Merge remote-tracking branches 'asoc/topic/nau8824', 'asoc/topic/omap',...

Merge remote-tracking branches 'asoc/topic/nau8824', 'asoc/topic/omap', 'asoc/topic/pxa' and 'asoc/topic/qcom' into asoc-next

Documentation/devicetree/bindings/sound/nau8824.txt

+Nuvoton NAU8824 audio codec
+
+This device supports I2C only.
+
+Required properties:
+  - compatible : Must be "nuvoton,nau8824"
+
+  - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1).
+
+Optional properties:
+  - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low.
+
+  - nuvoton,vref-impedance: VREF Impedance selection
+      0 - Open
+      1 - 25 kOhm
+      2 - 125 kOhm
+      3 - 2.5 kOhm
+
+  - nuvoton,micbias-voltage: Micbias voltage level.
+      0 - VDDA
+      1 - VDDA
+      2 - VDDA * 1.1
+      3 - VDDA * 1.2
+      4 - VDDA * 1.3
+      5 - VDDA * 1.4
+      6 - VDDA * 1.53
+      7 - VDDA * 1.53
+
+  - nuvoton,sar-threshold-num: Number of buttons supported
+  - nuvoton,sar-threshold: Impedance threshold for each button. Array that contains up to 8 buttons configuration. SAR value is calculated as
+    SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R)
+    where MICBIAS is configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by 'nuvoton,sar-voltage', R - button impedance.
+    Refer datasheet section 10.2 for more information about threshold calculation.
+
+  - nuvoton,sar-hysteresis: Button impedance measurement hysteresis.
+
+  - nuvoton,sar-voltage: Reference voltage for button impedance measurement.
+      0 - VDDA
+      1 - VDDA
+      2 - VDDA * 1.1
+      3 - VDDA * 1.2
+      4 - VDDA * 1.3
+      5 - VDDA * 1.4
+      6 - VDDA * 1.53
+      7 - VDDA * 1.53
+
+  - nuvoton,sar-compare-time: SAR compare time
+      0 - 500 ns
+      1 - 1 us
+      2 - 2 us
+      3 - 4 us
+
+  - nuvoton,sar-sampling-time: SAR sampling time
+      0 - 2 us
+      1 - 4 us
+      2 - 8 us
+      3 - 16 us
+
+  - nuvoton,short-key-debounce: Button short key press debounce time.
+      0 - 30 ms
+      1 - 50 ms
+      2 - 100 ms
+
+  - nuvoton,jack-eject-debounce: Jack ejection debounce time.
+      0 - 0 ms
+      1 - 1 ms
+      2 - 10 ms
+
+
+Example:
+
+  headset: nau8824@1a {
+      compatible = "nuvoton,nau8824";
+      reg = <0x1a>;
+      interrupt-parent = <&gpio>;
+      interrupts = <TEGRA_GPIO(E, 6) IRQ_TYPE_LEVEL_LOW>;
+      nuvoton,vref-impedance = <2>;
+      nuvoton,micbias-voltage = <6>;
+      // Setup 4 buttons impedance according to Android specification
+      nuvoton,sar-threshold-num = <4>;
+      nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>;
+      nuvoton,sar-hysteresis = <0>;
+      nuvoton,sar-voltage = <6>;
+      nuvoton,sar-compare-time = <1>;
+      nuvoton,sar-sampling-time = <1>;
+      nuvoton,short-key-debounce = <0>;
+      nuvoton,jack-eject-debounce = <1>;
+  };

sound/soc/codecs/Kconfig

@@ -101,6 +101,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_ML26124 if I2C
 	select SND_SOC_NAU8540 if I2C
 	select SND_SOC_NAU8810 if I2C
+	select SND_SOC_NAU8824 if I2C
 	select SND_SOC_NAU8825 if I2C
 	select SND_SOC_HDMI_CODEC
 	select SND_SOC_PCM1681 if I2C
@@ -1137,6 +1138,10 @@ config SND_SOC_NAU8810
 	tristate "Nuvoton Technology Corporation NAU88C10 CODEC"
 	depends on I2C
 
