Merge tag 'sound-6.4-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small fixes:
- HD-audio runtime PM bug fix
- A couple of HD-audio quirks
- Fix series of ASoC Intel AVS drivers
- ASoC DPCM fix for a bug found on new Intel systems
- A few other ASoC device-specific small fixes"
* tag 'sound-6.4-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek: Enable headset onLenovo M70/M90
ASoC: dwc: move DMA init to snd_soc_dai_driver probe()
ASoC: cs35l41: Fix default regmap values for some registers
ALSA: hda: Fix unhandled register update during auto-suspend period
ASoC: dt-bindings: tlv320aic32x4: Fix supply names
ASoC: Intel: avs: Add missing checks on FE startup
ASoC: Intel: avs: Fix avs_path_module::instance_id size
ASoC: Intel: avs: Account for UID of ACPI device
ASoC: Intel: avs: Fix declaration of enum avs_channel_config
ASoC: Intel: Skylake: Fix declaration of enum skl_ch_cfg
ASoC: Intel: avs: Access path components under lock
ASoC: Intel: avs: Fix module lookup
ALSA: hda/ca0132: add quirk for EVGA X299 DARK
ASoC: soc-pcm: test if a BE can be prepared
ASoC: rt5682: Disable jack detection interrupt during suspend
ASoC: lpass: Fix for KASAN use_after_free out of bounds
diff --git a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt
index f59125b..0b4e21b 100644
--- a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt
+++ b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt
@@ -8,7 +8,7 @@
"ti,tlv320aic32x6" TLV320AIC3206, TLV320AIC3256
"ti,tas2505" TAS2505, TAS2521
- reg: I2C slave address
- - supply-*: Required supply regulators are:
+ - *-supply: Required supply regulators are:
"iov" - digital IO power supply
"ldoin" - LDO power supply
"dv" - Digital core power supply
diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h
index b38fd25..5282790 100644
--- a/include/sound/soc-acpi.h
+++ b/include/sound/soc-acpi.h
@@ -170,6 +170,7 @@
/* Descriptor for SST ASoC machine driver */
struct snd_soc_acpi_mach {
u8 id[ACPI_ID_LEN];
+ const char *uid;
const struct snd_soc_acpi_codecs *comp_ids;
const u32 link_mask;
const struct snd_soc_acpi_link_adr *links;
diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h
index 4d6ac76..ebd2475 100644
--- a/include/sound/soc-dpcm.h
+++ b/include/sound/soc-dpcm.h
@@ -122,6 +122,10 @@
int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe,
struct snd_soc_pcm_runtime *be, int stream);
+/* can this BE perform prepare */
+int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe,
+ struct snd_soc_pcm_runtime *be, int stream);
+
/* is the current PCM operation for this FE ? */
int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream);
diff --git a/include/uapi/sound/skl-tplg-interface.h b/include/uapi/sound/skl-tplg-interface.h
index f29899b..4bf9c4f 100644
--- a/include/uapi/sound/skl-tplg-interface.h
+++ b/include/uapi/sound/skl-tplg-interface.h
@@ -66,7 +66,8 @@
SKL_CH_CFG_DUAL_MONO = 9,
SKL_CH_CFG_I2S_DUAL_STEREO_0 = 10,
SKL_CH_CFG_I2S_DUAL_STEREO_1 = 11,
- SKL_CH_CFG_4_CHANNEL = 12,
+ SKL_CH_CFG_7_1 = 12,
+ SKL_CH_CFG_4_CHANNEL = SKL_CH_CFG_7_1,
SKL_CH_CFG_INVALID
};
diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c
index accc9d2..6c043fb 100644
--- a/sound/hda/hdac_device.c
+++ b/sound/hda/hdac_device.c
@@ -611,7 +611,7 @@
int snd_hdac_keep_power_up(struct hdac_device *codec)
{
if (!atomic_inc_not_zero(&codec->in_pm)) {
- int ret = pm_runtime_get_if_in_use(&codec->dev);
