Merge remote-tracking branches 'asoc/topic/dpcm', 'asoc/topic/es8328',...

Merge remote-tracking branches 'asoc/topic/dpcm', 'asoc/topic/es8328', 'asoc/topic/extcon' and 'asoc/topic/fsl' into asoc-next

drivers/extcon/extcon-arizona.c

@@ -274,9 +274,10 @@ static void arizona_extcon_pulse_micbias(struct arizona_extcon_info *info)
 	struct arizona *arizona = info->arizona;
 	const char *widget = arizona_extcon_get_micbias(info);
 	struct snd_soc_dapm_context *dapm = arizona->dapm;
+	struct snd_soc_component *component = snd_soc_dapm_to_component(dapm);
 	int ret;
 
-	ret = snd_soc_dapm_force_enable_pin(dapm, widget);
+	ret = snd_soc_component_force_enable_pin(component, widget);
 	if (ret != 0)
 		dev_warn(arizona->dev, "Failed to enable %s: %d\n",
 			 widget, ret);
@@ -284,7 +285,7 @@ static void arizona_extcon_pulse_micbias(struct arizona_extcon_info *info)
 	snd_soc_dapm_sync(dapm);
 
 	if (!arizona->pdata.micd_force_micbias) {
-		ret = snd_soc_dapm_disable_pin(arizona->dapm, widget);
+		ret = snd_soc_component_disable_pin(component, widget);
 		if (ret != 0)
 			dev_warn(arizona->dev, "Failed to disable %s: %d\n",
 				 widget, ret);
@@ -349,6 +350,7 @@ static void arizona_stop_mic(struct arizona_extcon_info *info)
 	struct arizona *arizona = info->arizona;
 	const char *widget = arizona_extcon_get_micbias(info);
 	struct snd_soc_dapm_context *dapm = arizona->dapm;
+	struct snd_soc_component *component = snd_soc_dapm_to_component(dapm);
 	bool change;
 	int ret;
 
@@ -356,7 +358,7 @@ static void arizona_stop_mic(struct arizona_extcon_info *info)
 				 ARIZONA_MICD_ENA, 0,
 				 &change);
 
-	ret = snd_soc_dapm_disable_pin(dapm, widget);
+	ret = snd_soc_component_disable_pin(component, widget);
 	if (ret != 0)
 		dev_warn(arizona->dev,
 			 "Failed to disable %s: %d\n",

sound/soc/codecs/ak4641.c

@@ -27,7 +27,27 @@
 #include <sound/tlv.h>
 #include <sound/ak4641.h>
 
-#include "ak4641.h"
+/* AK4641 register space */
+#define AK4641_PM1		0x00
+#define AK4641_PM2		0x01
+#define AK4641_SIG1		0x02
+#define AK4641_SIG2		0x03
+#define AK4641_MODE1		0x04
+#define AK4641_MODE2		0x05
+#define AK4641_DAC		0x06
+#define AK4641_MIC		0x07
+#define AK4641_TIMER		0x08
+#define AK4641_ALC1		0x09
+#define AK4641_ALC2		0x0a
+#define AK4641_PGA		0x0b
+#define AK4641_LATT		0x0c
+#define AK4641_RATT		0x0d
+#define AK4641_VOL		0x0e
+#define AK4641_STATUS		0x0f
+#define AK4641_EQLO		0x10
+#define AK4641_EQMID		0x11
+#define AK4641_EQHI		0x12
+#define AK4641_BTIF		0x13
 
 /* codec private data */
 struct ak4641_priv {

sound/soc/codecs/ak4641.h

-/*
- * ak4641.h  --  AK4641 SoC Audio driver
- *
- * Copyright 2008 Harald Welte <laforge@gnufiish.org>
- *
- * Based on ak4535.h
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
-
-#ifndef _AK4641_H
-#define _AK4641_H
-
-/* AK4641 register space */
-
-#define AK4641_PM1		0x00
-#define AK4641_PM2		0x01
-#define AK4641_SIG1		0x02
-#define AK4641_SIG2		0x03
-#define AK4641_MODE1		0x04
-#define AK4641_MODE2		0x05
-#define AK4641_DAC		0x06
-#define AK4641_MIC		0x07
-#define AK4641_TIMER		0x08
-#define AK4641_ALC1		0x09
-#define AK4641_ALC2		0x0a
-#define AK4641_PGA		0x0b
-#define AK4641_LATT		0x0c
-#define AK4641_RATT		0x0d
-#define AK4641_VOL		0x0e
-#define AK4641_STATUS		0x0f
-#define AK4641_EQLO		0x10
-#define AK4641_EQMID		0x11
-#define AK4641_EQHI		0x12
-#define AK4641_BTIF		0x13
-
-#define AK4641_CACHEREGNUM	0x14
-
-
-
-#define AK4641_DAI_HIFI		0
-#define AK4641_DAI_VOICE	1
-
-
-#endif

