variable AR register array is not read */
/* then get panic info */
+ if (xoops->arch_hdr.totalsize > EXCEPT_MAX_HDR_SIZE) {
+ dev_err(sdev->dev, "invalid header size 0x%x. FW oops is bogus\n",
+ xoops->arch_hdr.totalsize);
+ return;
+ }
offset += xoops->arch_hdr.totalsize;
sof_mailbox_read(sdev, offset, panic_info, sizeof(*panic_info));
@@ -451,6 +457,7 @@ static int bdw_probe(struct snd_sof_dev *sdev)
/* TODO: add offsets */
sdev->mmio_bar = BDW_DSP_BAR;
sdev->mailbox_bar = BDW_DSP_BAR;
+ sdev->dsp_oops_offset = MBOX_OFFSET;
/* PCI base */
mmio = platform_get_resource(pdev, IORESOURCE_MEM,
diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c
index 5e7a6aaa627a..a1e514f71739 100644
--- a/sound/soc/sof/intel/byt.c
+++ b/sound/soc/sof/intel/byt.c
@@ -28,6 +28,7 @@
#define MBOX_OFFSET 0x144000
#define MBOX_SIZE 0x1000
#define EXCEPT_OFFSET 0x800
+#define EXCEPT_MAX_HDR_SIZE 0x400
/* DSP peripherals */
#define DMAC0_OFFSET 0x098000
@@ -126,6 +127,11 @@ static void byt_get_registers(struct snd_sof_dev *sdev,
/* note: variable AR register array is not read */
/* then get panic info */
+ if (xoops->arch_hdr.totalsize > EXCEPT_MAX_HDR_SIZE) {
+ dev_err(sdev->dev, "invalid header size 0x%x. FW oops is bogus\n",
+ xoops->arch_hdr.totalsize);
+ return;
+ }
offset += xoops->arch_hdr.totalsize;
sof_mailbox_read(sdev, offset, panic_info, sizeof(*panic_info));
diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c
index bc41028a7a01..df1909e1d950 100644
--- a/sound/soc/sof/intel/hda-ctrl.c
+++ b/sound/soc/sof/intel/hda-ctrl.c
@@ -139,20 +139,16 @@ void hda_dsp_ctrl_misc_clock_gating(struct snd_sof_dev *sdev, bool enable)
*/
int hda_dsp_ctrl_clock_power_gating(struct snd_sof_dev *sdev, bool enable)
{
-#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
- struct hdac_bus *bus = sof_to_bus(sdev);
-#endif
u32 val;
/* enable/disable audio dsp clock gating */
val = enable ? PCI_CGCTL_ADSPDCGE : 0;
snd_sof_pci_update_bits(sdev, PCI_CGCTL, PCI_CGCTL_ADSPDCGE, val);
-#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
- /* enable/disable L1 support */
- val = enable ? SOF_HDA_VS_EM2_L1SEN : 0;
- snd_hdac_chip_updatel(bus, VS_EM2, SOF_HDA_VS_EM2_L1SEN, val);
-#endif
+ /* enable/disable DMI Link L1 support */
+ val = enable ? HDA_VS_INTEL_EM2_L1SEN : 0;
+ snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, HDA_VS_INTEL_EM2,
+ HDA_VS_INTEL_EM2_L1SEN, val);
/* enable/disable audio dsp power gating */
val = enable ? 0 : PCI_PGCTL_ADSPPGD;
diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c
index 6427f0b3a2f1..65c2af3fcaab 100644
--- a/sound/soc/sof/intel/hda-loader.c
+++ b/sound/soc/sof/intel/hda-loader.c
@@ -44,6 +44,7 @@ static int cl_stream_prepare(struct snd_sof_dev *sdev, unsigned int format,
return -ENODEV;
}
hstream = &dsp_stream->hstream;
+ hstream->substream = NULL;
/* allocate DMA buffer */
ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, &pci->dev, size, dmab);
diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c
index ad8d41f22e92..2c7447188402 100644
--- a/sound/soc/sof/intel/hda-stream.c
+++ b/sound/soc/sof/intel/hda-stream.c
@@ -185,6 +185,17 @@ hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction)
direction == SNDRV_PCM_STREAM_PLAYBACK ?
"playback" : "capture");
+ /*
+ * Disable DMI Link L1 entry when capture stream is opened.
+ * Workaround to address a known issue with host DMA that results
+ * in xruns during pause/release in capture scenarios.
+ */
+ if (!IS_ENABLED(SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1))
+ if (stream && direction == SNDRV_PCM_STREAM_CAPTURE)
+ snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR,
+ HDA_VS_INTEL_EM2,
+ HDA_VS_INTEL_EM2_L1SEN, 0);
+
return stream;
}
@@ -193,23 +204,43 @@ int hda_dsp_stream_put(struct snd_sof_dev *sdev, int direction, int stream_tag)
{
struct hdac_bus *bus = sof_to_bus(sdev);
struct hdac_stream *s;
+ bool active_capture_stream = false;
+ bool found = false;
spin_lock_irq(&bus->reg_lock);
- /* find used stream */
+ /*
+ * close stream matching the stream tag
+ * and check if there are any open capture streams.