+config SND_SOC_NAU8824
+	tristate "Nuvoton Technology Corporation NAU88L24 CODEC"
+	depends on I2C
+
 config SND_SOC_NAU8825
 	tristate
 

sound/soc/codecs/Makefile

@@ -95,6 +95,7 @@ snd-soc-msm8916-analog-objs := msm8916-wcd-analog.o
 snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o
 snd-soc-nau8540-objs := nau8540.o
 snd-soc-nau8810-objs := nau8810.o
+snd-soc-nau8824-objs := nau8824.o
 snd-soc-nau8825-objs := nau8825.o
 snd-soc-hdmi-codec-objs := hdmi-codec.o
 snd-soc-pcm1681-objs := pcm1681.o
@@ -328,6 +329,7 @@ obj-$(CONFIG_SND_SOC_MSM8916_WCD_ANALOG) +=snd-soc-msm8916-analog.o
 obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o
 obj-$(CONFIG_SND_SOC_NAU8540)   += snd-soc-nau8540.o
 obj-$(CONFIG_SND_SOC_NAU8810)   += snd-soc-nau8810.o
+obj-$(CONFIG_SND_SOC_NAU8824)   += snd-soc-nau8824.o
 obj-$(CONFIG_SND_SOC_NAU8825)   += snd-soc-nau8825.o
 obj-$(CONFIG_SND_SOC_HDMI_CODEC)	+= snd-soc-hdmi-codec.o
 obj-$(CONFIG_SND_SOC_PCM1681)	+= snd-soc-pcm1681.o

sound/soc/omap/am3517evm.c

@@ -49,7 +49,7 @@ static int am3517evm_hw_params(struct snd_pcm_substream *substream,
 	return ret;
 }
 
-static struct snd_soc_ops am3517evm_ops = {
+static const struct snd_soc_ops am3517evm_ops = {
 	.hw_params = am3517evm_hw_params,
 };
 

sound/soc/omap/n810.c

@@ -124,7 +124,7 @@ static int n810_hw_params(struct snd_pcm_substream *substream,
 	return err;
 }
 
-static struct snd_soc_ops n810_ops = {
+static const struct snd_soc_ops n810_ops = {
 	.startup = n810_startup,
 	.hw_params = n810_hw_params,
 	.shutdown = n810_shutdown,

sound/soc/omap/omap-abe-twl6040.c

@@ -70,7 +70,7 @@ static int omap_abe_hw_params(struct snd_pcm_substream *substream,
 	return ret;
 }
 
-static struct snd_soc_ops omap_abe_ops = {
+static const struct snd_soc_ops omap_abe_ops = {
 	.hw_params = omap_abe_hw_params,
 };
 

sound/soc/omap/omap-twl4030.c

@@ -73,7 +73,7 @@ static int omap_twl4030_hw_params(struct snd_pcm_substream *substream,
 	return snd_soc_runtime_set_dai_fmt(rtd, fmt);
 }
 
-static struct snd_soc_ops omap_twl4030_ops = {
+static const struct snd_soc_ops omap_twl4030_ops = {
 	.hw_params = omap_twl4030_hw_params,
 };
 

sound/soc/omap/omap3pandora.c

@@ -184,7 +184,7 @@ static int omap3pandora_in_init(struct snd_soc_pcm_runtime *rtd)
 	return 0;
 }
 
-static struct snd_soc_ops omap3pandora_ops = {
+static const struct snd_soc_ops omap3pandora_ops = {
 	.hw_params = omap3pandora_hw_params,
 };
 

sound/soc/omap/osk5912.c

@@ -68,7 +68,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream,
 	return err;
 }
 
-static struct snd_soc_ops osk_ops = {
+static const struct snd_soc_ops osk_ops = {
 	.startup = osk_startup,
 	.hw_params = osk_hw_params,
 	.shutdown = osk_shutdown,

sound/soc/omap/rx51.c

@@ -123,7 +123,7 @@ static int rx51_hw_params(struct snd_pcm_substream *substream,
 				      SND_SOC_CLOCK_IN);
 }
 
-static struct snd_soc_ops rx51_ops = {
+static const struct snd_soc_ops rx51_ops = {
 	.startup = rx51_startup,
 	.hw_params = rx51_hw_params,
 };
@@ -433,10 +433,9 @@ static int rx51_soc_probe(struct platform_device *pdev)
 	}
 
 	pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL);
-	if (pdata == NULL) {
-		dev_err(card->dev, "failed to create private data\n");
+	if (pdata == NULL)
 		return -ENOMEM;
-	}
+
 	snd_soc_card_set_drvdata(card, pdata);
 
 	pdata->tvout_selection_gpio = devm_gpiod_get(card->dev,

sound/soc/pxa/brownstone.c

@@ -74,7 +74,7 @@ static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream,
 }
 
 /* machine stream operations */
-static struct snd_soc_ops brownstone_ops = {
+static const struct snd_soc_ops brownstone_ops = {
 	.hw_params = brownstone_wm8994_hw_params,
 };
 

sound/soc/pxa/corgi.c

@@ -154,7 +154,7 @@ static int corgi_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_soc_ops corgi_ops = {
+static const struct snd_soc_ops corgi_ops = {
 	.startup = corgi_startup,
 	.hw_params = corgi_hw_params,
 	.shutdown = corgi_shutdown,

sound/soc/pxa/e750_wm9705.c

@@ -81,7 +81,7 @@ static struct snd_soc_dai_link e750_dai[] = {
 		.name = "AC97 Aux",
 		.stream_name = "AC97 Aux",
 		.cpu_dai_name = "pxa2xx-ac97-aux",
-		.codec_dai_name ="wm9705-aux",
+		.codec_dai_name = "wm9705-aux",
 		.platform_name = "pxa-pcm-audio",
 		.codec_name = "wm9705-codec",
 	},

sound/soc/pxa/e800_wm9712.c

@@ -81,7 +81,7 @@ static struct snd_soc_dai_link e800_dai[] = {
 		.name = "AC97 Aux",
 		.stream_name = "AC97 Aux",
 		.cpu_dai_name = "pxa2xx-ac97-aux",
-		.codec_dai_name ="wm9712-aux",
+		.codec_dai_name = "wm9712-aux",
 		.platform_name = "pxa-pcm-audio",
 		.codec_name = "wm9712-codec",
 	},

sound/soc/pxa/em-x270.c

@@ -43,7 +43,7 @@ static struct snd_soc_dai_link em_x270_dai[] = {
 		.name = "AC97 Aux",
 		.stream_name = "AC97 Aux",
 		.cpu_dai_name = "pxa2xx-ac97-aux",
-		.codec_dai_name ="wm9712-aux",
+		.codec_dai_name = "wm9712-aux",
 		.platform_name = "pxa-pcm-audio",
 		.codec_name = "wm9712-codec",
 	},

sound/soc/pxa/hx4700.c

@@ -79,7 +79,7 @@ static int hx4700_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_soc_ops hx4700_ops = {
+static const struct snd_soc_ops hx4700_ops = {
 	.hw_params = hx4700_hw_params,
 };
 

sound/soc/pxa/imote2.c

@@ -42,7 +42,7 @@ static int imote2_asoc_hw_params(struct snd_pcm_substream *substream,
 	return ret;
 }
 
-static struct snd_soc_ops imote2_asoc_ops = {
+static const struct snd_soc_ops imote2_asoc_ops = {
 	.hw_params = imote2_asoc_hw_params,
 };
 

sound/soc/pxa/magician.c

@@ -255,12 +255,12 @@ static int magician_capture_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_soc_ops magician_capture_ops = {
+static const struct snd_soc_ops magician_capture_ops = {
 	.startup = magician_startup,
 	.hw_params = magician_capture_hw_params,
 };
 
-static struct snd_soc_ops magician_playback_ops = {
+static const struct snd_soc_ops magician_playback_ops = {
 	.startup = magician_startup,
 	.hw_params = magician_playback_hw_params,
 };

sound/soc/pxa/mioa701_wm9713.c

@@ -157,7 +157,7 @@ static struct snd_soc_dai_link mioa701_dai[] = {
 		.name = "AC97 Aux",
 		.stream_name = "AC97 Aux",
 		.cpu_dai_name = "pxa2xx-ac97-aux",
-		.codec_dai_name ="wm9713-aux",
+		.codec_dai_name = "wm9713-aux",
 		.codec_name = "wm9713-codec",
 		.platform_name = "pxa-pcm-audio",
 		.ops = &mioa701_ops,

sound/soc/pxa/mmp-pcm.c

@@ -166,7 +166,6 @@ static void mmp_pcm_free_dma_buffers(struct snd_pcm *pcm)
 		buf->area = NULL;
 	}
 
