Merge remote-tracking branches 'asoc/topic/atmel', 'asoc/topic/cirrus' and...

Merge remote-tracking branches 'asoc/topic/atmel', 'asoc/topic/cirrus' and 'asoc/topic/cleanup' into asoc-next

drivers/misc/atmel-ssc.c

@@ -83,16 +83,26 @@ EXPORT_SYMBOL(ssc_free);
 
 static struct atmel_ssc_platform_data at91rm9200_config = {
 	.use_dma = 0,
+	.has_fslen_ext = 0,
+};
+
+static struct atmel_ssc_platform_data at91sam9rl_config = {
+	.use_dma = 0,
+	.has_fslen_ext = 1,
 };
 
 static struct atmel_ssc_platform_data at91sam9g45_config = {
 	.use_dma = 1,
+	.has_fslen_ext = 1,
 };
 
 static const struct platform_device_id atmel_ssc_devtypes[] = {
 	{
 		.name = "at91rm9200_ssc",
 		.driver_data = (unsigned long) &at91rm9200_config,
+	}, {
+		.name = "at91sam9rl_ssc",
+		.driver_data = (unsigned long) &at91sam9rl_config,
 	}, {
 		.name = "at91sam9g45_ssc",
 		.driver_data = (unsigned long) &at91sam9g45_config,
@@ -106,6 +116,9 @@ static const struct of_device_id atmel_ssc_dt_ids[] = {
 	{
 		.compatible = "atmel,at91rm9200-ssc",
 		.data = &at91rm9200_config,
+	}, {
+		.compatible = "atmel,at91sam9rl-ssc",
+		.data = &at91sam9rl_config,
 	}, {
 		.compatible = "atmel,at91sam9g45-ssc",
 		.data = &at91sam9g45_config,

include/linux/atmel-ssc.h

@@ -7,6 +7,7 @@
 
 struct atmel_ssc_platform_data {
 	int			use_dma;
+	int			has_fslen_ext;
 };
 
 struct ssc_device {
@@ -71,6 +72,12 @@ void ssc_free(struct ssc_device *ssc);
 #define SSC_RFMR_DATNB_OFFSET			 8
 #define SSC_RFMR_FSEDGE_SIZE			 1
 #define SSC_RFMR_FSEDGE_OFFSET			24
+/*
+ * The FSLEN_EXT exist on at91sam9rl, at91sam9g10,
+ * at91sam9g20, and at91sam9g45 and newer SoCs
+ */
+#define SSC_RFMR_FSLEN_EXT_SIZE			 4
+#define SSC_RFMR_FSLEN_EXT_OFFSET		28
 #define SSC_RFMR_FSLEN_SIZE			 4
 #define SSC_RFMR_FSLEN_OFFSET			16
 #define SSC_RFMR_FSOS_SIZE			 4
@@ -109,6 +116,12 @@ void ssc_free(struct ssc_device *ssc);
 #define SSC_TFMR_FSDEN_OFFSET			23
 #define SSC_TFMR_FSEDGE_SIZE			 1
 #define SSC_TFMR_FSEDGE_OFFSET			24
+/*
+ * The FSLEN_EXT exist on at91sam9rl, at91sam9g10,
+ * at91sam9g20, and at91sam9g45 and newer SoCs
+ */
+#define SSC_TFMR_FSLEN_EXT_SIZE			 4
+#define SSC_TFMR_FSLEN_EXT_OFFSET		28
 #define SSC_TFMR_FSLEN_SIZE			 4
 #define SSC_TFMR_FSLEN_OFFSET			16
 #define SSC_TFMR_FSOS_SIZE			 3

sound/soc/atmel/atmel_ssc_dai.c

@@ -347,6 +347,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
 	u32 tfmr, rfmr, tcmr, rcmr;
 	int start_event;
 	int ret;
+	int fslen, fslen_ext;
 