+ int ret = pm_runtime_get_if_active(&codec->dev, true);
if (!ret)
return -1;
if (ret < 0)
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 099722eb..748a3c4 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -1306,6 +1306,7 @@
SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI),
SND_PCI_QUIRK(0x1458, 0xA036, "Gigabyte GA-Z170X-Gaming 7", QUIRK_R3DI),
SND_PCI_QUIRK(0x3842, 0x1038, "EVGA X99 Classified", QUIRK_R3DI),
+ SND_PCI_QUIRK(0x3842, 0x104b, "EVGA X299 Dark", QUIRK_R3DI),
SND_PCI_QUIRK(0x3842, 0x1055, "EVGA Z390 DARK", QUIRK_R3DI),
SND_PCI_QUIRK(0x1102, 0x0013, "Recon3D", QUIRK_R3D),
SND_PCI_QUIRK(0x1102, 0x0018, "Recon3D", QUIRK_R3D),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a7e4765..7b5f194 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -11719,6 +11719,8 @@
SND_PCI_QUIRK(0x17aa, 0x32cb, "Lenovo ThinkCentre M70", ALC897_FIXUP_HEADSET_MIC_PIN),
SND_PCI_QUIRK(0x17aa, 0x32cf, "Lenovo ThinkCentre M950", ALC897_FIXUP_HEADSET_MIC_PIN),
SND_PCI_QUIRK(0x17aa, 0x32f7, "Lenovo ThinkCentre M90", ALC897_FIXUP_HEADSET_MIC_PIN),
+ SND_PCI_QUIRK(0x17aa, 0x3321, "Lenovo ThinkCentre M70 Gen4", ALC897_FIXUP_HEADSET_MIC_PIN),
+ SND_PCI_QUIRK(0x17aa, 0x331b, "Lenovo ThinkCentre M90 Gen4", ALC897_FIXUP_HEADSET_MIC_PIN),
SND_PCI_QUIRK(0x17aa, 0x3742, "Lenovo TianYi510Pro-14IOB", ALC897_FIXUP_HEADSET_MIC_PIN2),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD),
diff --git a/sound/soc/codecs/cs35l41-lib.c b/sound/soc/codecs/cs35l41-lib.c
index 8538e28..1e42052 100644
--- a/sound/soc/codecs/cs35l41-lib.c
+++ b/sound/soc/codecs/cs35l41-lib.c
@@ -46,7 +46,7 @@
{ CS35L41_DSP1_RX5_SRC, 0x00000020 },
{ CS35L41_DSP1_RX6_SRC, 0x00000021 },
{ CS35L41_DSP1_RX7_SRC, 0x0000003A },
- { CS35L41_DSP1_RX8_SRC, 0x00000001 },
+ { CS35L41_DSP1_RX8_SRC, 0x0000003B },
{ CS35L41_NGATE1_SRC, 0x00000008 },
{ CS35L41_NGATE2_SRC, 0x00000009 },
{ CS35L41_AMP_DIG_VOL_CTRL, 0x00008000 },
@@ -58,8 +58,8 @@
{ CS35L41_IRQ1_MASK2, 0xFFFFFFFF },
{ CS35L41_IRQ1_MASK3, 0xFFFF87FF },
{ CS35L41_IRQ1_MASK4, 0xFEFFFFFF },
- { CS35L41_GPIO1_CTRL1, 0xE1000001 },
- { CS35L41_GPIO2_CTRL1, 0xE1000001 },
+ { CS35L41_GPIO1_CTRL1, 0x81000001 },
+ { CS35L41_GPIO2_CTRL1, 0x81000001 },
{ CS35L41_MIXER_NGATE_CFG, 0x00000000 },
{ CS35L41_MIXER_NGATE_CH1_CFG, 0x00000303 },
{ CS35L41_MIXER_NGATE_CH2_CFG, 0x00000303 },
diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c
index da6fcf7..de978c3 100644
--- a/sound/soc/codecs/lpass-tx-macro.c
+++ b/sound/soc/codecs/lpass-tx-macro.c
@@ -746,6 +746,8 @@
struct tx_macro *tx = snd_soc_component_get_drvdata(component);
val = ucontrol->value.enumerated.item[0];
+ if (val >= e->items)
+ return -EINVAL;
switch (e->reg) {
case CDC_TX_INP_MUX_ADC_MUX0_CFG0:
@@ -772,6 +774,9 @@
case CDC_TX_INP_MUX_ADC_MUX7_CFG0:
mic_sel_reg = CDC_TX7_TX_PATH_CFG0;
break;
+ default:
+ dev_err(component->dev, "Error in configuration!!\n");
+ return -EINVAL;
}
if (val != 0) {
diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c
index 2935c1b..5bc46b0 100644
--- a/sound/soc/codecs/rt5682-i2c.c
+++ b/sound/soc/codecs/rt5682-i2c.c
@@ -267,7 +267,9 @@
ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL,
rt5682_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING
| IRQF_ONESHOT, "rt5682", rt5682);
- if (ret)
+ if (!ret)
+ rt5682->irq = i2c->irq;
+ else
dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret);
}
