diff --git a/drivers/extcon/extcon-arizona.c b/drivers/extcon/extcon-arizona.c index 56e6c4c7c60d..d836d4ce5ee4 100644 --- a/drivers/extcon/extcon-arizona.c +++ b/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", diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index c91717d08513..ebdaf56c1d61 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -27,7 +27,27 @@ #include #include -#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 { diff --git a/sound/soc/codecs/ak4641.h b/sound/soc/codecs/ak4641.h deleted file mode 100644 index 4a263248efea..000000000000 --- a/sound/soc/codecs/ak4641.h +++ /dev/null @@ -1,47 +0,0 @@ -/* - * ak4641.h -- AK4641 SoC Audio driver - * - * Copyright 2008 Harald Welte - * - * 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 diff --git a/sound/soc/codecs/es8328.h b/sound/soc/codecs/es8328.h index 1a736e72a929..8930322d712b 100644 --- a/sound/soc/codecs/es8328.h +++ b/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 diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 19bdcac71775..37f9b6201918 100644 --- a/sound/soc/fsl/Kconfig +++ b/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. diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 13631003cb7c..a002ab892772 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/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; diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index ecbf2873b7ff..85483049b916 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -27,8 +27,6 @@ #include #include "pxa2xx-i2s.h" -#include "../codecs/ak4641.h" - static struct snd_soc_jack hs_jack; /* Headphones jack detection DAPM pin */ diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d56a16a0f6fa..e7a1eaa2772f 100644 --- a/sound/soc/soc-pcm.c +++ b/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: