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-rw-r--r--arch/arm/mach-shmobile/board-ag5evm.c10
-rw-r--r--arch/arm/mach-shmobile/board-ap4evb.c13
-rw-r--r--arch/arm/mach-shmobile/board-mackerel.c13
-rw-r--r--arch/arm/mach-tegra/include/mach/harmony_audio.h22
-rw-r--r--arch/sh/boards/mach-ecovec24/setup.c6
-rw-r--r--arch/sh/boards/mach-se/7724/setup.c6
-rw-r--r--include/sound/cs4271.h25
-rw-r--r--include/sound/sh_fsi.h76
-rw-r--r--include/sound/soc-dapm.h16
-rw-r--r--include/sound/soc.h41
-rw-r--r--include/sound/wm8903.h20
-rw-r--r--include/trace/events/asoc.h25
-rw-r--r--sound/soc/atmel/snd-soc-afeb9260.c2
-rw-r--r--sound/soc/blackfin/bf5xx-ssm2602.c2
-rw-r--r--sound/soc/codecs/Kconfig4
-rw-r--r--sound/soc/codecs/Makefile2
-rw-r--r--sound/soc/codecs/ak4642.c24
-rw-r--r--sound/soc/codecs/cq93vc.c2
-rw-r--r--sound/soc/codecs/cs4271.c659
-rw-r--r--sound/soc/codecs/cx20442.c3
-rw-r--r--sound/soc/codecs/sn95031.c281
-rw-r--r--sound/soc/codecs/wm8903.c127
-rw-r--r--sound/soc/codecs/wm8962.c2
-rw-r--r--sound/soc/codecs/wm8994.c2
-rw-r--r--sound/soc/codecs/wm8995.c2
-rw-r--r--sound/soc/codecs/wm_hubs.c15
-rw-r--r--sound/soc/davinci/davinci-evm.c2
-rw-r--r--sound/soc/ep93xx/Kconfig9
-rw-r--r--sound/soc/ep93xx/Makefile2
-rw-r--r--sound/soc/ep93xx/edb93xx.c142
-rw-r--r--sound/soc/fsl/mpc8610_hpcd.c6
-rw-r--r--sound/soc/fsl/p1022_ds.c6
-rw-r--r--sound/soc/mid-x86/mfld_machine.c3
-rw-r--r--sound/soc/mid-x86/sst_platform.c6
-rw-r--r--sound/soc/omap/ams-delta.c2
-rw-r--r--sound/soc/pxa/corgi.c4
-rw-r--r--sound/soc/pxa/poodle.c2
-rw-r--r--sound/soc/pxa/raumfeld.c4
-rw-r--r--sound/soc/pxa/spitz.c4
-rw-r--r--sound/soc/pxa/tosa.c4
-rw-r--r--sound/soc/pxa/zylonite.c9
-rw-r--r--sound/soc/samsung/neo1973_gta02_wm8753.c6
-rw-r--r--sound/soc/samsung/neo1973_wm8753.c7
-rw-r--r--sound/soc/samsung/s3c24xx_simtec_hermes.c4
-rw-r--r--sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c4
-rw-r--r--sound/soc/samsung/s3c24xx_uda134x.c2
-rw-r--r--sound/soc/sh/fsi-ak4642.c13
-rw-r--r--sound/soc/sh/fsi-da7210.c13
-rw-r--r--sound/soc/sh/fsi-hdmi.c77
-rw-r--r--sound/soc/sh/fsi.c203
-rw-r--r--sound/soc/soc-cache.c50
-rw-r--r--sound/soc/soc-core.c114
-rw-r--r--sound/soc/soc-dapm.c71
-rw-r--r--sound/soc/soc-utils.c23
-rw-r--r--sound/soc/tegra/harmony.c170
-rw-r--r--sound/soc/tegra/tegra_asoc_utils.c91
-rw-r--r--sound/soc/tegra/tegra_asoc_utils.h20
-rw-r--r--sound/soc/tegra/tegra_i2s.c2
58 files changed, 2044 insertions, 431 deletions
diff --git a/arch/arm/mach-shmobile/board-ag5evm.c b/arch/arm/mach-shmobile/board-ag5evm.c
index c18a740a4159..343362d02075 100644
--- a/arch/arm/mach-shmobile/board-ag5evm.c
+++ b/arch/arm/mach-shmobile/board-ag5evm.c
@@ -118,13 +118,6 @@ static struct platform_device keysc_device = {
};
/* FSI A */
-static struct sh_fsi_platform_info fsi_info = {
- .porta_flags = SH_FSI_OUT_SLAVE_MODE |
- SH_FSI_IN_SLAVE_MODE |
- SH_FSI_OFMT(I2S) |
- SH_FSI_IFMT(I2S),
-};
-
static struct resource fsi_resources[] = {
[0] = {
.name = "FSI",
@@ -143,9 +136,6 @@ static struct platform_device fsi_device = {
.id = -1,
.num_resources = ARRAY_SIZE(fsi_resources),
.resource = fsi_resources,
- .dev = {
- .platform_data = &fsi_info,
- },
};
static struct resource sh_mmcif_resources[] = {
diff --git a/arch/arm/mach-shmobile/board-ap4evb.c b/arch/arm/mach-shmobile/board-ap4evb.c
index 3cf0951caa2d..17f528a76a1c 100644
--- a/arch/arm/mach-shmobile/board-ap4evb.c
+++ b/arch/arm/mach-shmobile/board-ap4evb.c
@@ -673,16 +673,12 @@ static int fsi_set_rate(struct device *dev, int is_porta, int rate, int enable)
}
static struct sh_fsi_platform_info fsi_info = {
- .porta_flags = SH_FSI_BRS_INV |
- SH_FSI_OUT_SLAVE_MODE |
- SH_FSI_IN_SLAVE_MODE |
- SH_FSI_OFMT(PCM) |
- SH_FSI_IFMT(PCM),
+ .porta_flags = SH_FSI_BRS_INV,
.portb_flags = SH_FSI_BRS_INV |
SH_FSI_BRM_INV |
SH_FSI_LRS_INV |
- SH_FSI_OFMT(SPDIF),
+ SH_FSI_FMT_SPDIF,
.set_rate = fsi_set_rate,
};
@@ -783,6 +779,10 @@ static struct platform_device hdmi_device = {
},
};
+static struct platform_device fsi_hdmi_device = {
+ .name = "sh_fsi2_b_hdmi",
+};
+
static long ap4evb_clk_optimize(unsigned long target, unsigned long *best_freq,
unsigned long *parent_freq)
{
@@ -936,6 +936,7 @@ static struct platform_device *ap4evb_devices[] __initdata = {
&usb1_host_device,
&fsi_device,
&fsi_ak4643_device,
+ &fsi_hdmi_device,
&sh_mmcif_device,
&lcdc1_device,
&lcdc_device,
diff --git a/arch/arm/mach-shmobile/board-mackerel.c b/arch/arm/mach-shmobile/board-mackerel.c
index 7b15d21f0f68..73b8c90b5072 100644
--- a/arch/arm/mach-shmobile/board-mackerel.c
+++ b/arch/arm/mach-shmobile/board-mackerel.c
@@ -400,6 +400,10 @@ static struct platform_device hdmi_device = {
},
};
+static struct platform_device fsi_hdmi_device = {
+ .name = "sh_fsi2_b_hdmi",
+};
+
static int __init hdmi_init_pm_clock(void)
{
struct clk *hdmi_ick = clk_get(&hdmi_device.dev, "ick");
@@ -610,16 +614,12 @@ fsi_set_rate_end:
}
static struct sh_fsi_platform_info fsi_info = {
- .porta_flags = SH_FSI_BRS_INV |
- SH_FSI_OUT_SLAVE_MODE |
- SH_FSI_IN_SLAVE_MODE |
- SH_FSI_OFMT(PCM) |
- SH_FSI_IFMT(PCM),
+ .porta_flags = SH_FSI_BRS_INV,
.portb_flags = SH_FSI_BRS_INV |
SH_FSI_BRM_INV |
SH_FSI_LRS_INV |
- SH_FSI_OFMT(SPDIF),
+ SH_FSI_FMT_SPDIF,
.set_rate = fsi_set_rate,
};
@@ -922,6 +922,7 @@ static struct platform_device *mackerel_devices[] __initdata = {
&leds_device,
&fsi_device,
&fsi_ak4643_device,
+ &fsi_hdmi_device,
&sdhi0_device,
#if !defined(CONFIG_MMC_SH_MMCIF)
&sdhi1_device,
diff --git a/arch/arm/mach-tegra/include/mach/harmony_audio.h b/arch/arm/mach-tegra/include/mach/harmony_audio.h
new file mode 100644
index 000000000000..af086500ab7d
--- /dev/null
+++ b/arch/arm/mach-tegra/include/mach/harmony_audio.h
@@ -0,0 +1,22 @@
+/*
+ * arch/arm/mach-tegra/include/mach/harmony_audio.h
+ *
+ * Copyright 2011 NVIDIA, Inc.
+ *
+ * This software is licensed under the terms of the GNU General Public
+ * License version 2, as published by the Free Software Foundation, and
+ * may be copied, distributed, and modified under those terms.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ */
+
+struct harmony_audio_platform_data {
+ int gpio_spkr_en;
+ int gpio_hp_det;
+ int gpio_int_mic_en;
+ int gpio_ext_mic_en;
+};
diff --git a/arch/sh/boards/mach-ecovec24/setup.c b/arch/sh/boards/mach-ecovec24/setup.c
index 33b662999fc6..b96b79b970b2 100644
--- a/arch/sh/boards/mach-ecovec24/setup.c
+++ b/arch/sh/boards/mach-ecovec24/setup.c
@@ -723,11 +723,7 @@ static struct platform_device camera_devices[] = {
/* FSI */
static struct sh_fsi_platform_info fsi_info = {
- .portb_flags = SH_FSI_BRS_INV |
- SH_FSI_OUT_SLAVE_MODE |
- SH_FSI_IN_SLAVE_MODE |
- SH_FSI_OFMT(I2S) |
- SH_FSI_IFMT(I2S),
+ .portb_flags = SH_FSI_BRS_INV,
};
static struct resource fsi_resources[] = {
diff --git a/arch/sh/boards/mach-se/7724/setup.c b/arch/sh/boards/mach-se/7724/setup.c
index 527679394a25..c8bcf6a19b55 100644
--- a/arch/sh/boards/mach-se/7724/setup.c
+++ b/arch/sh/boards/mach-se/7724/setup.c
@@ -286,11 +286,7 @@ static struct platform_device ceu1_device = {
/* FSI */
/* change J20, J21, J22 pin to 1-2 connection to use slave mode */
static struct sh_fsi_platform_info fsi_info = {
- .porta_flags = SH_FSI_BRS_INV |
- SH_FSI_OUT_SLAVE_MODE |
- SH_FSI_IN_SLAVE_MODE |
- SH_FSI_OFMT(PCM) |
- SH_FSI_IFMT(PCM),
+ .porta_flags = SH_FSI_BRS_INV,
};
static struct resource fsi_resources[] = {
diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h
new file mode 100644
index 000000000000..16f8d325d3dc
--- /dev/null
+++ b/include/sound/cs4271.h
@@ -0,0 +1,25 @@
+/*
+ * Definitions for CS4271 ASoC codec driver
+ *
+ * Copyright (c) 2010 Alexander Sverdlin <subaparts@yandex.ru>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#ifndef __CS4271_H
+#define __CS4271_H
+
+struct cs4271_platform_data {
+ int gpio_nreset; /* GPIO driving Reset pin, if any */
+ int gpio_disable; /* GPIO that disable serial bus, if any */
+};
+
+#endif /* __CS4271_H */
diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h
index d79894192ae3..9a155f9d0a12 100644
--- a/include/sound/sh_fsi.h
+++ b/include/sound/sh_fsi.h
@@ -15,67 +15,29 @@
#define FSI_PORT_A 0
#define FSI_PORT_B 1
-/* flags format
-
- * 0xABCDEEFF
- *
- * A: channel size for TDM (input)
- * B: channel size for TDM (ooutput)
- * C: inversion
- * D: mode
- * E: input format
- * F: output format
- */
-
#include <linux/clk.h>
#include <sound/soc.h>
-/* TDM channel */
-#define SH_FSI_SET_CH_I(x) ((x & 0xF) << 28)
-#define SH_FSI_SET_CH_O(x) ((x & 0xF) << 24)
-
-#define SH_FSI_CH_IMASK 0xF0000000
-#define SH_FSI_CH_OMASK 0x0F000000
-#define SH_FSI_GET_CH_I(x) ((x & SH_FSI_CH_IMASK) >> 28)
-#define SH_FSI_GET_CH_O(x) ((x & SH_FSI_CH_OMASK) >> 24)
-
-/* clock inversion */
-#define SH_FSI_INVERSION_MASK 0x00F00000
-#define SH_FSI_LRM_INV (1 << 20)
-#define SH_FSI_BRM_INV (1 << 21)
-#define SH_FSI_LRS_INV (1 << 22)
-#define SH_FSI_BRS_INV (1 << 23)
-
-/* mode */
-#define SH_FSI_MODE_MASK 0x000F0000
-#define SH_FSI_IN_SLAVE_MODE (1 << 16) /* default master mode */
-#define SH_FSI_OUT_SLAVE_MODE (1 << 17) /* default master mode */
-
-/* DI format */
-#define SH_FSI_FMT_MASK 0x000000FF
-#define SH_FSI_IFMT(x) (((SH_FSI_FMT_ ## x) & SH_FSI_FMT_MASK) << 8)
-#define SH_FSI_OFMT(x) (((SH_FSI_FMT_ ## x) & SH_FSI_FMT_MASK) << 0)
-#define SH_FSI_GET_IFMT(x) ((x >> 8) & SH_FSI_FMT_MASK)
-#define SH_FSI_GET_OFMT(x) ((x >> 0) & SH_FSI_FMT_MASK)
-
-#define SH_FSI_FMT_MONO 0
-#define SH_FSI_FMT_MONO_DELAY 1
-#define SH_FSI_FMT_PCM 2
-#define SH_FSI_FMT_I2S 3
-#define SH_FSI_FMT_TDM 4
-#define SH_FSI_FMT_TDM_DELAY 5
-#define SH_FSI_FMT_SPDIF 6
-
-
-#define SH_FSI_IFMT_TDM_CH(x) \
- (SH_FSI_IFMT(TDM) | SH_FSI_SET_CH_I(x))
-#define SH_FSI_IFMT_TDM_DELAY_CH(x) \
- (SH_FSI_IFMT(TDM_DELAY) | SH_FSI_SET_CH_I(x))
+/*
+ * flags format
+ *
+ * 0x000000BA
+ *
+ * A: inversion
+ * B: format mode
+ */
-#define SH_FSI_OFMT_TDM_CH(x) \
- (SH_FSI_OFMT(TDM) | SH_FSI_SET_CH_O(x))
-#define SH_FSI_OFMT_TDM_DELAY_CH(x) \
- (SH_FSI_OFMT(TDM_DELAY) | SH_FSI_SET_CH_O(x))
+/* A: clock inversion */
+#define SH_FSI_INVERSION_MASK 0x0000000F
+#define SH_FSI_LRM_INV (1 << 0)
+#define SH_FSI_BRM_INV (1 << 1)
+#define SH_FSI_LRS_INV (1 << 2)
+#define SH_FSI_BRS_INV (1 << 3)
+
+/* B: format mode */
+#define SH_FSI_FMT_MASK 0x000000F0
+#define SH_FSI_FMT_DAI (0 << 4)
+#define SH_FSI_FMT_SPDIF (1 << 4)
/*
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 8031769ac485..979ed84e07d6 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -157,6 +157,18 @@
.invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1, \
.event = wevent, .event_flags = wflags}
+/* additional sequencing control within an event type */
+#define SND_SOC_DAPM_PGA_S(wname, wsubseq, wreg, wshift, winvert, \
+ wevent, wflags) \
+{ .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \
+ .invert = winvert, .event = wevent, .event_flags = wflags, \
+ .subseq = wsubseq}
+#define SND_SOC_DAPM_SUPPLY_S(wname, wsubseq, wreg, wshift, winvert, wevent, \
+ wflags) \
+{ .id = snd_soc_dapm_supply, .name = wname, .reg = wreg, \
+ .shift = wshift, .invert = winvert, .event = wevent, \
+ .event_flags = wflags, .subseq = wsubseq}
+
/* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */
#define SOC_PGA_E_ARRAY(wname, wreg, wshift, winvert, wcontrols, \
wevent, wflags) \
@@ -450,6 +462,7 @@ struct snd_soc_dapm_widget {
unsigned char ext:1; /* has external widgets */
unsigned char force:1; /* force state */
unsigned char ignore_suspend:1; /* kept enabled over suspend */
+ int subseq; /* sort within widget type */
int (*power_check)(struct snd_soc_dapm_widget *w);
@@ -487,6 +500,9 @@ struct snd_soc_dapm_context {
struct snd_soc_dapm_update *update;
+ void (*seq_notifier)(struct snd_soc_dapm_context *,
+ enum snd_soc_dapm_type, int);
+
struct device *dev; /* from parent - for debug */
struct snd_soc_codec *codec; /* parent codec */
struct snd_soc_card *card; /* parent card */
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 9952254974b3..4b6c0a8c332f 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -260,6 +260,9 @@ enum snd_soc_compress_type {
int snd_soc_register_card(struct snd_soc_card *card);
int snd_soc_unregister_card(struct snd_soc_card *card);
+int snd_soc_suspend(struct device *dev);
+int snd_soc_resume(struct device *dev);
+int snd_soc_poweroff(struct device *dev);
int snd_soc_register_platform(struct device *dev,
struct snd_soc_platform_driver *platform_drv);
void snd_soc_unregister_platform(struct device *dev);
@@ -267,7 +270,8 @@ int snd_soc_register_codec(struct device *dev,
const struct snd_soc_codec_driver *codec_drv,
struct snd_soc_dai_driver *dai_drv, int num_dai);
void snd_soc_unregister_codec(struct device *dev);
-int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg);
+int snd_soc_codec_volatile_register(struct snd_soc_codec *codec,
+ unsigned int reg);
int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
int addr_bits, int data_bits,
enum snd_soc_control_type control);
@@ -488,14 +492,15 @@ struct snd_soc_codec {
/* runtime */
struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */
unsigned int active;
- unsigned int cache_only:1; /* Suppress writes to hardware */
- unsigned int cache_sync:1; /* Cache needs to be synced to hardware */
+ unsigned int cache_bypass:1; /* Suppress access to the cache */
unsigned int suspended:1; /* Codec is in suspend PM state */
unsigned int probed:1; /* Codec has been probed */
unsigned int ac97_registered:1; /* Codec has been AC97 registered */
unsigned int ac97_created:1; /* Codec has been created by SoC */
unsigned int sysfs_registered:1; /* codec has been sysfs registered */
unsigned int cache_init:1; /* codec cache has been initialized */
+ u32 cache_only; /* Suppress writes to hardware */
+ u32 cache_sync; /* Cache needs to be synced to hardware */
/* codec IO */
void *control_data; /* codec control (i2c/3wire) data */
@@ -546,6 +551,9 @@ struct snd_soc_codec_driver {
/* codec bias level */
int (*set_bias_level)(struct snd_soc_codec *,
enum snd_soc_bias_level level);
+
+ void (*seq_notifier)(struct snd_soc_dapm_context *,
+ enum snd_soc_dapm_type, int);
};
/* SoC platform interface */
@@ -644,15 +652,15 @@ struct snd_soc_card {
bool instantiated;
- int (*probe)(struct platform_device *pdev);
- int (*remove)(struct platform_device *pdev);
+ int (*probe)(struct snd_soc_card *card);
+ int (*remove)(struct snd_soc_card *card);
/* the pre and post PM functions are used to do any PM work before and
* after the codec and DAI's do any PM work. */
- int (*suspend_pre)(struct platform_device *pdev, pm_message_t state);
- int (*suspend_post)(struct platform_device *pdev, pm_message_t state);
- int (*resume_pre)(struct platform_device *pdev);
- int (*resume_post)(struct platform_device *pdev);
+ int (*suspend_pre)(struct snd_soc_card *card);
+ int (*suspend_post)(struct snd_soc_card *card);
+ int (*resume_pre)(struct snd_soc_card *card);
+ int (*resume_post)(struct snd_soc_card *card);
/* callbacks */
int (*set_bias_level)(struct snd_soc_card *,
@@ -697,6 +705,8 @@ struct snd_soc_card {
struct dentry *debugfs_pop_time;
#endif
u32 pop_time;
+
+ void *drvdata;
};
/* SoC machine DAI configuration, glues a codec and cpu DAI together */
@@ -748,6 +758,17 @@ unsigned int snd_soc_write(struct snd_soc_codec *codec,
/* device driver data */
+static inline void snd_soc_card_set_drvdata(struct snd_soc_card *card,
+ void *data)
+{
+ card->drvdata = data;
+}
+
+static inline void *snd_soc_card_get_drvdata(struct snd_soc_card *card)
+{
+ return card->drvdata;
+}
+
static inline void snd_soc_codec_set_drvdata(struct snd_soc_codec *codec,
void *data)
{
@@ -797,4 +818,6 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card)
extern struct dentry *snd_soc_debugfs_root;
#endif
+extern const struct dev_pm_ops snd_soc_pm_ops;
+
#endif
diff --git a/include/sound/wm8903.h b/include/sound/wm8903.h
index b4a0db2307ef..86172cf4339f 100644
--- a/include/sound/wm8903.h
+++ b/include/sound/wm8903.h
@@ -37,6 +37,21 @@
#define WM8903_MICBIAS_ENA_WIDTH 1 /* MICBIAS_ENA */
/*
+ * WM8903_GPn_FN values
+ *
+ * See datasheets for list of valid values per pin
+ */
+#define WM8903_GPn_FN_GPIO_OUTPUT 0
+#define WM8903_GPn_FN_BCLK 1
+#define WM8903_GPn_FN_IRQ_OUTPT 2
+#define WM8903_GPn_FN_GPIO_INPUT 3
+#define WM8903_GPn_FN_MICBIAS_CURRENT_DETECT 4
+#define WM8903_GPn_FN_MICBIAS_SHORT_DETECT 5
+#define WM8903_GPn_FN_DMIC_LR_CLK_OUTPUT 6
+#define WM8903_GPn_FN_FLL_LOCK_OUTPUT 8
+#define WM8903_GPn_FN_FLL_CLOCK_OUTPUT 9
+
+/*
* R116 (0x74) - GPIO Control 1
*/
#define WM8903_GP1_FN_MASK 0x1F00 /* GP1_FN - [12:8] */
@@ -231,6 +246,8 @@
#define WM8903_GP5_DB_SHIFT 0 /* GP5_DB */
#define WM8903_GP5_DB_WIDTH 1 /* GP5_DB */
+#define WM8903_NUM_GPIO 5
+
struct wm8903_platform_data {
bool irq_active_low; /* Set if IRQ active low, default high */
@@ -243,7 +260,8 @@ struct wm8903_platform_data {
int micdet_delay; /* Delay after microphone detection (ms) */
- u32 gpio_cfg[5]; /* Default register values for GPIO pin mux */
+ int gpio_base;
+ u32 gpio_cfg[WM8903_NUM_GPIO]; /* Default register values for GPIO pin mux */
};
#endif
diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h
index 186e84db4b54..ae973d2e27a1 100644
--- a/include/trace/events/asoc.h
+++ b/include/trace/events/asoc.h
@@ -229,6 +229,31 @@ TRACE_EVENT(snd_soc_jack_notify,
TP_printk("jack=%s %x", __get_str(name), (int)__entry->val)
);
+TRACE_EVENT(snd_soc_cache_sync,
+
+ TP_PROTO(struct snd_soc_codec *codec, const char *type,
+ const char *status),
+
+ TP_ARGS(codec, type, status),
+
+ TP_STRUCT__entry(
+ __string( name, codec->name )
+ __string( status, status )
+ __string( type, type )
+ __field( int, id )
+ ),
+
+ TP_fast_assign(
+ __assign_str(name, codec->name);
+ __assign_str(status, status);
+ __assign_str(type, type);
+ __entry->id = codec->id;
+ ),
+
+ TP_printk("codec=%s.%d type=%s status=%s", __get_str(name),
+ (int)__entry->id, __get_str(type), __get_str(status))
+);
+
#endif /* _TRACE_ASOC_H */
/* This part must be outside protection */
diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c
index da2208e06b0d..5e4d499d8434 100644
--- a/sound/soc/atmel/snd-soc-afeb9260.c
+++ b/sound/soc/atmel/snd-soc-afeb9260.c
@@ -129,7 +129,7 @@ static struct snd_soc_dai_link afeb9260_dai = {
.cpu_dai_name = "atmel-ssc-dai.0",
.codec_dai_name = "tlv320aic23-hifi",
.platform_name = "atmel_pcm-audio",
- .codec_name = "tlv320aic23-codec.0-0x1a",
+ .codec_name = "tlv320aic23-codec.0-001a",
.init = afeb9260_tlv320aic23_init,
.ops = &afeb9260_ops,
};
diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c
index e902b24c1856..ad28663f5bbd 100644
--- a/sound/soc/blackfin/bf5xx-ssm2602.c
+++ b/sound/soc/blackfin/bf5xx-ssm2602.c
@@ -119,7 +119,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai = {
.cpu_dai_name = "bf5xx-i2s",
.codec_dai_name = "ssm2602-hifi",
.platform_name = "bf5xx-pcm-audio",
- .codec_name = "ssm2602-codec.0-0x1b",
+ .codec_name = "ssm2602-codec.0-001b",
.ops = &bf5xx_ssm2602_ops,
};
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index a9cb2a04ad56..e239345a4d5d 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -26,6 +26,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC
select SND_SOC_CS42L51 if I2C
select SND_SOC_CS4270 if I2C
+ select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI
select SND_SOC_CX20442
select SND_SOC_DA7210 if I2C
select SND_SOC_JZ4740_CODEC if SOC_JZ4740
@@ -157,6 +158,9 @@ config SND_SOC_CS4270_VD33_ERRATA
bool
depends on SND_SOC_CS4270
+config SND_SOC_CS4271
+ tristate
+
config SND_SOC_CX20442
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 68e76af894b9..83b7accd7037 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -12,6 +12,7 @@ snd-soc-ak4671-objs := ak4671.o
snd-soc-cq93vc-objs := cq93vc.o
snd-soc-cs42l51-objs := cs42l51.o
snd-soc-cs4270-objs := cs4270.o
+snd-soc-cs4271-objs := cs4271.o
snd-soc-cx20442-objs := cx20442.o
snd-soc-da7210-objs := da7210.o
snd-soc-dmic-objs := dmic.o
@@ -93,6 +94,7 @@ obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o
obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o
obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
+obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o
obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o
obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o
obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index f00eba313dfd..4be0570e3f1f 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -116,6 +116,12 @@
#define BCKO_MASK (1 << 3)
#define BCKO_64 BCKO_MASK
+#define DIF_MASK (3 << 0)
+#define DSP (0 << 0)
+#define RIGHT_J (1 << 0)
+#define LEFT_J (2 << 0)
+#define I2S (3 << 0)
+
/* MD_CTL2 */
#define FS0 (1 << 0)
#define FS1 (1 << 1)
@@ -354,6 +360,24 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
snd_soc_update_bits(codec, PW_MGMT2, MS, data);
snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko);
+ /* format type */
+ data = 0;
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_LEFT_J:
+ data = LEFT_J;
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ data = I2S;
+ break;
+ /* FIXME
+ * Please add RIGHT_J / DSP support here
+ */
+ default:
+ return -EINVAL;
+ break;
+ }
+ snd_soc_update_bits(codec, MD_CTL1, DIF_MASK, data);
+
return 0;
}
diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c
index 46dbfd067f79..347a567b01e1 100644
--- a/sound/soc/codecs/cq93vc.c
+++ b/sound/soc/codecs/cq93vc.c
@@ -153,7 +153,7 @@ static int cq93vc_resume(struct snd_soc_codec *codec)
static int cq93vc_probe(struct snd_soc_codec *codec)
{
- struct davinci_vc *davinci_vc = codec->dev->platform_data;
+ struct davinci_vc *davinci_vc = snd_soc_codec_get_drvdata(codec);
davinci_vc->cq93vc.codec = codec;
codec->control_data = davinci_vc;
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
new file mode 100644
index 000000000000..9c5b7db0ce6a
--- /dev/null
+++ b/sound/soc/codecs/cs4271.c
@@ -0,0 +1,659 @@
+/*
+ * CS4271 ASoC codec driver
+ *
+ * Copyright (c) 2010 Alexander Sverdlin <subaparts@yandex.ru>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * This driver support CS4271 codec being master or slave, working
+ * in control port mode, connected either via SPI or I2C.
+ * The data format accepted is I2S or left-justified.
+ * DAPM support not implemented.
+ */
+
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/delay.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include <linux/gpio.h>
+#include <linux/i2c.h>
+#include <linux/spi/spi.h>
+#include <sound/cs4271.h>
+
+#define CS4271_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+/*
+ * CS4271 registers
+ * High byte represents SPI chip address (0x10) + write command (0)
+ * Low byte - codec register address
+ */
+#define CS4271_MODE1 0x2001 /* Mode Control 1 */
+#define CS4271_DACCTL 0x2002 /* DAC Control */
+#define CS4271_DACVOL 0x2003 /* DAC Volume & Mixing Control */
+#define CS4271_VOLA 0x2004 /* DAC Channel A Volume Control */
+#define CS4271_VOLB 0x2005 /* DAC Channel B Volume Control */
+#define CS4271_ADCCTL 0x2006 /* ADC Control */
+#define CS4271_MODE2 0x2007 /* Mode Control 2 */
+#define CS4271_CHIPID 0x2008 /* Chip ID */
+
+#define CS4271_FIRSTREG CS4271_MODE1
+#define CS4271_LASTREG CS4271_MODE2
+#define CS4271_NR_REGS ((CS4271_LASTREG & 0xFF) + 1)
+
+/* Bit masks for the CS4271 registers */
+#define CS4271_MODE1_MODE_MASK 0xC0
+#define CS4271_MODE1_MODE_1X 0x00
+#define CS4271_MODE1_MODE_2X 0x80
+#define CS4271_MODE1_MODE_4X 0xC0
+
+#define CS4271_MODE1_DIV_MASK 0x30
+#define CS4271_MODE1_DIV_1 0x00
+#define CS4271_MODE1_DIV_15 0x10
+#define CS4271_MODE1_DIV_2 0x20
+#define CS4271_MODE1_DIV_3 0x30
+
+#define CS4271_MODE1_MASTER 0x08
+
+#define CS4271_MODE1_DAC_DIF_MASK 0x07
+#define CS4271_MODE1_DAC_DIF_LJ 0x00
+#define CS4271_MODE1_DAC_DIF_I2S 0x01
+#define CS4271_MODE1_DAC_DIF_RJ16 0x02
+#define CS4271_MODE1_DAC_DIF_RJ24 0x03
+#define CS4271_MODE1_DAC_DIF_RJ20 0x04
+#define CS4271_MODE1_DAC_DIF_RJ18 0x05
+
+#define CS4271_DACCTL_AMUTE 0x80
+#define CS4271_DACCTL_IF_SLOW 0x40
+
+#define CS4271_DACCTL_DEM_MASK 0x30
+#define CS4271_DACCTL_DEM_DIS 0x00
+#define CS4271_DACCTL_DEM_441 0x10
+#define CS4271_DACCTL_DEM_48 0x20
+#define CS4271_DACCTL_DEM_32 0x30
+
+#define CS4271_DACCTL_SVRU 0x08
+#define CS4271_DACCTL_SRD 0x04
+#define CS4271_DACCTL_INVA 0x02
+#define CS4271_DACCTL_INVB 0x01
+
+#define CS4271_DACVOL_BEQUA 0x40
+#define CS4271_DACVOL_SOFT 0x20
+#define CS4271_DACVOL_ZEROC 0x10
+
+#define CS4271_DACVOL_ATAPI_MASK 0x0F
+#define CS4271_DACVOL_ATAPI_M_M 0x00
+#define CS4271_DACVOL_ATAPI_M_BR 0x01
+#define CS4271_DACVOL_ATAPI_M_BL 0x02
+#define CS4271_DACVOL_ATAPI_M_BLR2 0x03
+#define CS4271_DACVOL_ATAPI_AR_M 0x04
+#define CS4271_DACVOL_ATAPI_AR_BR 0x05
+#define CS4271_DACVOL_ATAPI_AR_BL 0x06
+#define CS4271_DACVOL_ATAPI_AR_BLR2 0x07
+#define CS4271_DACVOL_ATAPI_AL_M 0x08
+#define CS4271_DACVOL_ATAPI_AL_BR 0x09
+#define CS4271_DACVOL_ATAPI_AL_BL 0x0A
+#define CS4271_DACVOL_ATAPI_AL_BLR2 0x0B
+#define CS4271_DACVOL_ATAPI_ALR2_M 0x0C
+#define CS4271_DACVOL_ATAPI_ALR2_BR 0x0D
+#define CS4271_DACVOL_ATAPI_ALR2_BL 0x0E
+#define CS4271_DACVOL_ATAPI_ALR2_BLR2 0x0F
+
+#define CS4271_VOLA_MUTE 0x80
+#define CS4271_VOLA_VOL_MASK 0x7F
+#define CS4271_VOLB_MUTE 0x80
+#define CS4271_VOLB_VOL_MASK 0x7F
+
+#define CS4271_ADCCTL_DITHER16 0x20
+
+#define CS4271_ADCCTL_ADC_DIF_MASK 0x10
+#define CS4271_ADCCTL_ADC_DIF_LJ 0x00
+#define CS4271_ADCCTL_ADC_DIF_I2S 0x10
+
+#define CS4271_ADCCTL_MUTEA 0x08
+#define CS4271_ADCCTL_MUTEB 0x04
+#define CS4271_ADCCTL_HPFDA 0x02
+#define CS4271_ADCCTL_HPFDB 0x01
+
+#define CS4271_MODE2_LOOP 0x10
+#define CS4271_MODE2_MUTECAEQUB 0x08
+#define CS4271_MODE2_FREEZE 0x04
+#define CS4271_MODE2_CPEN 0x02
+#define CS4271_MODE2_PDN 0x01
+
+#define CS4271_CHIPID_PART_MASK 0xF0
+#define CS4271_CHIPID_REV_MASK 0x0F
+
+/*
+ * Default CS4271 power-up configuration
+ * Array contains non-existing in hw register at address 0
+ * Array do not include Chip ID, as codec driver does not use
+ * registers read operations at all
+ */
+static const u8 cs4271_dflt_reg[CS4271_NR_REGS] = {
+ 0,
+ 0,
+ CS4271_DACCTL_AMUTE,
+ CS4271_DACVOL_SOFT | CS4271_DACVOL_ATAPI_AL_BR,
+ 0,
+ 0,
+ 0,
+ 0,
+};
+
+struct cs4271_private {
+ /* SND_SOC_I2C or SND_SOC_SPI */
+ enum snd_soc_control_type bus_type;
+ void *control_data;
+ unsigned int mclk;
+ bool master;
+ bool deemph;
+ /* Current sample rate for de-emphasis control */
+ int rate;
+ /* GPIO driving Reset pin, if any */
+ int gpio_nreset;
+ /* GPIO that disable serial bus, if any */
+ int gpio_disable;
+};
+
+struct cs4271_clk_cfg {
+ unsigned int ratio; /* MCLK / sample rate */
+ u8 speed_mode; /* codec speed mode: 1x, 2x, 4x */
+ u8 mclk_master; /* ratio bit mask for Master mode */
+ u8 mclk_slave; /* ratio bit mask for Slave mode */
+};
+
+static struct cs4271_clk_cfg cs4271_clk_tab[] = {
+ {64, CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1},
+ {96, CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1},
+ {128, CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1},
+ {192, CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1},
+ {256, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1},
+ {384, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1},
+ {512, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_2, CS4271_MODE1_DIV_1},
+ {768, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3, CS4271_MODE1_DIV_3},
+ {1024, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3, CS4271_MODE1_DIV_3}
+};
+
+#define CS4171_NR_RATIOS ARRAY_SIZE(cs4271_clk_tab)
+
+/*
+ * @freq is the desired MCLK rate
+ * MCLK rate should (c) be the sample rate, multiplied by one of the
+ * ratios listed in cs4271_mclk_fs_ratios table
+ */
+static int cs4271_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
+
+ cs4271->mclk = freq;
+ return 0;
+}
+
+static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int format)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
+ unsigned int val = 0;
+ int ret;
+
+ switch (format & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ cs4271->master = 0;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ cs4271->master = 1;
+ val |= CS4271_MODE1_MASTER;
+ break;
+ default:
+ dev_err(codec->dev, "Invalid DAI format\n");
+ return -EINVAL;
+ }
+
+ switch (format & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_LEFT_J:
+ val |= CS4271_MODE1_DAC_DIF_LJ;
+ ret = snd_soc_update_bits(codec, CS4271_ADCCTL,
+ CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_LJ);
+ if (ret < 0)
+ return ret;
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ val |= CS4271_MODE1_DAC_DIF_I2S;
+ ret = snd_soc_update_bits(codec, CS4271_ADCCTL,
+ CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_I2S);
+ if (ret < 0)
+ return ret;
+ break;
+ default:
+ dev_err(codec->dev, "Invalid DAI format\n");
+ return -EINVAL;
+ }
+
+ ret = snd_soc_update_bits(codec, CS4271_MODE1,
+ CS4271_MODE1_DAC_DIF_MASK | CS4271_MODE1_MASTER, val);
+ if (ret < 0)
+ return ret;
+ return 0;
+}
+
+static int cs4271_deemph[] = {0, 44100, 48000, 32000};
+
+static int cs4271_set_deemph(struct snd_soc_codec *codec)
+{
+ struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
+ int i, ret;
+ int val = CS4271_DACCTL_DEM_DIS;
+
+ if (cs4271->deemph) {
+ /* Find closest de-emphasis freq */
+ val = 1;
+ for (i = 2; i < ARRAY_SIZE(cs4271_deemph); i++)
+ if (abs(cs4271_deemph[i] - cs4271->rate) <
+ abs(cs4271_deemph[val] - cs4271->rate))
+ val = i;
+ val <<= 4;
+ }
+
+ ret = snd_soc_update_bits(codec, CS4271_DACCTL,
+ CS4271_DACCTL_DEM_MASK, val);
+ if (ret < 0)
+ return ret;
+ return 0;
+}
+
+static int cs4271_get_deemph(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
+
+ ucontrol->value.enumerated.item[0] = cs4271->deemph;
+ return 0;
+}
+
+static int cs4271_put_deemph(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
+
+ cs4271->deemph = ucontrol->value.enumerated.item[0];
+ return cs4271_set_deemph(codec);
+}
+
+static int cs4271_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->codec;
+ struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
+ int i, ret;
+ unsigned int ratio, val;
+
+ cs4271->rate = params_rate(params);