+ */
list_for_each_entry(s, &bus->stream_list, list) {
- if (s->direction == direction &&
- s->opened && s->stream_tag == stream_tag) {
+ if (!s->opened)
+ continue;
+
+ if (s->direction == direction && s->stream_tag == stream_tag) {
s->opened = false;
- spin_unlock_irq(&bus->reg_lock);
- return 0;
+ found = true;
+ } else if (s->direction == SNDRV_PCM_STREAM_CAPTURE) {
+ active_capture_stream = true;
}
}
spin_unlock_irq(&bus->reg_lock);
- dev_dbg(sdev->dev, "stream_tag %d not opened!\n", stream_tag);
- return -ENODEV;
+ /* Enable DMI L1 entry if there are no capture streams open */
+ if (!IS_ENABLED(SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1))
+ if (!active_capture_stream)
+ snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR,
+ HDA_VS_INTEL_EM2,
+ HDA_VS_INTEL_EM2_L1SEN,
+ HDA_VS_INTEL_EM2_L1SEN);
+
+ if (!found) {
+ dev_dbg(sdev->dev, "stream_tag %d not opened!\n", stream_tag);
+ return -ENODEV;
+ }
+
+ return 0;
}
int hda_dsp_stream_trigger(struct snd_sof_dev *sdev,
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index c72e9a09eee1..06e84679087b 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -35,6 +35,8 @@
#define IS_CFL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa348)
#define IS_CNL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9dc8)
+#define EXCEPT_MAX_HDR_SIZE 0x400
+
/*
* Debug
*/
@@ -131,6 +133,11 @@ static void hda_dsp_get_registers(struct snd_sof_dev *sdev,
/* note: variable AR register array is not read */
/* then get panic info */
+ if (xoops->arch_hdr.totalsize > EXCEPT_MAX_HDR_SIZE) {
+ dev_err(sdev->dev, "invalid header size 0x%x. FW oops is bogus\n",
+ xoops->arch_hdr.totalsize);
+ return;
+ }
offset += xoops->arch_hdr.totalsize;
sof_block_read(sdev, sdev->mmio_bar, offset,
panic_info, sizeof(*panic_info));
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 5591841a1b6f..23e430d3e056 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -39,7 +39,6 @@
#define SOF_HDA_WAKESTS 0x0E
#define SOF_HDA_WAKESTS_INT_MASK ((1 << 8) - 1)
#define SOF_HDA_RIRBSTS 0x5d
-#define SOF_HDA_VS_EM2_L1SEN BIT(13)
/* SOF_HDA_GCTL register bist */
#define SOF_HDA_GCTL_RESET BIT(0)
@@ -228,6 +227,10 @@
#define HDA_DSP_REG_HIPCIE (HDA_DSP_IPC_BASE + 0x0C)
#define HDA_DSP_REG_HIPCCTL (HDA_DSP_IPC_BASE + 0x10)
+/* Intel Vendor Specific Registers */
+#define HDA_VS_INTEL_EM2 0x1030
+#define HDA_VS_INTEL_EM2_L1SEN BIT(13)
+
/* HIPCI */
#define HDA_DSP_REG_HIPCI_BUSY BIT(31)
#define HDA_DSP_REG_HIPCI_MSG_MASK 0x7FFFFFFF
diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c
index d7f32745fefe..9a9a381a908d 100644
--- a/sound/soc/sof/loader.c
+++ b/sound/soc/sof/loader.c
@@ -546,10 +546,10 @@ int snd_sof_run_firmware(struct snd_sof_dev *sdev)
msecs_to_jiffies(sdev->boot_timeout));
if (ret == 0) {
dev_err(sdev->dev, "error: firmware boot failure\n");
- /* after this point FW_READY msg should be ignored */
- sdev->boot_complete = true;
snd_sof_dsp_dbg_dump(sdev, SOF_DBG_REGS | SOF_DBG_MBOX |
SOF_DBG_TEXT | SOF_DBG_PCI);
+ /* after this point FW_READY msg should be ignored */
+ sdev->boot_complete = true;
return -EIO;
}
diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c
index e3f6a6dc0f36..2b876d497447 100644
--- a/sound/soc/sof/pcm.c
+++ b/sound/soc/sof/pcm.c
@@ -244,7 +244,7 @@ static int sof_pcm_hw_free(struct snd_pcm_substream *substream)
snd_soc_rtdcom_lookup(rtd, DRV_NAME);
struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component);
struct snd_sof_pcm *spcm;
- int ret;
+ int ret, err = 0;
/* nothing to do for BE */
if (rtd->dai_link->no_pcm)
@@ -254,26 +254,26 @@ static int sof_pcm_hw_free(struct snd_pcm_substream *substream)
if (!spcm)
return -EINVAL;
- if (!spcm->prepared[substream->stream])
- return 0;
-
dev_dbg(sdev->dev, "pcm: free stream %d dir %d\n", spcm->pcm.pcm_id,
substream->stream);
- ret = sof_pcm_dsp_pcm_free(substream, sdev, spcm);
+ if (spcm->prepared[substream->stream]) {
+ ret = sof_pcm_dsp_pcm_free(substream, sdev, spcm);
+ if (ret < 0)
+ err = ret;
+ }
snd_pcm_lib_free_pages(substream);