sound/soc/codecs/es8328.h

@@ -278,43 +278,6 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
 
 #define ES8328_REG_MAX		0x35
 
-#define ES8328_PLL1		0
-#define ES8328_PLL2		1
-
-/* clock inputs */
-#define ES8328_MCLK		0
-#define ES8328_PCMCLK		1
-
-/* clock divider id's */
-#define ES8328_PCMDIV		0
-#define ES8328_BCLKDIV		1
-#define ES8328_VXCLKDIV		2
-
-/* PCM clock dividers */
-#define ES8328_PCM_DIV_1	(0 << 6)
-#define ES8328_PCM_DIV_3	(2 << 6)
-#define ES8328_PCM_DIV_5_5	(3 << 6)
-#define ES8328_PCM_DIV_2	(4 << 6)
-#define ES8328_PCM_DIV_4	(5 << 6)
-#define ES8328_PCM_DIV_6	(6 << 6)
-#define ES8328_PCM_DIV_8	(7 << 6)
-
-/* BCLK clock dividers */
-#define ES8328_BCLK_DIV_1	(0 << 7)
-#define ES8328_BCLK_DIV_2	(1 << 7)
-#define ES8328_BCLK_DIV_4	(2 << 7)
-#define ES8328_BCLK_DIV_8	(3 << 7)
-
-/* VXCLK clock dividers */
-#define ES8328_VXCLK_DIV_1	(0 << 6)
-#define ES8328_VXCLK_DIV_2	(1 << 6)
-#define ES8328_VXCLK_DIV_4	(2 << 6)
-#define ES8328_VXCLK_DIV_8	(3 << 6)
-#define ES8328_VXCLK_DIV_16	(4 << 6)
-
-#define ES8328_DAI_HIFI		0
-#define ES8328_DAI_VOICE	1
-
 #define ES8328_1536FS		1536
 #define ES8328_1024FS		1024
 #define ES8328_768FS		768

sound/soc/fsl/Kconfig

@@ -40,6 +40,7 @@ config SND_SOC_FSL_SPDIF
 	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 && (MXC_TZIC || MXC_AVIC)
+	select BITREVERSE
 	help
 	  Say Y if you want to add Sony/Philips Digital Interface (SPDIF)
 	  support for the Freescale CPUs.

sound/soc/mxs/mxs-saif.c

@@ -735,6 +735,11 @@ static int mxs_saif_probe(struct platform_device *pdev)
 	else
 		saif->id = ret;
 
+	if (saif->id >= ARRAY_SIZE(mxs_saif)) {
+		dev_err(&pdev->dev, "get wrong saif id\n");
+		return -EINVAL;
+	}
+
 	/*
 	 * If there is no "fsl,saif-master" phandle, it's a saif
 	 * master.  Otherwise, it's a slave and its phandle points
@@ -749,11 +754,11 @@ static int mxs_saif_probe(struct platform_device *pdev)
 			return ret;
 		else
 			saif->master_id = ret;
-	}
 
-	if (saif->master_id >= ARRAY_SIZE(mxs_saif)) {
-		dev_err(&pdev->dev, "get wrong master id\n");
-		return -EINVAL;
+		if (saif->master_id >= ARRAY_SIZE(mxs_saif)) {
+			dev_err(&pdev->dev, "get wrong master id\n");
+			return -EINVAL;
+		}
 	}
 
 	mxs_saif[saif->id] = saif;

sound/soc/pxa/hx4700.c

@@ -27,8 +27,6 @@
 #include <asm/mach-types.h>
 #include "pxa2xx-i2s.h"
 
-#include "../codecs/ak4641.h"
-
 static struct snd_soc_jack hs_jack;
 
 /* Headphones jack detection DAPM pin */

sound/soc/soc-pcm.c

@@ -2882,7 +2882,7 @@ int snd_soc_platform_trigger(struct snd_pcm_substream *substream,
 EXPORT_SYMBOL_GPL(snd_soc_platform_trigger);
 
 #ifdef CONFIG_DEBUG_FS
-static char *dpcm_state_string(enum snd_soc_dpcm_state state)
+static const char *dpcm_state_string(enum snd_soc_dpcm_state state)
 {
 	switch (state) {
 	case SND_SOC_DPCM_STATE_NEW: