Merge remote-tracking branches 'asoc/topic/fsl', 'asoc/topic/fsl-asrc',...

Merge remote-tracking branches 'asoc/topic/fsl', 'asoc/topic/fsl-asrc', 'asoc/topic/fsl-spdif' and 'asoc/topic/imx-audmux' into asoc-next

Documentation/devicetree/bindings/sound/fsl,asrc.txt

+Freescale Asynchronous Sample Rate Converter (ASRC) Controller
+
+The Asynchronous Sample Rate Converter (ASRC) converts the sampling rate of a
+signal associated with an input clock into a signal associated with a different
+output clock. The driver currently works as a Front End of DPCM with other Back
+Ends Audio controller such as ESAI, SSI and SAI. It has three pairs to support
+three substreams within totally 10 channels.
+
+Required properties:
+
+  - compatible		: Contains "fsl,imx35-asrc" or "fsl,imx53-asrc".
+
+  - reg			: Offset and length of the register set for the device.
+
+  - interrupts		: Contains the spdif interrupt.
+
+  - dmas		: Generic dma devicetree binding as described in
+			  Documentation/devicetree/bindings/dma/dma.txt.
+
+  - dma-names		: Contains "rxa", "rxb", "rxc", "txa", "txb" and "txc".
+
+  - clocks		: Contains an entry for each entry in clock-names.
+
+  - clock-names		: Contains the following entries
+	"mem"		  Peripheral access clock to access registers.
+	"ipg"		  Peripheral clock to driver module.
+	"asrck_<0-f>"	  Clock sources for input and output clock.
+
+   - big-endian		: If this property is absent, the little endian mode
+			  will be in use as default. Otherwise, the big endian
+			  mode will be in use for all the device registers.
+
+   - fsl,asrc-rate	: Defines a mutual sample rate used by DPCM Back Ends.
+
+   - fsl,asrc-width	: Defines a mutual sample width used by DPCM Back Ends.
+
+Example:
+
+asrc: asrc@02034000 {
+	compatible = "fsl,imx53-asrc";
+	reg = <0x02034000 0x4000>;
+	interrupts = <0 50 IRQ_TYPE_LEVEL_HIGH>;
+	clocks = <&clks 107>, <&clks 107>, <&clks 0>,
+	       <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>,
+	       <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>,
+	       <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>,
+	       <&clks 107>, <&clks 0>, <&clks 0>;
+	clock-names = "mem", "ipg", "asrck0",
+		"asrck_1", "asrck_2", "asrck_3", "asrck_4",
+		"asrck_5", "asrck_6", "asrck_7", "asrck_8",
+		"asrck_9", "asrck_a", "asrck_b", "asrck_c",
+		"asrck_d", "asrck_e", "asrck_f";
+	dmas = <&sdma 17 23 1>, <&sdma 18 23 1>, <&sdma 19 23 1>,
+	     <&sdma 20 23 1>, <&sdma 21 23 1>, <&sdma 22 23 1>;
+	dma-names = "rxa", "rxb", "rxc",
+		"txa", "txb", "txc";
+	fsl,asrc-rate  = <48000>;
+	fsl,asrc-width = <16>;
+	status = "okay";
+};

include/linux/platform_data/dma-imx.h

@@ -50,6 +50,7 @@ enum imx_dma_prio {
 
 struct imx_dma_data {
 	int dma_request; /* DMA request line */
+	int dma_request2; /* secondary DMA request line */
 	enum sdma_peripheral_type peripheral_type;
 	int priority;
 };

sound/soc/fsl/Kconfig

@@ -2,9 +2,20 @@ menu "SoC Audio for Freescale CPUs"
 
 comment "Common SoC Audio options for Freescale CPUs:"
 