+ ratio = cs4271->mclk / cs4271->rate;
+ for (i = 0; i < CS4171_NR_RATIOS; i++)
+ if (cs4271_clk_tab[i].ratio == ratio)
+ break;
+
+ if ((i == CS4171_NR_RATIOS) || ((ratio == 1024) && cs4271->master)) {
+ dev_err(codec->dev, "Invalid sample rate\n");
+ return -EINVAL;
+ }
+
+ /* Configure DAC */
+ val = cs4271_clk_tab[i].speed_mode;
+
+ if (cs4271->master)
+ val |= cs4271_clk_tab[i].mclk_master;
+ else
+ val |= cs4271_clk_tab[i].mclk_slave;
+
+ ret = snd_soc_update_bits(codec, CS4271_MODE1,
+ CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK, val);
+ if (ret < 0)
+ return ret;
+
+ return cs4271_set_deemph(codec);
+}
+
+static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ int ret;
+ int val_a = 0;
+ int val_b = 0;
+
+ if (mute) {
+ val_a = CS4271_VOLA_MUTE;
+ val_b = CS4271_VOLB_MUTE;
+ }
+
+ ret = snd_soc_update_bits(codec, CS4271_VOLA, CS4271_VOLA_MUTE, val_a);
+ if (ret < 0)
+ return ret;
+ ret = snd_soc_update_bits(codec, CS4271_VOLB, CS4271_VOLB_MUTE, val_b);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+/* CS4271 controls */
+static DECLARE_TLV_DB_SCALE(cs4271_dac_tlv, -12700, 100, 0);
+
+static const struct snd_kcontrol_new cs4271_snd_controls[] = {
+ SOC_DOUBLE_R_TLV("Master Playback Volume", CS4271_VOLA, CS4271_VOLB,
+ 0, 0x7F, 1, cs4271_dac_tlv),
+ SOC_SINGLE("Digital Loopback Switch", CS4271_MODE2, 4, 1, 0),
+ SOC_SINGLE("Soft Ramp Switch", CS4271_DACVOL, 5, 1, 0),
+ SOC_SINGLE("Zero Cross Switch", CS4271_DACVOL, 4, 1, 0),
+ SOC_SINGLE_BOOL_EXT("De-emphasis Switch", 0,
+ cs4271_get_deemph, cs4271_put_deemph),
+ SOC_SINGLE("Auto-Mute Switch", CS4271_DACCTL, 7, 1, 0),
+ SOC_SINGLE("Slow Roll Off Filter Switch", CS4271_DACCTL, 6, 1, 0),
+ SOC_SINGLE("Soft Volume Ramp-Up Switch", CS4271_DACCTL, 3, 1, 0),
+ SOC_SINGLE("Soft Ramp-Down Switch", CS4271_DACCTL, 2, 1, 0),
+ SOC_SINGLE("Left Channel Inversion Switch", CS4271_DACCTL, 1, 1, 0),
+ SOC_SINGLE("Right Channel Inversion Switch", CS4271_DACCTL, 0, 1, 0),
+ SOC_DOUBLE("Master Capture Switch", CS4271_ADCCTL, 3, 2, 1, 1),
+ SOC_SINGLE("Dither 16-Bit Data Switch", CS4271_ADCCTL, 5, 1, 0),
+ SOC_DOUBLE("High Pass Filter Switch", CS4271_ADCCTL, 1, 0, 1, 1),
+ SOC_DOUBLE_R("Master Playback Switch", CS4271_VOLA, CS4271_VOLB,
+ 7, 1, 1),
+};
+
+static struct snd_soc_dai_ops cs4271_dai_ops = {
+ .hw_params = cs4271_hw_params,
+ .set_sysclk = cs4271_set_dai_sysclk,
+ .set_fmt = cs4271_set_dai_fmt,
+ .digital_mute = cs4271_digital_mute,
+};
+
+static struct snd_soc_dai_driver cs4271_dai = {
+ .name = "cs4271-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .formats = CS4271_PCM_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .formats = CS4271_PCM_FORMATS,
+ },
+ .ops = &cs4271_dai_ops,
+ .symmetric_rates = 1,
+};
+
+#ifdef CONFIG_PM
+static int cs4271_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg)
+{
+ int ret;
+ /* Set power-down bit */
+ ret = snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN);
+ if (ret < 0)
+ return ret;
+ return 0;
+}
+
+static int cs4271_soc_resume(struct snd_soc_codec *codec)
+{
+ int ret;
+ /* Restore codec state */
+ ret = snd_soc_cache_sync(codec);
+ if (ret < 0)
+ return ret;
+ /* then disable the power-down bit */
+ ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0);
+ if (ret < 0)
+ return ret;
+ return 0;
+}
+#else
+#define cs4271_soc_suspend NULL
+#define cs4271_soc_resume NULL
+#endif /* CONFIG_PM */
+
+static int cs4271_probe(struct snd_soc_codec *codec)
+{
+ struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
+ struct cs4271_platform_data *cs4271plat = codec->dev->platform_data;
+ int ret;
+ int gpio_nreset = -EINVAL;
+ int gpio_disable = -EINVAL;
+
+ codec->control_data = cs4271->control_data;
+
+ if (cs4271plat) {
+ if (gpio_is_valid(cs4271plat->gpio_nreset))
+ gpio_nreset = cs4271plat->gpio_nreset;
+ if (gpio_is_valid(cs4271plat->gpio_disable))
+ gpio_disable = cs4271plat->gpio_disable;
+ }
+
+ if (gpio_disable >= 0)
+ if (gpio_request(gpio_disable, "CS4271 Disable"))
+ gpio_disable = -EINVAL;
+ if (gpio_disable >= 0)
+ gpio_direction_output(gpio_disable, 0);
+
+ if (gpio_nreset >= 0)
+ if (gpio_request(gpio_nreset, "CS4271 Reset"))
+ gpio_nreset = -EINVAL;
+ if (gpio_nreset >= 0) {
+ /* Reset codec */
+ gpio_direction_output(gpio_nreset, 0);
+ udelay(1);
+ gpio_set_value(gpio_nreset, 1);
+ /* Give the codec time to wake up */
+ udelay(1);
+ }
+
+ cs4271->gpio_nreset = gpio_nreset;
+ cs4271->gpio_disable = gpio_disable;
+
+ /*
+ * In case of I2C, chip address specified in board data.
+ * So cache IO operations use 8 bit codec register address.
+ * In case of SPI, chip address and register address
+ * passed together as 16 bit value.
+ * Anyway, register address is masked with 0xFF inside
+ * soc-cache code.
+ */
+ if (cs4271->bus_type == SND_SOC_SPI)
+ ret = snd_soc_codec_set_cache_io(codec, 16, 8,
+ cs4271->bus_type);
+ else
+ ret = snd_soc_codec_set_cache_io(codec, 8, 8,
+ cs4271->bus_type);
+ if (ret) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_update_bits(codec, CS4271_MODE2, 0,
+ CS4271_MODE2_PDN | CS4271_MODE2_CPEN);
+ if (ret < 0)
+ return ret;
+ ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0);
+ if (ret < 0)
+ return ret;
+ /* Power-up sequence requires 85 uS */
+ udelay(85);
+
+ return snd_soc_add_controls(codec, cs4271_snd_controls,
+ ARRAY_SIZE(cs4271_snd_controls));
+}
+
+static int cs4271_remove(struct snd_soc_codec *codec)
+{
+ struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
+ int gpio_nreset, gpio_disable;
+
+ gpio_nreset = cs4271->gpio_nreset;
+ gpio_disable = cs4271->gpio_disable;
+
+ if (gpio_is_valid(gpio_nreset)) {
+ /* Set codec to the reset state */
+ gpio_set_value(gpio_nreset, 0);
+ gpio_free(gpio_nreset);
+ }
+
+ if (gpio_is_valid(gpio_disable))
+ gpio_free(gpio_disable);
+
+ return 0;
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_cs4271 = {
+ .probe = cs4271_probe,
+ .remove = cs4271_remove,
+ .suspend = cs4271_soc_suspend,
+ .resume = cs4271_soc_resume,
+ .reg_cache_default = cs4271_dflt_reg,
+ .reg_cache_size = ARRAY_SIZE(cs4271_dflt_reg),
+ .reg_word_size = sizeof(cs4271_dflt_reg[0]),
+ .compress_type = SND_SOC_FLAT_COMPRESSION,
+};
+
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit cs4271_spi_probe(struct spi_device *spi)
+{
+ struct cs4271_private *cs4271;
+
+ cs4271 = devm_kzalloc(&spi->dev, sizeof(*cs4271), GFP_KERNEL);
+ if (!cs4271)
+ return -ENOMEM;
+
+ spi_set_drvdata(spi, cs4271);
+ cs4271->control_data = spi;
+ cs4271->bus_type = SND_SOC_SPI;
+
+ return snd_soc_register_codec(&spi->dev, &soc_codec_dev_cs4271,
+ &cs4271_dai, 1);
+}
+
+static int __devexit cs4271_spi_remove(struct spi_device *spi)
+{
+ snd_soc_unregister_codec(&spi->dev);
+ return 0;
+}
+
+static struct spi_driver cs4271_spi_driver = {
+ .driver = {
+ .name = "cs4271",
+ .owner = THIS_MODULE,
+ },
+ .probe = cs4271_spi_probe,
+ .remove = __devexit_p(cs4271_spi_remove),
+};
+#endif /* defined(CONFIG_SPI_MASTER) */
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static struct i2c_device_id cs4271_i2c_id[] = {
+ {"cs4271", 0},
+ {}
+};
+MODULE_DEVICE_TABLE(i2c, cs4271_i2c_id);
+
+static int __devinit cs4271_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ struct cs4271_private *cs4271;
+
+ cs4271 = devm_kzalloc(&client->dev, sizeof(*cs4271), GFP_KERNEL);
+ if (!cs4271)
+ return -ENOMEM;
+
+ i2c_set_clientdata(client, cs4271);
+ cs4271->control_data = client;
+ cs4271->bus_type = SND_SOC_I2C;
+
+ return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4271,
+ &cs4271_dai, 1);
+}
+
+static int __devexit cs4271_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ return 0;
+}
+
+static struct i2c_driver cs4271_i2c_driver = {
+ .driver = {
+ .name = "cs4271",
+ .owner = THIS_MODULE,
+ },
+ .id_table = cs4271_i2c_id,
+ .probe = cs4271_i2c_probe,
+ .remove = __devexit_p(cs4271_i2c_remove),
+};
+#endif /* defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) */
+
+/*
+ * We only register our serial bus driver here without
+ * assignment to particular chip. So if any of the below
+ * fails, there is some problem with I2C or SPI subsystem.
+ * In most cases this module will be compiled with support
+ * of only one serial bus.
+ */
+static int __init cs4271_modinit(void)
+{
+ int ret;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ ret = i2c_add_driver(&cs4271_i2c_driver);
+ if (ret) {
+ pr_err("Failed to register CS4271 I2C driver: %d\n", ret);
+ return ret;
+ }
+#endif
+
+#if defined(CONFIG_SPI_MASTER)
+ ret = spi_register_driver(&cs4271_spi_driver);
+ if (ret) {
+ pr_err("Failed to register CS4271 SPI driver: %d\n", ret);
+ return ret;
+ }
+#endif
+
+ return 0;
+}
+module_init(cs4271_modinit);
+
+static void __exit cs4271_modexit(void)
+{
+#if defined(CONFIG_SPI_MASTER)
+ spi_unregister_driver(&cs4271_spi_driver);
+#endif
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_del_driver(&cs4271_i2c_driver);
+#endif
+}
+module_exit(cs4271_modexit);
+
+MODULE_AUTHOR("Alexander Sverdlin <subaparts@yandex.ru>");
+MODULE_DESCRIPTION("Cirrus Logic CS4271 ALSA SoC Codec Driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index 03d1e860d229..bb4bf65b9e7e 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -367,9 +367,12 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec)
return 0;
}
+static const u8 cx20442_reg = CX20442_TELOUT | CX20442_MIC;
+
static struct snd_soc_codec_driver cx20442_codec_dev = {
.probe = cx20442_codec_probe,
.remove = cx20442_codec_remove,
+ .reg_cache_default = &cx20442_reg,
.reg_cache_size = 1,
.reg_word_size = sizeof(u8),
.read = cx20442_read_reg_cache,
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index 593632cf791d..40e285df9ae5 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -33,6 +33,7 @@
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/initval.h>
+#include <sound/tlv.h>
#include "sn95031.h"
#define SN95031_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100)
@@ -145,6 +146,129 @@ static int sn95031_vihf_event(struct snd_soc_dapm_widget *w,
return 0;
}
+static int sn95031_dmic12_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ unsigned int ldo = 0, clk_dir = 0, data_dir = 0;
+
+ if (SND_SOC_DAPM_EVENT_ON(event)) {
+ ldo = BIT(5)|BIT(4);
+ clk_dir = BIT(0);
+ data_dir = BIT(7);
+ }
+ /* program DMIC LDO, clock and set clock */
+ snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(5)|BIT(4), ldo);
+ snd_soc_update_bits(w->codec, SN95031_DMICBUF0123, BIT(0), clk_dir);
+ snd_soc_update_bits(w->codec, SN95031_DMICBUF0123, BIT(7), data_dir);
+ return 0;
+}
+
+static int sn95031_dmic34_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ unsigned int ldo = 0, clk_dir = 0, data_dir = 0;
+
+ if (SND_SOC_DAPM_EVENT_ON(event)) {
+ ldo = BIT(5)|BIT(4);
+ clk_dir = BIT(2);
+ data_dir = BIT(1);
+ }
+ /* program DMIC LDO, clock and set clock */
+ snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(5)|BIT(4), ldo);
+ snd_soc_update_bits(w->codec, SN95031_DMICBUF0123, BIT(2), clk_dir);
+ snd_soc_update_bits(w->codec, SN95031_DMICBUF45, BIT(1), data_dir);
+ return 0;
+}
+
+static int sn95031_dmic56_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ unsigned int ldo = 0;
+
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ ldo = BIT(7)|BIT(6);
+
+ /* program DMIC LDO */
+ snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(7)|BIT(6), ldo);
+ return 0;
+}
+
+/* mux controls */
+static const char *sn95031_mic_texts[] = { "AMIC", "LineIn" };
+
+static const struct soc_enum sn95031_micl_enum =
+ SOC_ENUM_SINGLE(SN95031_ADCCONFIG, 1, 2, sn95031_mic_texts);
+
+static const struct snd_kcontrol_new sn95031_micl_mux_control =
+ SOC_DAPM_ENUM("Route", sn95031_micl_enum);
+
+static const struct soc_enum sn95031_micr_enum =
+ SOC_ENUM_SINGLE(SN95031_ADCCONFIG, 3, 2, sn95031_mic_texts);
+
+static const struct snd_kcontrol_new sn95031_micr_mux_control =
+ SOC_DAPM_ENUM("Route", sn95031_micr_enum);
+
+static const char *sn95031_input_texts[] = { "DMIC1", "DMIC2", "DMIC3",
+ "DMIC4", "DMIC5", "DMIC6",
+ "ADC Left", "ADC Right" };
+
+static const struct soc_enum sn95031_input1_enum =
+ SOC_ENUM_SINGLE(SN95031_AUDIOMUX12, 0, 8, sn95031_input_texts);
+
+static const struct snd_kcontrol_new sn95031_input1_mux_control =
+ SOC_DAPM_ENUM("Route", sn95031_input1_enum);
+
+static const struct soc_enum sn95031_input2_enum =
+ SOC_ENUM_SINGLE(SN95031_AUDIOMUX12, 4, 8, sn95031_input_texts);
+
+static const struct snd_kcontrol_new sn95031_input2_mux_control =
+ SOC_DAPM_ENUM("Route", sn95031_input2_enum);
+
+static const struct soc_enum sn95031_input3_enum =
+ SOC_ENUM_SINGLE(SN95031_AUDIOMUX34, 0, 8, sn95031_input_texts);
+
+static const struct snd_kcontrol_new sn95031_input3_mux_control =
+ SOC_DAPM_ENUM("Route", sn95031_input3_enum);
+
+static const struct soc_enum sn95031_input4_enum =
+ SOC_ENUM_SINGLE(SN95031_AUDIOMUX34, 4, 8, sn95031_input_texts);
+
+static const struct snd_kcontrol_new sn95031_input4_mux_control =
+ SOC_DAPM_ENUM("Route", sn95031_input4_enum);
+
+/* capture path controls */
+
+static const char *sn95031_micmode_text[] = {"Single Ended", "Differential"};
+
+/* 0dB to 30dB in 10dB steps */
+static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 10, 30);
+
+static const struct soc_enum sn95031_micmode1_enum =
+ SOC_ENUM_SINGLE(SN95031_MICAMP1, 1, 2, sn95031_micmode_text);
+static const struct soc_enum sn95031_micmode2_enum =
+ SOC_ENUM_SINGLE(SN95031_MICAMP2, 1, 2, sn95031_micmode_text);
+
+static const char *sn95031_dmic_cfg_text[] = {"GPO", "DMIC"};
+
+static const struct soc_enum sn95031_dmic12_cfg_enum =
+ SOC_ENUM_SINGLE(SN95031_DMICMUX, 0, 2, sn95031_dmic_cfg_text);
+static const struct soc_enum sn95031_dmic34_cfg_enum =
+ SOC_ENUM_SINGLE(SN95031_DMICMUX, 1, 2, sn95031_dmic_cfg_text);
+static const struct soc_enum sn95031_dmic56_cfg_enum =
+ SOC_ENUM_SINGLE(SN95031_DMICMUX, 2, 2, sn95031_dmic_cfg_text);
+
+static const struct snd_kcontrol_new sn95031_snd_controls[] = {
+ SOC_ENUM("Mic1Mode Capture Route", sn95031_micmode1_enum),
+ SOC_ENUM("Mic2Mode Capture Route", sn95031_micmode2_enum),
+ SOC_ENUM("DMIC12 Capture Route", sn95031_dmic12_cfg_enum),
+ SOC_ENUM("DMIC34 Capture Route", sn95031_dmic34_cfg_enum),
+ SOC_ENUM("DMIC56 Capture Route", sn95031_dmic56_cfg_enum),
+ SOC_SINGLE_TLV("Mic1 Capture Volume", SN95031_MICAMP1,
+ 2, 4, 0, mic_tlv),
+ SOC_SINGLE_TLV("Mic2 Capture Volume", SN95031_MICAMP2,
+ 2, 4, 0, mic_tlv),
+};
+
/* DAPM widgets */
static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = {
@@ -159,6 +283,36 @@ static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("VIB1OUT"),
SND_SOC_DAPM_OUTPUT("VIB2OUT"),
+ SND_SOC_DAPM_INPUT("AMIC1"), /* headset mic */
+ SND_SOC_DAPM_INPUT("AMIC2"),
+ SND_SOC_DAPM_INPUT("DMIC1"),
+ SND_SOC_DAPM_INPUT("DMIC2"),
+ SND_SOC_DAPM_INPUT("DMIC3"),
+ SND_SOC_DAPM_INPUT("DMIC4"),
+ SND_SOC_DAPM_INPUT("DMIC5"),
+ SND_SOC_DAPM_INPUT("DMIC6"),
+ SND_SOC_DAPM_INPUT("LINEINL"),
+ SND_SOC_DAPM_INPUT("LINEINR"),
+
+ SND_SOC_DAPM_MICBIAS("AMIC1Bias", SN95031_MICBIAS, 2, 0),
+ SND_SOC_DAPM_MICBIAS("AMIC2Bias", SN95031_MICBIAS, 3, 0),
+ SND_SOC_DAPM_MICBIAS("DMIC12Bias", SN95031_DMICMUX, 3, 0),
