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authorMarcel Ziswiler <marcel.ziswiler@toradex.com>2019-12-18 22:52:20 +0100
committerMarcel Ziswiler <marcel.ziswiler@toradex.com>2019-12-18 22:52:20 +0100
commit1ddf624b0b268fdc0b80b1de618b98f8d117afea (patch)
tree3d3218332bcb34cb0afa01d6ad996058a3dbcb77 /sound/soc
parent6b774eec1f9d3064e9b33634dfa99d5666d0a73a (diff)
parentbfb9e5c03076a446b1f4f6a523ddc8d723c907a6 (diff)
Merge tag 'v4.14.159' into 4.14-2.0.x-imx
This is the 4.14.159 stable release Conflicts: arch/arm/Kconfig.debug arch/arm/boot/dts/imx7s.dtsi arch/arm/mach-imx/cpuidle-imx6sx.c drivers/crypto/caam/caamalg.c drivers/crypto/mxs-dcp.c drivers/dma/imx-sdma.c drivers/input/keyboard/imx_keypad.c drivers/net/can/flexcan.c drivers/net/can/rx-offload.c drivers/net/wireless/ath/ath10k/pci.c drivers/pci/dwc/pci-imx6.c drivers/spi/spi-fsl-lpspi.c drivers/usb/dwc3/gadget.c
Diffstat (limited to 'sound/soc')
-rw-r--r--sound/soc/codecs/cs4265.c2
-rw-r--r--sound/soc/codecs/cs42xx8.c1
-rw-r--r--sound/soc/codecs/es8316.c7
-rw-r--r--sound/soc/codecs/hdac_hdmi.c6
-rw-r--r--sound/soc/codecs/max98090.c16
-rw-r--r--sound/soc/codecs/msm8916-wcd-analog.c4
-rw-r--r--sound/soc/codecs/nau8540.c2
-rw-r--r--sound/soc/codecs/rt274.c3
-rw-r--r--sound/soc/codecs/sgtl5000.c247
-rw-r--r--sound/soc/codecs/tlv320aic31xx.c30
-rw-r--r--sound/soc/codecs/wm_adsp.c3
-rw-r--r--sound/soc/davinci/davinci-mcasp.c64
-rw-r--r--sound/soc/fsl/fsl_asrc.c4
-rw-r--r--sound/soc/fsl/fsl_ssi.c5
-rw-r--r--sound/soc/intel/common/sst-ipc.c2
-rw-r--r--sound/soc/intel/skylake/skl-debug.c2
-rw-r--r--sound/soc/intel/skylake/skl-nhlt.c2
-rw-r--r--sound/soc/kirkwood/kirkwood-i2s.c8
-rw-r--r--sound/soc/rockchip/rockchip_i2s.c2
-rw-r--r--sound/soc/sh/rcar/adg.c21
-rw-r--r--sound/soc/sh/rcar/core.c13
-rw-r--r--sound/soc/sh/rcar/rsnd.h1
-rw-r--r--sound/soc/sh/rcar/ssi.c4
-rw-r--r--sound/soc/soc-dapm.c26
-rw-r--r--sound/soc/soc-generic-dmaengine-pcm.c6
-rw-r--r--sound/soc/soc-jack.c3
-rw-r--r--sound/soc/soc-pcm.c5
-rw-r--r--sound/soc/stm/stm32_i2s.c29
-rw-r--r--sound/soc/sunxi/sun4i-i2s.c6
-rw-r--r--sound/soc/tegra/tegra_sgtl5000.c17
30 files changed, 421 insertions, 120 deletions
diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c
index 6e8eb1f5a041..bed64723e5d9 100644
--- a/sound/soc/codecs/cs4265.c
+++ b/sound/soc/codecs/cs4265.c
@@ -60,7 +60,7 @@ static const struct reg_default cs4265_reg_defaults[] = {
static bool cs4265_readable_register(struct device *dev, unsigned int reg)
{
switch (reg) {
- case CS4265_CHIP_ID ... CS4265_SPDIF_CTL2:
+ case CS4265_CHIP_ID ... CS4265_MAX_REGISTER:
return true;
default:
return false;
diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c
index 2e772427b48a..cedddee67199 100644
--- a/sound/soc/codecs/cs42xx8.c
+++ b/sound/soc/codecs/cs42xx8.c
@@ -668,6 +668,7 @@ static int cs42xx8_runtime_resume(struct device *dev)
CS42XX8_PWRCTL_PDN_MASK, 0);
regcache_cache_only(cs42xx8->regmap, false);
+ regcache_mark_dirty(cs42xx8->regmap);
ret = regcache_sync(cs42xx8->regmap);
if (ret) {
diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c
index da2d353af5ba..949dbdc0445e 100644
--- a/sound/soc/codecs/es8316.c
+++ b/sound/soc/codecs/es8316.c
@@ -46,7 +46,10 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9600, 50, 1);
static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_max_gain_tlv, -650, 150, 0);
static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_min_gain_tlv, -1200, 150, 0);
static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_target_tlv, -1650, 150, 0);
-static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(hpmixer_gain_tlv, -1200, 150, 0);
+static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpmixer_gain_tlv,
+ 0, 4, TLV_DB_SCALE_ITEM(-1200, 150, 0),
+ 8, 11, TLV_DB_SCALE_ITEM(-450, 150, 0),
+);
static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv,
0, 0, TLV_DB_SCALE_ITEM(-350, 0, 0),
@@ -84,7 +87,7 @@ static const struct snd_kcontrol_new es8316_snd_controls[] = {
SOC_DOUBLE_TLV("Headphone Playback Volume", ES8316_CPHP_ICAL_VOL,
4, 0, 3, 1, hpout_vol_tlv),
SOC_DOUBLE_TLV("Headphone Mixer Volume", ES8316_HPMIX_VOL,
- 0, 4, 7, 0, hpmixer_gain_tlv),
+ 0, 4, 11, 0, hpmixer_gain_tlv),
SOC_ENUM("Playback Polarity", dacpol),
SOC_DOUBLE_R_TLV("DAC Playback Volume", ES8316_DAC_VOLL,
diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index e824d47cc22b..1c3626347e12 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -1408,6 +1408,12 @@ static int hdac_hdmi_create_dais(struct hdac_device *hdac,
if (ret)
return ret;
+ /* Filter out 44.1, 88.2 and 176.4Khz */
+ rates &= ~(SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_176400);
+ if (!rates)
+ return -EINVAL;
+
sprintf(dai_name, "intel-hdmi-hifi%d", i+1);
hdmi_dais[i].name = devm_kstrdup(&hdac->dev,
dai_name, GFP_KERNEL);
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index cc66ea5cc776..3fe09828745a 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -1924,6 +1924,21 @@ static int max98090_configure_dmic(struct max98090_priv *max98090,
return 0;
}
+static int max98090_dai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ struct max98090_priv *max98090 = snd_soc_component_get_drvdata(component);
