Merge branch 'for-3.1' into for-3.2
diff --git a/Documentation/devicetree/bindings/sound/soc/codecs/fsl-sgtl5000.txt b/Documentation/devicetree/bindings/sound/soc/codecs/fsl-sgtl5000.txt
new file mode 100644
index 0000000..2c3cd41
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/soc/codecs/fsl-sgtl5000.txt
@@ -0,0 +1,11 @@
+* Freescale SGTL5000 Stereo Codec
+
+Required properties:
+- compatible : "fsl,sgtl5000".
+
+Example:
+
+codec: sgtl5000@0a {
+ compatible = "fsl,sgtl5000";
+ reg = <0x0a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8510.txt b/Documentation/devicetree/bindings/sound/wm8510.txt
new file mode 100644
index 0000000..fa1a32b
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8510.txt
@@ -0,0 +1,18 @@
+WM8510 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+ - compatible : "wlf,wm8510"
+
+ - reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+Example:
+
+codec: wm8510@1a {
+ compatible = "wlf,wm8510";
+ reg = <0x1a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8523.txt b/Documentation/devicetree/bindings/sound/wm8523.txt
new file mode 100644
index 0000000..0474618
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8523.txt
@@ -0,0 +1,16 @@
+WM8523 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+ - compatible : "wlf,wm8523"
+
+ - reg : the I2C address of the device.
+
+Example:
+
+codec: wm8523@1a {
+ compatible = "wlf,wm8523";
+ reg = <0x1a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8580.txt b/Documentation/devicetree/bindings/sound/wm8580.txt
new file mode 100644
index 0000000..7d9821f
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8580.txt
@@ -0,0 +1,16 @@
+WM8580 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+ - compatible : "wlf,wm8580"
+
+ - reg : the I2C address of the device.
+
+Example:
+
+codec: wm8580@1a {
+ compatible = "wlf,wm8580";
+ reg = <0x1a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8711.txt b/Documentation/devicetree/bindings/sound/wm8711.txt
new file mode 100644
index 0000000..8ed9998
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8711.txt
@@ -0,0 +1,18 @@
+WM8711 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+ - compatible : "wlf,wm8711"
+
+ - reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+Example:
+
+codec: wm8711@1a {
+ compatible = "wlf,wm8711";
+ reg = <0x1a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8728.txt b/Documentation/devicetree/bindings/sound/wm8728.txt
new file mode 100644
index 0000000..a8b5c36
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8728.txt
@@ -0,0 +1,18 @@
+WM8728 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+ - compatible : "wlf,wm8728"
+
+ - reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+Example:
+
+codec: wm8728@1a {
+ compatible = "wlf,wm8728";
+ reg = <0x1a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8731.txt b/Documentation/devicetree/bindings/sound/wm8731.txt
new file mode 100644
index 0000000..15f7004
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8731.txt
@@ -0,0 +1,18 @@
+WM8731 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+ - compatible : "wlf,wm8731"
+
+ - reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+Example:
+
+codec: wm8731@1a {
+ compatible = "wlf,wm8731";
+ reg = <0x1a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8737.txt b/Documentation/devicetree/bindings/sound/wm8737.txt
new file mode 100644
index 0000000..4bc2cea