+ SND_SOC_DAPM_MICBIAS("DMIC34Bias", SN95031_DMICMUX, 4, 0),
+ SND_SOC_DAPM_MICBIAS("DMIC56Bias", SN95031_DMICMUX, 5, 0),
+
+ SND_SOC_DAPM_SUPPLY("DMIC12supply", SN95031_DMICLK, 0, 0,
+ sn95031_dmic12_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SUPPLY("DMIC34supply", SN95031_DMICLK, 1, 0,
+ sn95031_dmic34_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SUPPLY("DMIC56supply", SN95031_DMICLK, 2, 0,
+ sn95031_dmic56_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_AIF_OUT("PCM_Out", "Capture", 0,
+ SND_SOC_NOPM, 0, 0),
+
SND_SOC_DAPM_SUPPLY("Headset Rail", SND_SOC_NOPM, 0, 0,
sn95031_vhs_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
@@ -209,6 +363,40 @@ static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = {
SN95031_VIB1C5, 1, 0),
SND_SOC_DAPM_DAC("Vibra2 DAC", "Vibra2",
SN95031_VIB2C5, 1, 0),
+
+ /* capture widgets */
+ SND_SOC_DAPM_PGA("LineIn Enable Left", SN95031_MICAMP1,
+ 7, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("LineIn Enable Right", SN95031_MICAMP2,
+ 7, 0, NULL, 0),
+
+ SND_SOC_DAPM_PGA("MIC1 Enable", SN95031_MICAMP1, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("MIC2 Enable", SN95031_MICAMP2, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("TX1 Enable", SN95031_AUDIOTXEN, 2, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("TX2 Enable", SN95031_AUDIOTXEN, 3, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("TX3 Enable", SN95031_AUDIOTXEN, 4, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("TX4 Enable", SN95031_AUDIOTXEN, 5, 0, NULL, 0),
+
+ /* ADC have null stream as they will be turned ON by TX path */
+ SND_SOC_DAPM_ADC("ADC Left", NULL,
+ SN95031_ADCCONFIG, 0, 0),
+ SND_SOC_DAPM_ADC("ADC Right", NULL,
+ SN95031_ADCCONFIG, 2, 0),
+
+ SND_SOC_DAPM_MUX("Mic_InputL Capture Route",
+ SND_SOC_NOPM, 0, 0, &sn95031_micl_mux_control),
+ SND_SOC_DAPM_MUX("Mic_InputR Capture Route",
+ SND_SOC_NOPM, 0, 0, &sn95031_micr_mux_control),
+
+ SND_SOC_DAPM_MUX("Txpath1 Capture Route",
+ SND_SOC_NOPM, 0, 0, &sn95031_input1_mux_control),
+ SND_SOC_DAPM_MUX("Txpath2 Capture Route",
+ SND_SOC_NOPM, 0, 0, &sn95031_input2_mux_control),
+ SND_SOC_DAPM_MUX("Txpath3 Capture Route",
+ SND_SOC_NOPM, 0, 0, &sn95031_input3_mux_control),
+ SND_SOC_DAPM_MUX("Txpath4 Capture Route",
+ SND_SOC_NOPM, 0, 0, &sn95031_input4_mux_control),
+
};
static const struct snd_soc_dapm_route sn95031_audio_map[] = {
@@ -250,6 +438,87 @@ static const struct snd_soc_dapm_route sn95031_audio_map[] = {
{ "Lineout Right Playback", NULL, "Headset Right Filter"},
{ "Lineout Right Playback", NULL, "Speaker Right Filter"},
{ "Lineout Right Playback", NULL, "Vibra2 DAC"},
+
+ /* Headset (AMIC1) mic */
+ { "AMIC1Bias", NULL, "AMIC1"},
+ { "MIC1 Enable", NULL, "AMIC1Bias"},
+ { "Mic_InputL Capture Route", "AMIC", "MIC1 Enable"},
+
+ /* AMIC2 */
+ { "AMIC2Bias", NULL, "AMIC2"},
+ { "MIC2 Enable", NULL, "AMIC2Bias"},
+ { "Mic_InputR Capture Route", "AMIC", "MIC2 Enable"},
+
+
+ /* Linein */
+ { "LineIn Enable Left", NULL, "LINEINL"},
+ { "LineIn Enable Right", NULL, "LINEINR"},
+ { "Mic_InputL Capture Route", "LineIn", "LineIn Enable Left"},
+ { "Mic_InputR Capture Route", "LineIn", "LineIn Enable Right"},
+
+ /* ADC connection */
+ { "ADC Left", NULL, "Mic_InputL Capture Route"},
+ { "ADC Right", NULL, "Mic_InputR Capture Route"},
+
+ /*DMIC connections */
+ { "DMIC1", NULL, "DMIC12supply"},
+ { "DMIC2", NULL, "DMIC12supply"},
+ { "DMIC3", NULL, "DMIC34supply"},
+ { "DMIC4", NULL, "DMIC34supply"},
+ { "DMIC5", NULL, "DMIC56supply"},
+ { "DMIC6", NULL, "DMIC56supply"},
+
+ { "DMIC12Bias", NULL, "DMIC1"},
+ { "DMIC12Bias", NULL, "DMIC2"},
+ { "DMIC34Bias", NULL, "DMIC3"},
+ { "DMIC34Bias", NULL, "DMIC4"},
+ { "DMIC56Bias", NULL, "DMIC5"},
+ { "DMIC56Bias", NULL, "DMIC6"},
+
+ /*TX path inputs*/
+ { "Txpath1 Capture Route", "ADC Left", "ADC Left"},
+ { "Txpath2 Capture Route", "ADC Left", "ADC Left"},
+ { "Txpath3 Capture Route", "ADC Left", "ADC Left"},
+ { "Txpath4 Capture Route", "ADC Left", "ADC Left"},
+ { "Txpath1 Capture Route", "ADC Right", "ADC Right"},
+ { "Txpath2 Capture Route", "ADC Right", "ADC Right"},
+ { "Txpath3 Capture Route", "ADC Right", "ADC Right"},
+ { "Txpath4 Capture Route", "ADC Right", "ADC Right"},
+ { "Txpath1 Capture Route", NULL, "DMIC1"},
+ { "Txpath2 Capture Route", NULL, "DMIC1"},
+ { "Txpath3 Capture Route", NULL, "DMIC1"},
+ { "Txpath4 Capture Route", NULL, "DMIC1"},
+ { "Txpath1 Capture Route", NULL, "DMIC2"},
+ { "Txpath2 Capture Route", NULL, "DMIC2"},
+ { "Txpath3 Capture Route", NULL, "DMIC2"},
+ { "Txpath4 Capture Route", NULL, "DMIC2"},
+ { "Txpath1 Capture Route", NULL, "DMIC3"},
+ { "Txpath2 Capture Route", NULL, "DMIC3"},
+ { "Txpath3 Capture Route", NULL, "DMIC3"},
+ { "Txpath4 Capture Route", NULL, "DMIC3"},
+ { "Txpath1 Capture Route", NULL, "DMIC4"},
+ { "Txpath2 Capture Route", NULL, "DMIC4"},
+ { "Txpath3 Capture Route", NULL, "DMIC4"},
+ { "Txpath4 Capture Route", NULL, "DMIC4"},
+ { "Txpath1 Capture Route", NULL, "DMIC5"},
+ { "Txpath2 Capture Route", NULL, "DMIC5"},
+ { "Txpath3 Capture Route", NULL, "DMIC5"},
+ { "Txpath4 Capture Route", NULL, "DMIC5"},
+ { "Txpath1 Capture Route", NULL, "DMIC6"},
+ { "Txpath2 Capture Route", NULL, "DMIC6"},
+ { "Txpath3 Capture Route", NULL, "DMIC6"},
+ { "Txpath4 Capture Route", NULL, "DMIC6"},
+
+ /* tx path */
+ { "TX1 Enable", NULL, "Txpath1 Capture Route"},
+ { "TX2 Enable", NULL, "Txpath2 Capture Route"},
+ { "TX3 Enable", NULL, "Txpath3 Capture Route"},
+ { "TX4 Enable", NULL, "Txpath4 Capture Route"},
+ { "PCM_Out", NULL, "TX1 Enable"},
+ { "PCM_Out", NULL, "TX2 Enable"},
+ { "PCM_Out", NULL, "TX3 Enable"},
+ { "PCM_Out", NULL, "TX4 Enable"},
+
};
/* speaker and headset mutes, for audio pops and clicks */
@@ -339,6 +608,13 @@ struct snd_soc_dai_driver sn95031_dais[] = {
.rates = SN95031_RATES,
.formats = SN95031_FORMATS,
},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 5,
+ .rates = SN95031_RATES,
+ .formats = SN95031_FORMATS,
+ },
.ops = &sn95031_headset_dai_ops,
},
{ .name = "SN95031 Speaker",
@@ -390,6 +666,8 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec)
snd_soc_write(codec, SN95031_PCM2RXSLOT01, 0x10);
snd_soc_write(codec, SN95031_PCM2RXSLOT23, 0x32);
snd_soc_write(codec, SN95031_PCM2RXSLOT45, 0x54);
+ snd_soc_write(codec, SN95031_PCM2TXSLOT01, 0x10);
+ snd_soc_write(codec, SN95031_PCM2TXSLOT23, 0x32);
/* pcm port setting
* This sets the pcm port to slave and clock at 19.2Mhz which
* can support 6slots, sampling rate set per stream in hw-params
@@ -423,6 +701,9 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec)
snd_soc_write(codec, SN95031_SSR2, 0x10);
snd_soc_write(codec, SN95031_SSR3, 0x40);
+ snd_soc_add_controls(codec, sn95031_snd_controls,
+ ARRAY_SIZE(sn95031_snd_controls));
+
ret = snd_soc_dapm_new_controls(&codec->dapm, sn95031_dapm_widgets,
ARRAY_SIZE(sn95031_dapm_widgets));
if (ret)
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index a2a446cb1807..3d4c55f3c7b5 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -2,6 +2,7 @@
* wm8903.c -- WM8903 ALSA SoC Audio driver
*
* Copyright 2008 Wolfson Microelectronics
+ * Copyright 2011 NVIDIA, Inc.
*
* Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
*
@@ -19,6 +20,7 @@
#include <linux/init.h>
#include <linux/completion.h>
#include <linux/delay.h>
+#include <linux/gpio.h>
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
@@ -213,6 +215,7 @@ static u16 wm8903_reg_defaults[] = {
};
struct wm8903_priv {
+ struct snd_soc_codec *codec;
int sysclk;
int irq;
@@ -230,6 +233,10 @@ struct wm8903_priv {
int mic_short;
int mic_last_report;
int mic_delay;
+
+#ifdef CONFIG_GPIOLIB
+ struct gpio_chip gpio_chip;
+#endif
};
static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int reg)
@@ -1635,6 +1642,120 @@ static int wm8903_resume(struct snd_soc_codec *codec)
return 0;
}
+#ifdef CONFIG_GPIOLIB
+static inline struct wm8903_priv *gpio_to_wm8903(struct gpio_chip *chip)
+{
+ return container_of(chip, struct wm8903_priv, gpio_chip);
+}
+
+static int wm8903_gpio_request(struct gpio_chip *chip, unsigned offset)
+{
+ if (offset >= WM8903_NUM_GPIO)
+ return -EINVAL;
+
+ return 0;
+}
+
+static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset)
+{
+ struct wm8903_priv *wm8903 = gpio_to_wm8903(chip);
+ struct snd_soc_codec *codec = wm8903->codec;
+ unsigned int mask, val;
+
+ mask = WM8903_GP1_FN_MASK | WM8903_GP1_DIR_MASK;
+ val = (WM8903_GPn_FN_GPIO_INPUT << WM8903_GP1_FN_SHIFT) |
+ WM8903_GP1_DIR;
+
+ return snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset,
+ mask, val);
+}
+
+static int wm8903_gpio_get(struct gpio_chip *chip, unsigned offset)
+{
+ struct wm8903_priv *wm8903 = gpio_to_wm8903(chip);
+ struct snd_soc_codec *codec = wm8903->codec;
+ int reg;
+
+ reg = snd_soc_read(codec, WM8903_GPIO_CONTROL_1 + offset);
+
+ return (reg & WM8903_GP1_LVL_MASK) >> WM8903_GP1_LVL_SHIFT;
+}
+
+static int wm8903_gpio_direction_out(struct gpio_chip *chip,
+ unsigned offset, int value)
+{
+ struct wm8903_priv *wm8903 = gpio_to_wm8903(chip);
+ struct snd_soc_codec *codec = wm8903->codec;
+ unsigned int mask, val;
+
+ mask = WM8903_GP1_FN_MASK | WM8903_GP1_DIR_MASK | WM8903_GP1_LVL_MASK;
+ val = (WM8903_GPn_FN_GPIO_OUTPUT << WM8903_GP1_FN_SHIFT) |
+ (value << WM8903_GP2_LVL_SHIFT);
+
+ return snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset,
+ mask, val);
+}
+
+static void wm8903_gpio_set(struct gpio_chip *chip, unsigned offset, int value)
+{
+ struct wm8903_priv *wm8903 = gpio_to_wm8903(chip);
+ struct snd_soc_codec *codec = wm8903->codec;
+
+ snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset,
+ WM8903_GP1_LVL_MASK,
+ !!value << WM8903_GP1_LVL_SHIFT);
+}
+
+static struct gpio_chip wm8903_template_chip = {
+ .label = "wm8903",
+ .owner = THIS_MODULE,
+ .request = wm8903_gpio_request,
+ .direction_input = wm8903_gpio_direction_in,
+ .get = wm8903_gpio_get,
+ .direction_output = wm8903_gpio_direction_out,
+ .set = wm8903_gpio_set,
+ .can_sleep = 1,
+};
+
+static void wm8903_init_gpio(struct snd_soc_codec *codec)
+{
+ struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
+ struct wm8903_platform_data *pdata = dev_get_platdata(codec->dev);
+ int ret;
+
+ wm8903->gpio_chip = wm8903_template_chip;
+ wm8903->gpio_chip.ngpio = WM8903_NUM_GPIO;
+ wm8903->gpio_chip.dev = codec->dev;
+
+ if (pdata && pdata->gpio_base)
+ wm8903->gpio_chip.base = pdata->gpio_base;
+ else
+ wm8903->gpio_chip.base = -1;
+
+ ret = gpiochip_add(&wm8903->gpio_chip);
+ if (ret != 0)
+ dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret);
+}
+
+static void wm8903_free_gpio(struct snd_soc_codec *codec)
+{
+ struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ ret = gpiochip_remove(&wm8903->gpio_chip);
+ if (ret != 0)
+ dev_err(codec->dev, "Failed to remove GPIOs: %d\n", ret);
+}
+#else
+static void wm8903_init_gpio(struct snd_soc_codec *codec)
+{
+}
+
+static void wm8903_free_gpio(struct snd_soc_codec *codec)
+{
+}
+#endif
+
static int wm8903_probe(struct snd_soc_codec *codec)
{
struct wm8903_platform_data *pdata = dev_get_platdata(codec->dev);
@@ -1643,6 +1764,7 @@ static int wm8903_probe(struct snd_soc_codec *codec)
int trigger, irq_pol;
u16 val;
+ wm8903->codec = codec;
init_completion(&wm8903->wseq);
ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C);
@@ -1667,7 +1789,7 @@ static int wm8903_probe(struct snd_soc_codec *codec)
/* Set up GPIOs and microphone detection */
if (pdata) {
for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) {
- if (!pdata->gpio_cfg[i])
+ if (pdata->gpio_cfg[i] == WM8903_GPIO_NO_CONFIG)
continue;
snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i,
@@ -1749,12 +1871,15 @@ static int wm8903_probe(struct snd_soc_codec *codec)
ARRAY_SIZE(wm8903_snd_controls));
wm8903_add_widgets(codec);
+ wm8903_init_gpio(codec);
+
return ret;
}
/* power down chip */
static int wm8903_remove(struct snd_soc_codec *codec)
{
+ wm8903_free_gpio(codec);
wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 5c7b730a864f..3b71dd65c966 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3635,7 +3635,7 @@ static void wm8962_gpio_set(struct gpio_chip *chip, unsigned offset, int value)
struct snd_soc_codec *codec = wm8962->codec;
snd_soc_update_bits(codec, WM8962_GPIO_BASE + offset,
- WM8962_GP2_LVL, value << WM8962_GP2_LVL_SHIFT);
+ WM8962_GP2_LVL, !!value << WM8962_GP2_LVL_SHIFT);
}
static int wm8962_gpio_direction_out(struct gpio_chip *chip,
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 0bb0bb40b842..0ca81d3c64e8 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2386,7 +2386,7 @@ static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate)
else
val = 0;
- return snd_soc_update_bits(codec, reg, mask, reg);
+ return snd_soc_update_bits(codec, reg, mask, val);
}
#define WM8994_RATES SNDRV_PCM_RATE_8000_96000
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 7d563413df3e..67eaaecbb42e 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -1265,7 +1265,7 @@ static int wm8995_set_tristate(struct snd_soc_dai *codec_dai, int tristate)
else
val = 0;
- return snd_soc_update_bits(codec, reg, mask, reg);
+ return snd_soc_update_bits(codec, reg, mask, val);
}
/* The size in bits of the FLL divide multiplied by 10
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index c466982eed23..613df5db0b32 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -91,6 +91,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op)
static void calibrate_dc_servo(struct snd_soc_codec *codec)
{
struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
+ s8 offset;
u16 reg, reg_l, reg_r, dcs_cfg;
/* If we're using a digital only path and have a previously
@@ -149,16 +150,14 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
hubs->dcs_codes);
/* HPOUT1L */
- if (reg_l + hubs->dcs_codes > 0 &&
- reg_l + hubs->dcs_codes < 0xff)
- reg_l += hubs->dcs_codes;
- dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
+ offset = reg_l;
+ offset += hubs->dcs_codes;
+ dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
/* HPOUT1R */
- if (reg_r + hubs->dcs_codes > 0 &&
- reg_r + hubs->dcs_codes < 0xff)
- reg_r += hubs->dcs_codes;
- dcs_cfg |= reg_r;
+ offset = reg_r;
+ offset += hubs->dcs_codes;
+ dcs_cfg |= (u8)offset;
dev_dbg(codec->dev, "DCS result: %x\n", dcs_cfg);
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 0c2d6bacc681..b36f0b39b090 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -223,7 +223,7 @@ static struct snd_soc_dai_link da8xx_evm_dai = {
.stream_name = "AIC3X",
.cpu_dai_name= "davinci-mcasp.0",
.codec_dai_name = "tlv320aic3x-hifi",
- .codec_name = "tlv320aic3x-codec.0-001a",
+ .codec_name = "tlv320aic3x-codec.1-0018",
.platform_name = "davinci-pcm-audio",
.init = evm_aic3x_init,
.ops = &evm_ops,
diff --git a/sound/soc/ep93xx/Kconfig b/sound/soc/ep93xx/Kconfig
index 57429041189c..91a28de94109 100644
--- a/sound/soc/ep93xx/Kconfig
+++ b/sound/soc/ep93xx/Kconfig
@@ -30,3 +30,12 @@ config SND_EP93XX_SOC_SIMONE
help
Say Y or M here if you want to add support for AC97 audio on the
Simplemachines Sim.One board.
