ASoC: Fixes for v6.4

A collection of fixes for v6.4, mostly driver specific but there's also one fix for DPCM to avoid incorrectly repeated calls to prepare() which can trigger issues on some systems. -----BEGIN PGP SIGNATURE----- iQEzBAABCgAdFiEEreZoqmdXGLWf4p/qJNaLcl1Uh9AFAmRt4skACgkQJNaLcl1U h9Cl2gf8CnINVsSuzSgRqvpaxSpsEufghFJrlVtNPJxJz6z+1X+sz/bsq3ceuIWl 56kJ9V0FgdcFJn8PtzJWSMvY/iwyMjAVrCl7cPqPlEIRrelQUihEteDQiL+lGrPn m0zkFWywW+r+KuzgoEosqL6/cDx7iR2NsELm/z/6aobsJRwVii7DigGAym9TNvwf dKvxy3uSX+XbCR7BTadO4wn4iHZ0/BkgYxUu4KckTgFDqHQHwrnbZNLhr0w214el bQuQOD0o6n8ZhBabOYv6j+TpwMO9qI/PjqJtn+fOZRi4pSmF1HlHL+8HLbukzblA FcsQJdXyKNzEbgHKTamOul9hgx4q4w== =H2J9 -----END PGP SIGNATURE----- Merge tag 'asoc-fix-v6.4-rc3' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v6.4 A collection of fixes for v6.4, mostly driver specific but there's also one fix for DPCM to avoid incorrectly repeated calls to prepare() which can trigger issues on some systems.
2024-11-27 06:04:23 +08:00 · 2023-05-24 12:14:24 +02:00 · 2023-05-24 12:14:24 +02:00 · bac4d8220b
commit bac4d8220b
parent 81302b1c7c 011a8719d6
20 changed files with 101 additions and 59 deletions
--- a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt
+++ b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt
@ -8,7 +8,7 @@ Required properties:
 	"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
--- a/include/sound/soc-acpi.h
+++ b/include/sound/soc-acpi.h
@ -170,6 +170,7 @@ struct snd_soc_acpi_link_adr {
 /* 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;
--- a/include/sound/soc-dpcm.h
+++ b/include/sound/soc-dpcm.h
@ -122,6 +122,10 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
 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);

--- a/include/uapi/sound/skl-tplg-interface.h
+++ b/include/uapi/sound/skl-tplg-interface.h
@ -66,7 +66,8 @@ enum skl_ch_cfg {
 	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
 };

--- a/sound/soc/codecs/cs35l41-lib.c
+++ b/sound/soc/codecs/cs35l41-lib.c
@ -46,7 +46,7 @@ static const struct reg_default cs35l41_reg[] = {
 	{ 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 @@ static const struct reg_default cs35l41_reg[] = {
 	{ 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 },
--- a/sound/soc/codecs/lpass-tx-macro.c
+++ b/sound/soc/codecs/lpass-tx-macro.c
@ -746,6 +746,8 @@ static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol,
 	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 @@ static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol,
 	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) {
--- a/sound/soc/codecs/rt5682-i2c.c
+++ b/sound/soc/codecs/rt5682-i2c.c
@ -267,7 +267,9 @@ static int rt5682_i2c_probe(struct i2c_client *i2c)
 		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);
 	}

--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@ -2959,6 +2959,9 @@ static int rt5682_suspend(struct snd_soc_component *component)
 	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 @@ static int rt5682_resume(struct snd_soc_component *component)
 	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
--- a/sound/soc/codecs/rt5682.h
+++ b/sound/soc/codecs/rt5682.h
@ -1462,6 +1462,7 @@ struct rt5682_priv {
 	int pll_out[RT5682_PLLS];

 	int jack_type;
+	int irq;
 	int irq_work_delay_time;
 };

--- a/sound/soc/dwc/dwc-i2s.c
+++ b/sound/soc/dwc/dwc-i2s.c
@ -183,30 +183,6 @@ static void i2s_stop(struct dw_i2s_dev *dev,
 	}
 }

-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 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream,
 	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 int dw_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
 }

 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_configure_dai_by_dt(struct dw_i2s_dev *dev,

 }

+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 @@ static int dw_i2s_probe(struct platform_device *pdev)
 		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))
--- a/sound/soc/intel/avs/apl.c
+++ b/sound/soc/intel/avs/apl.c
@ -169,6 +169,7 @@ static bool apl_lp_streaming(struct avs_dev *adev)
 {
 	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 @@ static bool apl_lp_streaming(struct avs_dev *adev)
 				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;
 }
--- a/sound/soc/intel/avs/avs.h
+++ b/sound/soc/intel/avs/avs.h
@ -283,8 +283,8 @@ void avs_release_firmwares(struct avs_dev *adev);

 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);
--- a/sound/soc/intel/avs/board_selection.c
+++ b/sound/soc/intel/avs/board_selection.c
@ -443,7 +443,7 @@ static int avs_register_i2s_boards(struct avs_dev *adev)
 	}

 	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)
--- a/sound/soc/intel/avs/control.c
+++ b/sound/soc/intel/avs/control.c
@ -21,17 +21,25 @@ static struct avs_dev *avs_get_kcontrol_adev(struct snd_kcontrol *kcontrol)
 	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 @@ int avs_control_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_va
 	/* 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 @@ int avs_control_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_va
 		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;
--- a/sound/soc/intel/avs/dsp.c
+++ b/sound/soc/intel/avs/dsp.c
@ -225,7 +225,7 @@ err:

 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 @@ err_mod_entry:
 	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;
--- a/sound/soc/intel/avs/messages.h
+++ b/sound/soc/intel/avs/messages.h
@ -619,7 +619,7 @@ enum avs_channel_config {
 	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
 };

--- a/sound/soc/intel/avs/path.h
+++ b/sound/soc/intel/avs/path.h
@ -37,7 +37,7 @@ struct avs_path_pipeline {

 struct avs_path_module {
 	u16 module_id;
-	u16 instance_id;
+	u8 instance_id;
 	union avs_gtw_attributes gtw_attrs;

 	struct avs_tplg_module *template;
--- a/sound/soc/intel/avs/pcm.c
+++ b/sound/soc/intel/avs/pcm.c
@ -468,21 +468,34 @@ static int avs_dai_fe_startup(struct snd_pcm_substream *substream, struct snd_so

 	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)
--- a/sound/soc/intel/avs/probes.c
+++ b/sound/soc/intel/avs/probes.c
@ -18,7 +18,7 @@ static int avs_dsp_init_probe(struct avs_dev *adev, union avs_connector_node_id
 {
 	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);

--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@ -2405,6 +2405,9 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream)
 		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 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe,
 	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);