+config SND_SOC_FSL_ASRC
+	tristate "Asynchronous Sample Rate Converter (ASRC) module support"
+	select REGMAP_MMIO
+	select SND_SOC_GENERIC_DMAENGINE_PCM
+	help
+	  Say Y if you want to add Asynchronous Sample Rate Converter (ASRC)
+	  support for the Freescale CPUs.
+	  This option is only useful for out-of-tree drivers since
+	  in-tree drivers select it automatically.
+
 config SND_SOC_FSL_SAI
 	tristate "Synchronous Audio Interface (SAI) module support"
 	select REGMAP_MMIO
+	select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n
 	select SND_SOC_GENERIC_DMAENGINE_PCM
 	help
 	  Say Y if you want to add Synchronous Audio Interface (SAI)
@@ -15,7 +26,7 @@ config SND_SOC_FSL_SAI
 config SND_SOC_FSL_SSI
 	tristate "Synchronous Serial Interface module support"
 	select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n
-	select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && ARCH_MXC
+	select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC)
 	select REGMAP_MMIO
 	help
 	  Say Y if you want to add Synchronous Serial Interface (SSI)
@@ -27,7 +38,7 @@ config SND_SOC_FSL_SPDIF
 	tristate "Sony/Philips Digital Interface module support"
 	select REGMAP_MMIO
 	select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n
-	select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && ARCH_MXC
+	select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC)
 	help
 	  Say Y if you want to add Sony/Philips Digital Interface (SPDIF)
 	  support for the Freescale CPUs.
@@ -37,6 +48,7 @@ config SND_SOC_FSL_SPDIF
 config SND_SOC_FSL_ESAI
 	tristate "Enhanced Serial Audio Interface (ESAI) module support"
 	select REGMAP_MMIO
+	select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n
 	select SND_SOC_FSL_UTILS
 	help
 	  Say Y if you want to add Enhanced Synchronous Audio Interface

sound/soc/fsl/Makefile

@@ -11,6 +11,7 @@ snd-soc-p1022-rdk-objs := p1022_rdk.o
 obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o
 
 # Freescale SSI/DMA/SAI/SPDIF Support
+snd-soc-fsl-asrc-objs := fsl_asrc.o fsl_asrc_dma.o
 snd-soc-fsl-sai-objs := fsl_sai.o
 snd-soc-fsl-ssi-y := fsl_ssi.o
 snd-soc-fsl-ssi-$(CONFIG_DEBUG_FS) += fsl_ssi_dbg.o
@@ -18,6 +19,7 @@ snd-soc-fsl-spdif-objs := fsl_spdif.o
 snd-soc-fsl-esai-objs := fsl_esai.o
 snd-soc-fsl-utils-objs := fsl_utils.o
 snd-soc-fsl-dma-objs := fsl_dma.o
+obj-$(CONFIG_SND_SOC_FSL_ASRC) += snd-soc-fsl-asrc.o
 obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o
 obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o
 obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o

sound/soc/fsl/fsl_esai.c

@@ -624,12 +624,14 @@ static int fsl_esai_dai_probe(struct snd_soc_dai *dai)
 static struct snd_soc_dai_driver fsl_esai_dai = {
 	.probe = fsl_esai_dai_probe,
 	.playback = {
+		.stream_name = "CPU-Playback",
 		.channels_min = 1,
 		.channels_max = 12,
 		.rates = FSL_ESAI_RATES,
 		.formats = FSL_ESAI_FORMATS,
 	},
 	.capture = {
+		.stream_name = "CPU-Capture",
 		.channels_min = 1,
 		.channels_max = 8,
 		.rates = FSL_ESAI_RATES,

sound/soc/fsl/fsl_sai.c

@@ -455,12 +455,14 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai)
 static struct snd_soc_dai_driver fsl_sai_dai = {
 	.probe = fsl_sai_dai_probe,
 	.playback = {
+		.stream_name = "CPU-Playback",
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000_96000,
 		.formats = FSL_SAI_FORMATS,
 	},
 	.capture = {
+		.stream_name = "CPU-Capture",
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000_96000,

sound/soc/fsl/fsl_spdif.c

@@ -33,9 +33,12 @@
 #define FSL_SPDIF_RXFIFO_WML	0x8
 
 #define INTR_FOR_PLAYBACK	(INT_TXFIFO_RESYNC)
-#define INTR_FOR_CAPTURE (INT_SYM_ERR | INT_BIT_ERR | INT_URX_FUL | INT_URX_OV|\
-		INT_QRX_FUL | INT_QRX_OV | INT_UQ_SYNC | INT_UQ_ERR |\
-		INT_RXFIFO_RESYNC | INT_LOSS_LOCK | INT_DPLL_LOCKED)
+#define INTR_FOR_CAPTURE	(INT_SYM_ERR | INT_BIT_ERR | INT_URX_FUL |\
+				INT_URX_OV | INT_QRX_FUL | INT_QRX_OV |\
+				INT_UQ_SYNC | INT_UQ_ERR | INT_RXFIFO_RESYNC |\
+				INT_LOSS_LOCK | INT_DPLL_LOCKED)
+
+#define SIE_INTR_FOR(tx)	(tx ? INTR_FOR_PLAYBACK : INTR_FOR_CAPTURE)
 