-	return;
 }
 
 static int mmp_pcm_preallocate_dma_buffer(struct snd_pcm_substream *substream,

sound/soc/pxa/mmp-sspa.c

@@ -119,7 +119,6 @@ static void mmp_sspa_shutdown(struct snd_pcm_substream *substream,
 	clk_disable(priv->sspa->clk);
 	clk_disable(priv->sysclk);
 
-	return;
 }
 
 /*

sound/soc/pxa/poodle.c

@@ -129,7 +129,7 @@ static int poodle_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_soc_ops poodle_ops = {
+static const struct snd_soc_ops poodle_ops = {
 	.startup = poodle_startup,
 	.hw_params = poodle_hw_params,
 	.shutdown = poodle_shutdown,

sound/soc/pxa/pxa-ssp.c

@@ -354,6 +354,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
 		if (ssp->type == PXA3xx_SSP) {
 			u32 val;
 			u64 tmp = 19968;
+
 			tmp *= 1000000;
 			do_div(tmp, freq_out);
 			val = tmp;
@@ -590,13 +591,13 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 
 		if ((pxa_ssp_get_scr(ssp) == 4) && (width == 16)) {
 			/* This is a special case where the bitclk is 64fs
-			* and we're not dealing with 2*32 bits of audio
-			* samples.
-			*
-			* The SSP values used for that are all found out by
-			* trying and failing a lot; some of the registers
-			* needed for that mode are only available on PXA3xx.
-			*/
+			 * and we're not dealing with 2*32 bits of audio
+			 * samples.
+			 *
+			 * The SSP values used for that are all found out by
+			 * trying and failing a lot; some of the registers
+			 * needed for that mode are only available on PXA3xx.
+			 */
 			if (ssp->type != PXA3xx_SSP)
 				return -EINVAL;
 

sound/soc/pxa/pxa2xx-ac97.c

@@ -140,9 +140,8 @@ static int pxa2xx_ac97_mic_startup(struct snd_pcm_substream *substream,
 {
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		return -ENODEV;
-	else
-		snd_soc_dai_set_dma_data(cpu_dai, substream,
-					 &pxa2xx_ac97_pcm_mic_mono_in);
+	snd_soc_dai_set_dma_data(cpu_dai, substream,
+				 &pxa2xx_ac97_pcm_mic_mono_in);
 
 	return 0;
 }

sound/soc/pxa/pxa2xx-i2s.c

@@ -46,10 +46,10 @@
 #define SACR0_STRF	(1 << 5)	/* FIFO Select for EFWR Special Function */
 #define SACR0_EFWR	(1 << 4)	/* Enable EFWR Function  */
 #define SACR0_RST	(1 << 3)	/* FIFO, i2s Register Reset */
-#define SACR0_BCKD	(1 << 2) 	/* Bit Clock Direction */
+#define SACR0_BCKD	(1 << 2)	/* Bit Clock Direction */
 #define SACR0_ENB	(1 << 0)	/* Enable I2S Link */
 #define SACR1_ENLBF	(1 << 5)	/* Enable Loopback */
-#define SACR1_DRPL	(1 << 4) 	/* Disable Replaying Function */
+#define SACR1_DRPL	(1 << 4)	/* Disable Replaying Function */
 #define SACR1_DREC	(1 << 3)	/* Disable Recording Function */
 #define SACR1_AMSL	(1 << 0)	/* Specify Alternate Mode */
 
@@ -60,7 +60,7 @@
 #define SASR0_TFS	(1 << 3)	/* Tx FIFO Service Request */
 #define SASR0_BSY	(1 << 2)	/* I2S Busy */
 #define SASR0_RNE	(1 << 1)	/* Rx FIFO Not Empty */
-#define SASR0_TNF	(1 << 0) 	/* Tx FIFO Not Empty */
+#define SASR0_TNF	(1 << 0)	/* Tx FIFO Not Empty */
 
 #define SAICR_ROR	(1 << 6)	/* Clear Rx FIFO Overrun Interrupt */
 #define SAICR_TUR	(1 << 5)	/* Clear Tx FIFO Underrun Interrupt */
@@ -119,7 +119,7 @@ static int pxa_i2s_wait(void)
 	int i;
 