+
+config SND_EP93XX_SOC_EDB93XX
+ tristate "SoC Audio support for Cirrus Logic EDB93xx boards"
+ depends on SND_EP93XX_SOC && (MACH_EDB9301 || MACH_EDB9302 || MACH_EDB9302A || MACH_EDB9307A || MACH_EDB9315A)
+ select SND_EP93XX_SOC_I2S
+ select SND_SOC_CS4271
+ help
+ Say Y or M here if you want to add support for I2S audio on the
+ Cirrus Logic EDB93xx boards.
diff --git a/sound/soc/ep93xx/Makefile b/sound/soc/ep93xx/Makefile
index 8e7977fb6b7d..5514146cbdf0 100644
--- a/sound/soc/ep93xx/Makefile
+++ b/sound/soc/ep93xx/Makefile
@@ -10,6 +10,8 @@ obj-$(CONFIG_SND_EP93XX_SOC_AC97) += snd-soc-ep93xx-ac97.o
# EP93XX Machine Support
snd-soc-snappercl15-objs := snappercl15.o
snd-soc-simone-objs := simone.o
+snd-soc-edb93xx-objs := edb93xx.o
obj-$(CONFIG_SND_EP93XX_SOC_SNAPPERCL15) += snd-soc-snappercl15.o
obj-$(CONFIG_SND_EP93XX_SOC_SIMONE) += snd-soc-simone.o
+obj-$(CONFIG_SND_EP93XX_SOC_EDB93XX) += snd-soc-edb93xx.o
diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c
new file mode 100644
index 000000000000..b270085227f3
--- /dev/null
+++ b/sound/soc/ep93xx/edb93xx.c
@@ -0,0 +1,142 @@
+/*
+ * SoC audio for EDB93xx
+ *
+ * Copyright (c) 2010 Alexander Sverdlin <subaparts@yandex.ru>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * This driver support CS4271 codec being master or slave, working
+ * in control port mode, connected either via SPI or I2C.
+ * The data format accepted is I2S or left-justified.
+ * DAPM support not implemented.
+ */
+
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include "ep93xx-pcm.h"
+
+#define edb93xx_has_audio() (machine_is_edb9301() || \
+ machine_is_edb9302() || \
+ machine_is_edb9302a() || \
+ machine_is_edb9307a() || \
+ machine_is_edb9315a())
+
+static int edb93xx_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ int err;
+ unsigned int rate = params_rate(params);
+ /*
+ * We set LRCLK equal to `rate' and SCLK = LRCLK * 64,
+ * because our sample size is 32 bit * 2 channels.
+ * I2S standard permits us to transmit more bits than
+ * the codec uses.
+ * MCLK = SCLK * 4 is the best recommended value,
+ * but we have to fall back to ratio 2 for higher
+ * sample rates.
+ */
+ unsigned int mclk_rate = rate * 64 * ((rate <= 48000) ? 4 : 2);
+
+ err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_IF |
+ SND_SOC_DAIFMT_CBS_CFS);
+ if (err)
+ return err;
+
+ err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_IF |
+ SND_SOC_DAIFMT_CBS_CFS);
+ if (err)
+ return err;
+
+ err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk_rate,
+ SND_SOC_CLOCK_IN);
+ if (err)
+ return err;
+
+ return snd_soc_dai_set_sysclk(cpu_dai, 0, mclk_rate,
+ SND_SOC_CLOCK_OUT);
+}
+
+static struct snd_soc_ops edb93xx_ops = {
+ .hw_params = edb93xx_hw_params,
+};
+
+static struct snd_soc_dai_link edb93xx_dai = {
+ .name = "CS4271",
+ .stream_name = "CS4271 HiFi",
+ .platform_name = "ep93xx-pcm-audio",
+ .cpu_dai_name = "ep93xx-i2s",
+ .codec_name = "spi0.0",
+ .codec_dai_name = "cs4271-hifi",
+ .ops = &edb93xx_ops,
+};
+
+static struct snd_soc_card snd_soc_edb93xx = {
+ .name = "EDB93XX",
+ .dai_link = &edb93xx_dai,
+ .num_links = 1,
+};
+
+static struct platform_device *edb93xx_snd_device;
+
+static int __init edb93xx_init(void)
+{
+ int ret;
+
+ if (!edb93xx_has_audio())
+ return -ENODEV;
+
+ ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97,
+ EP93XX_SYSCON_I2SCLKDIV_ORIDE |
+ EP93XX_SYSCON_I2SCLKDIV_SPOL);
+ if (ret)
+ return ret;
+
+ edb93xx_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!edb93xx_snd_device) {
+ ret = -ENOMEM;
+ goto free_i2s;
+ }
+
+ platform_set_drvdata(edb93xx_snd_device, &snd_soc_edb93xx);
+ ret = platform_device_add(edb93xx_snd_device);
+ if (ret)
+ goto device_put;
+
+ return 0;
+
+device_put:
+ platform_device_put(edb93xx_snd_device);
+free_i2s:
+ ep93xx_i2s_release();
+ return ret;
+}
+module_init(edb93xx_init);
+
+static void __exit edb93xx_exit(void)
+{
+ platform_device_unregister(edb93xx_snd_device);
+ ep93xx_i2s_release();
+}
+module_exit(edb93xx_exit);
+
+MODULE_AUTHOR("Alexander Sverdlin <subaparts@yandex.ru>");
+MODULE_DESCRIPTION("ALSA SoC EDB93xx");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index 7d7847a1e66b..c16c6b2eff95 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -53,9 +53,8 @@ struct mpc8610_hpcd_data {
*
* Here we program the DMACR and PMUXCR registers.
*/
-static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device)
+static int mpc8610_hpcd_machine_probe(struct snd_soc_card *card)
{
- struct snd_soc_card *card = platform_get_drvdata(sound_device);
struct mpc8610_hpcd_data *machine_data =
container_of(card, struct mpc8610_hpcd_data, card);
struct ccsr_guts_86xx __iomem *guts;
@@ -138,9 +137,8 @@ static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream)
* This function is called to remove the sound device for one SSI. We
* de-program the DMACR and PMUXCR register.
*/
-static int mpc8610_hpcd_machine_remove(struct platform_device *sound_device)
+static int mpc8610_hpcd_machine_remove(struct snd_soc_card *card)
{
- struct snd_soc_card *card = platform_get_drvdata(sound_device);
struct mpc8610_hpcd_data *machine_data =
container_of(card, struct mpc8610_hpcd_data, card);
struct ccsr_guts_86xx __iomem *guts;
diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c
index 026b756961e0..66e0b68af147 100644
--- a/sound/soc/fsl/p1022_ds.c
+++ b/sound/soc/fsl/p1022_ds.c
@@ -85,9 +85,8 @@ struct machine_data {
*
* Here we program the DMACR and PMUXCR registers.
*/
-static int p1022_ds_machine_probe(struct platform_device *sound_device)
+static int p1022_ds_machine_probe(struct snd_soc_card *card)
{
- struct snd_soc_card *card = platform_get_drvdata(sound_device);
struct machine_data *mdata =
container_of(card, struct machine_data, card);
struct ccsr_guts_85xx __iomem *guts;
@@ -160,9 +159,8 @@ static int p1022_ds_startup(struct snd_pcm_substream *substream)
* This function is called to remove the sound device for one SSI. We
* de-program the DMACR and PMUXCR register.
*/
-static int p1022_ds_machine_remove(struct platform_device *sound_device)
+static int p1022_ds_machine_remove(struct snd_soc_card *card)
{
- struct snd_soc_card *card = platform_get_drvdata(sound_device);
struct machine_data *mdata =
container_of(card, struct machine_data, card);
struct ccsr_guts_85xx __iomem *guts;
diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c
index 1a330be1a01e..7925851a5de1 100644
--- a/sound/soc/mid-x86/mfld_machine.c
+++ b/sound/soc/mid-x86/mfld_machine.c
@@ -182,6 +182,9 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime)
snd_soc_dapm_disable_pin(dapm, "LINEOUTR");
lo_dac = 3;
hs_switch = 0;
+ /* we dont use linein in this so set to NC */
+ snd_soc_dapm_disable_pin(dapm, "LINEINL");
+ snd_soc_dapm_disable_pin(dapm, "LINEINR");
return snd_soc_dapm_sync(dapm);
}
diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c
index 1d1f5447b338..96e6e9c9c5f4 100644
--- a/sound/soc/mid-x86/sst_platform.c
+++ b/sound/soc/mid-x86/sst_platform.c
@@ -73,6 +73,12 @@ struct snd_soc_dai_driver sst_platform_dai[] = {
.rates = SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S24_LE,
},
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 5,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S24_LE,
+ },
},
{
.name = "Speaker-cpu-dai",
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 2101bdcee21f..3167be689621 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -507,8 +507,6 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
/* Set up digital mute if not provided by the codec */
if (!codec_dai->driver->ops) {
codec_dai->driver->ops = &ams_delta_dai_ops;
- } else if (!codec_dai->driver->ops->digital_mute) {
- codec_dai->driver->ops->digital_mute = ams_delta_digital_mute;
} else {
ams_delta_ops.startup = ams_delta_startup;
ams_delta_ops.shutdown = ams_delta_shutdown;
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index fc592f0d5fc7..784cff5f67e8 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -307,10 +307,10 @@ static int corgi_wm8731_init(struct snd_soc_pcm_runtime *rtd)
static struct snd_soc_dai_link corgi_dai = {
.name = "WM8731",
.stream_name = "WM8731",
- .cpu_dai_name = "pxa-is2-dai",
+ .cpu_dai_name = "pxa2xx-i2s",
.codec_dai_name = "wm8731-hifi",
.platform_name = "pxa-pcm-audio",
- .codec_name = "wm8731-codec-0.001a",
+ .codec_name = "wm8731-codec-0.001b",
.init = corgi_wm8731_init,
.ops = &corgi_ops,
};
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 6298ee115e27..a7d4999f9b24 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -276,7 +276,7 @@ static struct snd_soc_dai_link poodle_dai = {
.cpu_dai_name = "pxa2xx-i2s",
.codec_dai_name = "wm8731-hifi",
.platform_name = "pxa-pcm-audio",
- .codec_name = "wm8731-codec.0-001a",
+ .codec_name = "wm8731-codec.0-001b",
.init = poodle_wm8731_init,
.ops = &poodle_ops,
};
diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c
index 0fd60f423036..db1dd560a585 100644
--- a/sound/soc/pxa/raumfeld.c
+++ b/sound/soc/pxa/raumfeld.c
@@ -151,13 +151,13 @@ static struct snd_soc_ops raumfeld_cs4270_ops = {
.hw_params = raumfeld_cs4270_hw_params,
};
-static int raumfeld_line_suspend(struct platform_device *pdev, pm_message_t state)
+static int raumfeld_line_suspend(struct snd_soc_card *card)
{
raumfeld_enable_audio(false);
return 0;
}
-static int raumfeld_line_resume(struct platform_device *pdev)
+static int raumfeld_line_resume(struct snd_soc_card *card)
{
raumfeld_enable_audio(true);
return 0;
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index c2acb69b957a..8e1571350630 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -315,10 +315,10 @@ static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd)
static struct snd_soc_dai_link spitz_dai = {
.name = "wm8750",
.stream_name = "WM8750",
- .cpu_dai_name = "pxa-is2",
+ .cpu_dai_name = "pxa2xx-i2s",
.codec_dai_name = "wm8750-hifi",
.platform_name = "pxa-pcm-audio",
- .codec_name = "wm8750-codec.0-001a",
+ .codec_name = "wm8750-codec.0-001b",
.init = spitz_wm8750_init,
.ops = &spitz_ops,
};
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index f75804ef0897..489139a31cf9 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -237,7 +237,7 @@ static struct snd_soc_dai_link tosa_dai[] = {
},
};
-static int tosa_probe(struct platform_device *dev)
+static int tosa_probe(struct snd_soc_card *card)
{
int ret;
@@ -251,7 +251,7 @@ static int tosa_probe(struct platform_device *dev)
return ret;
}
-static int tosa_remove(struct platform_device *dev)
+static int tosa_remove(struct snd_soc_card *card)
{
gpio_free(TOSA_GPIO_L_MUTE);
return 0;
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index b222a7d72027..c5858296b48a 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -189,7 +189,7 @@ static struct snd_soc_dai_link zylonite_dai[] = {
},
};
-static int zylonite_probe(struct platform_device *pdev)
+static int zylonite_probe(struct snd_soc_card *card)
{
int ret;
@@ -216,7 +216,7 @@ static int zylonite_probe(struct platform_device *pdev)
return 0;
}
-static int zylonite_remove(struct platform_device *pdev)
+static int zylonite_remove(struct snd_soc_card *card)
{
if (clk_pout) {
clk_disable(pout);
@@ -226,8 +226,7 @@ static int zylonite_remove(struct platform_device *pdev)
return 0;
}
-static int zylonite_suspend_post(struct platform_device *pdev,
- pm_message_t state)
+static int zylonite_suspend_post(struct snd_soc_card *card)
{
if (clk_pout)
clk_disable(pout);
@@ -235,7 +234,7 @@ static int zylonite_suspend_post(struct platform_device *pdev,
return 0;
}
-static int zylonite_resume_pre(struct platform_device *pdev)
+static int zylonite_resume_pre(struct snd_soc_card *card)
{
int ret = 0;
diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c
index 69e08fd1ff00..95ebf812b146 100644
--- a/sound/soc/samsung/neo1973_gta02_wm8753.c
+++ b/sound/soc/samsung/neo1973_gta02_wm8753.c
@@ -387,11 +387,11 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = {
{ /* Hifi Playback - for similatious use with voice below */
.name = "WM8753",
.stream_name = "WM8753 HiFi",
- .cpu_dai_name = "s3c24xx-i2s",
+ .cpu_dai_name = "s3c24xx-iis",
.codec_dai_name = "wm8753-hifi",
.init = neo1973_gta02_wm8753_init,
.platform_name = "samsung-audio",
- .codec_name = "wm8753-codec.0-0x1a",
+ .codec_name = "wm8753-codec.0-001a",
.ops = &neo1973_gta02_hifi_ops,
},
{ /* Voice via BT */
@@ -400,7 +400,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = {
.cpu_dai_name = "bluetooth-dai",
.codec_dai_name = "wm8753-voice",
.ops = &neo1973_gta02_voice_ops,
- .codec_name = "wm8753-codec.0-0x1a",
+ .codec_name = "wm8753-codec.0-001a",
.platform_name = "samsung-audio",
},
};
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index c7a24514beb5..d3cd6888a810 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -24,7 +24,6 @@
#include <sound/tlv.h>
#include <asm/mach-types.h>
-#include <asm/hardware/scoop.h>
#include <mach/regs-clock.h>
#include <mach/regs-gpio.h>
#include <mach/hardware.h>
@@ -559,9 +558,9 @@ static struct snd_soc_dai_link neo1973_dai[] = {
.name = "WM8753",
.stream_name = "WM8753 HiFi",
.platform_name = "samsung-audio",
- .cpu_dai_name = "s3c24xx-i2s",
+ .cpu_dai_name = "s3c24xx-iis",
.codec_dai_name = "wm8753-hifi",
- .codec_name = "wm8753-codec.0-0x1a",
+ .codec_name = "wm8753-codec.0-001a",
.init = neo1973_wm8753_init,
.ops = &neo1973_hifi_ops,
},
@@ -571,7 +570,7 @@ static struct snd_soc_dai_link neo1973_dai[] = {
.platform_name = "samsung-audio",
.cpu_dai_name = "bluetooth-dai",
.codec_dai_name = "wm8753-voice",
- .codec_name = "wm8753-codec.0-0x1a",
+ .codec_name = "wm8753-codec.0-001a",
.ops = &neo1973_voice_ops,
},
};
diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c
index d7b3e6e9d783..ce6aef604179 100644
--- a/sound/soc/samsung/s3c24xx_simtec_hermes.c
+++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c
@@ -84,8 +84,8 @@ static int simtec_hermes_init(struct snd_soc_pcm_runtime *rtd)
static struct snd_soc_dai_link simtec_dai_aic33 = {
.name = "tlv320aic33",
.stream_name = "TLV320AIC33",
- .codec_name = "tlv320aic3x-codec.0-0x1a",
- .cpu_dai_name = "s3c24xx-i2s",
+ .codec_name = "tlv320aic3x-codec.0-001a",
+ .cpu_dai_name = "s3c24xx-iis",
.codec_dai_name = "tlv320aic3x-hifi",
.platform_name = "samsung-audio",
.init = simtec_hermes_init,
diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
index ff6168f5f2e0..a7ef7db54687 100644
--- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
+++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
@@ -73,8 +73,8 @@ static int simtec_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
static struct snd_soc_dai_link simtec_dai_aic23 = {
.name = "tlv320aic23",
.stream_name = "TLV320AIC23",
- .codec_name = "tlv320aic3x-codec.0-0x1a",
- .cpu_dai_name = "s3c24xx-i2s",
+ .codec_name = "tlv320aic3x-codec.0-001a",
+ .cpu_dai_name = "s3c24xx-iis",
.codec_dai_name = "tlv320aic3x-hifi",
.platform_name = "samsung-audio",
.init = simtec_tlv320aic23_init,
diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c
index ce749a10ec07..3cb700751078 100644
--- a/sound/soc/samsung/s3c24xx_uda134x.c
+++ b/sound/soc/samsung/s3c24xx_uda134x.c
@@ -221,7 +221,7 @@ static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = {
.stream_name = "UDA134X",
.codec_name = "uda134x-hifi",
.codec_dai_name = "uda134x-hifi",
- .cpu_dai_name = "s3c24xx-i2s",
+ .cpu_dai_name = "s3c24xx-iis",
.ops = &s3c24xx_uda134x_ops,
.platform_name = "samsung-audio",
};
diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c
index 56cd34223100..d6f4703b3c07 100644
--- a/sound/soc/sh/fsi-ak4642.c
+++ b/sound/soc/sh/fsi-ak4642.c
@@ -23,14 +23,21 @@ struct fsi_ak4642_data {
static int fsi_ak4642_dai_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_dai *dai = rtd->codec_dai;
+ struct snd_soc_dai *codec = rtd->codec_dai;
+ struct snd_soc_dai *cpu = rtd->cpu_dai;
int ret;
- ret = snd_soc_dai_set_fmt(dai, SND_SOC_DAIFMT_CBM_CFM);
+ ret = snd_soc_dai_set_fmt(codec, SND_SOC_DAIFMT_LEFT_J |
+ SND_SOC_DAIFMT_CBM_CFM);
if (ret < 0)
return ret;
- ret = snd_soc_dai_set_sysclk(dai, 0, 11289600, 0);
+ ret = snd_soc_dai_set_sysclk(codec, 0, 11289600, 0);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_LEFT_J |
+ SND_SOC_DAIFMT_CBS_CFS);
return ret;
}
diff --git a/sound/soc/sh/fsi-da7210.c b/sound/soc/sh/fsi-da7210.c
index e8df9da92f71..dbafd7ac5590 100644
--- a/sound/soc/sh/fsi-da7210.c
+++ b/sound/soc/sh/fsi-da7210.c
@@ -15,11 +15,20 @@
static int fsi_da7210_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_dai *dai = rtd->codec_dai;
+ struct snd_soc_dai *codec = rtd->codec_dai;
+ struct snd_soc_dai *cpu = rtd->cpu_dai;
+ int ret;
- return snd_soc_dai_set_fmt(dai,
+ ret = snd_soc_dai_set_fmt(codec,
SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_CBS_CFS);
+
+ return ret;
}
static struct snd_soc_dai_link fsi_da7210_dai = {
diff --git a/sound/soc/sh/fsi-hdmi.c b/sound/soc/sh/fsi-hdmi.c
index a52dd8ec71d3..9719985eb82d 100644
--- a/sound/soc/sh/fsi-hdmi.c
+++ b/sound/soc/sh/fsi-hdmi.c
@@ -12,31 +12,59 @@
#include <linux/platform_device.h>
#include <sound/sh_fsi.h>
+struct fsi_hdmi_data {
+ const char *cpu_dai;
+ const char *card;
+ int id;
+};
+
+static int fsi_hdmi_dai_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_dai *cpu = rtd->cpu_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_CBM_CFM);
+
+ return ret;
+}
+
static struct snd_soc_dai_link fsi_dai_link = {
.name = "HDMI",
.stream_name = "HDMI",
- .cpu_dai_name = "fsib-dai", /* fsi B */
.codec_dai_name = "sh_mobile_hdmi-hifi",
.platform_name = "sh_fsi2",
.codec_name = "sh-mobile-hdmi",
+ .init = fsi_hdmi_dai_init,
};
static struct snd_soc_card fsi_soc_card = {
- .name = "FSI (SH MOBILE HDMI)",
.dai_link = &fsi_dai_link,
.num_links = 1,
};
static struct platform_device *fsi_snd_device;
-static int __init fsi_hdmi_init(void)
+static int fsi_hdmi_probe(struct platform_device *pdev)
{
int ret = -ENOMEM;
+ const struct platform_device_id *id_entry;
+ struct fsi_hdmi_data *pdata;
+
+ id_entry = pdev->id_entry;
+ if (!id_entry) {
+ dev_err(&pdev->dev, "unknown fsi hdmi\n");
+ return -ENODEV;
+ }
- fsi_snd_device = platform_device_alloc("soc-audio", FSI_PORT_B);
+ pdata = (struct fsi_hdmi_data *)id_entry->driver_data;
+
+ fsi_snd_device = platform_device_alloc("soc-audio", pdata->id);
if (!fsi_snd_device)
goto out;
+ fsi_dai_link.cpu_dai_name = pdata->cpu_dai;
+ fsi_soc_card.name = pdata->card;
+
platform_set_drvdata(fsi_snd_device, &fsi_soc_card);
ret = platform_device_add(fsi_snd_device);
@@ -47,9 +75,48 @@ out:
return ret;
}
-static void __exit fsi_hdmi_exit(void)
+static int fsi_hdmi_remove(struct platform_device *pdev)
{
platform_device_unregister(fsi_snd_device);
+ return 0;
+}
+
+static struct fsi_hdmi_data fsi2_a_hdmi = {
+ .cpu_dai = "fsia-dai",
+ .card = "FSI2A (SH MOBILE HDMI)",
+ .id = FSI_PORT_A,
+};
+
+static struct fsi_hdmi_data fsi2_b_hdmi = {
+ .cpu_dai = "fsib-dai",
+ .card = "FSI2B (SH MOBILE HDMI)",
+ .id = FSI_PORT_B,
+};
+
+static struct platform_device_id fsi_id_table[] = {
+ /* FSI 2 */
+ { "sh_fsi2_a_hdmi", (kernel_ulong_t)&fsi2_a_hdmi },
+ { "sh_fsi2_b_hdmi", (kernel_ulong_t)&fsi2_b_hdmi },
+ {},
+};
+
+static struct platform_driver fsi_hdmi = {
+ .driver = {
+ .name = "fsi-hdmi-audio",
+ },
+ .probe = fsi_hdmi_probe,
+ .remove = fsi_hdmi_remove,
+ .id_table = fsi_id_table,
+};
+
+static int __init fsi_hdmi_init(void)
+{
+ return platform_driver_register(&fsi_hdmi);
+}
+
+static void __exit fsi_hdmi_exit(void)
+{
+ platform_driver_unregister(&fsi_hdmi);
}
module_init(fsi_hdmi_init);
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 2b06402801ef..0c9997e2d8c0 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -78,6 +78,8 @@
/* CKG1 */
#define ACKMD_MASK 0x00007000
#define BPFMD_MASK 0x00000700
+#define DIMD (1 << 4)
+#define DOMD (1 << 0)
/* A/B MST_CTLR */
#define BP (1 << 4) /* Fix the signal of Biphase output */
@@ -111,6 +113,8 @@
#define FSI_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE)
+typedef int (*set_rate_func)(struct device *dev, int is_porta, int rate, int enable);
+
/*
* FSI driver use below type name for variable
*
@@ -128,7 +132,6 @@ struct fsi_stream {
struct snd_pcm_substream *substream;
int fifo_max_num;
- int chan_num;
int buff_offset;
int buff_len;
@@ -143,6 +146,7 @@ struct fsi_priv {
void __iomem *base;
struct fsi_master *master;
+ int chan_num;
struct fsi_stream playback;
struct fsi_stream capture;
@@ -252,9 +256,8 @@ static struct snd_soc_dai *fsi_get_dai(struct snd_pcm_substream *substream)
return rtd->cpu_dai;
}
-static struct fsi_priv *fsi_get_priv(struct snd_pcm_substream *substream)
+static struct fsi_priv *fsi_get_priv_frm_dai(struct snd_soc_dai *dai)
{
- struct snd_soc_dai *dai = fsi_get_dai(substream);
struct fsi_master *master = snd_soc_dai_get_drvdata(dai);
if (dai->id == 0)
@@ -263,11 +266,27 @@ static struct fsi_priv *fsi_get_priv(struct snd_pcm_substream *substream)
return &master->fsib;
}
+static struct fsi_priv *fsi_get_priv(struct snd_pcm_substream *substream)
+{
+ return fsi_get_priv_frm_dai(fsi_get_dai(substream));
+}
+
+static set_rate_func fsi_get_info_set_rate(struct fsi_master *master)
+{
+ if (!master->info)
+ return NULL;
+
+ return master->info->set_rate;
+}
+
static u32 fsi_get_info_flags(struct fsi_priv *fsi)
{
int is_porta = fsi_is_port_a(fsi);
struct fsi_master *master = fsi_get_master(fsi);
+ if (!master->info)
+ return 0;
+
return is_porta ? master->info->porta_flags :
master->info->portb_flags;
}
@@ -288,21 +307,6 @@ static inline struct fsi_stream *fsi_get_stream(struct fsi_priv *fsi,
return is_play ? &fsi->playback : &fsi->capture;
}
-static int fsi_is_master_mode(struct fsi_priv *fsi, int is_play)
-{
- u32 mode;
- u32 flags = fsi_get_info_flags(fsi);
-
- mode = is_play ? SH_FSI_OUT_SLAVE_MODE : SH_FSI_IN_SLAVE_MODE;
-
- /* return
- * 1 : master mode
- * 0 : slave mode
- */
-
- return (mode & flags) != mode;
-}
-
static u32 fsi_get_port_shift(struct fsi_priv *fsi, int is_play)
{
int is_porta = fsi_is_port_a(fsi);
@@ -357,7 +361,6 @@ static void fsi_stream_pop(struct fsi_priv *fsi, int is_play)
static int fsi_get_fifo_data_num(struct fsi_priv *fsi, int is_play)
{
u32 status;
- struct fsi_stream *io = fsi_get_stream(fsi, is_play);
int data_num;
status = is_play ?
@@ -365,7 +368,7 @@ static int fsi_get_fifo_data_num(struct fsi_priv *fsi, int is_play)
fsi_reg_read(fsi, DIFF_ST);
data_num = 0x1ff & (status >> 8);
- data_num *= io->chan_num;
+ data_num *= fsi->chan_num;
return data_num;
}
@@ -387,7 +390,7 @@ static int fsi_get_frame_width(struct fsi_priv *fsi, int is_play)
struct snd_pcm_substream *substream = io->substream;
struct snd_pcm_runtime *runtime = substream->runtime;
- return frames_to_bytes(runtime, 1) / io->chan_num;
+ return frames_to_bytes(runtime, 1) / fsi->chan_num;
}
static void fsi_count_fifo_err(struct fsi_priv *fsi)
@@ -580,10 +583,10 @@ static void fsi_fifo_init(struct fsi_priv *fsi,
* 7 channels: 32 ( 32 x 7 = 224)
* 8 channels: 32 ( 32 x 8 = 256)
*/
- for (i = 1; i < io->chan_num; i <<= 1)
+ for (i = 1; i < fsi->chan_num; i <<= 1)
io->fifo_max_num >>= 1;
dev_dbg(dai->dev, "%d channel %d store\n",
- io->chan_num, io->fifo_max_num);
+ fsi->chan_num, io->fifo_max_num);
/*
* set interrupt generation factor
@@ -659,7 +662,7 @@ static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, int stream)
* data_num_max : number of FSI fifo free space
* data_num : number of ALSA residue data
*/
- data_num_max = io->fifo_max_num * io->chan_num;
+ data_num_max = io->fifo_max_num * fsi->chan_num;
data_num_max -= fsi_get_fifo_data_num(fsi, is_play);
data_num = data_residue_num;
@@ -754,25 +757,12 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct fsi_priv *fsi = fsi_get_priv(substream);
- struct fsi_master *master = fsi_get_master(fsi);
- struct fsi_stream *io;
u32 flags = fsi_get_info_flags(fsi);
- u32 fmt;
u32 data;
int is_play = fsi_is_play(substream);
- int is_master;
-
- io = fsi_get_stream(fsi, is_play);
pm_runtime_get_sync(dai->dev);
- /* CKG1 */
- data = is_play ? (1 << 0) : (1 << 4);
- is_master = fsi_is_master_mode(fsi, is_play);
- if (is_master)
- fsi_reg_mask_set(fsi, CKG1, data, data);
- else
- fsi_reg_mask_set(fsi, CKG1, data, 0);
/* clock inversion (CKG2) */
data = 0;
@@ -787,54 +777,6 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream,
fsi_reg_write(fsi, CKG2, data);
- /* do fmt, di fmt */
- data = 0;
- fmt = is_play ? SH_FSI_GET_OFMT(flags) : SH_FSI_GET_IFMT(flags);
- switch (fmt) {
- case SH_FSI_FMT_MONO:
- data = CR_MONO;
- io->chan_num = 1;
- break;
- case SH_FSI_FMT_MONO_DELAY:
- data = CR_MONO_D;
- io->chan_num = 1;
- break;
- case SH_FSI_FMT_PCM:
- data = CR_PCM;
- io->chan_num = 2;
- break;
- case SH_FSI_FMT_I2S:
- data = CR_I2S;
- io->chan_num = 2;
- break;
- case SH_FSI_FMT_TDM:
- io->chan_num = is_play ?
- SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags);
- data = CR_TDM | (io->chan_num - 1);
- break;
- case SH_FSI_FMT_TDM_DELAY:
- io->chan_num = is_play ?
- SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags);
- data = CR_TDM_D | (io->chan_num - 1);
- break;
- case SH_FSI_FMT_SPDIF:
- if (master->core->ver < 2) {
- dev_err(dai->dev, "This FSI can not use SPDIF\n");
- return -EINVAL;
- }
- data = CR_BWS_16 | CR_DTMD_SPDIF_PCM | CR_PCM;
- io->chan_num = 2;
- fsi_spdif_clk_ctrl(fsi, 1);
- fsi_reg_mask_set(fsi, OUT_SEL, DMMD, DMMD);
- break;
- default:
- dev_err(dai->dev, "unknown format.\n");
- return -EINVAL;
- }
- is_play ?
- fsi_reg_write(fsi, DO_FMT, data) :
- fsi_reg_write(fsi, DI_FMT, data);
-
/* irq clear */
fsi_irq_disable(fsi, is_play);
fsi_irq_clear_status(fsi);
@@ -851,12 +793,12 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream,
struct fsi_priv *fsi = fsi_get_priv(substream);
int is_play = fsi_is_play(substream);
struct fsi_master *master = fsi_get_master(fsi);
- int (*set_rate)(struct device *dev, int is_porta, int rate, int enable);
+ set_rate_func set_rate;
fsi_irq_disable(fsi, is_play);
fsi_clk_ctrl(fsi, 0);
- set_rate = master->info->set_rate;
+ set_rate = fsi_get_info_set_rate(master);
if (set_rate && fsi->rate)
set_rate(dai->dev, fsi_is_port_a(fsi), fsi->rate, 0);
fsi->rate = 0;
@@ -889,18 +831,100 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd,
return ret;
}
+static int fsi_set_fmt_dai(struct fsi_priv *fsi, unsigned int fmt)
+{
+ u32 data = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ data = CR_I2S;
+ fsi->chan_num = 2;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ data = CR_PCM;
+ fsi->chan_num = 2;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ fsi_reg_write(fsi, DO_FMT, data);
+ fsi_reg_write(fsi, DI_FMT, data);
+
+ return 0;
+}
+
+static int fsi_set_fmt_spdif(struct fsi_priv *fsi)
+{
+ struct fsi_master *master = fsi_get_master(fsi);
+ u32 data = 0;
+
+ if (master->core->ver < 2)
+ return -EINVAL;
+
+ data = CR_BWS_16 | CR_DTMD_SPDIF_PCM | CR_PCM;
+ fsi->chan_num = 2;
+ fsi_spdif_clk_ctrl(fsi, 1);
+ fsi_reg_mask_set(fsi, OUT_SEL, DMMD, DMMD);
+
+ fsi_reg_write(fsi, DO_FMT, data);
+ fsi_reg_write(fsi, DI_FMT, data);
+
+ return 0;
+}
+
+static int fsi_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct fsi_priv *fsi = fsi_get_priv_frm_dai(dai);
+ u32 flags = fsi_get_info_flags(fsi);
+ u32 data = 0;
+ int ret;
+
+ pm_runtime_get_sync(dai->dev);
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ data = DIMD | DOMD;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ ret = -EINVAL;
+ goto set_fmt_exit;
+ }
+ fsi_reg_mask_set(fsi, CKG1, (DIMD | DOMD), data);
+
+ /* set format */
+ switch (flags & SH_FSI_FMT_MASK) {
+ case SH_FSI_FMT_DAI:
+ ret = fsi_set_fmt_dai(fsi, fmt & SND_SOC_DAIFMT_FORMAT_MASK);
+ break;
+ case SH_FSI_FMT_SPDIF:
+ ret = fsi_set_fmt_spdif(fsi);
+ break;
+ default:
+ ret = -EINVAL;
+ }
+
+set_fmt_exit:
+ pm_runtime_put_sync(dai->dev);
+
+ return ret;
+}
+
static int fsi_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct fsi_priv *fsi = fsi_get_priv(substream);
struct fsi_master *master = fsi_get_master(fsi);
- int (*set_rate)(struct device *dev, int is_porta, int rate, int enable);
+ set_rate_func set_rate;
int fsi_ver = master->core->ver;
long rate = params_rate(params);
int ret;
- set_rate = master->info->set_rate;
+ set_rate = fsi_get_info_set_rate(master);
if (!set_rate)
return 0;
@@ -975,6 +999,7 @@ static struct snd_soc_dai_ops fsi_dai_ops = {
.startup = fsi_dai_startup,
.shutdown = fsi_dai_shutdown,
.trigger = fsi_dai_trigger,
+ .set_fmt = fsi_dai_set_fmt,
.hw_params = fsi_dai_hw_params,
};
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index d97a59f6a249..db66dc44add2 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -18,6 +18,8 @@
#include <linux/bitmap.h>
#include <linux/rbtree.h>
+#include <trace/events/asoc.h>
+
static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec,
unsigned int reg)
{
@@ -25,7 +27,8 @@ static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec,
unsigned int val;
if (reg >= codec->driver->reg_cache_size ||
- snd_soc_codec_volatile_register(codec, reg)) {
+ snd_soc_codec_volatile_register(codec, reg) ||
+ codec->cache_bypass) {
if (codec->cache_only)
return -1;
@@ -49,7 +52,8 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg,
data[1] = value & 0x00ff;
if (!snd_soc_codec_volatile_register(codec, reg) &&
- reg < codec->driver->reg_cache_size) {
+ reg < codec->driver->reg_cache_size &&
+ !codec->cache_bypass) {
ret = snd_soc_cache_write(codec, reg, value);
if (ret < 0)
return -1;
@@ -106,7 +110,8 @@ static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec,
unsigned int val;
if (reg >= codec->driver->reg_cache_size ||
- snd_soc_codec_volatile_register(codec, reg)) {
+ snd_soc_codec_volatile_register(codec, reg) ||
+ codec->cache_bypass) {
if (codec->cache_only)
return -1;
@@ -130,7 +135,8 @@ static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg,
data[1] = value & 0x00ff;
if (!snd_soc_codec_volatile_register(codec, reg) &&
- reg < codec->driver->reg_cache_size) {
+ reg < codec->driver->reg_cache_size &&
+ !codec->cache_bypass) {
ret = snd_soc_cache_write(codec, reg, value);
if (ret < 0)
return -1;
@@ -191,7 +197,8 @@ static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg,
data[1] = value & 0xff;
if (!snd_soc_codec_volatile_register(codec, reg) &&
- reg < codec->driver->reg_cache_size) {
+ reg < codec->driver->reg_cache_size &&
+ !codec->cache_bypass) {
ret = snd_soc_cache_write(codec, reg, value);
if (ret < 0)
return -1;
@@ -216,7 +223,8 @@ static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec,
reg &= 0xff;
if (reg >= codec->driver->reg_cache_size ||
- snd_soc_codec_volatile_register(codec, reg)) {
+ snd_soc_codec_volatile_register(codec, reg) ||
+ codec->cache_bypass) {
if (codec->cache_only)
return -1;
@@ -271,7 +279,8 @@ static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg,
data[2] = value & 0xff;
if (!snd_soc_codec_volatile_register(codec, reg) &&
- reg < codec->driver->reg_cache_size) {
+ reg < codec->driver->reg_cache_size &&
+ !codec->cache_bypass) {
ret = snd_soc_cache_write(codec, reg, value);
if (ret < 0)
return -1;
@@ -295,7 +304,8 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec,
unsigned int val;
if (reg >= codec->driver->reg_cache_size ||
- snd_soc_codec_volatile_register(codec, reg)) {
+ snd_soc_codec_volatile_register(codec, reg) ||
+ codec->cache_bypass) {
if (codec->cache_only)
return -1;
@@ -450,7 +460,8 @@ static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec,
reg &= 0xff;
if (reg >= codec->driver->reg_cache_size ||
- snd_soc_codec_volatile_register(codec, reg)) {
+ snd_soc_codec_volatile_register(codec, reg) ||
+ codec->cache_bypass) {
if (codec->cache_only)
return -1;
@@ -476,7 +487,8 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg,
reg &= 0xff;
if (!snd_soc_codec_volatile_register(codec, reg) &&
- reg < codec->driver->reg_cache_size) {
+ reg < codec->driver->reg_cache_size &&
+ !codec->cache_bypass) {
ret = snd_soc_cache_write(codec, reg, value);
if (ret < 0)
return -1;
@@ -568,7 +580,8 @@ static unsigned int snd_soc_16_16_read(struct snd_soc_codec *codec,
unsigned int val;
if (reg >= codec->driver->reg_cache_size ||
- snd_soc_codec_volatile_register(codec, reg)) {
+ snd_soc_codec_volatile_register(codec, reg) ||
+ codec->cache_bypass) {
if (codec->cache_only)
return -1;
@@ -595,7 +608,8 @@ static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg,
data[3] = value & 0xff;
if (!snd_soc_codec_volatile_register(codec, reg) &&
- reg < codec->driver->reg_cache_size) {
+ reg < codec->driver->reg_cache_size &&
+ !codec->cache_bypass) {
ret = snd_soc_cache_write(codec, reg, value);
if (ret < 0)
return -1;
@@ -878,7 +892,9 @@ static int snd_soc_rbtree_cache_sync(struct snd_soc_codec *codec)
ret = snd_soc_cache_read(codec, rbnode->reg, &val);
if (ret)
return ret;
+ codec->cache_bypass = 1;
ret = snd_soc_write(codec, rbnode->reg, val);
+ codec->cache_bypass = 0;
if (ret)
return ret;
dev_dbg(codec->dev, "Synced register %#x, value = %#x\n",
@@ -1136,7 +1152,9 @@ static int snd_soc_lzo_cache_sync(struct snd_soc_codec *codec)
ret = snd_soc_cache_read(codec, i, &val);
if (ret)
return ret;
+ codec->cache_bypass = 1;
ret = snd_soc_write(codec, i, val);
+ codec->cache_bypass = 0;
if (ret)
return ret;
dev_dbg(codec->dev, "Synced register %#x, value = %#x\n",
@@ -1585,18 +1603,26 @@ EXPORT_SYMBOL_GPL(snd_soc_cache_write);
int snd_soc_cache_sync(struct snd_soc_codec *codec)