 /* Index list for the values that has if (DPLL Locked) condition */
 static u8 srpc_dpll_locked[] = { 0x0, 0x1, 0x2, 0x3, 0x4, 0xa, 0xb };
@@ -96,7 +99,7 @@ struct fsl_spdif_priv {
 	struct platform_device *pdev;
 	struct regmap *regmap;
 	bool dpll_locked;
-	u16 txrate[SPDIF_TXRATE_MAX];
+	u32 txrate[SPDIF_TXRATE_MAX];
 	u8 txclk_df[SPDIF_TXRATE_MAX];
 	u8 sysclk_df[SPDIF_TXRATE_MAX];
 	u8 txclk_src[SPDIF_TXRATE_MAX];
@@ -137,10 +140,9 @@ static void spdif_irq_sym_error(struct fsl_spdif_priv *spdif_priv)
 
 	dev_dbg(&pdev->dev, "isr: receiver found illegal symbol\n");
 
-	if (!spdif_priv->dpll_locked) {
-		/* DPLL unlocked seems no audio stream */
+	/* Clear illegal symbol if DPLL unlocked since no audio stream */
+	if (!spdif_priv->dpll_locked)
 		regmap_update_bits(regmap, REG_SPDIF_SIE, INT_SYM_ERR, 0);
-	}
 }
 
 /* U/Q Channel receive register full */
@@ -390,6 +392,14 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream,
 		rate = SPDIF_TXRATE_48000;
 		csfs = IEC958_AES3_CON_FS_48000;
 		break;
+	case 96000:
+		rate = SPDIF_TXRATE_96000;
+		csfs = IEC958_AES3_CON_FS_96000;
+		break;
+	case 192000:
+		rate = SPDIF_TXRATE_192000;
+		csfs = IEC958_AES3_CON_FS_192000;
+		break;
 	default:
 		dev_err(&pdev->dev, "unsupported sample rate %d\n", sample_rate);
 		return -EINVAL;
@@ -433,13 +443,12 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream,
 	spdif_set_cstatus(ctrl, IEC958_AES3_CON_FS, csfs);
 
 	/* select clock source and divisor */
-	stc = STC_TXCLK_ALL_EN | STC_TXCLK_SRC_SET(clk) | STC_TXCLK_DF(txclk_df);
-	mask = STC_TXCLK_ALL_EN_MASK | STC_TXCLK_SRC_MASK | STC_TXCLK_DF_MASK;
+	stc = STC_TXCLK_ALL_EN | STC_TXCLK_SRC_SET(clk) |
+	      STC_TXCLK_DF(txclk_df) | STC_SYSCLK_DF(sysclk_df);
+	mask = STC_TXCLK_ALL_EN_MASK | STC_TXCLK_SRC_MASK |
+	       STC_TXCLK_DF_MASK | STC_SYSCLK_DF_MASK;
 	regmap_update_bits(regmap, REG_SPDIF_STC, mask, stc);
 
-	regmap_update_bits(regmap, REG_SPDIF_STC,
-			   STC_SYSCLK_DF_MASK, STC_SYSCLK_DF(sysclk_df));
-
 	dev_dbg(&pdev->dev, "set sample rate to %dHz for %dHz playback\n",
 			spdif_priv->txrate[rate], sample_rate);
 
@@ -569,9 +578,9 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
 	struct regmap *regmap = spdif_priv->regmap;
-	int is_playack = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
-	u32 intr = is_playack ? INTR_FOR_PLAYBACK : INTR_FOR_CAPTURE;
-	u32 dmaen = is_playack ? SCR_DMA_TX_EN : SCR_DMA_RX_EN;;
+	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+	u32 intr = SIE_INTR_FOR(tx);
+	u32 dmaen = SCR_DMA_xX_EN(tx);
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
@@ -662,9 +671,8 @@ static int fsl_spdif_capture_get(struct snd_kcontrol *kcontrol,
 	u32 cstatus, val;
 
 	regmap_read(regmap, REG_SPDIF_SIS, &val);
-	if (!(val & INT_CNEW)) {
+	if (!(val & INT_CNEW))
 		return -EAGAIN;
-	}
 
 	regmap_read(regmap, REG_SPDIF_SRCSH, &cstatus);
 	ucontrol->value.iec958.status[0] = (cstatus >> 16) & 0xFF;
@@ -693,15 +701,14 @@ static int fsl_spdif_subcode_get(struct snd_kcontrol *kcontrol,
 	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
 	struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
 	unsigned long flags;
-	int ret = 0;
+	int ret = -EAGAIN;
 
 	spin_lock_irqsave(&ctrl->ctl_lock, flags);
 	if (ctrl->ready_buf) {
 		int idx = (ctrl->ready_buf - 1) * SPDIF_UBITS_SIZE;
 		memcpy(&ucontrol->value.iec958.subcode[0],
 				&ctrl->subcode[idx], SPDIF_UBITS_SIZE);
-	} else {
-		ret = -EAGAIN;
+		ret = 0;
 	}
 	spin_unlock_irqrestore(&ctrl->ctl_lock, flags);
 
