Merge tag 'asoc-fix-v6.3-rc1' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

tiwai · tiwai · commit c24ff3ec55f0 · 2023-03-10T06:52:15.000+01:00
ASoC: Fixes for v6.3

More fixes that came in since -rc1, a lot from Intel - looks like
they've been busy test.  Everything is driver specific.
diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c
@@ -968,6 +968,8 @@ int da7219_aad_init(struct snd_soc_component *component)
 	INIT_WORK(&da7219_aad->hptest_work, da7219_aad_hptest_work);
 	INIT_WORK(&da7219_aad->jack_det_work, da7219_aad_jack_det_work);
 
+	mutex_init(&da7219_aad->jack_det_mutex);
+
 	ret = request_threaded_irq(da7219_aad->irq, da7219_aad_pre_irq_thread,
 				   da7219_aad_irq_thread,
 				   IRQF_TRIGGER_LOW | IRQF_ONESHOT,
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
@@ -428,8 +428,13 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream,
 {
 	struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+	bool has_capture = !hcp->hcd.no_i2s_capture;
+	bool has_playback = !hcp->hcd.no_i2s_playback;
 	int ret = 0;
 
+	if (!((has_playback && tx) || (has_capture && !tx)))
+		return 0;
+
 	mutex_lock(&hcp->lock);
 	if (hcp->busy) {
 		dev_err(dai->dev, "Only one simultaneous stream supported!\n");
@@ -468,6 +473,12 @@ static void hdmi_codec_shutdown(struct snd_pcm_substream *substream,
 				struct snd_soc_dai *dai)
 {
 	struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+	bool has_capture = !hcp->hcd.no_i2s_capture;
+	bool has_playback = !hcp->hcd.no_i2s_playback;
+
+	if (!((has_playback && tx) || (has_capture && !tx)))
+		return;
 
 	hcp->chmap_idx = HDMI_CODEC_CHMAP_IDX_UNKNOWN;
 	hcp->hcd.ops->audio_shutdown(dai->dev->parent, hcp->hcd.data);
diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c
@@ -242,7 +242,7 @@ enum {
 
 struct tx_mute_work {
 	struct tx_macro *tx;
-	u32 decimator;
+	u8 decimator;
 	struct delayed_work dwork;
 };
 
@@ -635,7 +635,7 @@ static int tx_macro_mclk_enable(struct tx_macro *tx,
 	return 0;
 }
 
-static bool is_amic_enabled(struct snd_soc_component *component, int decimator)
+static bool is_amic_enabled(struct snd_soc_component *component, u8 decimator)
 {
 	u16 adc_mux_reg, adc_reg, adc_n;
 
@@ -849,7 +849,7 @@ static int tx_macro_enable_dec(struct snd_soc_dapm_widget *w,
 			       struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
-	unsigned int decimator;
+	u8 decimator;
 	u16 tx_vol_ctl_reg, dec_cfg_reg, hpf_gate_reg, tx_gain_ctl_reg;
 	u8 hpf_cut_off_freq;
 	int hpf_delay = TX_MACRO_DMIC_HPF_DELAY_MS;
@@ -1064,7 +1064,8 @@ static int tx_macro_hw_params(struct snd_pcm_substream *substream,
 			      struct snd_soc_dai *dai)
 {
 	struct snd_soc_component *component = dai->component;
-	u32 decimator, sample_rate;
+	u32 sample_rate;
+	u8 decimator;
 	int tx_fs_rate;
 	struct tx_macro *tx = snd_soc_component_get_drvdata(component);
 
@@ -1128,7 +1129,7 @@ static int tx_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream)
 {
 	struct snd_soc_component *component = dai->component;
 	struct tx_macro *tx = snd_soc_component_get_drvdata(component);
-	u16 decimator;
+	u8 decimator;
 
 	/* active decimator not set yet */
 	if (tx->active_decimator[dai->id] == -1)
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
@@ -294,6 +294,10 @@ config SND_SOC_IMX_SGTL5000
 	  Say Y if you want to add support for SoC audio on an i.MX board with
 	  a sgtl5000 codec.
 