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index f6c798b..5d99254 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -2959,6 +2959,9 @@
if (rt5682->is_sdw)
return 0;
+ if (rt5682->irq)
+ disable_irq(rt5682->irq);
+
cancel_delayed_work_sync(&rt5682->jack_detect_work);
cancel_delayed_work_sync(&rt5682->jd_check_work);
if (rt5682->hs_jack && (rt5682->jack_type & SND_JACK_HEADSET) == SND_JACK_HEADSET) {
@@ -3027,6 +3030,9 @@
mod_delayed_work(system_power_efficient_wq,
&rt5682->jack_detect_work, msecs_to_jiffies(0));
+ if (rt5682->irq)
+ enable_irq(rt5682->irq);
+
return 0;
}
#else
diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h
index d568c69..e8efd8a 100644
--- a/sound/soc/codecs/rt5682.h
+++ b/sound/soc/codecs/rt5682.h
@@ -1462,6 +1462,7 @@
int pll_out[RT5682_PLLS];
int jack_type;
+ int irq;
int irq_work_delay_time;
};
diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c
index ca20cad..399a489 100644
--- a/sound/soc/dwc/dwc-i2s.c
+++ b/sound/soc/dwc/dwc-i2s.c
@@ -183,30 +183,6 @@
}
}
-static int dw_i2s_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *cpu_dai)
-{
- struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai);
- union dw_i2s_snd_dma_data *dma_data = NULL;
-
- if (!(dev->capability & DWC_I2S_RECORD) &&
- (substream->stream == SNDRV_PCM_STREAM_CAPTURE))
- return -EINVAL;
-
- if (!(dev->capability & DWC_I2S_PLAY) &&
- (substream->stream == SNDRV_PCM_STREAM_PLAYBACK))
- return -EINVAL;
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- dma_data = &dev->play_dma_data;
- else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
- dma_data = &dev->capture_dma_data;
-
- snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)dma_data);
-
- return 0;
-}
-
static void dw_i2s_config(struct dw_i2s_dev *dev, int stream)
{
u32 ch_reg;
@@ -305,12 +281,6 @@
return 0;
}
-static void dw_i2s_shutdown(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- snd_soc_dai_set_dma_data(dai, substream, NULL);
-}
-
static int dw_i2s_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
@@ -382,8 +352,6 @@
}
static const struct snd_soc_dai_ops dw_i2s_dai_ops = {
- .startup = dw_i2s_startup,
- .shutdown = dw_i2s_shutdown,
.hw_params = dw_i2s_hw_params,
.prepare = dw_i2s_prepare,
.trigger = dw_i2s_trigger,
@@ -625,6 +593,14 @@
}
+static int dw_i2s_dai_probe(struct snd_soc_dai *dai)
+{
+ struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+ snd_soc_dai_init_dma_data(dai, &dev->play_dma_data, &dev->capture_dma_data);
+ return 0;
+}
+
static int dw_i2s_probe(struct platform_device *pdev)
{
const struct i2s_platform_data *pdata = pdev->dev.platform_data;
@@ -643,6 +619,7 @@
return -ENOMEM;
dw_i2s_dai->ops = &dw_i2s_dai_ops;
+ dw_i2s_dai->probe = dw_i2s_dai_probe;
dev->i2s_base = devm_platform_get_and_ioremap_resource(pdev, 0, &res);
if (IS_ERR(dev->i2s_base))
diff --git a/sound/soc/intel/avs/apl.c b/sound/soc/intel/avs/apl.c
index 02683dc..1860099 100644
--- a/sound/soc/intel/avs/apl.c
+++ b/sound/soc/intel/avs/apl.c
@@ -169,6 +169,7 @@
{
struct avs_path *path;
+ spin_lock(&adev->path_list_lock);
/* Any gateway without buffer allocated in LP area disqualifies D0IX. */
list_for_each_entry(path, &adev->path_list, node) {
struct avs_path_pipeline *ppl;
@@ -188,11 +189,14 @@
if (cfg->copier.dma_type == INVALID_OBJECT_ID)
continue;
- if (!mod->gtw_attrs.lp_buffer_alloc)
+ if (!mod->gtw_attrs.lp_buffer_alloc) {
+ spin_unlock(&adev->path_list_lock);
return false;
+ }
}
}