 	/*
 	 * Currently, there is only one set of dma params for
@@ -387,18 +388,6 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	/*
-	 * The SSC only supports up to 16-bit samples in I2S format, due
-	 * to the size of the Frame Mode Register FSLEN field.
-	 */
-	if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S
-		&& bits > 16) {
-		printk(KERN_WARNING
-				"atmel_ssc_dai: sample size %d "
-				"is too large for I2S\n", bits);
-		return -EINVAL;
-	}
-
 	/*
 	 * Compute SSC register settings.
 	 */
@@ -413,6 +402,17 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
 		 * from the MCK divider, and the BCLK signal
 		 * is output on the SSC TK line.
 		 */
+
+		if (bits > 16 && !ssc->pdata->has_fslen_ext) {
+			dev_err(dai->dev,
+				"sample size %d is too large for SSC device\n",
+				bits);
+			return -EINVAL;
+		}
+
+		fslen_ext = (bits - 1) / 16;
+		fslen = (bits - 1) % 16;
+
 		rcmr =	  SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period)
 			| SSC_BF(RCMR_STTDLY, START_DELAY)
 			| SSC_BF(RCMR_START, SSC_START_FALLING_RF)
@@ -420,9 +420,10 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
 			| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
 			| SSC_BF(RCMR_CKS, SSC_CKS_DIV);
 
-		rfmr =	  SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+		rfmr =    SSC_BF(RFMR_FSLEN_EXT, fslen_ext)
+			| SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
 			| SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE)
-			| SSC_BF(RFMR_FSLEN, (bits - 1))
+			| SSC_BF(RFMR_FSLEN, fslen)
 			| SSC_BF(RFMR_DATNB, (channels - 1))
 			| SSC_BIT(RFMR_MSBF)
 			| SSC_BF(RFMR_LOOP, 0)
@@ -435,10 +436,11 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
 			| SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS)
 			| SSC_BF(TCMR_CKS, SSC_CKS_DIV);
 
-		tfmr =	  SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+		tfmr =    SSC_BF(TFMR_FSLEN_EXT, fslen_ext)
+			| SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
 			| SSC_BF(TFMR_FSDEN, 0)
 			| SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE)
-			| SSC_BF(TFMR_FSLEN, (bits - 1))
+			| SSC_BF(TFMR_FSLEN, fslen)
 			| SSC_BF(TFMR_DATNB, (channels - 1))
 			| SSC_BIT(TFMR_MSBF)
 			| SSC_BF(TFMR_DATDEF, 0)

sound/soc/atmel/atmel_wm8904.c

@@ -18,10 +18,6 @@
 #include "../codecs/wm8904.h"
 #include "atmel_ssc_dai.h"
 
-#define MCLK_RATE 32768
-
-static struct clk *mclk;
-
 static const struct snd_soc_dapm_widget atmel_asoc_wm8904_dapm_widgets[] = {
 	SND_SOC_DAPM_HP("Headphone Jack", NULL),
 	SND_SOC_DAPM_MIC("Mic", NULL),
@@ -61,26 +57,6 @@ static struct snd_soc_ops atmel_asoc_wm8904_ops = {
 	.hw_params = atmel_asoc_wm8904_hw_params,
 };
 
-static int atmel_set_bias_level(struct snd_soc_card *card,
-		struct snd_soc_dapm_context *dapm,
-		enum snd_soc_bias_level level)
-{
-	if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
-		switch (level) {
-		case SND_SOC_BIAS_PREPARE:
-			clk_prepare_enable(mclk);
-			break;
-		case SND_SOC_BIAS_OFF:
-			clk_disable_unprepare(mclk);
-			break;
-		default:
-			break;
-		}
-	}
-
-	return 0;
-};
-
 static struct snd_soc_dai_link atmel_asoc_wm8904_dailink = {
 	.name = "WM8904",
 	.stream_name = "WM8904 PCM",
@@ -94,7 +70,6 @@ static struct snd_soc_dai_link atmel_asoc_wm8904_dailink = {
 static struct snd_soc_card atmel_asoc_wm8904_card = {
 	.name = "atmel_asoc_wm8904",
 	.owner = THIS_MODULE,
-	.set_bias_level = atmel_set_bias_level,
 	.dai_link = &atmel_asoc_wm8904_dailink,
 	.num_links = 1,
 	.dapm_widgets = atmel_asoc_wm8904_dapm_widgets,
@@ -153,7 +128,6 @@ static int atmel_asoc_wm8904_probe(struct platform_device *pdev)
 {
 	struct snd_soc_card *card = &atmel_asoc_wm8904_card;
 	struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink;
-	struct clk *clk_src;
 	int id, ret;
 