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8737.txt
@@ -0,0 +1,18 @@
+WM8737 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+ - compatible : "wlf,wm8737"
+
+ - reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+Example:
+
+codec: wm8737@1a {
+ compatible = "wlf,wm8737";
+ reg = <0x1a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8741.txt b/Documentation/devicetree/bindings/sound/wm8741.txt
new file mode 100644
index 0000000..74bda58
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8741.txt
@@ -0,0 +1,18 @@
+WM8741 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+ - compatible : "wlf,wm8741"
+
+ - reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+Example:
+
+codec: wm8741@1a {
+ compatible = "wlf,wm8741";
+ reg = <0x1a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8750.txt b/Documentation/devicetree/bindings/sound/wm8750.txt
new file mode 100644
index 0000000..8db239f
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8750.txt
@@ -0,0 +1,18 @@
+WM8750 and WM8987 audio CODECs
+
+These devices support both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+ - compatible : "wlf,wm8750" or "wlf,wm8987"
+
+ - reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+Example:
+
+codec: wm8750@1a {
+ compatible = "wlf,wm8750";
+ reg = <0x1a>;
+};
diff --git a/Documentation/devicetree/bindings/sound/wm8753.txt b/Documentation/devicetree/bindings/sound/wm8753.txt
new file mode 100644
index 0000000..e65277a
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8753.txt
@@ -0,0 +1,18 @@
+WM8753 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+ - compatible : "wlf,wm8753"
+
+ - reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+Example:
+
+codec: wm8737@1a {
+ compatible = "wlf,wm8753";
+ reg = <0x1a>;
+};
diff --git a/arch/mips/alchemy/devboards/db1200/platform.c b/arch/mips/alchemy/devboards/db1200/platform.c
index fbb5593..dda090b 100644
--- a/arch/mips/alchemy/devboards/db1200/platform.c
+++ b/arch/mips/alchemy/devboards/db1200/platform.c
@@ -422,6 +422,7 @@
},
};
+/* AC97 or I2S device */
static struct platform_device db1200_audio_dev = {
/* name assigned later based on switch setting */
.id = 1, /* PSC ID */
@@ -429,19 +430,32 @@
.resource = au1200_psc1_res,
};
+/* DB1200 ASoC card device */
+static struct platform_device db1200_sound_dev = {
+ /* name assigned later based on switch setting */
+ .id = 1, /* PSC ID */
+};
+
static struct platform_device db1200_stac_dev = {
.name = "ac97-codec",
.id = 1, /* on PSC1 */
};
+static struct platform_device db1200_audiodma_dev = {
+ .name = "au1xpsc-pcm",
+ .id = 1, /* PSC ID */
+};
+
static struct platform_device *db1200_devs[] __initdata = {
NULL, /* PSC0, selected by S6.8 */
&db1200_ide_dev,
&db1200_eth_dev,
&db1200_rtc_dev,
&db1200_nand_dev,
+ &db1200_audiodma_dev,
&db1200_audio_dev,
&db1200_stac_dev,
+ &db1200_sound_dev,
};
static int __init db1200_dev_init(void)
@@ -501,10 +515,12 @@
if (sw == BCSR_SWITCHES_DIP_8) {
bcsr_mod(BCSR_RESETS, 0, BCSR_RESETS_PSC1MUX);
db1200_audio_dev.name = "au1xpsc_i2s";
+ db1200_sound_dev.name = "db1200-i2s";
printk(KERN_INFO " S6.7 ON : PSC1 mode I2S\n");
} else {
bcsr_mod(BCSR_RESETS, BCSR_RESETS_PSC1MUX, 0);
db1200_audio_dev.name = "au1xpsc_ac97";
+ db1200_sound_dev.name = "db1200-ac97";
printk(KERN_INFO " S6.7 OFF: PSC1 mode AC97\n");
}
diff --git a/arch/mips/alchemy/devboards/db1x00/platform.c b/arch/mips/alchemy/devboards/db1x00/platform.c