{
int ret;
+ const char *name;
if (!codec->cache_sync) {
return 0;
}
+ if (codec->cache_ops->name)
+ name = codec->cache_ops->name;
+ else
+ name = "unknown";
+
if (codec->cache_ops && codec->cache_ops->sync) {
if (codec->cache_ops->name)
dev_dbg(codec->dev, "Syncing %s cache for %s codec\n",
codec->cache_ops->name, codec->name);
+ trace_snd_soc_cache_sync(codec, name, "start");
ret = codec->cache_ops->sync(codec);
if (!ret)
codec->cache_sync = 0;
+ trace_snd_soc_cache_sync(codec, name, "end");
return ret;
}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 9e68984423b2..205cbd7b149f 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -69,10 +69,32 @@ static int pmdown_time = 5000;
module_param(pmdown_time, int, 0);
MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)");
+/* returns the minimum number of bytes needed to represent
+ * a particular given value */
+static int min_bytes_needed(unsigned long val)
+{
+ int c = 0;
+ int i;
+
+ for (i = (sizeof val * 8) - 1; i >= 0; --i, ++c)
+ if (val & (1UL << i))
+ break;
+ c = (sizeof val * 8) - c;
+ if (!c || (c % 8))
+ c = (c + 8) / 8;
+ else
+ c /= 8;
+ return c;
+}
+
/* codec register dump */
static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf)
{
int ret, i, step = 1, count = 0;
+ int wordsize, regsize;
+
+ wordsize = codec->driver->reg_word_size * 2;
+ regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2;
if (!codec->driver->reg_cache_size)
return 0;
@@ -80,12 +102,11 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf)
if (codec->driver->reg_cache_step)
step = codec->driver->reg_cache_step;
- count += sprintf(buf, "%s registers\n", codec->name);
for (i = 0; i < codec->driver->reg_cache_size; i += step) {
if (codec->readable_register && !codec->readable_register(codec, i))
continue;
- count += sprintf(buf + count, "%2x: ", i);
+ count += sprintf(buf + count, "%.*x: ", regsize, i);
if (count >= PAGE_SIZE - 1)
break;
@@ -101,7 +122,7 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf)
if (ret >= 0)
count += snprintf(buf + count,
PAGE_SIZE - count,
- "%4x", ret);
+ "%.*x", wordsize, ret);
else
count += snprintf(buf + count,
PAGE_SIZE - count,
@@ -235,6 +256,11 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec)
return;
}
+ debugfs_create_bool("cache_sync", 0444, codec->debugfs_codec_root,
+ &codec->cache_sync);
+ debugfs_create_bool("cache_only", 0444, codec->debugfs_codec_root,
+ &codec->cache_only);
+
codec->debugfs_reg = debugfs_create_file("codec_reg", 0644,
codec->debugfs_codec_root,
codec, &codec_reg_fops);
@@ -965,12 +991,11 @@ static struct snd_pcm_ops soc_pcm_ops = {
.pointer = soc_pcm_pointer,
};
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
/* powers down audio subsystem for suspend */
-static int soc_suspend(struct device *dev)
+int snd_soc_suspend(struct device *dev)
{
- struct platform_device *pdev = to_platform_device(dev);
- struct snd_soc_card *card = platform_get_drvdata(pdev);
+ struct snd_soc_card *card = dev_get_drvdata(dev);
struct snd_soc_codec *codec;
int i;
@@ -1011,7 +1036,7 @@ static int soc_suspend(struct device *dev)
}
if (card->suspend_pre)
- card->suspend_pre(pdev, PMSG_SUSPEND);
+ card->suspend_pre(card);
for (i = 0; i < card->num_rtd; i++) {
struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai;
@@ -1078,10 +1103,11 @@ static int soc_suspend(struct device *dev)
}
if (card->suspend_post)
- card->suspend_post(pdev, PMSG_SUSPEND);
+ card->suspend_post(card);
return 0;
}
+EXPORT_SYMBOL_GPL(snd_soc_suspend);
/* deferred resume work, so resume can complete before we finished
* setting our codec back up, which can be very slow on I2C
@@ -1090,7 +1116,6 @@ static void soc_resume_deferred(struct work_struct *work)
{
struct snd_soc_card *card =
container_of(work, struct snd_soc_card, deferred_resume_work);
- struct platform_device *pdev = to_platform_device(card->dev);
struct snd_soc_codec *codec;
int i;
@@ -1104,7 +1129,7 @@ static void soc_resume_deferred(struct work_struct *work)
snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D2);
if (card->resume_pre)
- card->resume_pre(pdev);
+ card->resume_pre(card);
/* resume AC97 DAIs */
for (i = 0; i < card->num_rtd; i++) {
@@ -1179,7 +1204,7 @@ static void soc_resume_deferred(struct work_struct *work)
}
if (card->resume_post)
- card->resume_post(pdev);
+ card->resume_post(card);
dev_dbg(card->dev, "resume work completed\n");
@@ -1188,10 +1213,9 @@ static void soc_resume_deferred(struct work_struct *work)
}
/* powers up audio subsystem after a suspend */
-static int soc_resume(struct device *dev)
+int snd_soc_resume(struct device *dev)
{
- struct platform_device *pdev = to_platform_device(dev);
- struct snd_soc_card *card = platform_get_drvdata(pdev);
+ struct snd_soc_card *card = dev_get_drvdata(dev);
int i;
/* AC97 devices might have other drivers hanging off them so
@@ -1213,9 +1237,10 @@ static int soc_resume(struct device *dev)
return 0;
}
+EXPORT_SYMBOL_GPL(snd_soc_resume);
#else
-#define soc_suspend NULL
-#define soc_resume NULL
+#define snd_soc_suspend NULL
+#define snd_soc_resume NULL
#endif
static struct snd_soc_dai_ops null_dai_ops = {
@@ -1408,26 +1433,31 @@ static int soc_probe_codec(struct snd_soc_card *card,
codec->dapm.card = card;
soc_set_name_prefix(card, codec);
+ if (!try_module_get(codec->dev->driver->owner))
+ return -ENODEV;
+
if (codec->driver->probe) {
ret = codec->driver->probe(codec);
if (ret < 0) {
dev_err(codec->dev,
"asoc: failed to probe CODEC %s: %d\n",
codec->name, ret);
- return ret;
+ goto err_probe;
}
}
soc_init_codec_debugfs(codec);
/* mark codec as probed and add to card codec list */
- if (!try_module_get(codec->dev->driver->owner))
- return -ENODEV;
-
codec->probed = 1;
list_add(&codec->card_list, &card->codec_dev_list);
list_add(&codec->dapm.list, &card->dapm_list);
+ return 0;
+
+err_probe:
+ module_put(codec->dev->driver->owner);
+
return ret;
}
@@ -1545,19 +1575,19 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num)
/* probe the platform */
if (!platform->probed) {
+ if (!try_module_get(platform->dev->driver->owner))
+ return -ENODEV;
+
if (platform->driver->probe) {
ret = platform->driver->probe(platform);
if (ret < 0) {
printk(KERN_ERR "asoc: failed to probe platform %s\n",
platform->name);
+ module_put(platform->dev->driver->owner);
return ret;
}
}
/* mark platform as probed and add to card platform list */
-
- if (!try_module_get(platform->dev->driver->owner))
- return -ENODEV;
-
platform->probed = 1;
list_add(&platform->card_list, &card->platform_dev_list);
}
@@ -1666,9 +1696,6 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num)
goto out;
found:
- if (!try_module_get(codec->dev->driver->owner))
- return -ENODEV;
-
ret = soc_probe_codec(card, codec);
if (ret < 0)
return ret;
@@ -1718,7 +1745,6 @@ static int snd_soc_init_codec_cache(struct snd_soc_codec *codec,
static void snd_soc_instantiate_card(struct snd_soc_card *card)
{
- struct platform_device *pdev = to_platform_device(card->dev);
struct snd_soc_codec *codec;
struct snd_soc_codec_conf *codec_conf;
enum snd_soc_compress_type compress_type;
@@ -1782,7 +1808,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
/* initialise the sound card only once */
if (card->probe) {
- ret = card->probe(pdev);
+ ret = card->probe(card);
if (ret < 0)
goto card_probe_error;
}
@@ -1843,7 +1869,7 @@ probe_dai_err:
card_probe_error:
if (card->remove)
- card->remove(pdev);
+ card->remove(card);
snd_card_free(card->snd_card);
@@ -1876,7 +1902,6 @@ static int soc_probe(struct platform_device *pdev)
/* Bodge while we unpick instantiation */
card->dev = &pdev->dev;
- snd_soc_initialize_card_lists(card);
ret = snd_soc_register_card(card);
if (ret != 0) {
@@ -1889,7 +1914,6 @@ static int soc_probe(struct platform_device *pdev)
static int soc_cleanup_card_resources(struct snd_soc_card *card)
{
- struct platform_device *pdev = to_platform_device(card->dev);
int i;
/* make sure any delayed work runs */
@@ -1910,7 +1934,7 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card)
/* remove the card */
if (card->remove)
- card->remove(pdev);
+ card->remove(card);
kfree(card->rtd);
snd_card_free(card->snd_card);
@@ -1927,10 +1951,9 @@ static int soc_remove(struct platform_device *pdev)
return 0;
}
-static int soc_poweroff(struct device *dev)
+int snd_soc_poweroff(struct device *dev)
{
- struct platform_device *pdev = to_platform_device(dev);
- struct snd_soc_card *card = platform_get_drvdata(pdev);
+ struct snd_soc_card *card = dev_get_drvdata(dev);
int i;
if (!card->instantiated)
@@ -1947,11 +1970,12 @@ static int soc_poweroff(struct device *dev)
return 0;
}
+EXPORT_SYMBOL_GPL(snd_soc_poweroff);
-static const struct dev_pm_ops soc_pm_ops = {
- .suspend = soc_suspend,
- .resume = soc_resume,
- .poweroff = soc_poweroff,
+const struct dev_pm_ops snd_soc_pm_ops = {
+ .suspend = snd_soc_suspend,
+ .resume = snd_soc_resume,
+ .poweroff = snd_soc_poweroff,
};
/* ASoC platform driver */
@@ -1959,7 +1983,7 @@ static struct platform_driver soc_driver = {
.driver = {
.name = "soc-audio",
.owner = THIS_MODULE,
- .pm = &soc_pm_ops,
+ .pm = &snd_soc_pm_ops,
},
.probe = soc_probe,
.remove = soc_remove,
@@ -2029,7 +2053,8 @@ static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
*
* Boolean function indiciating if a CODEC register is volatile.
*/
-int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg)
+int snd_soc_codec_volatile_register(struct snd_soc_codec *codec,
+ unsigned int reg)
{
if (codec->volatile_register)
return codec->volatile_register(codec, reg);
@@ -3117,6 +3142,8 @@ int snd_soc_register_card(struct snd_soc_card *card)
if (!card->name || !card->dev)
return -EINVAL;
+ snd_soc_initialize_card_lists(card);
+
soc_init_card_debugfs(card);
card->rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime) *
@@ -3496,6 +3523,7 @@ int snd_soc_register_codec(struct device *dev,
codec->dapm.bias_level = SND_SOC_BIAS_OFF;
codec->dapm.dev = dev;
codec->dapm.codec = codec;
+ codec->dapm.seq_notifier = codec_drv->seq_notifier;
codec->dev = dev;
codec->driver = codec_drv;
codec->num_dai = num_dai;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 499730ab5638..d0342aab2c15 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -726,10 +726,23 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w)
static int dapm_seq_compare(struct snd_soc_dapm_widget *a,
struct snd_soc_dapm_widget *b,
- int sort[])
+ bool power_up)
{
+ int *sort;
+
+ if (power_up)
+ sort = dapm_up_seq;
+ else
+ sort = dapm_down_seq;
+
if (sort[a->id] != sort[b->id])
return sort[a->id] - sort[b->id];
+ if (a->subseq != b->subseq) {
+ if (power_up)
+ return a->subseq - b->subseq;
+ else
+ return b->subseq - a->subseq;
+ }
if (a->reg != b->reg)
return a->reg - b->reg;
if (a->dapm != b->dapm)
@@ -741,12 +754,12 @@ static int dapm_seq_compare(struct snd_soc_dapm_widget *a,
/* Insert a widget in order into a DAPM power sequence. */
static void dapm_seq_insert(struct snd_soc_dapm_widget *new_widget,
struct list_head *list,
- int sort[])
+ bool power_up)
{
struct snd_soc_dapm_widget *w;
list_for_each_entry(w, list, power_list)
- if (dapm_seq_compare(new_widget, w, sort) < 0) {
+ if (dapm_seq_compare(new_widget, w, power_up) < 0) {
list_add_tail(&new_widget->power_list, &w->power_list);
return;
}
@@ -857,26 +870,42 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm,
* handled.
*/
static void dapm_seq_run(struct snd_soc_dapm_context *dapm,
- struct list_head *list, int event, int sort[])
+ struct list_head *list, int event, bool power_up)
{
struct snd_soc_dapm_widget *w, *n;
LIST_HEAD(pending);
int cur_sort = -1;
+ int cur_subseq = -1;
int cur_reg = SND_SOC_NOPM;
struct snd_soc_dapm_context *cur_dapm = NULL;
- int ret;
+ int ret, i;
+ int *sort;
+
+ if (power_up)
+ sort = dapm_up_seq;
+ else
+ sort = dapm_down_seq;
list_for_each_entry_safe(w, n, list, power_list) {
ret = 0;
/* Do we need to apply any queued changes? */
if (sort[w->id] != cur_sort || w->reg != cur_reg ||
- w->dapm != cur_dapm) {
+ w->dapm != cur_dapm || w->subseq != cur_subseq) {
if (!list_empty(&pending))
dapm_seq_run_coalesced(cur_dapm, &pending);
+ if (cur_dapm && cur_dapm->seq_notifier) {
+ for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++)
+ if (sort[i] == cur_sort)
+ cur_dapm->seq_notifier(cur_dapm,
+ i,
+ cur_subseq);
+ }
+
INIT_LIST_HEAD(&pending);
cur_sort = -1;
+ cur_subseq = -1;
cur_reg = SND_SOC_NOPM;
cur_dapm = NULL;
}
@@ -921,6 +950,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm,
default:
/* Queue it up for application */
cur_sort = sort[w->id];
+ cur_subseq = w->subseq;
cur_reg = w->reg;
cur_dapm = w->dapm;
list_move(&w->power_list, &pending);
@@ -934,6 +964,13 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm,
if (!list_empty(&pending))
dapm_seq_run_coalesced(dapm, &pending);
+
+ if (cur_dapm && cur_dapm->seq_notifier) {
+ for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++)
+ if (sort[i] == cur_sort)
+ cur_dapm->seq_notifier(cur_dapm,
+ i, cur_subseq);
+ }
}
static void dapm_widget_update(struct snd_soc_dapm_context *dapm)
@@ -1002,10 +1039,10 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
list_for_each_entry(w, &card->widgets, list) {
switch (w->id) {
case snd_soc_dapm_pre:
- dapm_seq_insert(w, &down_list, dapm_down_seq);
+ dapm_seq_insert(w, &down_list, false);
break;
case snd_soc_dapm_post:
- dapm_seq_insert(w, &up_list, dapm_up_seq);
+ dapm_seq_insert(w, &up_list, true);
break;
default:
@@ -1025,9 +1062,9 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
trace_snd_soc_dapm_widget_power(w, power);
if (power)
- dapm_seq_insert(w, &up_list, dapm_up_seq);
+ dapm_seq_insert(w, &up_list, true);
else
- dapm_seq_insert(w, &down_list, dapm_down_seq);
+ dapm_seq_insert(w, &down_list, false);
w->power = power;
break;
@@ -1086,12 +1123,12 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
}
/* Power down widgets first; try to avoid amplifying pops. */
- dapm_seq_run(dapm, &down_list, event, dapm_down_seq);
+ dapm_seq_run(dapm, &down_list, event, false);
dapm_widget_update(dapm);
/* Now power up. */
- dapm_seq_run(dapm, &up_list, event, dapm_up_seq);
+ dapm_seq_run(dapm, &up_list, event, true);
list_for_each_entry(d, &dapm->card->dapm_list, list) {
/* If we just powered the last thing off drop to standby bias */
@@ -1742,7 +1779,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
int max = mc->max;
unsigned int mask = (1 << fls(max)) - 1;
unsigned int invert = mc->invert;
- unsigned int val, val_mask;
+ unsigned int val;
int connect, change;
struct snd_soc_dapm_update update;
@@ -1750,13 +1787,13 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
if (invert)
val = max - val;
- val_mask = mask << shift;
+ mask = mask << shift;
val = val << shift;
mutex_lock(&widget->codec->mutex);
widget->value = val;
- change = snd_soc_test_bits(widget->codec, reg, val_mask, val);
+ change = snd_soc_test_bits(widget->codec, reg, mask, val);
if (change) {
if (val)
/* new connection */
@@ -2372,7 +2409,7 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm)
if (w->dapm != dapm)
continue;
if (w->power) {
- dapm_seq_insert(w, &down_list, dapm_down_seq);
+ dapm_seq_insert(w, &down_list, false);
w->power = 0;
powerdown = 1;
}
@@ -2383,7 +2420,7 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm)
*/
if (powerdown) {
snd_soc_dapm_set_bias_level(NULL, dapm, SND_SOC_BIAS_PREPARE);
- dapm_seq_run(dapm, &down_list, 0, dapm_down_seq);
+ dapm_seq_run(dapm, &down_list, 0, false);
snd_soc_dapm_set_bias_level(NULL, dapm, SND_SOC_BIAS_STANDBY);
}
}
diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index 1d07b931f3d8..3f45e6a439bf 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -28,26 +28,9 @@ int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params)
{
int sample_size;
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
- case SNDRV_PCM_FORMAT_S16_BE:
- sample_size = 16;
- break;
- case SNDRV_PCM_FORMAT_S20_3LE:
- case SNDRV_PCM_FORMAT_S20_3BE:
- sample_size = 20;
- break;
- case SNDRV_PCM_FORMAT_S24_LE:
- case SNDRV_PCM_FORMAT_S24_BE:
- sample_size = 24;
- break;
- case SNDRV_PCM_FORMAT_S32_LE:
- case SNDRV_PCM_FORMAT_S32_BE:
- sample_size = 32;
- break;
- default:
- return -ENOTSUPP;
- }
+ sample_size = snd_pcm_format_width(params_format(params));
+ if (sample_size < 0)
+ return sample_size;
return snd_soc_calc_frame_size(sample_size, params_channels(params),
1);
diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c
index b160b7113f45..11e2cb825664 100644
--- a/sound/soc/tegra/harmony.c
+++ b/sound/soc/tegra/harmony.c
@@ -2,7 +2,7 @@
* harmony.c - Harmony machine ASoC driver
*
* Author: Stephen Warren <swarren@nvidia.com>
- * Copyright (C) 2010 - NVIDIA, Inc.
