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-rw-r--r--drivers/extcon/extcon-arizona.c8
-rw-r--r--sound/soc/codecs/ak4641.c22
-rw-r--r--sound/soc/codecs/ak4641.h47
-rw-r--r--sound/soc/codecs/es8328.h37
-rw-r--r--sound/soc/fsl/Kconfig1
-rw-r--r--sound/soc/mxs/mxs-saif.c13
-rw-r--r--sound/soc/pxa/hx4700.c2
-rw-r--r--sound/soc/soc-pcm.c2
8 files changed, 37 insertions, 95 deletions
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 <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 {
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 <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
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 <asm/mach-types.h>
#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:
generated by cgit v1.2.3 (git 2.39.0) at 2024-08-17 00:17:04 +0300