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authorTakashi Iwai <tiwai@suse.de>2016-02-12 09:48:51 +0100
committerTakashi Iwai <tiwai@suse.de>2016-02-12 09:48:51 +0100
commit86c2ee16704522a546c0ee1e8238096e3c391468 (patch)
treeeb8f363d8f89b4b434e651c44565a13f7dce2e34
parent397da2d0ab0d1398242622d80a74df7245395c63 (diff)
parentf11aec0d7c835c3b83e74e3fd60d2e48a850c857 (diff)
Merge tag 'asoc-fix-v4.5-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linussound-4.5-rc4
ASoC: Fixes for v4.5 A rather large batch of fixes here, almost all in the Intel driver. The changes that got merged in this merge window for Skylake were rather large and as well as issues that you'd expect in a large block of new code there were some problems created for older processors which needed fixing up. Things are largely settling down now hopefully.
-rw-r--r--Documentation/devicetree/bindings/sound/fsl-asoc-card.txt2
-rw-r--r--sound/soc/amd/acp-pcm-dma.c1
-rw-r--r--sound/soc/codecs/arizona.c43
-rw-r--r--sound/soc/codecs/rt286.c26
-rw-r--r--sound/soc/codecs/rt5645.c2
-rw-r--r--sound/soc/codecs/rt5659.c31
-rw-r--r--sound/soc/codecs/rt5659.h1
-rw-r--r--sound/soc/codecs/sigmadsp-i2c.c5
-rw-r--r--sound/soc/codecs/wm5110.c1
-rw-r--r--sound/soc/codecs/wm8960.c40
-rw-r--r--sound/soc/dwc/designware_i2s.c5
-rw-r--r--sound/soc/fsl/fsl_ssi.c42
-rw-r--r--sound/soc/fsl/imx-spdif.c2
-rw-r--r--sound/soc/generic/simple-card.c2
-rw-r--r--sound/soc/intel/Kconfig13
-rw-r--r--sound/soc/intel/atom/sst-mfld-platform-pcm.c1
-rw-r--r--sound/soc/intel/boards/skl_rt286.c5
-rw-r--r--sound/soc/intel/common/Makefile9
-rw-r--r--sound/soc/intel/common/sst-acpi.c4
-rw-r--r--sound/soc/intel/common/sst-match-acpi.c3
-rw-r--r--sound/soc/intel/skylake/skl-messages.c6
-rw-r--r--sound/soc/intel/skylake/skl-pcm.c1
-rw-r--r--sound/soc/intel/skylake/skl-topology.c75
-rw-r--r--sound/soc/intel/skylake/skl.c2
-rw-r--r--sound/soc/mediatek/Kconfig4
-rw-r--r--sound/soc/mxs/mxs-saif.c13
-rw-r--r--sound/soc/qcom/lpass-platform.c15
-rw-r--r--sound/soc/soc-dapm.c8
-rw-r--r--sound/soc/soc-pcm.c3
29 files changed, 221 insertions, 144 deletions
diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
index ce55c0a6f757..4da41bf1888e 100644
--- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
+++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
@@ -30,6 +30,8 @@ The compatible list for this generic sound card currently:
"fsl,imx-audio-sgtl5000"
(compatible with Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt)
+ "fsl,imx-audio-wm8960"
+
Required properties:
- compatible : Contains one of entries in the compatible list.
diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c
index 3191e0a7d273..d1fb035f44db 100644
--- a/sound/soc/amd/acp-pcm-dma.c
+++ b/sound/soc/amd/acp-pcm-dma.c
@@ -635,6 +635,7 @@ static int acp_dma_open(struct snd_pcm_substream *substream)
SNDRV_PCM_HW_PARAM_PERIODS);
if (ret < 0) {
dev_err(prtd->platform->dev, "set integer constraint failed\n");
+ kfree(adata);
return ret;
}
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 33143fe1de0b..91785318b283 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -1929,6 +1929,25 @@ static struct {
{ 1000000, 13500000, 0, 1 },
};
+static const unsigned int pseudo_fref_max[ARIZONA_FLL_MAX_FRATIO] = {
+ 13500000,
+ 6144000,
+ 6144000,
+ 3072000,
+ 3072000,
+ 2822400,
+ 2822400,
+ 1536000,
+ 1536000,
+ 1536000,
+ 1536000,
+ 1536000,
+ 1536000,
+ 1536000,
+ 1536000,
+ 768000,
+};
+
static struct {
unsigned int min;
unsigned int max;
@@ -2042,16 +2061,32 @@ static int arizona_calc_fratio(struct arizona_fll *fll,
/* Adjust FRATIO/refdiv to avoid integer mode if possible */
refdiv = cfg->refdiv;
+ arizona_fll_dbg(fll, "pseudo: initial ratio=%u fref=%u refdiv=%u\n",
+ init_ratio, Fref, refdiv);
+
while (div <= ARIZONA_FLL_MAX_REFDIV) {
for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO;
ratio++) {
if ((ARIZONA_FLL_VCO_CORNER / 2) /
- (fll->vco_mult * ratio) < Fref)
+ (fll->vco_mult * ratio) < Fref) {
+ arizona_fll_dbg(fll, "pseudo: hit VCO corner\n");
break;
+ }
+
+ if (Fref > pseudo_fref_max[ratio - 1]) {
+ arizona_fll_dbg(fll,
+ "pseudo: exceeded max fref(%u) for ratio=%u\n",
+ pseudo_fref_max[ratio - 1],
+ ratio);
+ break;
+ }
if (target % (ratio * Fref)) {
cfg->refdiv = refdiv;
cfg->fratio = ratio - 1;
+ arizona_fll_dbg(fll,
+ "pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n",
+ Fref, refdiv, div, ratio);
return ratio;
}
}
@@ -2060,6 +2095,9 @@ static int arizona_calc_fratio(struct arizona_fll *fll,
if (target % (ratio * Fref)) {
cfg->refdiv = refdiv;
cfg->fratio = ratio - 1;
+ arizona_fll_dbg(fll,
+ "pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n",
+ Fref, refdiv, div, ratio);
return ratio;
}
}
@@ -2068,6 +2106,9 @@ static int arizona_calc_fratio(struct arizona_fll *fll,
