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-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
25 files changed, 316 insertions, 93 deletions
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 806dfa0e5eae..05f4aed13af9 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);
generated by cgit v1.2.3 (git 2.25.1) at 2024-12-02 11:23:35 +0000