ASoC: Intel: avs: Machine board fixes

Merge series from Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>:

Series of adjustments to machine board files. Use fixed format in boards
that were not using one. Fix clock handling.

5 files changed, 72 insertions, 38 deletions

diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c
index acd43b6108e9..1a1d572cc1d0 100644
--- 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
index 921f42caf7e0..183123d08c5a 100644
--- 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
index b31fa931ba8b..b69fc5567135 100644
--- 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
index 473e9fe5d0bf..b2c2ba93dcb5 100644
--- 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
index c5db69612762..2b7f5ad92aca 100644
--- 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[] = {