Fref /= 2;
refdiv++;
init_ratio = arizona_find_fratio(Fref, NULL);
+ arizona_fll_dbg(fll,
+ "pseudo: change fref=%u refdiv=%d(%d) ratio=%u\n",
+ Fref, refdiv, div, init_ratio);
}
arizona_fll_warn(fll, "Falling back to integer mode operation\n");
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index bc08f0c5a5f6..1bd31644a782 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -266,6 +266,8 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic)
} else {
*mic = false;
regmap_write(rt286->regmap, RT286_SET_MIC1, 0x20);
+ regmap_update_bits(rt286->regmap,
+ RT286_CBJ_CTRL1, 0x0400, 0x0000);
}
} else {
regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf);
@@ -470,24 +472,6 @@ static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w,
return 0;
}
-static int rt286_vref_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *kcontrol, int event)
-{
- struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
-
- switch (event) {
- case SND_SOC_DAPM_PRE_PMU:
- snd_soc_update_bits(codec,
- RT286_CBJ_CTRL1, 0x0400, 0x0000);
- mdelay(50);
- break;
- default:
- return 0;
- }
-
- return 0;
-}
-
static int rt286_ldo2_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
@@ -536,7 +520,7 @@ static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY_S("HV", 1, RT286_POWER_CTRL1,
12, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY("VREF", RT286_POWER_CTRL1,
- 0, 1, rt286_vref_event, SND_SOC_DAPM_PRE_PMU),
+ 0, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY_S("LDO1", 1, RT286_POWER_CTRL2,
2, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY_S("LDO2", 2, RT286_POWER_CTRL1,
@@ -911,8 +895,6 @@ static int rt286_set_bias_level(struct snd_soc_codec *codec,
case SND_SOC_BIAS_ON:
mdelay(10);
snd_soc_update_bits(codec,
- RT286_CBJ_CTRL1, 0x0400, 0x0400);
- snd_soc_update_bits(codec,
RT286_DC_GAIN, 0x200, 0x0);
break;
@@ -920,8 +902,6 @@ static int rt286_set_bias_level(struct snd_soc_codec *codec,
case SND_SOC_BIAS_STANDBY:
snd_soc_write(codec,
RT286_SET_AUDIO_POWER, AC_PWRST_D3);
- snd_soc_update_bits(codec,
- RT286_CBJ_CTRL1, 0x0400, 0x0000);
break;
default:
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index c61d38b585fb..93e8c9017633 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -776,7 +776,7 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = {
/* IN1/IN2 Control */
SOC_SINGLE_TLV("IN1 Boost", RT5645_IN1_CTRL1,
- RT5645_BST_SFT1, 8, 0, bst_tlv),
+ RT5645_BST_SFT1, 12, 0, bst_tlv),
SOC_SINGLE_TLV("IN2 Boost", RT5645_IN2_CTRL,
RT5645_BST_SFT2, 8, 0, bst_tlv),
diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c
index 820d8fa62b5e..fb8ea05c0de1 100644
--- a/sound/soc/codecs/rt5659.c
+++ b/sound/soc/codecs/rt5659.c
@@ -3985,7 +3985,6 @@ static int rt5659_i2c_probe(struct i2c_client *i2c,
if (rt5659 == NULL)
return -ENOMEM;
- rt5659->i2c = i2c;
i2c_set_clientdata(i2c, rt5659);
if (pdata)
@@ -4157,24 +4156,17 @@ static int rt5659_i2c_probe(struct i2c_client *i2c,
INIT_DELAYED_WORK(&rt5659->jack_detect_work, rt5659_jack_detect_work);
- if (rt5659->i2c->irq) {
- ret = request_threaded_irq(rt5659->i2c->irq, NULL, rt5659_irq,
- IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING
+ if (i2c->irq) {
+ ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL,
+ rt5659_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING
| IRQF_ONESHOT, "rt5659", rt5659);
if (ret)
dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret);
}
- ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5659,
+ return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5659,
rt5659_dai, ARRAY_SIZE(rt5659_dai));
-
- if (ret) {
- if (rt5659->i2c->irq)
- free_irq(rt5659->i2c->irq, rt5659);
- }
-
- return 0;
}
static int rt5659_i2c_remove(struct i2c_client *i2c)
@@ -4191,24 +4183,29 @@ void rt5659_i2c_shutdown(struct i2c_client *client)
regmap_write(rt5659->regmap, RT5659_RESET, 0);
}
+#ifdef CONFIG_OF
static const struct of_device_id rt5659_of_match[] = {
{ .compatible = "realtek,rt5658", },
{ .compatible = "realtek,rt5659", },
- {},
+ { },
};
+MODULE_DEVICE_TABLE(of, rt5659_of_match);
+#endif
+#ifdef CONFIG_ACPI
static struct acpi_device_id rt5659_acpi_match[] = {
- { "10EC5658", 0},
- { "10EC5659", 0},
- { },
+ { "10EC5658", 0, },
+ { "10EC5659", 0, },
+ { },
};
MODULE_DEVICE_TABLE(acpi, rt5659_acpi_match);
+#endif
struct i2c_driver rt5659_i2c_driver = {
.driver = {
.name = "rt5659",
.owner = THIS_MODULE,
- .of_match_table = rt5659_of_match,
+ .of_match_table = of_match_ptr(rt5659_of_match),
.acpi_match_table = ACPI_PTR(rt5659_acpi_match),
},
.probe = rt5659_i2c_probe,
diff --git a/sound/soc/codecs/rt5659.h b/sound/soc/codecs/rt5659.h
index 8f07ee903eaa..d31c9e5bcec8 100644
--- a/sound/soc/codecs/rt5659.h
+++ b/sound/soc/codecs/rt5659.h
@@ -1792,7 +1792,6 @@ struct rt5659_priv {