}
+ spin_unlock(&adev->path_list_lock);
return true;
}
diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h
index d7fccdc..0cf38c9 100644
--- a/sound/soc/intel/avs/avs.h
+++ b/sound/soc/intel/avs/avs.h
@@ -283,8 +283,8 @@
int avs_dsp_init_module(struct avs_dev *adev, u16 module_id, u8 ppl_instance_id,
u8 core_id, u8 domain, void *param, u32 param_size,
- u16 *instance_id);
-void avs_dsp_delete_module(struct avs_dev *adev, u16 module_id, u16 instance_id,
+ u8 *instance_id);
+void avs_dsp_delete_module(struct avs_dev *adev, u16 module_id, u8 instance_id,
u8 ppl_instance_id, u8 core_id);
int avs_dsp_create_pipeline(struct avs_dev *adev, u16 req_size, u8 priority,
bool lp, u16 attributes, u8 *instance_id);
diff --git a/sound/soc/intel/avs/board_selection.c b/sound/soc/intel/avs/board_selection.c
index b2823c2..60f8fb0 100644
--- a/sound/soc/intel/avs/board_selection.c
+++ b/sound/soc/intel/avs/board_selection.c
@@ -443,7 +443,7 @@
}
for (mach = boards->machs; mach->id[0]; mach++) {
- if (!acpi_dev_present(mach->id, NULL, -1))
+ if (!acpi_dev_present(mach->id, mach->uid, -1))
continue;
if (mach->machine_quirk)
diff --git a/sound/soc/intel/avs/control.c b/sound/soc/intel/avs/control.c
index a8b14b78..3dfa2e9 100644
--- a/sound/soc/intel/avs/control.c
+++ b/sound/soc/intel/avs/control.c
@@ -21,17 +21,25 @@
return to_avs_dev(w->dapm->component->dev);
}
-static struct avs_path_module *avs_get_kcontrol_module(struct avs_dev *adev, u32 id)
+static struct avs_path_module *avs_get_volume_module(struct avs_dev *adev, u32 id)
{
struct avs_path *path;
struct avs_path_pipeline *ppl;
struct avs_path_module *mod;
- list_for_each_entry(path, &adev->path_list, node)
- list_for_each_entry(ppl, &path->ppl_list, node)
- list_for_each_entry(mod, &ppl->mod_list, node)
- if (mod->template->ctl_id && mod->template->ctl_id == id)
+ spin_lock(&adev->path_list_lock);
+ list_for_each_entry(path, &adev->path_list, node) {
+ list_for_each_entry(ppl, &path->ppl_list, node) {
+ list_for_each_entry(mod, &ppl->mod_list, node) {
+ if (guid_equal(&mod->template->cfg_ext->type, &AVS_PEAKVOL_MOD_UUID)
+ && mod->template->ctl_id == id) {
+ spin_unlock(&adev->path_list_lock);
return mod;
+ }
+ }
+ }
+ }
+ spin_unlock(&adev->path_list_lock);
return NULL;
}
@@ -49,7 +57,7 @@
/* prevent access to modules while path is being constructed */
mutex_lock(&adev->path_mutex);
- active_module = avs_get_kcontrol_module(adev, ctl_data->id);
+ active_module = avs_get_volume_module(adev, ctl_data->id);
if (active_module) {
ret = avs_ipc_peakvol_get_volume(adev, active_module->module_id,
active_module->instance_id, &dspvols,
@@ -89,7 +97,7 @@
changed = 1;
}
- active_module = avs_get_kcontrol_module(adev, ctl_data->id);
+ active_module = avs_get_volume_module(adev, ctl_data->id);
if (active_module) {
dspvol.channel_id = AVS_ALL_CHANNELS_MASK;
dspvol.target_volume = *volume;
diff --git a/sound/soc/intel/avs/dsp.c b/sound/soc/intel/avs/dsp.c
index b881100..aa03af4 100644
--- a/sound/soc/intel/avs/dsp.c
+++ b/sound/soc/intel/avs/dsp.c
@@ -225,7 +225,7 @@
int avs_dsp_init_module(struct avs_dev *adev, u16 module_id, u8 ppl_instance_id,
u8 core_id, u8 domain, void *param, u32 param_size,
- u16 *instance_id)
+ u8 *instance_id)
{
struct avs_module_entry mentry;
bool was_loaded = false;
@@ -272,7 +272,7 @@
return ret;
}
-void avs_dsp_delete_module(struct avs_dev *adev, u16 module_id, u16 instance_id,