@@ -726,15 +733,14 @@ static int fsl_spdif_qget(struct snd_kcontrol *kcontrol,
 	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
 	struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
 	unsigned long flags;
-	int ret = 0;
+	int ret = -EAGAIN;
 
 	spin_lock_irqsave(&ctrl->ctl_lock, flags);
 	if (ctrl->ready_buf) {
 		int idx = (ctrl->ready_buf - 1) * SPDIF_QSUB_SIZE;
 		memcpy(&ucontrol->value.bytes.data[0],
 				&ctrl->qsub[idx], SPDIF_QSUB_SIZE);
-	} else {
-		ret = -EAGAIN;
+		ret = 0;
 	}
 	spin_unlock_irqrestore(&ctrl->ctl_lock, flags);
 
@@ -799,10 +805,10 @@ static int spdif_get_rxclk_rate(struct fsl_spdif_priv *spdif_priv,
 	regmap_read(regmap, REG_SPDIF_SRPC, &phaseconf);
 
 	clksrc = (phaseconf >> SRPC_CLKSRC_SEL_OFFSET) & 0xf;
-	if (srpc_dpll_locked[clksrc] && (phaseconf & SRPC_DPLL_LOCKED)) {
+
 	/* Get bus clock from system */
+	if (srpc_dpll_locked[clksrc] && (phaseconf & SRPC_DPLL_LOCKED))
 		busclk_freq = clk_get_rate(spdif_priv->sysclk);
-	}
 
 	/* FreqMeas_CLK = (BUS_CLK * FreqMeas) / 2 ^ 10 / GAINSEL / 128 */
 	tmpval64 = (u64) busclk_freq * freqmeas;
@@ -826,12 +832,12 @@ static int fsl_spdif_rxrate_get(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
 	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
-	int rate = spdif_get_rxclk_rate(spdif_priv, SPDIF_DEFAULT_GAINSEL);
+	int rate = 0;
 
 	if (spdif_priv->dpll_locked)
+		rate = spdif_get_rxclk_rate(spdif_priv, SPDIF_DEFAULT_GAINSEL);
+
 	ucontrol->value.integer.value[0] = rate;
-	else
-		ucontrol->value.integer.value[0] = 0;
 
 	return 0;
 }
@@ -969,12 +975,14 @@ static int fsl_spdif_dai_probe(struct snd_soc_dai *dai)
 static struct snd_soc_dai_driver fsl_spdif_dai = {
 	.probe = &fsl_spdif_dai_probe,
 	.playback = {
+		.stream_name = "CPU-Playback",
 		.channels_min = 2,
 		.channels_max = 2,
 		.rates = FSL_SPDIF_RATES_PLAYBACK,
 		.formats = FSL_SPDIF_FORMATS_PLAYBACK,
 	},
 	.capture = {
+		.stream_name = "CPU-Capture",
 		.channels_min = 2,
 		.channels_max = 2,
 		.rates = FSL_SPDIF_RATES_CAPTURE,
@@ -1046,7 +1054,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv,
 				struct clk *clk, u64 savesub,
 				enum spdif_txrate index, bool round)
 {
-	const u32 rate[] = { 32000, 44100, 48000 };
+	const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 };
 	bool is_sysclk = clk == spdif_priv->sysclk;
 	u64 rate_ideal, rate_actual, sub;
 	u32 sysclk_dfmin, sysclk_dfmax;
@@ -1105,7 +1113,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv,
 static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv,
 				enum spdif_txrate index)
 {
-	const u32 rate[] = { 32000, 44100, 48000 };
+	const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 };
 	struct platform_device *pdev = spdif_priv->pdev;
 	struct device *dev = &pdev->dev;
 	u64 savesub = 100000, ret;
@@ -1238,12 +1246,12 @@ static int fsl_spdif_probe(struct platform_device *pdev)
 	spin_lock_init(&ctrl->ctl_lock);
 