+	  Note that this is an old driver. Consider enabling
+	  SND_SOC_FSL_ASOC_CARD and SND_SOC_SGTL5000 to use the newer
+	  driver.
+
 config SND_SOC_IMX_SPDIF
 	tristate "SoC Audio support for i.MX boards with S/PDIF"
 	select SND_SOC_IMX_PCM_DMA
diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c
@@ -117,6 +117,26 @@ static void avs_da7219_codec_exit(struct snd_soc_pcm_runtime *rtd)
 	snd_soc_component_set_jack(asoc_rtd_to_codec(rtd, 0)->component, NULL, NULL);
 }
 
+static int
+avs_da7219_be_fixup(struct snd_soc_pcm_runtime *runrime, struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate, *channels;
+	struct snd_mask *fmt;
+
+	rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+	channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+	fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+
+	/* The ADSP will convert the FE rate to 48k, stereo */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set SSP0 to 24 bit */
+	snd_mask_none(fmt);
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
+	return 0;
+}
+
 static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port,
 			       struct snd_soc_dai_link **dai_link)
 {
@@ -148,6 +168,7 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in
 	dl->num_platforms = 1;
 	dl->id = 0;
 	dl->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS;
+	dl->be_hw_params_fixup = avs_da7219_be_fixup;
 	dl->init = avs_da7219_codec_init;
 	dl->exit = avs_da7219_codec_exit;
 	dl->nonatomic = 1;
diff --git a/sound/soc/intel/avs/boards/max98357a.c b/sound/soc/intel/avs/boards/max98357a.c
@@ -8,6 +8,7 @@
 
 #include <linux/module.h>
 #include <linux/platform_device.h>
+#include <sound/pcm_params.h>
 #include <sound/soc.h>
 #include <sound/soc-acpi.h>
 #include <sound/soc-dapm.h>
@@ -24,6 +25,26 @@ static const struct snd_soc_dapm_route card_base_routes[] = {
 	{ "Spk", NULL, "Speaker" },
 };
 
+static int
+avs_max98357a_be_fixup(struct snd_soc_pcm_runtime *runrime, struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate, *channels;
+	struct snd_mask *fmt;
+
+	rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+	channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+	fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+
+	/* The ADSP will convert the FE rate to 48k, stereo */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set SSP0 to 16 bit */
+	snd_mask_none(fmt);
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
+	return 0;
+}
+
 static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port,
 			       struct snd_soc_dai_link **dai_link)
 {
@@ -55,6 +76,7 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in
 	dl->num_platforms = 1;
 	dl->id = 0;
 	dl->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS;
+	dl->be_hw_params_fixup = avs_max98357a_be_fixup;
 	dl->nonatomic = 1;
 	dl->no_pcm = 1;
 	dl->dpcm_playback = 1;
diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c
@@ -33,15 +33,15 @@ avs_nau8825_clock_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *co
 		return -EINVAL;
 	}
 
-	if (!SND_SOC_DAPM_EVENT_ON(event)) {
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_MCLK, 24000000,
+					     SND_SOC_CLOCK_IN);
+	else
 		ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN);
-		if (ret < 0) {
-			dev_err(card->dev, "set sysclk err = %d\n", ret);
-			return ret;
-		}
-	}
+	if (ret < 0)
+		dev_err(card->dev, "Set sysclk failed: %d\n", ret);
 
-	return 0;
+	return ret;
 }
 
 static const struct snd_kcontrol_new card_controls[] = {
diff --git a/sound/soc/intel/avs/boards/rt5682.c b/sound/soc/intel/avs/boards/rt5682.c
@@ -169,6 +169,27 @@ static const struct snd_soc_ops avs_rt5682_ops = {
 	.hw_params = avs_rt5682_hw_params,
 };
 
+static int
+avs_rt5682_be_fixup(struct snd_soc_pcm_runtime *runtime, struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate, *channels;
+	struct snd_mask *fmt;
+
+	rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+	channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+	fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+
+	/* The ADSP will convert the FE rate to 48k, stereo */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set SSPN to 24 bit */
+	snd_mask_none(fmt);
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
+
+	return 0;
+}
+
 static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port,
 			       struct snd_soc_dai_link **dai_link)
 {
@@ -201,6 +222,7 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in
 	dl->id = 0;
 	dl->init = avs_rt5682_codec_init;
 	dl->exit = avs_rt5682_codec_exit;
+	dl->be_hw_params_fixup = avs_rt5682_be_fixup;
 	dl->ops = &avs_rt5682_ops;
 	dl->nonatomic = 1;
 	dl->no_pcm = 1;
diff --git a/sound/soc/intel/avs/boards/ssm4567.c b/sound/soc/intel/avs/boards/ssm4567.c
@@ -15,7 +15,6 @@
 #include <sound/soc-acpi.h>
 #include "../../../codecs/nau8825.h"
 