 	/* flush the Rx FIFO */
-	for(i = 0; i < 16; i++)
+	for (i = 0; i < 16; i++)
 		SADR;
 	return 0;
 }

sound/soc/pxa/pxa2xx-pcm.c

@@ -85,7 +85,7 @@ static int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd)
 }
 
 static struct snd_soc_platform_driver pxa2xx_soc_platform = {
-	.ops 	= &pxa2xx_pcm_ops,
+	.ops		= &pxa2xx_pcm_ops,
 	.pcm_new	= pxa2xx_soc_pcm_new,
 	.pcm_free	= pxa2xx_pcm_free_dma_buffers,
 };

sound/soc/pxa/raumfeld.c

@@ -132,7 +132,7 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_soc_ops raumfeld_cs4270_ops = {
+static const struct snd_soc_ops raumfeld_cs4270_ops = {
 	.startup = raumfeld_cs4270_startup,
 	.shutdown = raumfeld_cs4270_shutdown,
 	.hw_params = raumfeld_cs4270_hw_params,
@@ -228,14 +228,12 @@ static struct snd_soc_ops raumfeld_ak4104_ops = {
 	.codec_name	= "spi0.0",			\
 }
 
-static struct snd_soc_dai_link snd_soc_raumfeld_connector_dai[] =
-{
+static struct snd_soc_dai_link snd_soc_raumfeld_connector_dai[] = {
 	DAI_LINK_CS4270,
 	DAI_LINK_AK4104,
 };
 
-static struct snd_soc_dai_link snd_soc_raumfeld_speaker_dai[] =
-{
+static struct snd_soc_dai_link snd_soc_raumfeld_speaker_dai[] = {
 	DAI_LINK_CS4270,
 };
 

sound/soc/pxa/spitz.c

@@ -156,7 +156,7 @@ static int spitz_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_soc_ops spitz_ops = {
+static const struct snd_soc_ops spitz_ops = {
 	.startup = spitz_startup,
 	.hw_params = spitz_hw_params,
 };
@@ -230,8 +230,8 @@ static const struct snd_soc_dapm_route spitz_audio_map[] = {
 	{"Headset Jack", NULL, "ROUT1"},
 
 	/* ext speaker connected to LOUT2, ROUT2  */
-	{"Ext Spk", NULL , "ROUT2"},
-	{"Ext Spk", NULL , "LOUT2"},
+	{"Ext Spk", NULL, "ROUT2"},
+	{"Ext Spk", NULL, "LOUT2"},
 
 	/* mic is connected to input 1 - with bias */
 	{"LINPUT1", NULL, "Mic Bias"},

sound/soc/pxa/tosa.c

@@ -85,7 +85,7 @@ static int tosa_startup(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static struct snd_soc_ops tosa_ops = {
+static const struct snd_soc_ops tosa_ops = {
 	.startup = tosa_startup,
 };
 
@@ -133,7 +133,7 @@ static int tosa_set_spk(struct snd_kcontrol *kcontrol,
 static int tosa_hp_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *k, int event)
 {
-	gpio_set_value(TOSA_GPIO_L_MUTE, SND_SOC_DAPM_EVENT_ON(event) ? 1 :0);
+	gpio_set_value(TOSA_GPIO_L_MUTE, SND_SOC_DAPM_EVENT_ON(event) ? 1 : 0);
 	return 0;
 }
 

sound/soc/pxa/z2.c

@@ -119,8 +119,8 @@ static const struct snd_soc_dapm_route z2_audio_map[] = {
 	{"Headphone Jack", NULL, "ROUT1"},
 
 	/* ext speaker connected to LOUT2, ROUT2  */
-	{"Ext Spk", NULL , "ROUT2"},
-	{"Ext Spk", NULL , "LOUT2"},
+	{"Ext Spk", NULL, "ROUT2"},
+	{"Ext Spk", NULL, "LOUT2"},
 
 	/* mic is connected to R input 2 - with bias */
 	{"RINPUT2", NULL, "Mic Bias"},
@@ -152,7 +152,7 @@ static int z2_wm8750_init(struct snd_soc_pcm_runtime *rtd)
 	return ret;
 }
 