+ unsigned int fmt = max98090->dai_fmt;
+
+ /* Remove 24-bit format support if it is not in right justified mode. */
+ if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_RIGHT_J) {
+ substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE;
+ snd_pcm_hw_constraint_msbits(substream->runtime, 0, 16, 16);
+ }
+ return 0;
+}
+
static int max98090_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
@@ -2331,6 +2346,7 @@ EXPORT_SYMBOL_GPL(max98090_mic_detect);
#define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE)
static const struct snd_soc_dai_ops max98090_dai_ops = {
+ .startup = max98090_dai_startup,
.set_sysclk = max98090_dai_set_sysclk,
.set_fmt = max98090_dai_set_fmt,
.set_tdm_slot = max98090_set_tdm_slot,
diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c
index 0b9b014b4bb6..969283737787 100644
--- a/sound/soc/codecs/msm8916-wcd-analog.c
+++ b/sound/soc/codecs/msm8916-wcd-analog.c
@@ -303,7 +303,7 @@ struct pm8916_wcd_analog_priv {
};
static const char *const adc2_mux_text[] = { "ZERO", "INP2", "INP3" };
-static const char *const rdac2_mux_text[] = { "ZERO", "RX2", "RX1" };
+static const char *const rdac2_mux_text[] = { "RX1", "RX2" };
static const char *const hph_text[] = { "ZERO", "Switch", };
static const struct soc_enum hph_enum = SOC_ENUM_SINGLE_VIRT(
@@ -318,7 +318,7 @@ static const struct soc_enum adc2_enum = SOC_ENUM_SINGLE_VIRT(
/* RDAC2 MUX */
static const struct soc_enum rdac2_mux_enum = SOC_ENUM_SINGLE(
- CDC_D_CDC_CONN_HPHR_DAC_CTL, 0, 3, rdac2_mux_text);
+ CDC_D_CDC_CONN_HPHR_DAC_CTL, 0, 2, rdac2_mux_text);
static const struct snd_kcontrol_new spkr_switch[] = {
SOC_DAPM_SINGLE("Switch", CDC_A_SPKR_DAC_CTL, 7, 1, 0)
diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c
index f9c9933acffb..c0c64f90a61b 100644
--- a/sound/soc/codecs/nau8540.c
+++ b/sound/soc/codecs/nau8540.c
@@ -548,7 +548,7 @@ static int nau8540_calc_fll_param(unsigned int fll_in,
fvco_max = 0;
fvco_sel = ARRAY_SIZE(mclk_src_scaling);
for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) {
- fvco = 256 * fs * 2 * mclk_src_scaling[i].param;
+ fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param;
if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX &&
fvco_max < fvco) {
fvco_max = fvco;
diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c
index cd048df76232..43086ac9ffec 100644
--- a/sound/soc/codecs/rt274.c
+++ b/sound/soc/codecs/rt274.c
@@ -398,6 +398,8 @@ static int rt274_mic_detect(struct snd_soc_codec *codec,
{
struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec);
+ rt274->jack = jack;
+
if (jack == NULL) {
/* Disable jack detection */
regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL,
@@ -405,7 +407,6 @@ static int rt274_mic_detect(struct snd_soc_codec *codec,
return 0;
}
- rt274->jack = jack;
regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL,
RT274_IRQ_EN, RT274_IRQ_EN);
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index f3ffa31b5bca..ca8a70ab22a8 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -35,6 +35,13 @@
#define SGTL5000_DAP_REG_OFFSET 0x0100
#define SGTL5000_MAX_REG_OFFSET 0x013A
+/* Delay for the VAG ramp up */
+#define SGTL5000_VAG_POWERUP_DELAY 500 /* ms */
+/* Delay for the VAG ramp down */
+#define SGTL5000_VAG_POWERDOWN_DELAY 500 /* ms */
+
+#define SGTL5000_OUTPUTS_MUTE (SGTL5000_HP_MUTE | SGTL5000_LINE_OUT_MUTE)
+
/* default value of sgtl5000 registers */
static const struct reg_default sgtl5000_reg_defaults[] = {
{ SGTL5000_CHIP_DIG_POWER, 0x0000 },
@@ -120,6 +127,13 @@ enum {
I2S_LRCLK_STRENGTH_HIGH,
};
+enum {
+ HP_POWER_EVENT,
+ DAC_POWER_EVENT,
+ ADC_POWER_EVENT,
+ LAST_POWER_EVENT = ADC_POWER_EVENT
+};
+
/* sgtl5000 private structure in codec */
struct sgtl5000_priv {
int sysclk; /* sysclk rate */
@@ -133,8 +147,117 @@ struct sgtl5000_priv {
u8 micbias_resistor;
u8 micbias_voltage;
u8 lrclk_strength;
+ u16 mute_state[LAST_POWER_EVENT + 1];
};
+static inline int hp_sel_input(struct snd_soc_component *component)
+{
+ unsigned int ana_reg = 0;
+
+ snd_soc_component_read(component, SGTL5000_CHIP_ANA_CTRL, &ana_reg);
+
+ return (ana_reg & SGTL5000_HP_SEL_MASK) >> SGTL5000_HP_SEL_SHIFT;
+}
+
+static inline u16 mute_output(struct snd_soc_component *component,
+ u16 mute_mask)
+{
+ unsigned int mute_reg = 0;
+
+ snd_soc_component_read(component, SGTL5000_CHIP_ANA_CTRL, &mute_reg);
+
+ snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_CTRL,
+ mute_mask, mute_mask);
+ return mute_reg;
+}
+
+static inline void restore_output(struct snd_soc_component *component,
+ u16 mute_mask, u16 mute_reg)
+{
+ snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_CTRL,
+ mute_mask, mute_reg);
+}
+
+static void vag_power_on(struct snd_soc_component *component, u32 source)
+{
+ unsigned int ana_reg = 0;
+
+ snd_soc_component_read(component, SGTL5000_CHIP_ANA_POWER, &ana_reg);
+
+ if (ana_reg & SGTL5000_VAG_POWERUP)
+ return;
+
+ snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_POWER,
+ SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP);
+
+ /* When VAG powering on to get local loop from Line-In, the sleep
+ * is required to avoid loud pop.