struct snd_soc_codec *codec;
struct rt5659_platform_data pdata;
struct regmap *regmap;
- struct i2c_client *i2c;
struct gpio_desc *gpiod_ldo1_en;
struct gpio_desc *gpiod_reset;
struct snd_soc_jack *hs_jack;
diff --git a/sound/soc/codecs/sigmadsp-i2c.c b/sound/soc/codecs/sigmadsp-i2c.c
index 21ca3a5e9f66..d374c18d4db7 100644
--- a/sound/soc/codecs/sigmadsp-i2c.c
+++ b/sound/soc/codecs/sigmadsp-i2c.c
@@ -31,7 +31,10 @@ static int sigmadsp_write_i2c(void *control_data,
kfree(buf);
- return ret;
+ if (ret < 0)
+ return ret;
+
+ return 0;
}
static int sigmadsp_read_i2c(void *control_data,
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 6088d30962a9..97c0f1e23886 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -2382,6 +2382,7 @@ error:
static int wm5110_remove(struct platform_device *pdev)
{
+ snd_soc_unregister_platform(&pdev->dev);
snd_soc_unregister_codec(&pdev->dev);
pm_runtime_disable(&pdev->dev);
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index ff237726775a..d7f444f87460 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -240,13 +240,13 @@ SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL,
SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL,
7, 1, 1),
-SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT3 Volume",
+SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT3 Volume",
WM8960_INBMIX1, 4, 7, 0, lineinboost_tlv),
-SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT2 Volume",
+SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT2 Volume",
WM8960_INBMIX1, 1, 7, 0, lineinboost_tlv),
-SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT3 Volume",
+SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT3 Volume",
WM8960_INBMIX2, 4, 7, 0, lineinboost_tlv),
-SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT2 Volume",
+SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT2 Volume",
WM8960_INBMIX2, 1, 7, 0, lineinboost_tlv),
SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT1 Volume",
WM8960_RINPATH, 4, 3, 0, micboost_tlv),
@@ -643,29 +643,31 @@ static int wm8960_configure_clocking(struct snd_soc_codec *codec)
return -EINVAL;
}
- /* check if the sysclk frequency is available. */
- for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) {
- if (sysclk_divs[i] == -1)
- continue;
- sysclk = freq_out / sysclk_divs[i];
- for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) {
- if (sysclk == dac_divs[j] * lrclk) {
+ if (wm8960->clk_id != WM8960_SYSCLK_PLL) {
+ /* check if the sysclk frequency is available. */
+ for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) {
+ if (sysclk_divs[i] == -1)
+ continue;
+ sysclk = freq_out / sysclk_divs[i];
+ for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) {
+ if (sysclk != dac_divs[j] * lrclk)
+ continue;
for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k)
if (sysclk == bclk * bclk_divs[k] / 10)
break;
if (k != ARRAY_SIZE(bclk_divs))
break;
}
+ if (j != ARRAY_SIZE(dac_divs))
+ break;
}
- if (j != ARRAY_SIZE(dac_divs))
- break;
- }
- if (i != ARRAY_SIZE(sysclk_divs)) {
- goto configure_clock;
- } else if (wm8960->clk_id != WM8960_SYSCLK_AUTO) {
- dev_err(codec->dev, "failed to configure clock\n");
- return -EINVAL;
+ if (i != ARRAY_SIZE(sysclk_divs)) {
+ goto configure_clock;
+ } else if (wm8960->clk_id != WM8960_SYSCLK_AUTO) {
+ dev_err(codec->dev, "failed to configure clock\n");
+ return -EINVAL;
+ }
}
/* get a available pll out frequency and set pll */
for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) {
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index ce664c239be3..bff258d7bcea 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -645,6 +645,8 @@ static int dw_i2s_probe(struct platform_device *pdev)
dev->dev = &pdev->dev;
+ dev->i2s_reg_comp1 = I2S_COMP_PARAM_1;
+ dev->i2s_reg_comp2 = I2S_COMP_PARAM_2;
if (pdata) {
dev->capability = pdata->cap;
clk_id = NULL;
@@ -652,9 +654,6 @@ static int dw_i2s_probe(struct platform_device *pdev)
if (dev->quirks & DW_I2S_QUIRK_COMP_REG_OFFSET) {
dev->i2s_reg_comp1 = pdata->i2s_reg_comp1;
dev->i2s_reg_comp2 = pdata->i2s_reg_comp2;
- } else {
- dev->i2s_reg_comp1 = I2S_COMP_PARAM_1;
- dev->i2s_reg_comp2 = I2S_COMP_PARAM_2;
}
ret = dw_configure_dai_by_pd(dev, dw_i2s_dai, res, pdata);
} else {
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 40dfd8a36484..ed8de1035cda 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -112,20 +112,6 @@ struct fsl_ssi_rxtx_reg_val {
struct fsl_ssi_reg_val tx;
};
-static const struct reg_default fsl_ssi_reg_defaults[] = {
- {CCSR_SSI_SCR, 0x00000000},
- {CCSR_SSI_SIER, 0x00003003},
- {CCSR_SSI_STCR, 0x00000200},
- {CCSR_SSI_SRCR, 0x00000200},
- {CCSR_SSI_STCCR, 0x00040000},
- {CCSR_SSI_SRCCR, 0x00040000},
- {CCSR_SSI_SACNT, 0x00000000},
- {CCSR_SSI_STMSK, 0x00000000},