-#define SKL_NUVOTON_CODEC_DAI	"nau8825-hifi"
 #define SKL_SSM_CODEC_DAI	"ssm4567-hifi"
 
 static struct snd_soc_codec_conf card_codec_conf[] = {
@@ -34,41 +33,11 @@ static const struct snd_kcontrol_new card_controls[] = {
 	SOC_DAPM_PIN_SWITCH("Right Speaker"),
 };
 
-static int
-platform_clock_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *control, int event)
-{
-	struct snd_soc_dapm_context *dapm = w->dapm;
-	struct snd_soc_card *card = dapm->card;
-	struct snd_soc_dai *codec_dai;
-	int ret;
-
-	codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI);
-	if (!codec_dai) {
-		dev_err(card->dev, "Codec dai not found\n");
-		return -EINVAL;
-	}
-
-	if (SND_SOC_DAPM_EVENT_ON(event)) {
-		ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_MCLK, 24000000,
-					     SND_SOC_CLOCK_IN);
-		if (ret < 0)
-			dev_err(card->dev, "set sysclk err = %d\n", ret);
-	} else {
-		ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN);
-		if (ret < 0)
-			dev_err(card->dev, "set sysclk err = %d\n", ret);
-	}
-
-	return ret;
-}
-
 static const struct snd_soc_dapm_widget card_widgets[] = {
 	SND_SOC_DAPM_SPK("Left Speaker", NULL),
 	SND_SOC_DAPM_SPK("Right Speaker", NULL),
 	SND_SOC_DAPM_SPK("DP1", NULL),
 	SND_SOC_DAPM_SPK("DP2", NULL),
-	SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, platform_clock_control,
-			    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
 };
 
 static const struct snd_soc_dapm_route card_base_routes[] = {
diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c
@@ -559,7 +559,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = {
 	{
 		.comp_ids = &essx_83x6,
 		.drv_name = "sof-essx8336",
-		.sof_tplg_filename = "sof-adl-es83x6", /* the tplg suffix is added at run time */
+		.sof_tplg_filename = "sof-adl-es8336", /* the tplg suffix is added at run time */
 		.tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_SSP_NUMBER |
 					SND_SOC_ACPI_TPLG_INTEL_SSP_MSB |
 					SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER,
diff --git a/sound/soc/qcom/qdsp6/q6prm.c b/sound/soc/qcom/qdsp6/q6prm.c
@@ -184,9 +184,9 @@ int q6prm_set_lpass_clock(struct device *dev, int clk_id, int clk_attr, int clk_
 			  unsigned int freq)
 {
 	if (freq)
-		return q6prm_request_lpass_clock(dev, clk_id, clk_attr, clk_attr, freq);
+		return q6prm_request_lpass_clock(dev, clk_id, clk_attr, clk_root, freq);
 
-	return q6prm_release_lpass_clock(dev, clk_id, clk_attr, clk_attr, freq);
+	return q6prm_release_lpass_clock(dev, clk_id, clk_attr, clk_root, freq);
 }
 EXPORT_SYMBOL_GPL(q6prm_set_lpass_clock);
 
diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c
@@ -196,12 +196,15 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev)
 		goto err;
 	}
 
+	usleep_range(500, 1000);
+
 	/* exit HDA controller reset */
 	ret = hda_dsp_ctrl_link_reset(sdev, false);
 	if (ret < 0) {
 		dev_err(sdev->dev, "error: failed to exit HDA controller reset\n");
 		goto err;
 	}
+	usleep_range(1000, 1200);
 
 	hda_codec_detect_mask(sdev);
 
diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
@@ -392,6 +392,12 @@ static int hda_dsp_update_d0i3c_register(struct snd_sof_dev *sdev, u8 value)
 	snd_sof_dsp_update8(sdev, HDA_DSP_HDA_BAR, chip->d0i3_offset,
 			    SOF_HDA_VS_D0I3C_I3, value);
 
+	/*
+	 * The value written to the D0I3C::I3 bit may not be taken into account immediately.
+	 * A delay is recommended before checking if D0I3C::CIP is cleared
+	 */
+	usleep_range(30, 40);
+
 	/* Wait for cmd in progress to be cleared before exiting the function */
 	ret = hda_dsp_wait_d0i3c_done(sdev);
 	if (ret < 0) {
@@ -400,6 +406,12 @@ static int hda_dsp_update_d0i3c_register(struct snd_sof_dev *sdev, u8 value)
 	}
 