+ */
+ if (hp_sel_input(component) == SGTL5000_HP_SEL_LINE_IN &&
+ source == HP_POWER_EVENT)
+ msleep(SGTL5000_VAG_POWERUP_DELAY);
+}
+
+static int vag_power_consumers(struct snd_soc_component *component,
+ u16 ana_pwr_reg, u32 source)
+{
+ int consumers = 0;
+
+ /* count dac/adc consumers unconditional */
+ if (ana_pwr_reg & SGTL5000_DAC_POWERUP)
+ consumers++;
+ if (ana_pwr_reg & SGTL5000_ADC_POWERUP)
+ consumers++;
+
+ /*
+ * If the event comes from HP and Line-In is selected,
+ * current action is 'DAC to be powered down'.
+ * As HP_POWERUP is not set when HP muxed to line-in,
+ * we need to keep VAG power ON.
+ */
+ if (source == HP_POWER_EVENT) {
+ if (hp_sel_input(component) == SGTL5000_HP_SEL_LINE_IN)
+ consumers++;
+ } else {
+ if (ana_pwr_reg & SGTL5000_HP_POWERUP)
+ consumers++;
+ }
+
+ return consumers;
+}
+
+static void vag_power_off(struct snd_soc_component *component, u32 source)
+{
+ unsigned int ana_pwr = SGTL5000_VAG_POWERUP;
+
+ snd_soc_component_read(component, SGTL5000_CHIP_ANA_POWER, &ana_pwr);
+
+ if (!(ana_pwr & SGTL5000_VAG_POWERUP))
+ return;
+
+ /*
+ * This function calls when any of VAG power consumers is disappearing.
+ * Thus, if there is more than one consumer at the moment, as minimum
+ * one consumer will definitely stay after the end of the current
+ * event.
+ * Don't clear VAG_POWERUP if 2 or more consumers of VAG present:
+ * - LINE_IN (for HP events) / HP (for DAC/ADC events)
+ * - DAC
+ * - ADC
+ * (the current consumer is disappearing right now)
+ */
+ if (vag_power_consumers(component, ana_pwr, source) >= 2)
+ return;
+
+ snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_POWER,
+ SGTL5000_VAG_POWERUP, 0);
+ /* In power down case, we need wait 400-1000 ms
+ * when VAG fully ramped down.
+ * As longer we wait, as smaller pop we've got.
+ */
+ msleep(SGTL5000_VAG_POWERDOWN_DELAY);
+}
+
/*
* mic_bias power on/off share the same register bits with
* output impedance of mic bias, when power on mic bias, we
@@ -166,36 +289,46 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w,
return 0;
}
-/*
- * As manual described, ADC/DAC only works when VAG powerup,
- * So enabled VAG before ADC/DAC up.
- * In power down case, we need wait 400ms when vag fully ramped down.
- */
-static int power_vag_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *kcontrol, int event)
+static int vag_and_mute_control(struct snd_soc_component *component,
+ int event, int event_source)
{
- struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
- const u32 mask = SGTL5000_DAC_POWERUP | SGTL5000_ADC_POWERUP;
+ static const u16 mute_mask[] = {
+ /*
+ * Mask for HP_POWER_EVENT.
+ * Muxing Headphones have to be wrapped with mute/unmute
+ * headphones only.
+ */
+ SGTL5000_HP_MUTE,
+ /*
+ * Masks for DAC_POWER_EVENT/ADC_POWER_EVENT.
+ * Muxing DAC or ADC block have to be wrapped with mute/unmute
+ * both headphones and line-out.
+ */
+ SGTL5000_OUTPUTS_MUTE,
+ SGTL5000_OUTPUTS_MUTE
+ };
+
+ struct sgtl5000_priv *sgtl5000 =
+ snd_soc_component_get_drvdata(component);
switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ sgtl5000->mute_state[event_source] =
+ mute_output(component, mute_mask[event_source]);
+ break;
case SND_SOC_DAPM_POST_PMU:
- snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
- SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP);
- msleep(400);
+ vag_power_on(component, event_source);
+ restore_output(component, mute_mask[event_source],
+ sgtl5000->mute_state[event_source]);
break;
-
case SND_SOC_DAPM_PRE_PMD:
- /*
- * Don't clear VAG_POWERUP, when both DAC and ADC are
- * operational to prevent inadvertently starving the
- * other one of them.
- */
- if ((snd_soc_read(codec, SGTL5000_CHIP_ANA_POWER) &
- mask) != mask) {
- snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
- SGTL5000_VAG_POWERUP, 0);
- msleep(400);
- }
+ sgtl5000->mute_state[event_source] =
+ mute_output(component, mute_mask[event_source]);
+ vag_power_off(component, event_source);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ restore_output(component, mute_mask[event_source],
+ sgtl5000->mute_state[event_source]);
break;
default:
break;
@@ -204,6 +337,41 @@ static int power_vag_event(struct snd_soc_dapm_widget *w,
return 0;
}
+/*
+ * Mute Headphone when power it up/down.
+ * Control VAG power on HP power path.
+ */
+static int headphone_pga_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_component *component =
+ snd_soc_dapm_to_component(w->dapm);
+
+ return vag_and_mute_control(component, event, HP_POWER_EVENT);
+}
+
+/* As manual describes, ADC/DAC powering up/down requires
+ * to mute outputs to avoid pops.
+ * Control VAG power on ADC/DAC power path.
+ */
+static int adc_updown_depop(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_component *component =
+ snd_soc_dapm_to_component(w->dapm);
+
+ return vag_and_mute_control(component, event, ADC_POWER_EVENT);
+}
+
+static int dac_updown_depop(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_component *component =
+ snd_soc_dapm_to_component(w->dapm);
+
+ return vag_and_mute_control(component, event, DAC_POWER_EVENT);
+}
+
/* input sources for ADC */
static const char *adc_mux_text[] = {
"MIC_IN", "LINE_IN"
@@ -239,7 +407,10 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = {
mic_bias_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
- SND_SOC_DAPM_PGA("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0),
+ SND_SOC_DAPM_PGA_E("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0,
+ headphone_pga_event,
+ SND_SOC_DAPM_PRE_POST_PMU |
+ SND_SOC_DAPM_PRE_POST_PMD),
SND_SOC_DAPM_PGA("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0),
SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux),
@@ -255,11 +426,12 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = {
0, SGTL5000_CHIP_DIG_POWER,
1, 0),
- SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0),
- SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0),
-
- SND_SOC_DAPM_PRE("VAG_POWER_PRE", power_vag_event),
- SND_SOC_DAPM_POST("VAG_POWER_POST", power_vag_event),
+ SND_SOC_DAPM_ADC_E("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0,
+ adc_updown_depop, SND_SOC_DAPM_PRE_POST_PMU |
+ SND_SOC_DAPM_PRE_POST_PMD),
+ SND_SOC_DAPM_DAC_E("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0,
+ dac_updown_depop, SND_SOC_DAPM_PRE_POST_PMU |
+ SND_SOC_DAPM_PRE_POST_PMD),
};
/* routes for sgtl5000 */
@@ -1084,12 +1256,17 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec)
SGTL5000_INT_OSC_EN);
/* Enable VDDC charge pump */
ana_pwr |= SGTL5000_VDDC_CHRGPMP_POWERUP;
- } else if (vddio >= 3100 && vdda >= 3100) {
+ } else {
ana_pwr &= ~SGTL5000_VDDC_CHRGPMP_POWERUP;
- /* VDDC use VDDIO rail */
- lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD;
- lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO <<
- SGTL5000_VDDC_MAN_ASSN_SHIFT;
+ /*
+ * if vddio == vdda the source of charge pump should be
+ * assigned manually to VDDIO
+ */
+ if (vddio == vdda) {
+ lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD;
+ lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO <<
+ SGTL5000_VDDC_MAN_ASSN_SHIFT;
+ }
}
snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, lreg_ctrl);
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index 54a87a905eb6..cc95c15ceceb 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -924,23 +924,31 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai,
return -EINVAL;
}
+ /* signal polarity */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface_reg2 |= AIC31XX_BCLKINV_MASK;
+ break;
+ default:
+ dev_err(codec->dev, "Invalid DAI clock signal polarity\n");
+ return -EINVAL;
+ }
+
/* interface format */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
break;
case SND_SOC_DAIFMT_DSP_A:
- dsp_a_val = 0x1;
+ dsp_a_val = 0x1; /* fall through */
case SND_SOC_DAIFMT_DSP_B:
- /* NOTE: BCLKINV bit value 1 equas NB and 0 equals IB */
- switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
- case SND_SOC_DAIFMT_NB_NF:
- iface_reg2 |= AIC31XX_BCLKINV_MASK;
- break;
- case SND_SOC_DAIFMT_IB_NF:
- break;
- default:
- return -EINVAL;
- }
+ /*
+ * NOTE: This CODEC samples on the falling edge of BCLK in
+ * DSP mode, this is inverted compared to what most DAIs
+ * expect, so we invert for this mode
+ */
+ iface_reg2 ^= AIC31XX_BCLKINV_MASK;
iface_reg1 |= (AIC31XX_DSP_MODE <<
AIC31XX_IFACE1_DATATYPE_SHIFT);
break;
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index d632a0511d62..158ce68bc9bf 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -1169,8 +1169,7 @@ 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)
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 9aa741d27279..07bac9ea65c4 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -1158,6 +1158,28 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream,
return ret;
}
+static int davinci_mcasp_hw_rule_slot_width(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct davinci_mcasp_ruledata *rd = rule->private;
+ struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+ struct snd_mask nfmt;
+ int i, slot_width;
+
+ snd_mask_none(&nfmt);
+ slot_width = rd->mcasp->slot_width;
+
+ for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
+ if (snd_mask_test(fmt, i)) {
+ if (snd_pcm_format_width(i) <= slot_width) {
+ snd_mask_set(&nfmt, i);
+ }
+ }
+ }
+
+ return snd_mask_refine(fmt, &nfmt);
+}
+
static const unsigned int davinci_mcasp_dai_rates[] = {
8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000,
88200, 96000, 176400, 192000,
@@ -1251,7 +1273,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
struct davinci_mcasp_ruledata *ruledata =
&mcasp->ruledata[substream->stream];
u32 max_channels = 0;
- int i, dir;
+ int i, dir, ret;
int tdm_slots = mcasp->tdm_slots;
/* Do not allow more then one stream per direction */
@@ -1280,6 +1302,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
max_channels++;
}
ruledata->serializers = max_channels;
+ ruledata->mcasp = mcasp;
max_channels *= tdm_slots;
/*
* If the already active stream has less channels than the calculated
@@ -1305,20 +1328,22 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
0, SNDRV_PCM_HW_PARAM_CHANNELS,
&mcasp->chconstr[substream->stream]);
- if (mcasp->slot_width)
- snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
- 8, mcasp->slot_width);
+ if (mcasp->slot_width) {
+ /* Only allow formats require <= slot_width bits on the bus */
+ ret = snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_FORMAT,
+ davinci_mcasp_hw_rule_slot_width,
+ ruledata,
+ SNDRV_PCM_HW_PARAM_FORMAT, -1);
+ if (ret)
+ return ret;
+ }
/*
* If we rely on implicit BCLK divider setting we should
* set constraints based on what we can provide.
*/
if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) {
- int ret;
-
- ruledata->mcasp = mcasp;
-
ret = snd_pcm_hw_rule_add(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
davinci_mcasp_hw_rule_rate,
@@ -1723,7 +1748,8 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp)
PTR_ERR(chan));
return PTR_ERR(chan);
}
- BUG_ON(!chan->device || !chan->device->dev);
+ if (WARN_ON(!chan->device || !chan->device->dev))
+ return -EINVAL;
if (chan->device->dev->of_node)
ret = of_property_read_string(chan->device->dev->of_node,
@@ -1869,6 +1895,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
if (irq >= 0) {
irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common",
dev_name(&pdev->dev));
+ if (!irq_name) {
+ ret = -ENOMEM;
+ goto err;
+ }
ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
davinci_mcasp_common_irq_handler,
IRQF_ONESHOT | IRQF_SHARED,
@@ -1886,6 +1916,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
if (irq >= 0) {
irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx",
dev_name(&pdev->dev));
+ if (!irq_name) {
+ ret = -ENOMEM;
+ goto err;
+ }
ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
davinci_mcasp_rx_irq_handler,
IRQF_ONESHOT, irq_name, mcasp);
@@ -1901,6 +1935,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
if (irq >= 0) {
irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx",
dev_name(&pdev->dev));
+ if (!irq_name) {
+ ret = -ENOMEM;
+ goto err;
+ }
ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
davinci_mcasp_tx_irq_handler,
IRQF_ONESHOT, irq_name, mcasp);
@@ -1984,8 +2022,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
GFP_KERNEL);
if (!mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list ||
- !mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list)
- return -ENOMEM;
+ !mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list) {
+ ret = -ENOMEM;
+ goto err;
+ }
ret = davinci_mcasp_set_ch_constraints(mcasp);
if (ret)
diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c
index 04e61aa6fc9b..f0d75177cde3 100644
--- a/sound/soc/fsl/fsl_asrc.c
+++ b/sound/soc/fsl/fsl_asrc.c
@@ -353,8 +353,8 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair, bool p2p_in, bool p2
return -EINVAL;
}
- if ((outrate > 8000 && outrate < 30000) &&
- (outrate/inrate > 24 || inrate/outrate > 8)) {
+ if ((outrate >= 8000 && outrate <= 30000) &&
+ (outrate > 24 * inrate || inrate > 8 * outrate)) {
pair_err("exceed supported ratio range [1/24, 8] for \
inrate/outrate: %d/%d\n", inrate, outrate);
return -EINVAL;
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 45e9de81cea9..1245db8451a1 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1459,6 +1459,7 @@ static int fsl_ssi_probe(struct platform_device *pdev)
struct fsl_ssi_private *ssi_private;
int ret = 0;
struct device_node *np = pdev->dev.of_node;
+ struct device_node *root;
const struct of_device_id *of_id;
const char *p, *sprop;
const uint32_t *iprop;
@@ -1648,7 +1649,9 @@ static int fsl_ssi_probe(struct platform_device *pdev)
* device tree. We also pass the address of the CPU DAI driver
* structure.
*/
- sprop = of_get_property(of_find_node_by_path("/"), "compatible", NULL);
+ root = of_find_node_by_path("/");
+ sprop = of_get_property(root, "compatible", NULL);
+ of_node_put(root);
/* Sometimes the compatible name has a "fsl," prefix, so we strip it. */
p = strrchr(sprop, ',');
if (p)
diff --git a/sound/soc/intel/common/sst-ipc.c b/sound/soc/intel/common/sst-ipc.c
index 62f3a8e0ec87..fedce78675e8 100644
--- a/sound/soc/intel/common/sst-ipc.c
+++ b/sound/soc/intel/common/sst-ipc.c
@@ -231,6 +231,8 @@ struct ipc_message *sst_ipc_reply_find_msg(struct sst_generic_ipc *ipc,
if (ipc->ops.reply_msg_match != NULL)
header = ipc->ops.reply_msg_match(header, &mask);
+ else
+ mask = (u64)-1;
if (list_empty(&ipc->rx_list)) {
dev_err(ipc->dev, "error: rx list empty but received 0x%llx\n",
diff --git a/sound/soc/intel/skylake/skl-debug.c b/sound/soc/intel/skylake/skl-debug.c
index dc20d91f62e6..1987f78ea91e 100644
--- a/sound/soc/intel/skylake/skl-debug.c
+++ b/sound/soc/intel/skylake/skl-debug.c
@@ -196,7 +196,7 @@ static ssize_t fw_softreg_read(struct file *file, char __user *user_buf,
memset(d->fw_read_buff, 0, FW_REG_BUF);
if (w0_stat_sz > 0)
- __iowrite32_copy(d->fw_read_buff, fw_reg_addr, w0_stat_sz >> 2);
+ __ioread32_copy(d->fw_read_buff, fw_reg_addr, w0_stat_sz >> 2);
for (offset = 0; offset < FW_REG_SIZE; offset += 16) {
ret += snprintf(tmp + ret, FW_REG_BUF - ret, "%#.4x: ", offset);
diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c
index 55859c5b456f..1b0129478a7f 100644
--- a/sound/soc/intel/skylake/skl-nhlt.c
+++ b/sound/soc/intel/skylake/skl-nhlt.c
@@ -215,7 +215,7 @@ int skl_nhlt_update_topology_bin(struct skl *skl)
struct hdac_bus *bus = ebus_to_hbus(&skl->ebus);
struct device *dev = bus->dev;
- dev_dbg(dev, "oem_id %.6s, oem_table_id %8s oem_revision %d\n",
+ dev_dbg(dev, "oem_id %.6s, oem_table_id %.8s oem_revision %d\n",
nhlt->header.oem_id, nhlt->header.oem_table_id,
nhlt->header.oem_revision);
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index 105a73cc5158..149b7cba10fb 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -569,10 +569,6 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
return PTR_ERR(priv->clk);
}
- err = clk_prepare_enable(priv->clk);
- if (err < 0)
- return err;
-
priv->extclk = devm_clk_get(&pdev->dev, "extclk");
if (IS_ERR(priv->extclk)) {
if (PTR_ERR(priv->extclk) == -EPROBE_DEFER)
@@ -588,6 +584,10 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
}
}
+ err = clk_prepare_enable(priv->clk);
+ if (err < 0)
+ return err;
+
/* Some sensible defaults - this reflects the powerup values */
priv->ctl_play = KIRKWOOD_PLAYCTL_SIZE_24;
priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24;
diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index 66fc13a2396a..0e07e3dea7de 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -676,7 +676,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev)
ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
if (ret) {
dev_err(&pdev->dev, "Could not register PCM\n");
- return ret;
+ goto err_suspend;
}
return 0;
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index eb7879bcc6a7..686401bcd1f5 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -33,6 +33,7 @@ struct rsnd_adg {
struct clk *clkout[CLKOUTMAX];
struct clk_onecell_data onecell;
struct rsnd_mod mod;
+ int clk_rate[CLKMAX];
u32 flags;
u32 ckr;
u32 rbga;
@@ -110,9 +111,9 @@ static void __rsnd_adg_get_timesel_ratio(struct rsnd_priv *priv,
unsigned int val, en;
unsigned int min, diff;
unsigned int sel_rate[] = {
- clk_get_rate(adg->clk[CLKA]), /* 0000: CLKA */
- clk_get_rate(adg->clk[CLKB]), /* 0001: CLKB */
- clk_get_rate(adg->clk[CLKC]), /* 0010: CLKC */
+ adg->clk_rate[CLKA], /* 0000: CLKA */
+ adg->clk_rate[CLKB], /* 0001: CLKB */
+ adg->clk_rate[CLKC], /* 0010: CLKC */
adg->rbga_rate_for_441khz, /* 0011: RBGA */
adg->rbgb_rate_for_48khz, /* 0100: RBGB */
};
@@ -328,7 +329,7 @@ int rsnd_adg_clk_query(struct rsnd_priv *priv, unsigned int rate)
* AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC/AUDIO_CLKI.
*/
for_each_rsnd_clk(clk, adg, i) {
- if (rate == clk_get_rate(clk))
+ if (rate == adg->clk_rate[i])
return sel_table[i];
}
@@ -394,10 +395,18 @@ void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable)
for_each_rsnd_clk(clk, adg, i) {
ret = 0;
- if (enable)
+ if (enable) {
ret = clk_prepare_enable(clk);
- else
+
+ /*
+ * We shouldn't use clk_get_rate() under
+ * atomic context. Let's keep it when
+ * rsnd_adg_clk_enable() was called
+ */
+ adg->clk_rate[i] = clk_get_rate(adg->clk[i]);
+ } else {
clk_disable_unprepare(clk);
+ }
if (ret < 0)
dev_warn(dev, "can't use clk %d\n", i);
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 710c01cd2ad2..f203c0878e69 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -676,6 +676,7 @@ 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;
@@ -1277,6 +1278,18 @@ int rsnd_kctrl_new(struct rsnd_mod *mod,
};
int ret;
+ /*
+ * 1) Avoid duplicate register for DVC with MIX case
+ * 2) Allow duplicate register for MIX
+ * 3) re-register if card was rebinded
+ */
+ list_for_each_entry(kctrl, &card->controls, list) {
+ struct rsnd_kctrl_cfg *c = kctrl->private_data;
+
+ if (c == cfg)
+ return 0;
+ }
+
if (size > RSND_MAX_CHANNELS)
return -EINVAL;
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 1768a0ae469d..c68b31483c7b 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -432,6 +432,7 @@ struct rsnd_dai_stream {
char name[RSND_DAI_NAME_SIZE];
struct snd_pcm_substream *substream;
struct rsnd_mod *mod[RSND_MOD_MAX];
+ struct rsnd_mod *dma;
struct rsnd_dai *rdai;
u32 parent_ssi_status;
};
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 60cc550c5a4c..cae9ed6a0cdb 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -66,7 +66,6 @@
struct rsnd_ssi {
struct rsnd_mod mod;
- struct rsnd_mod *dma;
u32 flags;
u32 cr_own;
@@ -868,7 +867,6 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod,
struct rsnd_dai_stream *io,
struct rsnd_priv *priv)
{
- struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
int ret;
/*
@@ -883,7 +881,7 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod,
return ret;
/* SSI probe might be called many times in MUX multi path */
- ret = rsnd_dma_attach(io, mod, &ssi->dma);
+ ret = rsnd_dma_attach(io, mod, &io->dma);
return ret;
}
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index e9f7c6287376..104d5f487c7d 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1152,8 +1152,8 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget,
list_add_tail(&widget->work_list, list);
if (custom_stop_condition && custom_stop_condition(widget, dir)) {
- widget->endpoints[dir] = 1;
- return widget->endpoints[dir];
+ list = NULL;
+ custom_stop_condition = NULL;
}
if ((widget->is_ep & SND_SOC_DAPM_DIR_TO_EP(dir)) && widget->connected) {
@@ -1190,8 +1190,8 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget,
*
* Optionally, can be supplied with a function acting as a stopping condition.