- {CCSR_SSI_SRMSK, 0x00000000},
- {CCSR_SSI_SACCEN, 0x00000000},
- {CCSR_SSI_SACCDIS, 0x00000000},
-};
-
static bool fsl_ssi_readable_reg(struct device *dev, unsigned int reg)
{
switch (reg) {
@@ -190,8 +176,7 @@ static const struct regmap_config fsl_ssi_regconfig = {
.val_bits = 32,
.reg_stride = 4,
.val_format_endian = REGMAP_ENDIAN_NATIVE,
- .reg_defaults = fsl_ssi_reg_defaults,
- .num_reg_defaults = ARRAY_SIZE(fsl_ssi_reg_defaults),
+ .num_reg_defaults_raw = CCSR_SSI_SACCDIS / sizeof(uint32_t) + 1,
.readable_reg = fsl_ssi_readable_reg,
.volatile_reg = fsl_ssi_volatile_reg,
.precious_reg = fsl_ssi_precious_reg,
@@ -201,6 +186,7 @@ static const struct regmap_config fsl_ssi_regconfig = {
struct fsl_ssi_soc_data {
bool imx;
+ bool imx21regs; /* imx21-class SSI - no SACC{ST,EN,DIS} regs */
bool offline_config;
u32 sisr_write_mask;
};
@@ -303,6 +289,7 @@ static struct fsl_ssi_soc_data fsl_ssi_mpc8610 = {
static struct fsl_ssi_soc_data fsl_ssi_imx21 = {
.imx = true,
+ .imx21regs = true,
.offline_config = true,
.sisr_write_mask = 0,
};
@@ -586,8 +573,12 @@ static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private)
*/
regmap_write(regs, CCSR_SSI_SACNT,
CCSR_SSI_SACNT_AC97EN | CCSR_SSI_SACNT_FV);
- regmap_write(regs, CCSR_SSI_SACCDIS, 0xff);
- regmap_write(regs, CCSR_SSI_SACCEN, 0x300);
+
+ /* no SACC{ST,EN,DIS} regs on imx21-class SSI */
+ if (!ssi_private->soc->imx21regs) {
+ regmap_write(regs, CCSR_SSI_SACCDIS, 0xff);
+ regmap_write(regs, CCSR_SSI_SACCEN, 0x300);
+ }
/*
* Enable SSI, Transmit and Receive. AC97 has to communicate with the
@@ -1397,6 +1388,7 @@ static int fsl_ssi_probe(struct platform_device *pdev)
struct resource *res;
void __iomem *iomem;
char name[64];
+ struct regmap_config regconfig = fsl_ssi_regconfig;
of_id = of_match_device(fsl_ssi_ids, &pdev->dev);
if (!of_id || !of_id->data)
@@ -1444,15 +1436,25 @@ static int fsl_ssi_probe(struct platform_device *pdev)
return PTR_ERR(iomem);
ssi_private->ssi_phys = res->start;
+ if (ssi_private->soc->imx21regs) {
+ /*
+ * According to datasheet imx21-class SSI
+ * don't have SACC{ST,EN,DIS} regs.
+ */
+ regconfig.max_register = CCSR_SSI_SRMSK;
+ regconfig.num_reg_defaults_raw =
+ CCSR_SSI_SRMSK / sizeof(uint32_t) + 1;
+ }
+
ret = of_property_match_string(np, "clock-names", "ipg");
if (ret < 0) {
ssi_private->has_ipg_clk_name = false;
ssi_private->regs = devm_regmap_init_mmio(&pdev->dev, iomem,
- &fsl_ssi_regconfig);
+ &regconfig);
} else {
ssi_private->has_ipg_clk_name = true;
ssi_private->regs = devm_regmap_init_mmio_clk(&pdev->dev,
- "ipg", iomem, &fsl_ssi_regconfig);
+ "ipg", iomem, &regconfig);
}
if (IS_ERR(ssi_private->regs)) {
dev_err(&pdev->dev, "Failed to init register map\n");
diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c
index a407e833c612..fb896b2c9ba3 100644
--- a/sound/soc/fsl/imx-spdif.c
+++ b/sound/soc/fsl/imx-spdif.c
@@ -72,8 +72,6 @@ static int imx_spdif_audio_probe(struct platform_device *pdev)
goto end;
}
- platform_set_drvdata(pdev, data);
-
end:
of_node_put(spdif_np);
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 1ded8811598e..2389ab47e25f 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -99,7 +99,7 @@ static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream,
if (ret && ret != -ENOTSUPP)
goto err;
}
-
+ return 0;
err:
return ret;
}
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index 803f95e40679..7d7c872c280d 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -30,11 +30,15 @@ config SND_SST_IPC_ACPI
config SND_SOC_INTEL_SST
tristate
select SND_SOC_INTEL_SST_ACPI if ACPI
+ select SND_SOC_INTEL_SST_MATCH if ACPI
depends on (X86 || COMPILE_TEST)
config SND_SOC_INTEL_SST_ACPI
tristate
+config SND_SOC_INTEL_SST_MATCH
+ tristate
+
config SND_SOC_INTEL_HASWELL
tristate
@@ -57,7 +61,7 @@ config SND_SOC_INTEL_HASWELL_MACH
config SND_SOC_INTEL_BYT_RT5640_MACH
tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec"
depends on X86_INTEL_LPSS && I2C
- depends on DW_DMAC_CORE=y && (SND_SOC_INTEL_BYTCR_RT5640_MACH = n)
+ depends on DW_DMAC_CORE=y && (SND_SST_IPC_ACPI = n)
select SND_SOC_INTEL_SST
select SND_SOC_INTEL_BAYTRAIL
select SND_SOC_RT5640
@@ -69,7 +73,7 @@ config SND_SOC_INTEL_BYT_RT5640_MACH
config SND_SOC_INTEL_BYT_MAX98090_MACH
tristate "ASoC Audio driver for Intel Baytrail with MAX98090 codec"
depends on X86_INTEL_LPSS && I2C
- depends on DW_DMAC_CORE=y
+ depends on DW_DMAC_CORE=y && (SND_SST_IPC_ACPI = n)
select SND_SOC_INTEL_SST
select SND_SOC_INTEL_BAYTRAIL
select SND_SOC_MAX98090
@@ -97,6 +101,7 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH
select SND_SOC_RT5640
select SND_SST_MFLD_PLATFORM
select SND_SST_IPC_ACPI
+ select SND_SOC_INTEL_SST_MATCH if ACPI
help
This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR
platforms with RT5640 audio codec.