 	reg = snd_sof_dsp_read8(sdev, HDA_DSP_HDA_BAR, chip->d0i3_offset);
+	/* Confirm d0i3 state changed with paranoia check */
+	if ((reg ^ value) & SOF_HDA_VS_D0I3C_I3) {
+		dev_err(sdev->dev, "failed to update D0I3C!\n");
+		return -EIO;
+	}
+
 	trace_sof_intel_D0I3C_updated(sdev, reg);
 
 	return 0;
diff --git a/sound/soc/sof/intel/pci-apl.c b/sound/soc/sof/intel/pci-apl.c
@@ -78,6 +78,7 @@ static const struct sof_dev_desc glk_desc = {
 	.nocodec_tplg_filename = "sof-glk-nocodec.tplg",
 	.ops = &sof_apl_ops,
 	.ops_init = sof_apl_ops_init,
+	.ops_free = hda_ops_free,
 };
 
 /* PCI IDs */
diff --git a/sound/soc/sof/intel/pci-cnl.c b/sound/soc/sof/intel/pci-cnl.c
@@ -48,6 +48,7 @@ static const struct sof_dev_desc cnl_desc = {
 	.nocodec_tplg_filename = "sof-cnl-nocodec.tplg",
 	.ops = &sof_cnl_ops,
 	.ops_init = sof_cnl_ops_init,
+	.ops_free = hda_ops_free,
 };
 
 static const struct sof_dev_desc cfl_desc = {
@@ -111,6 +112,7 @@ static const struct sof_dev_desc cml_desc = {
 	.nocodec_tplg_filename = "sof-cnl-nocodec.tplg",
 	.ops = &sof_cnl_ops,
 	.ops_init = sof_cnl_ops_init,
+	.ops_free = hda_ops_free,
 };
 
 /* PCI IDs */
diff --git a/sound/soc/sof/intel/pci-icl.c b/sound/soc/sof/intel/pci-icl.c
@@ -79,6 +79,7 @@ static const struct sof_dev_desc jsl_desc = {
 	.nocodec_tplg_filename = "sof-jsl-nocodec.tplg",
 	.ops = &sof_cnl_ops,
 	.ops_init = sof_cnl_ops_init,
+	.ops_free = hda_ops_free,
 };
 
 /* PCI IDs */
diff --git a/sound/soc/sof/intel/pci-mtl.c b/sound/soc/sof/intel/pci-mtl.c
@@ -46,6 +46,7 @@ static const struct sof_dev_desc mtl_desc = {
 	.nocodec_tplg_filename = "sof-mtl-nocodec.tplg",
 	.ops = &sof_mtl_ops,
 	.ops_init = sof_mtl_ops_init,
+	.ops_free = hda_ops_free,
 };
 
 /* PCI IDs */
diff --git a/sound/soc/sof/intel/pci-skl.c b/sound/soc/sof/intel/pci-skl.c
@@ -38,6 +38,7 @@ static struct sof_dev_desc skl_desc = {
 	.nocodec_tplg_filename = "sof-skl-nocodec.tplg",
 	.ops = &sof_skl_ops,
 	.ops_init = sof_skl_ops_init,
+	.ops_free = hda_ops_free,
 };
 
 static struct sof_dev_desc kbl_desc = {
@@ -61,6 +62,7 @@ static struct sof_dev_desc kbl_desc = {
 	.nocodec_tplg_filename = "sof-kbl-nocodec.tplg",
 	.ops = &sof_skl_ops,
 	.ops_init = sof_skl_ops_init,
+	.ops_free = hda_ops_free,
 };
 
 /* PCI IDs */
diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c
diff --git a/sound/soc/sof/intel/pci-tng.c b/sound/soc/sof/intel/pci-tng.c
diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c
diff --git a/sound/soc/sof/ipc3.c b/sound/soc/sof/ipc3.c
diff --git a/sound/soc/sof/ipc4-control.c b/sound/soc/sof/ipc4-control.c
diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h
diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c
diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c

Original file line number	Diff line number	Diff line change
`@@ -559,7 +559,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = {`
`559`	`559`	`{`
`560`	`560`	`.comp_ids = &essx_83x6,`
`561`	`561`	`.drv_name = "sof-essx8336",`
`562`		`- .sof_tplg_filename = "sof-adl-es83x6", /* the tplg suffix is added at run time */`
	`562`	`+ .sof_tplg_filename = "sof-adl-es8336", /* the tplg suffix is added at run time */`
`563`	`563`	`.tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_SSP_NUMBER \|`
`564`	`564`	`SND_SOC_ACPI_TPLG_INTEL_SSP_MSB \|`
`565`	`565`	`SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER,`