index 978d5ab..7057d28 100644
--- a/arch/mips/alchemy/devboards/db1x00/platform.c
+++ b/arch/mips/alchemy/devboards/db1x00/platform.c
@@ -19,8 +19,11 @@
*/
#include <linux/init.h>
+#include <linux/interrupt.h>
#include <linux/platform_device.h>
+#include <asm/mach-au1x00/au1000.h>
+#include <asm/mach-au1x00/au1000_dma.h>
#include <asm/mach-au1x00/au1xxx.h>
#include <asm/mach-db1x00/bcsr.h>
#include "../platform.h"
@@ -85,6 +88,45 @@
#endif
#endif
+static struct resource alchemy_ac97c_res[] = {
+ [0] = {
+ .start = AU1000_AC97_PHYS_ADDR,
+ .end = AU1000_AC97_PHYS_ADDR + 0xfff,
+ .flags = IORESOURCE_MEM,
+ },
+ [1] = {
+ .start = DMA_ID_AC97C_TX,
+ .end = DMA_ID_AC97C_TX,
+ .flags = IORESOURCE_DMA,
+ },
+ [2] = {
+ .start = DMA_ID_AC97C_RX,
+ .end = DMA_ID_AC97C_RX,
+ .flags = IORESOURCE_DMA,
+ },
+};
+
+static struct platform_device alchemy_ac97c_dev = {
+ .name = "alchemy-ac97c",
+ .id = -1,
+ .resource = alchemy_ac97c_res,
+ .num_resources = ARRAY_SIZE(alchemy_ac97c_res),
+};
+
+static struct platform_device alchemy_ac97c_dma_dev = {
+ .name = "alchemy-pcm-dma",
+ .id = 0,
+};
+
+static struct platform_device db1x00_codec_dev = {
+ .name = "ac97-codec",
+ .id = -1,
+};
+
+static struct platform_device db1x00_audio_dev = {
+ .name = "db1000-audio",
+};
+
static int __init db1xxx_dev_init(void)
{
#ifdef DB1XXX_HAS_PCMCIA
@@ -113,6 +155,12 @@
1);
#endif
db1x_register_norflash(BOARD_FLASH_SIZE, BOARD_FLASH_WIDTH, F_SWAPPED);
+
+ platform_device_register(&db1x00_codec_dev);
+ platform_device_register(&alchemy_ac97c_dma_dev);
+ platform_device_register(&alchemy_ac97c_dev);
+ platform_device_register(&db1x00_audio_dev);
+
return 0;
}
device_initcall(db1xxx_dev_init);
diff --git a/include/linux/mfd/wm8994/registers.h b/include/linux/mfd/wm8994/registers.h
index f3ee842..83ecdcd 100644
--- a/include/linux/mfd/wm8994/registers.h
+++ b/include/linux/mfd/wm8994/registers.h
@@ -72,6 +72,7 @@
#define WM8994_DC_SERVO_2 0x55
#define WM8994_DC_SERVO_4 0x57
#define WM8994_DC_SERVO_READBACK 0x58
+#define WM8994_DC_SERVO_4E 0x59
#define WM8994_ANALOGUE_HP_1 0x60
#define WM8958_MIC_DETECT_1 0xD0
#define WM8958_MIC_DETECT_2 0xD1
@@ -133,6 +134,8 @@
#define WM8994_AIF1_DAC1_FILTERS_2 0x421
#define WM8994_AIF1_DAC2_FILTERS_1 0x422
#define WM8994_AIF1_DAC2_FILTERS_2 0x423
+#define WM8958_AIF1_DAC1_NOISE_GATE 0x430
+#define WM8958_AIF1_DAC2_NOISE_GATE 0x431
#define WM8994_AIF1_DRC1_1 0x440
#define WM8994_AIF1_DRC1_2 0x441
#define WM8994_AIF1_DRC1_3 0x442
@@ -190,6 +193,7 @@
#define WM8994_AIF2_ADC_FILTERS 0x510
#define WM8994_AIF2_DAC_FILTERS_1 0x520
#define WM8994_AIF2_DAC_FILTERS_2 0x521
+#define WM8958_AIF2_DAC_NOISE_GATE 0x530
#define WM8994_AIF2_DRC_1 0x540
#define WM8994_AIF2_DRC_2 0x541
#define WM8994_AIF2_DRC_3 0x542
@@ -1921,6 +1925,44 @@
#define WM8994_LDO2_DISCH_WIDTH 1 /* LDO2_DISCH */
/*
+ * R61 (0x3D) - MICBIAS1
+ */
+#define WM8958_MICB1_RATE 0x0020 /* MICB1_RATE */
+#define WM8958_MICB1_RATE_MASK 0x0020 /* MICB1_RATE */
+#define WM8958_MICB1_RATE_SHIFT 5 /* MICB1_RATE */
+#define WM8958_MICB1_RATE_WIDTH 1 /* MICB1_RATE */
+#define WM8958_MICB1_MODE 0x0010 /* MICB1_MODE */
+#define WM8958_MICB1_MODE_MASK 0x0010 /* MICB1_MODE */
+#define WM8958_MICB1_MODE_SHIFT 4 /* MICB1_MODE */
+#define WM8958_MICB1_MODE_WIDTH 1 /* MICB1_MODE */
+#define WM8958_MICB1_LVL_MASK 0x000E /* MICB1_LVL - [3:1] */
+#define WM8958_MICB1_LVL_SHIFT 1 /* MICB1_LVL - [3:1] */