@@ -109,6 +114,7 @@ config SND_SOC_INTEL_BYTCR_RT5651_MACH
select SND_SOC_RT5651
select SND_SST_MFLD_PLATFORM
select SND_SST_IPC_ACPI
+ select SND_SOC_INTEL_SST_MATCH if ACPI
help
This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR
platforms with RT5651 audio codec.
@@ -121,6 +127,7 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH
select SND_SOC_RT5670
select SND_SST_MFLD_PLATFORM
select SND_SST_IPC_ACPI
+ select SND_SOC_INTEL_SST_MATCH if ACPI
help
This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell
platforms with RT5672 audio codec.
@@ -133,6 +140,7 @@ config SND_SOC_INTEL_CHT_BSW_RT5645_MACH
select SND_SOC_RT5645
select SND_SST_MFLD_PLATFORM
select SND_SST_IPC_ACPI
+ select SND_SOC_INTEL_SST_MATCH if ACPI
help
This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell
platforms with RT5645/5650 audio codec.
@@ -145,6 +153,7 @@ config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH
select SND_SOC_TS3A227E
select SND_SST_MFLD_PLATFORM
select SND_SST_IPC_ACPI
+ select SND_SOC_INTEL_SST_MATCH if ACPI
help
This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell
platforms with MAX98090 audio codec it also can support TI jack chip as aux device.
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index 55c33dc76ce4..52ed434cbca6 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -528,6 +528,7 @@ static struct snd_soc_dai_driver sst_platform_dai[] = {
.ops = &sst_compr_dai_ops,
.playback = {
.stream_name = "Compress Playback",
+ .channels_min = 1,
},
},
/* BE CPU Dais */
diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c
index 7396ddb427d8..2cbcbe412661 100644
--- a/sound/soc/intel/boards/skl_rt286.c
+++ b/sound/soc/intel/boards/skl_rt286.c
@@ -212,7 +212,10 @@ static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
{
struct snd_interval *channels = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
- channels->min = channels->max = 4;
+ if (params_channels(params) == 2)
+ channels->min = channels->max = 2;
+ else
+ channels->min = channels->max = 4;
return 0;
}
diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile
index 668fdeee195e..fbbb25c2ceed 100644
--- a/sound/soc/intel/common/Makefile
+++ b/sound/soc/intel/common/Makefile
@@ -1,13 +1,10 @@
snd-soc-sst-dsp-objs := sst-dsp.o
-ifneq ($(CONFIG_SND_SST_IPC_ACPI),)
-snd-soc-sst-acpi-objs := sst-match-acpi.o
-else
-snd-soc-sst-acpi-objs := sst-acpi.o sst-match-acpi.o
-endif
-
+snd-soc-sst-acpi-objs := sst-acpi.o
+snd-soc-sst-match-objs := sst-match-acpi.o
snd-soc-sst-ipc-objs := sst-ipc.o
snd-soc-sst-dsp-$(CONFIG_DW_DMAC_CORE) += sst-firmware.o
obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o
obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o
+obj-$(CONFIG_SND_SOC_INTEL_SST_MATCH) += snd-soc-sst-match.o
diff --git a/sound/soc/intel/common/sst-acpi.c b/sound/soc/intel/common/sst-acpi.c
index 7a85c576dad3..2c5eda14d510 100644
--- a/sound/soc/intel/common/sst-acpi.c
+++ b/sound/soc/intel/common/sst-acpi.c
@@ -215,6 +215,7 @@ static struct sst_acpi_desc sst_acpi_broadwell_desc = {
.dma_size = SST_LPT_DSP_DMA_SIZE,
};
+#if !IS_ENABLED(CONFIG_SND_SST_IPC_ACPI)
static struct sst_acpi_mach baytrail_machines[] = {
{ "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-48kHz_i2s_master", NULL, NULL, NULL },
{ "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-48kHz_i2s_master", NULL, NULL, NULL },
@@ -231,11 +232,14 @@ static struct sst_acpi_desc sst_acpi_baytrail_desc = {
.sst_id = SST_DEV_ID_BYT,
.resindex_dma_base = -1,
};
+#endif