+ * Copyright (C) 2010-2011 - NVIDIA, Inc.
*
* Based on code copyright/by:
*
@@ -29,7 +29,14 @@
*/
#include <asm/mach-types.h>
+
#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/gpio.h>
+
+#include <mach/harmony_audio.h>
+
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -40,9 +47,13 @@
#include "tegra_pcm.h"
#include "tegra_asoc_utils.h"
-#define PREFIX "ASoC Harmony: "
+#define DRV_NAME "tegra-snd-harmony"
-static struct platform_device *harmony_snd_device;
+struct tegra_harmony {
+ struct tegra_asoc_utils_data util_data;
+ struct harmony_audio_platform_data *pdata;
+ int gpio_spkr_en_requested;
+};
static int harmony_asoc_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
@@ -50,6 +61,9 @@ static int harmony_asoc_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_card *card = codec->card;
+ struct tegra_harmony *harmony = snd_soc_card_get_drvdata(card);
int srate, mclk, mclk_change;
int err;
@@ -68,9 +82,10 @@ static int harmony_asoc_hw_params(struct snd_pcm_substream *substream,
while (mclk < 6000000)
mclk *= 2;
- err = tegra_asoc_utils_set_rate(srate, mclk, &mclk_change);
+ err = tegra_asoc_utils_set_rate(&harmony->util_data, srate, mclk,
+ &mclk_change);
if (err < 0) {
- pr_err(PREFIX "Can't configure clocks\n");
+ dev_err(card->dev, "Can't configure clocks\n");
return err;
}
@@ -79,7 +94,7 @@ static int harmony_asoc_hw_params(struct snd_pcm_substream *substream,
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS);
if (err < 0) {
- pr_err(PREFIX "codec_dai fmt not set\n");
+ dev_err(card->dev, "codec_dai fmt not set\n");
return err;
}
@@ -88,16 +103,17 @@ static int harmony_asoc_hw_params(struct snd_pcm_substream *substream,
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS);
if (err < 0) {
- pr_err(PREFIX "cpu_dai fmt not set\n");
+ dev_err(card->dev, "cpu_dai fmt not set\n");
return err;
}
if (mclk_change) {
- err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, SND_SOC_CLOCK_IN);
- if (err < 0) {
- pr_err(PREFIX "codec_dai clock not set\n");
- return err;
- }
+ err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk,
+ SND_SOC_CLOCK_IN);
+ if (err < 0) {
+ dev_err(card->dev, "codec_dai clock not set\n");
+ return err;
+ }
}
return 0;
@@ -107,7 +123,22 @@ static struct snd_soc_ops harmony_asoc_ops = {
.hw_params = harmony_asoc_hw_params,
};
+static int harmony_event_int_spk(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_card *card = codec->card;
+ struct tegra_harmony *harmony = snd_soc_card_get_drvdata(card);
+ struct harmony_audio_platform_data *pdata = harmony->pdata;
+
+ gpio_set_value_cansleep(pdata->gpio_spkr_en,
+ SND_SOC_DAPM_EVENT_ON(event));
+
+ return 0;
+}
+
static const struct snd_soc_dapm_widget harmony_dapm_widgets[] = {
+ SND_SOC_DAPM_SPK("Int Spk", harmony_event_int_spk),
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_MIC("Mic Jack", NULL),
};
@@ -115,6 +146,10 @@ static const struct snd_soc_dapm_widget harmony_dapm_widgets[] = {
static const struct snd_soc_dapm_route harmony_audio_map[] = {
{"Headphone Jack", NULL, "HPOUTR"},
{"Headphone Jack", NULL, "HPOUTL"},
+ {"Int Spk", NULL, "ROP"},
+ {"Int Spk", NULL, "RON"},
+ {"Int Spk", NULL, "LOP"},
+ {"Int Spk", NULL, "LON"},
{"Mic Bias", NULL, "Mic Jack"},
{"IN1L", NULL, "Mic Bias"},
};
@@ -123,6 +158,19 @@ static int harmony_asoc_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
+ struct snd_soc_card *card = codec->card;
+ struct tegra_harmony *harmony = snd_soc_card_get_drvdata(card);
+ struct harmony_audio_platform_data *pdata = harmony->pdata;
+ int ret;
+
+ ret = gpio_request(pdata->gpio_spkr_en, "spkr_en");
+ if (ret) {
+ dev_err(card->dev, "cannot get spkr_en gpio\n");
+ return ret;
+ }
+ harmony->gpio_spkr_en_requested = 1;
+
+ gpio_direction_output(pdata->gpio_spkr_en, 0);
snd_soc_dapm_new_controls(dapm, harmony_dapm_widgets,
ARRAY_SIZE(harmony_dapm_widgets));
@@ -131,6 +179,7 @@ static int harmony_asoc_init(struct snd_soc_pcm_runtime *rtd)
ARRAY_SIZE(harmony_audio_map));
snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin(dapm, "Int Spk");
snd_soc_dapm_enable_pin(dapm, "Mic Jack");
snd_soc_dapm_sync(dapm);
@@ -154,56 +203,101 @@ static struct snd_soc_card snd_soc_harmony = {
.num_links = 1,
};
-static int __init harmony_soc_modinit(void)
+static __devinit int tegra_snd_harmony_probe(struct platform_device *pdev)
{
+ struct snd_soc_card *card = &snd_soc_harmony;
+ struct tegra_harmony *harmony;
+ struct harmony_audio_platform_data *pdata;
int ret;
if (!machine_is_harmony()) {
- pr_err(PREFIX "Not running on Tegra Harmony!\n");
+ dev_err(&pdev->dev, "Not running on Tegra Harmony!\n");
return -ENODEV;
}
- ret = tegra_asoc_utils_init();
- if (ret) {
- return ret;
+ pdata = pdev->dev.platform_data;
+ if (!pdata) {
+ dev_err(&pdev->dev, "no platform data supplied\n");
+ return -EINVAL;
}
- /*
- * Create and register platform device
- */
- harmony_snd_device = platform_device_alloc("soc-audio", -1);
- if (harmony_snd_device == NULL) {
- pr_err(PREFIX "platform_device_alloc failed\n");
- ret = -ENOMEM;
- goto err_clock_utils;
+ harmony = kzalloc(sizeof(struct tegra_harmony), GFP_KERNEL);
+ if (!harmony) {
+ dev_err(&pdev->dev, "Can't allocate tegra_harmony\n");
+ return -ENOMEM;
}
- platform_set_drvdata(harmony_snd_device, &snd_soc_harmony);
+ harmony->pdata = pdata;
+
+ ret = tegra_asoc_utils_init(&harmony->util_data, &pdev->dev);
+ if (ret)
+ goto err_free_harmony;
+
+ card->dev = &pdev->dev;
+ platform_set_drvdata(pdev, card);
+ snd_soc_card_set_drvdata(card, harmony);
- ret = platform_device_add(harmony_snd_device);
+ ret = snd_soc_register_card(card);
if (ret) {
- pr_err(PREFIX "platform_device_add failed (%d)\n",
+ dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n",
ret);
- goto err_device_put;
+ goto err_clear_drvdata;
}
return 0;
-err_device_put:
- platform_device_put(harmony_snd_device);
-err_clock_utils:
- tegra_asoc_utils_fini();
+err_clear_drvdata:
+ snd_soc_card_set_drvdata(card, NULL);
+ platform_set_drvdata(pdev, NULL);
+ card->dev = NULL;
+ tegra_asoc_utils_fini(&harmony->util_data);
+err_free_harmony:
+ kfree(harmony);
return ret;
}
-module_init(harmony_soc_modinit);
-static void __exit harmony_soc_modexit(void)
+static int __devexit tegra_snd_harmony_remove(struct platform_device *pdev)
{
- platform_device_unregister(harmony_snd_device);
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+ struct tegra_harmony *harmony = snd_soc_card_get_drvdata(card);
+ struct harmony_audio_platform_data *pdata = harmony->pdata;
+
+ snd_soc_unregister_card(card);
+
+ snd_soc_card_set_drvdata(card, NULL);
+ platform_set_drvdata(pdev, NULL);
+ card->dev = NULL;
- tegra_asoc_utils_fini();
+ tegra_asoc_utils_fini(&harmony->util_data);
+
+ if (harmony->gpio_spkr_en_requested)
+ gpio_free(pdata->gpio_spkr_en);
+
+ kfree(harmony);
+
+ return 0;
+}
+
+static struct platform_driver tegra_snd_harmony_driver = {
+ .driver = {
+ .name = DRV_NAME,
+ .owner = THIS_MODULE,
+ },
+ .probe = tegra_snd_harmony_probe,
+ .remove = __devexit_p(tegra_snd_harmony_remove),
+};
+
+static int __init snd_tegra_harmony_init(void)
+{
+ return platform_driver_register(&tegra_snd_harmony_driver);
+}
+module_init(snd_tegra_harmony_init);
+
+static void __exit snd_tegra_harmony_exit(void)
+{
+ platform_driver_unregister(&tegra_snd_harmony_driver);
}
-module_exit(harmony_soc_modexit);
+module_exit(snd_tegra_harmony_exit);
MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
MODULE_DESCRIPTION("Harmony machine ASoC driver");
diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c
index cfe2ea890dc0..cb4fc13c7d22 100644
--- a/sound/soc/tegra/tegra_asoc_utils.c
+++ b/sound/soc/tegra/tegra_asoc_utils.c
@@ -21,20 +21,14 @@
*/
#include <linux/clk.h>
+#include <linux/device.h>
#include <linux/err.h>
#include <linux/kernel.h>
#include "tegra_asoc_utils.h"
-#define PREFIX "ASoC Tegra: "
-
-static struct clk *clk_pll_a;
-static struct clk *clk_pll_a_out0;
-static struct clk *clk_cdev1;
-
-static int set_baseclock, set_mclk;
-
-int tegra_asoc_utils_set_rate(int srate, int mclk, int *mclk_change)
+int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate,
+ int mclk, int *mclk_change)
{
int new_baseclock;
int err;
@@ -58,95 +52,98 @@ int tegra_asoc_utils_set_rate(int srate, int mclk, int *mclk_change)
return -EINVAL;
}
- *mclk_change = ((new_baseclock != set_baseclock) ||
- (mclk != set_mclk));
+ *mclk_change = ((new_baseclock != data->set_baseclock) ||
+ (mclk != data->set_mclk));
if (!*mclk_change)
return 0;
- set_baseclock = 0;
- set_mclk = 0;
+ data->set_baseclock = 0;
+ data->set_mclk = 0;
- clk_disable(clk_cdev1);
- clk_disable(clk_pll_a_out0);
- clk_disable(clk_pll_a);
+ clk_disable(data->clk_cdev1);
+ clk_disable(data->clk_pll_a_out0);
+ clk_disable(data->clk_pll_a);
- err = clk_set_rate(clk_pll_a, new_baseclock);
+ err = clk_set_rate(data->clk_pll_a, new_baseclock);
if (err) {
- pr_err(PREFIX "Can't set pll_a rate: %d\n", err);
+ dev_err(data->dev, "Can't set pll_a rate: %d\n", err);
return err;
}
- err = clk_set_rate(clk_pll_a_out0, mclk);
+ err = clk_set_rate(data->clk_pll_a_out0, mclk);
if (err) {
- pr_err(PREFIX "Can't set pll_a_out0 rate: %d\n", err);
+ dev_err(data->dev, "Can't set pll_a_out0 rate: %d\n", err);
return err;
}
/* Don't set cdev1 rate; its locked to pll_a_out0 */
- err = clk_enable(clk_pll_a);
+ err = clk_enable(data->clk_pll_a);
if (err) {
- pr_err(PREFIX "Can't enable pll_a: %d\n", err);
+ dev_err(data->dev, "Can't enable pll_a: %d\n", err);
return err;
}
- err = clk_enable(clk_pll_a_out0);
+ err = clk_enable(data->clk_pll_a_out0);
if (err) {
- pr_err(PREFIX "Can't enable pll_a_out0: %d\n", err);
+ dev_err(data->dev, "Can't enable pll_a_out0: %d\n", err);
return err;
}
- err = clk_enable(clk_cdev1);
+ err = clk_enable(data->clk_cdev1);
if (err) {
- pr_err(PREFIX "Can't enable cdev1: %d\n", err);
+ dev_err(data->dev, "Can't enable cdev1: %d\n", err);
return err;
}
- set_baseclock = new_baseclock;
- set_mclk = mclk;
+ data->set_baseclock = new_baseclock;
+ data->set_mclk = mclk;
return 0;
}
-int tegra_asoc_utils_init(void)
+int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data,
+ struct device *dev)
{
int ret;
- clk_pll_a = clk_get_sys(NULL, "pll_a");
- if (IS_ERR(clk_pll_a)) {
- pr_err(PREFIX "Can't retrieve clk pll_a\n");
- ret = PTR_ERR(clk_pll_a);
+ data->dev = dev;
+
+ data->clk_pll_a = clk_get_sys(NULL, "pll_a");
+ if (IS_ERR(data->clk_pll_a)) {
+ dev_err(data->dev, "Can't retrieve clk pll_a\n");
+ ret = PTR_ERR(data->clk_pll_a);
goto err;
}
- clk_pll_a_out0 = clk_get_sys(NULL, "pll_a_out0");
- if (IS_ERR(clk_pll_a_out0)) {
- pr_err(PREFIX "Can't retrieve clk pll_a_out0\n");
- ret = PTR_ERR(clk_pll_a_out0);
+ data->clk_pll_a_out0 = clk_get_sys(NULL, "pll_a_out0");
+ if (IS_ERR(data->clk_pll_a_out0)) {
+ dev_err(data->dev, "Can't retrieve clk pll_a_out0\n");
+ ret = PTR_ERR(data->clk_pll_a_out0);
goto err_put_pll_a;
}
- clk_cdev1 = clk_get_sys(NULL, "cdev1");
- if (IS_ERR(clk_cdev1)) {
- pr_err(PREFIX "Can't retrieve clk cdev1\n");
- ret = PTR_ERR(clk_cdev1);
+ data->clk_cdev1 = clk_get_sys(NULL, "cdev1");
+ if (IS_ERR(data->clk_cdev1)) {
+ dev_err(data->dev, "Can't retrieve clk cdev1\n");
+ ret = PTR_ERR(data->clk_cdev1);
goto err_put_pll_a_out0;
}
return 0;
err_put_pll_a_out0:
- clk_put(clk_pll_a_out0);
+ clk_put(data->clk_pll_a_out0);
err_put_pll_a:
- clk_put(clk_pll_a);
+ clk_put(data->clk_pll_a);
err:
return ret;
}
-void tegra_asoc_utils_fini(void)
+void tegra_asoc_utils_fini(struct tegra_asoc_utils_data *data)
{
- clk_put(clk_cdev1);
- clk_put(clk_pll_a_out0);
- clk_put(clk_pll_a);
+ clk_put(data->clk_cdev1);
+ clk_put(data->clk_pll_a_out0);
+ clk_put(data->clk_pll_a);
}
diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h
index 855f8f6e44ca..bbba7afdfc2c 100644
--- a/sound/soc/tegra/tegra_asoc_utils.h
+++ b/sound/soc/tegra/tegra_asoc_utils.h
@@ -23,9 +23,23 @@
#ifndef __TEGRA_ASOC_UTILS_H__
#define __TEGRA_ASOC_UTILS_H_
-int tegra_asoc_utils_set_rate(int srate, int mclk_rate, int *mclk_change);
-int tegra_asoc_utils_init(void);
-void tegra_asoc_utils_fini(void);
+struct clk;
+struct device;
+
+struct tegra_asoc_utils_data {
+ struct device *dev;
+ struct clk *clk_pll_a;
+ struct clk *clk_pll_a_out0;
+ struct clk *clk_cdev1;
+ int set_baseclock;
+ int set_mclk;
+};
+
+int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate,
+ int mclk, int *mclk_change);
+int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data,
+ struct device *dev);
+void tegra_asoc_utils_fini(struct tegra_asoc_utils_data *data);
#endif
diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c
index 6d668785e9af..870ee361f757 100644
--- a/sound/soc/tegra/tegra_i2s.c
+++ b/sound/soc/tegra/tegra_i2s.c
@@ -386,7 +386,7 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev)
snprintf(clk_name, sizeof(clk_name), DRV_NAME ".%d", pdev->id);
i2s->clk_i2s = clk_get_sys(clk_name, NULL);
if (IS_ERR(i2s->clk_i2s)) {
- pr_err("Can't retrieve i2s clock\n");
+ dev_err(&pdev->dev, "Can't retrieve i2s clock\n");
ret = PTR_ERR(i2s->clk_i2s);
goto err_free;
}