 	card->dev = &pdev->dev;
@@ -170,30 +144,6 @@ static int atmel_asoc_wm8904_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	mclk = clk_get(NULL, "pck0");
-	if (IS_ERR(mclk)) {
-		dev_err(&pdev->dev, "failed to get pck0\n");
-		ret = PTR_ERR(mclk);
-		goto err_set_audio;
-	}
-
-	clk_src = clk_get(NULL, "clk32k");
-	if (IS_ERR(clk_src)) {
-		dev_err(&pdev->dev, "failed to get clk32k\n");
-		ret = PTR_ERR(clk_src);
-		goto err_set_audio;
-	}
-
-	ret = clk_set_parent(mclk, clk_src);
-	clk_put(clk_src);
-	if (ret != 0) {
-		dev_err(&pdev->dev, "failed to set MCLK parent\n");
-		goto err_set_audio;
-	}
-
-	dev_info(&pdev->dev, "setting pck0 to %dHz\n", MCLK_RATE);
-	clk_set_rate(mclk, MCLK_RATE);
-
 	ret = snd_soc_register_card(card);
 	if (ret) {
 		dev_err(&pdev->dev, "snd_soc_register_card failed\n");

sound/soc/codecs/cs42l52.c

@@ -399,15 +399,15 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
 			      CS42L52_MASTERB_VOL, 0, 0x34, 0xE4, hl_tlv),
 
 	SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L52_HPA_VOL,
-			      CS42L52_HPB_VOL, 0, 0x34, 0xCC, hpd_tlv),
+			      CS42L52_HPB_VOL, 0, 0x34, 0xC0, hpd_tlv),
 
 	SOC_ENUM("Headphone Analog Gain", hp_gain_enum),
 
 	SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL,
-			      CS42L52_SPKB_VOL, 0, 0x1, 0xff, hl_tlv),
+			      CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv),
 
 	SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL,
-			      CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv),
+			      CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv),
 
 	SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0),
 
@@ -417,10 +417,10 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
 	SOC_ENUM("MIC Bias Level", mic_bias_level_enum),
 
 	SOC_DOUBLE_R_SX_TLV("ADC Volume", CS42L52_ADCA_VOL,
-			      CS42L52_ADCB_VOL, 7, 0x80, 0xA0, ipd_tlv),
+			      CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv),
 	SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume",
 			     CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL,
-				6, 0x7f, 0x19, ipd_tlv),
+				0, 0x19, 0x7F, ipd_tlv),
 
 	SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0),
 
@@ -428,11 +428,11 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
 		     CS42L52_ADCB_MIXER_VOL, 7, 1, 1),
 
 	SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L52_PGAA_CTL,
-			    CS42L52_PGAB_CTL, 0, 0x28, 0x30, pga_tlv),
+			    CS42L52_PGAB_CTL, 0, 0x28, 0x24, pga_tlv),
 
 	SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume",
 			    CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL,
-				0, 0x7f, 0x19, mix_tlv),
+				0, 0x19, 0x7f, mix_tlv),
 	SOC_DOUBLE_R("PCM Mixer Switch",
 		     CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1),
 

sound/soc/codecs/cs42l56.c

@@ -421,15 +421,15 @@ static const struct soc_enum ng_delay_enum =
 static const struct snd_kcontrol_new cs42l56_snd_controls[] = {
 
 	SOC_DOUBLE_R_SX_TLV("Master Volume", CS42L56_MASTER_A_VOLUME,
-			      CS42L56_MASTER_B_VOLUME, 0, 0x34, 0xfd, adv_tlv),
+			      CS42L56_MASTER_B_VOLUME, 0, 0x34, 0xE4, adv_tlv),
 	SOC_DOUBLE("Master Mute Switch", CS42L56_DSP_MUTE_CTL, 0, 1, 1, 1),
 
 	SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L56_ADCA_MIX_VOLUME,
-			      CS42L56_ADCB_MIX_VOLUME, 0, 0x88, 0xa9, hl_tlv),
+			      CS42L56_ADCB_MIX_VOLUME, 0, 0x88, 0x90, hl_tlv),
 	SOC_DOUBLE("ADC Mixer Mute Switch", CS42L56_DSP_MUTE_CTL, 6, 7, 1, 1),
 