-static struct snd_soc_ops z2_ops = {
+static const struct snd_soc_ops z2_ops = {
 	.hw_params = z2_hw_params,
 };
 

sound/soc/pxa/zylonite.c

@@ -132,7 +132,7 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static struct snd_soc_ops zylonite_voice_ops = {
+static const struct snd_soc_ops zylonite_voice_ops = {
 	.hw_params = zylonite_voice_hw_params,
 };
 

sound/soc/qcom/lpass-apq8016.c

@@ -231,6 +231,18 @@ static struct lpass_variant apq8016_data = {
 	.wrdma_channels		= 2,
 	.dai_driver		= apq8016_lpass_cpu_dai_driver,
 	.num_dai		= ARRAY_SIZE(apq8016_lpass_cpu_dai_driver),
+	.dai_osr_clk_names	= (const char *[]) {
+				"mi2s-osr-clk0",
+				"mi2s-osr-clk1",
+				"mi2s-osr-clk2",
+				"mi2s-osr-clk3",
+				},
+	.dai_bit_clk_names	= (const char *[]) {
+				"mi2s-bit-clk0",
+				"mi2s-bit-clk1",
+				"mi2s-bit-clk2",
+				"mi2s-bit-clk3",
+				},
 	.init			= apq8016_lpass_init,
 	.exit			= apq8016_lpass_exit,
 	.alloc_dma_channel	= apq8016_lpass_alloc_dma_channel,

sound/soc/qcom/lpass-cpu.c

@@ -429,7 +429,6 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev)
 	struct lpass_variant *variant;
 	struct device *dev = &pdev->dev;
 	const struct of_device_id *match;
-	char clk_name[16];
 	int ret, i, dai_id;
 
 	dsp_of_node = of_parse_phandle(pdev->dev.of_node, "qcom,adsp", 0);
@@ -477,31 +476,24 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev)
 
 	for (i = 0; i < variant->num_dai; i++) {
 		dai_id = variant->dai_driver[i].id;
-		if (variant->num_dai > 1)
-			sprintf(clk_name, "mi2s-osr-clk%d", i);
-		else
-			sprintf(clk_name, "mi2s-osr-clk");
-
 		drvdata->mi2s_osr_clk[dai_id] = devm_clk_get(&pdev->dev,
-								clk_name);
+					     variant->dai_osr_clk_names[i]);
 		if (IS_ERR(drvdata->mi2s_osr_clk[dai_id])) {
 			dev_warn(&pdev->dev,
-				"error getting optional mi2s-osr-clk: %ld\n",
+				"%s() error getting optional %s: %ld\n",
+				__func__,
+				variant->dai_osr_clk_names[i],
 				PTR_ERR(drvdata->mi2s_osr_clk[dai_id]));
 
 			drvdata->mi2s_osr_clk[dai_id] = NULL;
 		}
 
-		if (variant->num_dai > 1)
-			sprintf(clk_name, "mi2s-bit-clk%d", i);
-		else
-			sprintf(clk_name, "mi2s-bit-clk");
-
 		drvdata->mi2s_bit_clk[dai_id] = devm_clk_get(&pdev->dev,
-							    clk_name);
+						variant->dai_bit_clk_names[i]);
 		if (IS_ERR(drvdata->mi2s_bit_clk[dai_id])) {
 			dev_err(&pdev->dev,
-				"error getting mi2s-bit-clk: %ld\n",
+				"error getting %s: %ld\n",
+				variant->dai_bit_clk_names[i],
 				PTR_ERR(drvdata->mi2s_bit_clk[dai_id]));
 			return PTR_ERR(drvdata->mi2s_bit_clk[dai_id]);
 		}

sound/soc/qcom/lpass-ipq806x.c

@@ -92,6 +92,12 @@ static struct lpass_variant ipq806x_data = {
 	.wrdma_channels		= 4,
 	.dai_driver		= &ipq806x_lpass_cpu_dai_driver,
 	.num_dai		= 1,
+	.dai_osr_clk_names	= (const char *[]) {
+				"mi2s-osr-clk",
+				},
+	.dai_bit_clk_names	= (const char *[]) {
+				"mi2s-bit-clk",
+				},
 	.alloc_dma_channel	= ipq806x_lpass_alloc_dma_channel,
 	.free_dma_channel	= ipq806x_lpass_free_dma_channel,
 };

sound/soc/qcom/lpass.h

@@ -91,6 +91,8 @@ struct lpass_variant {
 	/* SOC specific dais */
 	struct snd_soc_dai_driver *dai_driver;
 	int num_dai;
+	const char * const *dai_osr_clk_names;
+	const char * const *dai_bit_clk_names;
 };
 
 /* register the platform driver from the CPU DAI driver */