* This function takes the dapm widget currently being examined and the walk
- * direction as an arguments, it should return true if the walk should be
- * stopped and false otherwise.
+ * direction as an arguments, it should return true if widgets from that point
+ * in the graph onwards should not be added to the widget list.
*/
static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
struct list_head *list,
@@ -2120,23 +2120,25 @@ void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm,
{
struct dentry *d;
- if (!parent)
+ if (!parent || IS_ERR(parent))
return;
dapm->debugfs_dapm = debugfs_create_dir("dapm", parent);
- if (!dapm->debugfs_dapm) {
+ if (IS_ERR(dapm->debugfs_dapm)) {
dev_warn(dapm->dev,
- "ASoC: Failed to create DAPM debugfs directory\n");
+ "ASoC: Failed to create DAPM debugfs directory %ld\n",
+ PTR_ERR(dapm->debugfs_dapm));
return;
}
d = debugfs_create_file("bias_level", 0444,
dapm->debugfs_dapm, dapm,
&dapm_bias_fops);
- if (!d)
+ if (IS_ERR(d))
dev_warn(dapm->dev,
- "ASoC: Failed to create bias level debugfs file\n");
+ "ASoC: Failed to create bias level debugfs file: %ld\n",
+ PTR_ERR(d));
}
static void dapm_debugfs_add_widget(struct snd_soc_dapm_widget *w)
@@ -2150,10 +2152,10 @@ static void dapm_debugfs_add_widget(struct snd_soc_dapm_widget *w)
d = debugfs_create_file(w->name, 0444,
dapm->debugfs_dapm, w,
&dapm_widget_power_fops);
- if (!d)
+ if (IS_ERR(d))
dev_warn(w->dapm->dev,
- "ASoC: Failed to create %s debugfs file\n",
- w->name);
+ "ASoC: Failed to create %s debugfs file: %ld\n",
+ w->name, PTR_ERR(d));
}
static void dapm_debugfs_cleanup(struct snd_soc_dapm_context *dapm)
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index d53786498b61..052778c6afad 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -311,6 +311,12 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd)
if (!dmaengine_pcm_can_report_residue(dev, pcm->chan[i]))
pcm->flags |= SND_DMAENGINE_PCM_FLAG_NO_RESIDUE;
+
+ if (rtd->pcm->streams[i].pcm->name[0] == '\0') {
+ strncpy(rtd->pcm->streams[i].pcm->name,
+ rtd->pcm->streams[i].pcm->id,
+ sizeof(rtd->pcm->streams[i].pcm->name));
+ }
}
return 0;
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 99902ae1a2d9..b04ecc633da3 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -127,10 +127,9 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
unsigned int sync = 0;
int enable;
- trace_snd_soc_jack_report(jack, mask, status);
-
if (!jack)
return;
+ trace_snd_soc_jack_report(jack, mask, status);
dapm = &jack->card->dapm;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 6b290773250b..1bb3d0406c96 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1578,7 +1578,7 @@ static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime,
u64 formats)
{
runtime->hw.rate_min = stream->rate_min;
- runtime->hw.rate_max = stream->rate_max;
+ runtime->hw.rate_max = min_not_zero(stream->rate_max, UINT_MAX);
runtime->hw.channels_min = stream->channels_min;
runtime->hw.channels_max = stream->channels_max;
if (runtime->hw.formats)
@@ -2276,7 +2276,8 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream)
if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) &&
(be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) &&
- (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND))
+ (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND) &&
+ (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED))
continue;
dev_dbg(be->dev, "ASoC: prepare BE %s\n",
diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c
index 6d0bf78d114d..aa2b1196171a 100644
--- a/sound/soc/stm/stm32_i2s.c
+++ b/sound/soc/stm/stm32_i2s.c
@@ -246,8 +246,8 @@ static irqreturn_t stm32_i2s_isr(int irq, void *devid)
return IRQ_NONE;
}
- regmap_update_bits(i2s->regmap, STM32_I2S_IFCR_REG,
- I2S_IFCR_MASK, flags);
+ regmap_write_bits(i2s->regmap, STM32_I2S_IFCR_REG,
+ I2S_IFCR_MASK, flags);
if (flags & I2S_SR_OVR) {
dev_dbg(&pdev->dev, "Overrun\n");
@@ -276,7 +276,6 @@ static bool stm32_i2s_readable_reg(struct device *dev, unsigned int reg)
case STM32_I2S_CFG2_REG:
case STM32_I2S_IER_REG:
case STM32_I2S_SR_REG:
- case STM32_I2S_IFCR_REG:
case STM32_I2S_TXDR_REG:
case STM32_I2S_RXDR_REG:
case STM32_I2S_CGFR_REG:
@@ -488,7 +487,7 @@ static int stm32_i2s_configure(struct snd_soc_dai *cpu_dai,
{
struct stm32_i2s_data *i2s = snd_soc_dai_get_drvdata(cpu_dai);
int format = params_width(params);