cancel_work_sync(&spcm->stream[substream->stream].period_elapsed_work);
- if (ret < 0)
- return ret;
-
ret = snd_sof_pcm_platform_hw_free(sdev, substream);
- if (ret < 0)
+ if (ret < 0) {
dev_err(sdev->dev, "error: platform hw free failed\n");
+ err = ret;
+ }
- return ret;
+ return err;
}
static int sof_pcm_prepare(struct snd_pcm_substream *substream)
@@ -323,6 +323,7 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
struct sof_ipc_stream stream;
struct sof_ipc_reply reply;
bool reset_hw_params = false;
+ bool ipc_first = false;
int ret;
/* nothing to do for BE */
@@ -343,6 +344,7 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
switch (cmd) {
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_PAUSE;
+ ipc_first = true;
break;
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_RELEASE;
@@ -363,6 +365,7 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_STOP:
stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_STOP;
+ ipc_first = true;
reset_hw_params = true;
break;
default:
@@ -370,12 +373,22 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
return -EINVAL;
}
- snd_sof_pcm_platform_trigger(sdev, substream, cmd);
+ /*
+ * DMA and IPC sequence is different for start and stop. Need to send
+ * STOP IPC before stop DMA
+ */
+ if (!ipc_first)
+ snd_sof_pcm_platform_trigger(sdev, substream, cmd);
/* send IPC to the DSP */
ret = sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, &stream,
sizeof(stream), &reply, sizeof(reply));
+ /* need to STOP DMA even if STOP IPC failed */
+ if (ipc_first)
+ snd_sof_pcm_platform_trigger(sdev, substream, cmd);
+
+ /* free PCM if reset_hw_params is set and the STOP IPC is successful */
if (!ret && reset_hw_params)
ret = sof_pcm_dsp_pcm_free(substream, sdev, spcm);
diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c
index fc85efbad378..0aabb3190ddc 100644
--- a/sound/soc/sof/topology.c
+++ b/sound/soc/sof/topology.c
@@ -920,7 +920,9 @@ static void sof_parse_word_tokens(struct snd_soc_component *scomp,
for (j = 0; j < count; j++) {
/* match token type */
if (!(tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_WORD ||
- tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_SHORT))
+ tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_SHORT ||
+ tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_BYTE ||
+ tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_BOOL))
continue;
/* match token id */
diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c
index d7501f88aaa6..a4060813bc74 100644
--- a/sound/soc/stm/stm32_sai_sub.c
+++ b/sound/soc/stm/stm32_sai_sub.c
@@ -505,10 +505,20 @@ static int stm32_sai_set_sysclk(struct snd_soc_dai *cpu_dai,
if (dir == SND_SOC_CLOCK_OUT && sai->sai_mclk) {
ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX,
SAI_XCR1_NODIV,
- (unsigned int)~SAI_XCR1_NODIV);
+ freq ? 0 : SAI_XCR1_NODIV);
if (ret < 0)
return ret;
+ /* Assume shutdown if requested frequency is 0Hz */
+ if (!freq) {
+ /* Release mclk rate only if rate was actually set */
+ if (sai->mclk_rate) {
+ clk_rate_exclusive_put(sai->sai_mclk);
+ sai->mclk_rate = 0;
+ }
+ return 0;
+ }
+
/* If master clock is used, set parent clock now */
ret = stm32_sai_set_parent_clock(sai, freq);
if (ret)
@@ -1093,15 +1103,6 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream,
regmap_update_bits(sai->regmap, STM_SAI_IMR_REGX, SAI_XIMR_MASK, 0);
- regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_NODIV,
- SAI_XCR1_NODIV);
-
- /* Release mclk rate only if rate was actually set */
- if (sai->mclk_rate) {
- clk_rate_exclusive_put(sai->sai_mclk);
- sai->mclk_rate = 0;
- }
-
clk_disable_unprepare(sai->sai_ck);
spin_lock_irqsave(&sai->irq_lock, flags);
diff --git a/sound/usb/validate.c b/sound/usb/validate.c
index 3c8f73a0eb12..a5e584b60dcd 100644
--- a/sound/usb/validate.c
+++ b/sound/usb/validate.c
@@ -75,7 +75,7 @@ static bool validate_processing_unit(const void *p,
if (d->bLength < sizeof(*d))
return false;
- len = d->bLength < sizeof(*d) + d->bNrInPins;
+ len = sizeof(*d) + d->bNrInPins;
if (d->bLength < len)
return false;
switch (v->protocol) {