 	SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", CS42L56_PCMA_MIX_VOLUME,
-			      CS42L56_PCMB_MIX_VOLUME, 0, 0x88, 0xa9, hl_tlv),
+			      CS42L56_PCMB_MIX_VOLUME, 0, 0x88, 0x90, hl_tlv),
 	SOC_DOUBLE("PCM Mixer Mute Switch", CS42L56_DSP_MUTE_CTL, 4, 5, 1, 1),
 
 	SOC_SINGLE_TLV("Analog Advisory Volume",
@@ -438,16 +438,16 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = {
 			  CS42L56_DIGINPUT_ADV_VOLUME, 0, 0x00, 1, adv_tlv),
 
 	SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L56_PGAA_MUX_VOLUME,
-			      CS42L56_PGAB_MUX_VOLUME, 0, 0x34, 0xfd, pga_tlv),
+			      CS42L56_PGAB_MUX_VOLUME, 0, 0x34, 0x24, pga_tlv),
 	SOC_DOUBLE_R_TLV("ADC Volume", CS42L56_ADCA_ATTENUATOR,
 			      CS42L56_ADCB_ATTENUATOR, 0, 0x00, 1, adc_tlv),
 	SOC_DOUBLE("ADC Mute Switch", CS42L56_MISC_ADC_CTL, 2, 3, 1, 1),
 	SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1),
 
 	SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME,
-			      CS42L56_HPB_VOLUME, 0, 0x44, 0x55, hl_tlv),
+			      CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv),
 	SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME,
-			      CS42L56_LOB_VOLUME, 0, 0x44, 0x55, hl_tlv),
+			      CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv),
 
 	SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL,
 			0, 0x00, 1, tone_tlv),

sound/soc/codecs/cs42l73.c

@@ -401,7 +401,7 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = {
 			CS42L73_LOBAVOL, 0, 0x41, 0x4B, hpaloa_tlv),
 
 	SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL,
-			CS42L73_MICBPREPGABVOL, 5, 0x34,
+			CS42L73_MICBPREPGABVOL, 0, 0x34,
 			0x24, micpga_tlv),
 
 	SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL,

sound/soc/codecs/twl4030.c

@@ -344,17 +344,16 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
 static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable)
 {
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
-	int status = -1;
 
 	if (enable) {
 		twl4030->apll_enabled++;
 		if (twl4030->apll_enabled == 1)
-			status = twl4030_audio_enable_resource(
+			twl4030_audio_enable_resource(
 							TWL4030_AUDIO_RES_APLL);
 	} else {
 		twl4030->apll_enabled--;
 		if (!twl4030->apll_enabled)
-			status = twl4030_audio_disable_resource(
+			twl4030_audio_disable_resource(
 							TWL4030_AUDIO_RES_APLL);
 	}
 }

sound/soc/codecs/wm8350.c

@@ -1341,21 +1341,18 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which,
 {
 	struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec);
 	struct wm8350 *wm8350 = priv->wm8350;
-	int irq;
 	int ena;
 
 	switch (which) {
 	case WM8350_JDL:
 		priv->hpl.jack = jack;
 		priv->hpl.report = report;
-		irq = WM8350_IRQ_CODEC_JCK_DET_L;
 		ena = WM8350_JDL_ENA;
 		break;
 
 	case WM8350_JDR:
 		priv->hpr.jack = jack;
 		priv->hpr.report = report;
-		irq = WM8350_IRQ_CODEC_JCK_DET_R;
 		ena = WM8350_JDR_ENA;
 		break;
 

sound/soc/codecs/wm8996.c

@@ -620,15 +620,12 @@ static int bg_event(struct snd_soc_dapm_widget *w,
 static int cp_event(struct snd_soc_dapm_widget *w,
 		    struct snd_kcontrol *kcontrol, int event)
 {
-	int ret = 0;
-
 	switch (event) {
 	case SND_SOC_DAPM_POST_PMU:
 		msleep(5);
 		break;
 	default:
 		WARN(1, "Invalid event %d\n", event);
-		ret = -EINVAL;
 	}
 
 	return 0;