- u32 cfgr, cfgr_mask, cfg1, cfg1_mask;
+ u32 cfgr, cfgr_mask, cfg1;
unsigned int fthlv;
int ret;
@@ -501,7 +500,7 @@ static int stm32_i2s_configure(struct snd_soc_dai *cpu_dai,
switch (format) {
case 16:
cfgr = I2S_CGFR_DATLEN_SET(I2S_I2SMOD_DATLEN_16);
- cfgr_mask = I2S_CGFR_DATLEN_MASK;
+ cfgr_mask = I2S_CGFR_DATLEN_MASK | I2S_CGFR_CHLEN;
break;
case 32:
cfgr = I2S_CGFR_DATLEN_SET(I2S_I2SMOD_DATLEN_32) |
@@ -529,15 +528,11 @@ static int stm32_i2s_configure(struct snd_soc_dai *cpu_dai,
if (ret < 0)
return ret;
- cfg1 = I2S_CFG1_RXDMAEN | I2S_CFG1_TXDMAEN;
- cfg1_mask = cfg1;
-
fthlv = STM32_I2S_FIFO_SIZE * I2S_FIFO_TH_ONE_QUARTER / 4;
- cfg1 |= I2S_CFG1_FTHVL_SET(fthlv - 1);
- cfg1_mask |= I2S_CFG1_FTHVL_MASK;
+ cfg1 = I2S_CFG1_FTHVL_SET(fthlv - 1);
return regmap_update_bits(i2s->regmap, STM32_I2S_CFG1_REG,
- cfg1_mask, cfg1);
+ I2S_CFG1_FTHVL_MASK, cfg1);
}
static int stm32_i2s_startup(struct snd_pcm_substream *substream,
@@ -551,8 +546,8 @@ static int stm32_i2s_startup(struct snd_pcm_substream *substream,
i2s->refcount++;
spin_unlock(&i2s->lock_fd);
- return regmap_update_bits(i2s->regmap, STM32_I2S_IFCR_REG,
- I2S_IFCR_MASK, I2S_IFCR_MASK);
+ return regmap_write_bits(i2s->regmap, STM32_I2S_IFCR_REG,
+ I2S_IFCR_MASK, I2S_IFCR_MASK);
}
static int stm32_i2s_hw_params(struct snd_pcm_substream *substream,
@@ -589,6 +584,10 @@ static int stm32_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
/* Enable i2s */
dev_dbg(cpu_dai->dev, "start I2S\n");
+ cfg1_mask = I2S_CFG1_RXDMAEN | I2S_CFG1_TXDMAEN;
+ regmap_update_bits(i2s->regmap, STM32_I2S_CFG1_REG,
+ cfg1_mask, cfg1_mask);
+
ret = regmap_update_bits(i2s->regmap, STM32_I2S_CR1_REG,
I2S_CR1_SPE, I2S_CR1_SPE);
if (ret < 0) {
@@ -603,8 +602,8 @@ static int stm32_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
return ret;
}
- regmap_update_bits(i2s->regmap, STM32_I2S_IFCR_REG,
- I2S_IFCR_MASK, I2S_IFCR_MASK);
+ regmap_write_bits(i2s->regmap, STM32_I2S_IFCR_REG,
+ I2S_IFCR_MASK, I2S_IFCR_MASK);
if (playback_flg) {
ier = I2S_IER_UDRIE;
diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index b4af5ce78ecb..da0a2083e12a 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -110,7 +110,7 @@
#define SUN8I_I2S_TX_CHAN_MAP_REG 0x44
#define SUN8I_I2S_TX_CHAN_SEL_REG 0x34
-#define SUN8I_I2S_TX_CHAN_OFFSET_MASK GENMASK(13, 11)
+#define SUN8I_I2S_TX_CHAN_OFFSET_MASK GENMASK(13, 12)
#define SUN8I_I2S_TX_CHAN_OFFSET(offset) (offset << 12)
#define SUN8I_I2S_TX_CHAN_EN_MASK GENMASK(11, 4)
#define SUN8I_I2S_TX_CHAN_EN(num_chan) (((1 << num_chan) - 1) << 4)
@@ -442,6 +442,10 @@ static int sun4i_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
regmap_update_bits(i2s->regmap, SUN8I_I2S_TX_CHAN_SEL_REG,
SUN8I_I2S_TX_CHAN_OFFSET_MASK,
SUN8I_I2S_TX_CHAN_OFFSET(offset));
+
+ regmap_update_bits(i2s->regmap, SUN8I_I2S_RX_CHAN_SEL_REG,
+ SUN8I_I2S_TX_CHAN_OFFSET_MASK,
+ SUN8I_I2S_TX_CHAN_OFFSET(offset));
}
regmap_field_write(i2s->field_fmt_mode, val);
diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c
index 45a4aa9d2a47..901457da25ec 100644
--- a/sound/soc/tegra/tegra_sgtl5000.c
+++ b/sound/soc/tegra/tegra_sgtl5000.c
@@ -149,14 +149,14 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev)
dev_err(&pdev->dev,
"Property 'nvidia,i2s-controller' missing/invalid\n");
ret = -EINVAL;
- goto err;
+ goto err_put_codec_of_node;
}
tegra_sgtl5000_dai.platform_of_node = tegra_sgtl5000_dai.cpu_of_node;
ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev);
if (ret)
- goto err;
+ goto err_put_cpu_of_node;
ret = snd_soc_register_card(card);
if (ret) {
@@ -169,6 +169,13 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev)
err_fini_utils:
tegra_asoc_utils_fini(&machine->util_data);
+err_put_cpu_of_node:
+ of_node_put(tegra_sgtl5000_dai.cpu_of_node);
+ tegra_sgtl5000_dai.cpu_of_node = NULL;
+ tegra_sgtl5000_dai.platform_of_node = NULL;
+err_put_codec_of_node:
+ of_node_put(tegra_sgtl5000_dai.codec_of_node);
+ tegra_sgtl5000_dai.codec_of_node = NULL;
err:
return ret;
}
@@ -183,6 +190,12 @@ static int tegra_sgtl5000_driver_remove(struct platform_device *pdev)
tegra_asoc_utils_fini(&machine->util_data);
+ of_node_put(tegra_sgtl5000_dai.cpu_of_node);
+ tegra_sgtl5000_dai.cpu_of_node = NULL;
+ tegra_sgtl5000_dai.platform_of_node = NULL;
+ of_node_put(tegra_sgtl5000_dai.codec_of_node);
+ tegra_sgtl5000_dai.codec_of_node = NULL;
+
return ret;
}