 	/* Init tx channel status default value */
-	ctrl->ch_status[0] =
-		IEC958_AES0_CON_NOT_COPYRIGHT | IEC958_AES0_CON_EMPHASIS_5015;
+	ctrl->ch_status[0] = IEC958_AES0_CON_NOT_COPYRIGHT |
+			     IEC958_AES0_CON_EMPHASIS_5015;
 	ctrl->ch_status[1] = IEC958_AES1_CON_DIGDIGCONV_ID;
 	ctrl->ch_status[2] = 0x00;
-	ctrl->ch_status[3] =
-		IEC958_AES3_CON_FS_44100 | IEC958_AES3_CON_CLOCK_1000PPM;
+	ctrl->ch_status[3] = IEC958_AES3_CON_FS_44100 |
+			     IEC958_AES3_CON_CLOCK_1000PPM;
 
 	spdif_priv->dpll_locked = false;
 

sound/soc/fsl/fsl_spdif.h

@@ -93,6 +93,8 @@
 #define SCR_USRC_SEL_RECV		(0x1 << SCR_USRC_SEL_OFFSET)
 #define SCR_USRC_SEL_CHIP		(0x3 << SCR_USRC_SEL_OFFSET)
 
+#define SCR_DMA_xX_EN(tx)		(tx ? SCR_DMA_TX_EN : SCR_DMA_RX_EN)
+
 /* SPDIF CDText control */
 #define SRCD_CD_USER_OFFSET		1
 #define SRCD_CD_USER			(1 << SRCD_CD_USER_OFFSET)
@@ -164,8 +166,10 @@ enum spdif_txrate {
 	SPDIF_TXRATE_32000 = 0,
 	SPDIF_TXRATE_44100,
 	SPDIF_TXRATE_48000,
+	SPDIF_TXRATE_96000,
+	SPDIF_TXRATE_192000,
 };
-#define SPDIF_TXRATE_MAX		(SPDIF_TXRATE_48000 + 1)
+#define SPDIF_TXRATE_MAX		(SPDIF_TXRATE_192000 + 1)
 
 
 #define SPDIF_CSTATUS_BYTE		6
@@ -175,7 +179,9 @@ enum spdif_txrate {
 
 #define FSL_SPDIF_RATES_PLAYBACK	(SNDRV_PCM_RATE_32000 |	\
 					 SNDRV_PCM_RATE_44100 |	\
-					 SNDRV_PCM_RATE_48000)
+					 SNDRV_PCM_RATE_48000 |	\
+					 SNDRV_PCM_RATE_96000 |	\
+					 SNDRV_PCM_RATE_192000)
 
 #define FSL_SPDIF_RATES_CAPTURE		(SNDRV_PCM_RATE_16000 | \
 					 SNDRV_PCM_RATE_32000 |	\

sound/soc/fsl/fsl_ssi.c

@@ -1032,12 +1032,14 @@ static const struct snd_soc_dai_ops fsl_ssi_dai_ops = {
 static struct snd_soc_dai_driver fsl_ssi_dai_template = {
 	.probe = fsl_ssi_dai_probe,
 	.playback = {
+		.stream_name = "CPU-Playback",
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = FSLSSI_I2S_RATES,
 		.formats = FSLSSI_I2S_FORMATS,
 	},
 	.capture = {
+		.stream_name = "CPU-Capture",
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = FSLSSI_I2S_RATES,

sound/soc/fsl/imx-audmux.c

@@ -67,7 +67,7 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf,
 {
 	ssize_t ret;
 	char *buf;
-	int port = (int)file->private_data;
+	uintptr_t port = (uintptr_t)file->private_data;
 	u32 pdcr, ptcr;
 
 	if (audmux_clk) {
@@ -147,7 +147,7 @@ static const struct file_operations audmux_debugfs_fops = {
 
 static void audmux_debugfs_init(void)
 {
-	int i;
+	uintptr_t i;
 	char buf[20];
 
 	audmux_debugfs_root = debugfs_create_dir("audmux", NULL);
@@ -157,10 +157,10 @@ static void audmux_debugfs_init(void)
 	}
 
 	for (i = 0; i < MX31_AUDMUX_PORT7_SSI_PINS_7 + 1; i++) {
-		snprintf(buf, sizeof(buf), "ssi%d", i);
+		snprintf(buf, sizeof(buf), "ssi%lu", i);
 		if (!debugfs_create_file(buf, 0444, audmux_debugfs_root,
 					 (void *)i, &audmux_debugfs_fops))
-			pr_warning("Failed to create AUDMUX port %d debugfs file\n",
+			pr_warning("Failed to create AUDMUX port %lu debugfs file\n",
 				   i);
 	}
 }