static const struct acpi_device_id sst_acpi_match[] = {
{ "INT33C8", (unsigned long)&sst_acpi_haswell_desc },
{ "INT3438", (unsigned long)&sst_acpi_broadwell_desc },
+#if !IS_ENABLED(CONFIG_SND_SST_IPC_ACPI)
{ "80860F28", (unsigned long)&sst_acpi_baytrail_desc },
+#endif
{ }
};
MODULE_DEVICE_TABLE(acpi, sst_acpi_match);
diff --git a/sound/soc/intel/common/sst-match-acpi.c b/sound/soc/intel/common/sst-match-acpi.c
index dd077e116d25..3b4539d21492 100644
--- a/sound/soc/intel/common/sst-match-acpi.c
+++ b/sound/soc/intel/common/sst-match-acpi.c
@@ -41,3 +41,6 @@ struct sst_acpi_mach *sst_acpi_find_machine(struct sst_acpi_mach *machines)
return NULL;
}
EXPORT_SYMBOL_GPL(sst_acpi_find_machine);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Common ACPI Match module");
diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c
index de6dac496a0d..4629372d7c8e 100644
--- a/sound/soc/intel/skylake/skl-messages.c
+++ b/sound/soc/intel/skylake/skl-messages.c
@@ -688,14 +688,14 @@ int skl_unbind_modules(struct skl_sst *ctx,
/* get src queue index */
src_index = skl_get_queue_index(src_mcfg->m_out_pin, dst_id, out_max);
if (src_index < 0)
- return -EINVAL;
+ return 0;
msg.src_queue = src_index;
/* get dst queue index */
dst_index = skl_get_queue_index(dst_mcfg->m_in_pin, src_id, in_max);
if (dst_index < 0)
- return -EINVAL;
+ return 0;
msg.dst_queue = dst_index;
@@ -747,7 +747,7 @@ int skl_bind_modules(struct skl_sst *ctx,
skl_dump_bind_info(ctx, src_mcfg, dst_mcfg);
- if (src_mcfg->m_state < SKL_MODULE_INIT_DONE &&
+ if (src_mcfg->m_state < SKL_MODULE_INIT_DONE ||
dst_mcfg->m_state < SKL_MODULE_INIT_DONE)
return 0;
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index f3553258091a..b6e6b61d10ec 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -863,6 +863,7 @@ static int skl_get_delay_from_lpib(struct hdac_ext_bus *ebus,
else
delay += hstream->bufsize;
}
+ delay = (hstream->bufsize == delay) ? 0 : delay;
if (delay >= hstream->period_bytes) {
dev_info(bus->dev,
diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index 4624556f486d..a294fee431f0 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -54,12 +54,9 @@ static int is_skl_dsp_widget_type(struct snd_soc_dapm_widget *w)
/*
* Each pipelines needs memory to be allocated. Check if we have free memory
- * from available pool. Then only add this to pool
- * This is freed when pipe is deleted
- * Note: DSP does actual memory management we only keep track for complete
- * pool
+ * from available pool.
*/
-static bool skl_tplg_alloc_pipe_mem(struct skl *skl,
+static bool skl_is_pipe_mem_avail(struct skl *skl,
struct skl_module_cfg *mconfig)
{
struct skl_sst *ctx = skl->skl_sst;
@@ -74,10 +71,20 @@ static bool skl_tplg_alloc_pipe_mem(struct skl *skl,
"exceeds ppl memory available %d mem %d\n",
skl->resource.max_mem, skl->resource.mem);
return false;
+ } else {
+ return true;
}
+}
+/*
+ * Add the mem to the mem pool. This is freed when pipe is deleted.
+ * Note: DSP does actual memory management we only keep track for complete
+ * pool
+ */
+static void skl_tplg_alloc_pipe_mem(struct skl *skl,
+ struct skl_module_cfg *mconfig)
+{
skl->resource.mem += mconfig->pipe->memory_pages;
- return true;
}
/*
@@ -85,10 +92,10 @@ static bool skl_tplg_alloc_pipe_mem(struct skl *skl,
* quantified in MCPS (Million Clocks Per Second) required for module/pipe
*
* Each pipelines needs mcps to be allocated. Check if we have mcps for this
- * pipe. This adds the mcps to driver counter
- * This is removed on pipeline delete
+ * pipe.