+void avs_dsp_delete_module(struct avs_dev *adev, u16 module_id, u8 instance_id,
u8 ppl_instance_id, u8 core_id)
{
struct avs_module_entry mentry;
diff --git a/sound/soc/intel/avs/messages.h b/sound/soc/intel/avs/messages.h
index d3b60ae..7f23a30 100644
--- a/sound/soc/intel/avs/messages.h
+++ b/sound/soc/intel/avs/messages.h
@@ -619,7 +619,7 @@
AVS_CHANNEL_CONFIG_DUAL_MONO = 9,
AVS_CHANNEL_CONFIG_I2S_DUAL_STEREO_0 = 10,
AVS_CHANNEL_CONFIG_I2S_DUAL_STEREO_1 = 11,
- AVS_CHANNEL_CONFIG_4_CHANNEL = 12,
+ AVS_CHANNEL_CONFIG_7_1 = 12,
AVS_CHANNEL_CONFIG_INVALID
};
diff --git a/sound/soc/intel/avs/path.h b/sound/soc/intel/avs/path.h
index 197222c..657f7b0 100644
--- a/sound/soc/intel/avs/path.h
+++ b/sound/soc/intel/avs/path.h
@@ -37,7 +37,7 @@
struct avs_path_module {
u16 module_id;
- u16 instance_id;
+ u8 instance_id;
union avs_gtw_attributes gtw_attrs;
struct avs_tplg_module *template;
diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c
index 31c032a..1fbb2c2 100644
--- a/sound/soc/intel/avs/pcm.c
+++ b/sound/soc/intel/avs/pcm.c
@@ -468,21 +468,34 @@
host_stream = snd_hdac_ext_stream_assign(bus, substream, HDAC_EXT_STREAM_TYPE_HOST);
if (!host_stream) {
- kfree(data);
- return -EBUSY;
+ ret = -EBUSY;
+ goto err;
}
data->host_stream = host_stream;
- snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
+ ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ goto err;
+
/* avoid wrap-around with wall-clock */
- snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_TIME, 20, 178000000);
- snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_rates);
+ ret = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_TIME, 20, 178000000);
+ if (ret < 0)
+ goto err;
+
+ ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_rates);
+ if (ret < 0)
+ goto err;
+
snd_pcm_set_sync(substream);
dev_dbg(dai->dev, "%s fe STARTUP tag %d str %p",
__func__, hdac_stream(host_stream)->stream_tag, substream);
return 0;
+
+err:
+ kfree(data);
+ return ret;
}
static void avs_dai_fe_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
diff --git a/sound/soc/intel/avs/probes.c b/sound/soc/intel/avs/probes.c
index 70a94201..2759282 100644
--- a/sound/soc/intel/avs/probes.c
+++ b/sound/soc/intel/avs/probes.c
@@ -18,7 +18,7 @@
{
struct avs_probe_cfg cfg = {{0}};
struct avs_module_entry mentry;
- u16 dummy;
+ u8 dummy;
avs_get_module_entry(adev, &AVS_PROBE_MOD_UUID, &mentry);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index adb69d7..4fb1ac8 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2405,6 +2405,9 @@
if (!snd_soc_dpcm_be_can_update(fe, be, stream))
continue;
+ if (!snd_soc_dpcm_can_be_prepared(fe, be, stream))
+ continue;
+
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) &&
@@ -3042,3 +3045,20 @@
return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state));
}
EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params);
+
+/*
+ * We can only prepare a BE DAI if any of it's FE are not prepared,
+ * running or paused for the specified stream direction.
+ */
+int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe,
+ struct snd_soc_pcm_runtime *be, int stream)
+{
+ const enum snd_soc_dpcm_state state[] = {
+ SND_SOC_DPCM_STATE_START,
+ SND_SOC_DPCM_STATE_PAUSED,
+ SND_SOC_DPCM_STATE_PREPARE,
+ };
+
+ return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state));
+}
+EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_prepared);