*/
-static bool skl_tplg_alloc_pipe_mcps(struct skl *skl,
+
+static bool skl_is_pipe_mcps_avail(struct skl *skl,
struct skl_module_cfg *mconfig)
{
struct skl_sst *ctx = skl->skl_sst;
@@ -98,13 +105,18 @@ static bool skl_tplg_alloc_pipe_mcps(struct skl *skl,
"%s: module_id %d instance %d\n", __func__,
mconfig->id.module_id, mconfig->id.instance_id);
dev_err(ctx->dev,
- "exceeds ppl memory available %d > mem %d\n",
+ "exceeds ppl mcps available %d > mem %d\n",
skl->resource.max_mcps, skl->resource.mcps);
return false;
+ } else {
+ return true;
}
+}
+static void skl_tplg_alloc_pipe_mcps(struct skl *skl,
+ struct skl_module_cfg *mconfig)
+{
skl->resource.mcps += mconfig->mcps;
- return true;
}
/*
@@ -411,7 +423,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe)
mconfig = w->priv;
/* check resource available */
- if (!skl_tplg_alloc_pipe_mcps(skl, mconfig))
+ if (!skl_is_pipe_mcps_avail(skl, mconfig))
return -ENOMEM;
if (mconfig->is_loadable && ctx->dsp->fw_ops.load_mod) {
@@ -435,6 +447,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe)
ret = skl_tplg_set_module_params(w, ctx);
if (ret < 0)
return ret;
+ skl_tplg_alloc_pipe_mcps(skl, mconfig);
}
return 0;
@@ -477,10 +490,10 @@ static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w,
struct skl_sst *ctx = skl->skl_sst;
/* check resource available */
- if (!skl_tplg_alloc_pipe_mcps(skl, mconfig))
+ if (!skl_is_pipe_mcps_avail(skl, mconfig))
return -EBUSY;
- if (!skl_tplg_alloc_pipe_mem(skl, mconfig))
+ if (!skl_is_pipe_mem_avail(skl, mconfig))
return -ENOMEM;
/*
@@ -526,11 +539,15 @@ static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w,
src_module = dst_module;
}
+ skl_tplg_alloc_pipe_mem(skl, mconfig);
+ skl_tplg_alloc_pipe_mcps(skl, mconfig);
+
return 0;
}
static int skl_tplg_bind_sinks(struct snd_soc_dapm_widget *w,
struct skl *skl,
+ struct snd_soc_dapm_widget *src_w,
struct skl_module_cfg *src_mconfig)
{
struct snd_soc_dapm_path *p;
@@ -547,6 +564,10 @@ static int skl_tplg_bind_sinks(struct snd_soc_dapm_widget *w,
dev_dbg(ctx->dev, "%s: sink widget=%s\n", __func__, p->sink->name);
next_sink = p->sink;
+
+ if (!is_skl_dsp_widget_type(p->sink))
+ return skl_tplg_bind_sinks(p->sink, skl, src_w, src_mconfig);
+
/*
* here we will check widgets in sink pipelines, so that
* can be any widgets type and we are only interested if
@@ -576,7 +597,7 @@ static int skl_tplg_bind_sinks(struct snd_soc_dapm_widget *w,
}
if (!sink)
- return skl_tplg_bind_sinks(next_sink, skl, src_mconfig);
+ return skl_tplg_bind_sinks(next_sink, skl, src_w, src_mconfig);
return 0;
}
@@ -605,7 +626,7 @@ static int skl_tplg_pga_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w,
* if sink is not started, start sink pipe first, then start
* this pipe
*/
- ret = skl_tplg_bind_sinks(w, skl, src_mconfig);
+ ret = skl_tplg_bind_sinks(w, skl, w, src_mconfig);
if (ret)
return ret;
@@ -773,10 +794,7 @@ static int skl_tplg_mixer_dapm_post_pmd_event(struct snd_soc_dapm_widget *w,
continue;
}
- ret = skl_unbind_modules(ctx, src_module, dst_module);
- if (ret < 0)
- return ret;
-
+ skl_unbind_modules(ctx, src_module, dst_module);
src_module = dst_module;
}
@@ -814,9 +832,6 @@ static int skl_tplg_pga_dapm_post_pmd_event(struct snd_soc_dapm_widget *w,
* This is a connecter and if path is found that means
* unbind between source and sink has not happened yet
*/
- ret = skl_stop_pipe(ctx, sink_mconfig->pipe);
- if (ret < 0)
- return ret;
ret = skl_unbind_modules(ctx, src_mconfig,
sink_mconfig);
}
@@ -842,6 +857,12 @@ static int skl_tplg_vmixer_event(struct snd_soc_dapm_widget *w,
case SND_SOC_DAPM_PRE_PMU:
return skl_tplg_mixer_dapm_pre_pmu_event(w, skl);
+ case SND_SOC_DAPM_POST_PMU:
+ return skl_tplg_mixer_dapm_post_pmu_event(w, skl);
+
+ case SND_SOC_DAPM_PRE_PMD:
+ return skl_tplg_mixer_dapm_pre_pmd_event(w, skl);
+
case SND_SOC_DAPM_POST_PMD:
return skl_tplg_mixer_dapm_post_pmd_event(w, skl);
}
@@ -916,6 +937,13 @@ static int skl_tplg_tlv_control_get(struct snd_kcontrol *kcontrol,
skl_get_module_params(skl->skl_sst, (u32 *)bc->params,
bc->max, bc->param_id, mconfig);
+ /* decrement size for TLV header */
+ size -= 2 * sizeof(u32);
+
+ /* check size as we don't want to send kernel data */
+ if (size > bc->max)
+ size = bc->max;
+
if (bc->params) {
if (copy_to_user(data, &bc->param_id, sizeof(u32)))
return -EFAULT;
@@ -1510,6 +1538,7 @@ int skl_tplg_init(struct snd_soc_platform *platform, struct hdac_ext_bus *ebus)
&skl_tplg_ops, fw, 0);
if (ret < 0) {
dev_err(bus->dev, "tplg component load failed%d\n", ret);
+ release_firmware(fw);
return -EINVAL;
}
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index 443a15de94b5..092705e73db4 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -614,8 +614,6 @@ static int skl_probe(struct pci_dev *pci,
goto out_unregister;
/*configure PM */
- pm_runtime_set_autosuspend_delay(bus->dev, SKL_SUSPEND_DELAY);
- pm_runtime_use_autosuspend(bus->dev);
pm_runtime_put_noidle(bus->dev);
pm_runtime_allow(bus->dev);
diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig
index 15c04e2eae34..976967675387 100644
--- a/sound/soc/mediatek/Kconfig
+++ b/sound/soc/mediatek/Kconfig
@@ -9,7 +9,7 @@ config SND_SOC_MEDIATEK
config SND_SOC_MT8173_MAX98090
tristate "ASoC Audio driver for MT8173 with MAX98090 codec"
- depends on SND_SOC_MEDIATEK
+ depends on SND_SOC_MEDIATEK && I2C
select SND_SOC_MAX98090
help
This adds ASoC driver for Mediatek MT8173 boards
@@ -19,7 +19,7 @@ config SND_SOC_MT8173_MAX98090
config SND_SOC_MT8173_RT5650_RT5676
tristate "ASoC Audio driver for MT8173 with RT5650 RT5676 codecs"
- depends on SND_SOC_MEDIATEK
+ depends on SND_SOC_MEDIATEK && I2C
select SND_SOC_RT5645
select SND_SOC_RT5677
help
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index c866ade28ad0..a6c7b8d87cd2 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -381,9 +381,19 @@ static int mxs_saif_startup(struct snd_pcm_substream *substream,
__raw_writel(BM_SAIF_CTRL_CLKGATE,
saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+ clk_prepare(saif->clk);
+
return 0;
}
+static void mxs_saif_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+
+ clk_unprepare(saif->clk);
+}
+
/*
* Should only be called when port is inactive.
* although can be called multiple times by upper layers.
@@ -424,8 +434,6 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream,
return ret;
}
- /* prepare clk in hw_param, enable in trigger */
- clk_prepare(saif->clk);
if (saif != master_saif) {
/*
* Set an initial clock rate for the saif internal logic to work
@@ -611,6 +619,7 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd,
static const struct snd_soc_dai_ops mxs_saif_dai_ops = {
.startup = mxs_saif_startup,
+ .shutdown = mxs_saif_shutdown,
.trigger = mxs_saif_trigger,
.prepare = mxs_saif_prepare,
.hw_params = mxs_saif_hw_params,
diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c
index 79688aa1941a..4aeb8e1a7160 100644
--- a/sound/soc/qcom/lpass-platform.c
+++ b/sound/soc/qcom/lpass-platform.c
@@ -440,18 +440,18 @@ static irqreturn_t lpass_platform_lpaif_irq(int irq, void *data)
}
static int lpass_platform_alloc_buffer(struct snd_pcm_substream *substream,
- struct snd_soc_pcm_runtime *soc_runtime)
+ struct snd_soc_pcm_runtime *rt)
{
struct snd_dma_buffer *buf = &substream->dma_buffer;
size_t size = lpass_platform_pcm_hardware.buffer_bytes_max;
buf->dev.type = SNDRV_DMA_TYPE_DEV;
- buf->dev.dev = soc_runtime->dev;
+ buf->dev.dev = rt->platform->dev;
buf->private_data = NULL;
- buf->area = dma_alloc_coherent(soc_runtime->dev, size, &buf->addr,
+ buf->area = dma_alloc_coherent(rt->platform->dev, size, &buf->addr,
GFP_KERNEL);
if (!buf->area) {
- dev_err(soc_runtime->dev, "%s: Could not allocate DMA buffer\n",
+ dev_err(rt->platform->dev, "%s: Could not allocate DMA buffer\n",
__func__);
return -ENOMEM;
}
@@ -461,12 +461,12 @@ static int lpass_platform_alloc_buffer(struct snd_pcm_substream *substream,
}
static void lpass_platform_free_buffer(struct snd_pcm_substream *substream,
- struct snd_soc_pcm_runtime *soc_runtime)
+ struct snd_soc_pcm_runtime *rt)
{
struct snd_dma_buffer *buf = &substream->dma_buffer;
if (buf->area) {
- dma_free_coherent(soc_runtime->dev, buf->bytes, buf->area,
+ dma_free_coherent(rt->dev, buf->bytes, buf->area,
buf->addr);
}
buf->area = NULL;
@@ -499,9 +499,6 @@ static int lpass_platform_pcm_new(struct snd_soc_pcm_runtime *soc_runtime)
snd_soc_pcm_set_drvdata(soc_runtime, data);
- soc_runtime->dev->coherent_dma_mask = DMA_BIT_MASK(32);
- soc_runtime->dev->dma_mask = &soc_runtime->dev->coherent_dma_mask;
-
ret = lpass_platform_alloc_buffer(substream, soc_runtime);
if (ret)
return ret;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 5a2812fa8946..0d3707987900 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -310,7 +310,7 @@ struct dapm_kcontrol_data {
};
static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
- struct snd_kcontrol *kcontrol)
+ struct snd_kcontrol *kcontrol, const char *ctrl_name)
{
struct dapm_kcontrol_data *data;
struct soc_mixer_control *mc;
@@ -333,7 +333,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
if (mc->autodisable) {
struct snd_soc_dapm_widget template;
- name = kasprintf(GFP_KERNEL, "%s %s", kcontrol->id.name,
+ name = kasprintf(GFP_KERNEL, "%s %s", ctrl_name,
"Autodisable");
if (!name) {
ret = -ENOMEM;
@@ -371,7 +371,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
if (e->autodisable) {
struct snd_soc_dapm_widget template;
- name = kasprintf(GFP_KERNEL, "%s %s", kcontrol->id.name,
+ name = kasprintf(GFP_KERNEL, "%s %s", ctrl_name,
"Autodisable");
if (!name) {
ret = -ENOMEM;
@@ -871,7 +871,7 @@ static int dapm_create_or_share_kcontrol(struct snd_soc_dapm_widget *w,
kcontrol->private_free = dapm_kcontrol_free;
- ret = dapm_kcontrol_data_alloc(w, kcontrol);
+ ret = dapm_kcontrol_data_alloc(w, kcontrol, name);
if (ret) {
snd_ctl_free_one(kcontrol);
goto exit_free;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index e898b427be7e..1af4f23697a7 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1810,7 +1810,8 @@ int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream)
(be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) &&
(be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) &&
(be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) &&
- (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP))
+ (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) &&
+ (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND))
continue;
dev_dbg(be->dev, "ASoC: hw_free BE %s\n",