+#define WM8958_MICB1_LVL_WIDTH 3 /* MICB1_LVL - [3:1] */
+#define WM8958_MICB1_DISCH 0x0001 /* MICB1_DISCH */
+#define WM8958_MICB1_DISCH_MASK 0x0001 /* MICB1_DISCH */
+#define WM8958_MICB1_DISCH_SHIFT 0 /* MICB1_DISCH */
+#define WM8958_MICB1_DISCH_WIDTH 1 /* MICB1_DISCH */
+
+/*
+ * R62 (0x3E) - MICBIAS2
+ */
+#define WM8958_MICB2_RATE 0x0020 /* MICB2_RATE */
+#define WM8958_MICB2_RATE_MASK 0x0020 /* MICB2_RATE */
+#define WM8958_MICB2_RATE_SHIFT 5 /* MICB2_RATE */
+#define WM8958_MICB2_RATE_WIDTH 1 /* MICB2_RATE */
+#define WM8958_MICB2_MODE 0x0010 /* MICB2_MODE */
+#define WM8958_MICB2_MODE_MASK 0x0010 /* MICB2_MODE */
+#define WM8958_MICB2_MODE_SHIFT 4 /* MICB2_MODE */
+#define WM8958_MICB2_MODE_WIDTH 1 /* MICB2_MODE */
+#define WM8958_MICB2_LVL_MASK 0x000E /* MICB2_LVL - [3:1] */
+#define WM8958_MICB2_LVL_SHIFT 1 /* MICB2_LVL - [3:1] */
+#define WM8958_MICB2_LVL_WIDTH 3 /* MICB2_LVL - [3:1] */
+#define WM8958_MICB2_DISCH 0x0001 /* MICB2_DISCH */
+#define WM8958_MICB2_DISCH_MASK 0x0001 /* MICB2_DISCH */
+#define WM8958_MICB2_DISCH_SHIFT 0 /* MICB2_DISCH */
+#define WM8958_MICB2_DISCH_WIDTH 1 /* MICB2_DISCH */
+
+/*
* R76 (0x4C) - Charge Pump (1)
*/
#define WM8994_CP_ENA 0x8000 /* CP_ENA */
@@ -2949,6 +2991,34 @@
#define WM8994_AIF1DAC2_3D_ENA_WIDTH 1 /* AIF1DAC2_3D_ENA */
/*
+ * R1072 (0x430) - AIF1 DAC1 Noise Gate
+ */
+#define WM8958_AIF1DAC1_NG_HLD_MASK 0x0060 /* AIF1DAC1_NG_HLD - [6:5] */
+#define WM8958_AIF1DAC1_NG_HLD_SHIFT 5 /* AIF1DAC1_NG_HLD - [6:5] */
+#define WM8958_AIF1DAC1_NG_HLD_WIDTH 2 /* AIF1DAC1_NG_HLD - [6:5] */
+#define WM8958_AIF1DAC1_NG_THR_MASK 0x000E /* AIF1DAC1_NG_THR - [3:1] */
+#define WM8958_AIF1DAC1_NG_THR_SHIFT 1 /* AIF1DAC1_NG_THR - [3:1] */
+#define WM8958_AIF1DAC1_NG_THR_WIDTH 3 /* AIF1DAC1_NG_THR - [3:1] */
+#define WM8958_AIF1DAC1_NG_ENA 0x0001 /* AIF1DAC1_NG_ENA */
+#define WM8958_AIF1DAC1_NG_ENA_MASK 0x0001 /* AIF1DAC1_NG_ENA */
+#define WM8958_AIF1DAC1_NG_ENA_SHIFT 0 /* AIF1DAC1_NG_ENA */
+#define WM8958_AIF1DAC1_NG_ENA_WIDTH 1 /* AIF1DAC1_NG_ENA */
+
+/*
+ * R1073 (0x431) - AIF1 DAC2 Noise Gate
+ */
+#define WM8958_AIF1DAC2_NG_HLD_MASK 0x0060 /* AIF1DAC2_NG_HLD - [6:5] */
+#define WM8958_AIF1DAC2_NG_HLD_SHIFT 5 /* AIF1DAC2_NG_HLD - [6:5] */
+#define WM8958_AIF1DAC2_NG_HLD_WIDTH 2 /* AIF1DAC2_NG_HLD - [6:5] */
+#define WM8958_AIF1DAC2_NG_THR_MASK 0x000E /* AIF1DAC2_NG_THR - [3:1] */
+#define WM8958_AIF1DAC2_NG_THR_SHIFT 1 /* AIF1DAC2_NG_THR - [3:1] */
+#define WM8958_AIF1DAC2_NG_THR_WIDTH 3 /* AIF1DAC2_NG_THR - [3:1] */
+#define WM8958_AIF1DAC2_NG_ENA 0x0001 /* AIF1DAC2_NG_ENA */
+#define WM8958_AIF1DAC2_NG_ENA_MASK 0x0001 /* AIF1DAC2_NG_ENA */
+#define WM8958_AIF1DAC2_NG_ENA_SHIFT 0 /* AIF1DAC2_NG_ENA */
+#define WM8958_AIF1DAC2_NG_ENA_WIDTH 1 /* AIF1DAC2_NG_ENA */
+
+/*
* R1088 (0x440) - AIF1 DRC1 (1)
*/
#define WM8994_AIF1DRC1_SIG_DET_RMS_MASK 0xF800 /* AIF1DRC1_SIG_DET_RMS - [15:11] */
@@ -3560,6 +3630,20 @@
#define WM8994_AIF2DAC_3D_ENA_WIDTH 1 /* AIF2DAC_3D_ENA */
/*
+ * R1328 (0x530) - AIF2 DAC Noise Gate
+ */
+#define WM8958_AIF2DAC_NG_HLD_MASK 0x0060 /* AIF2DAC_NG_HLD - [6:5] */
+#define WM8958_AIF2DAC_NG_HLD_SHIFT 5 /* AIF2DAC_NG_HLD - [6:5] */
+#define WM8958_AIF2DAC_NG_HLD_WIDTH 2 /* AIF2DAC_NG_HLD - [6:5] */
+#define WM8958_AIF2DAC_NG_THR_MASK 0x000E /* AIF2DAC_NG_THR - [3:1] */
+#define WM8958_AIF2DAC_NG_THR_SHIFT 1 /* AIF2DAC_NG_THR - [3:1] */
+#define WM8958_AIF2DAC_NG_THR_WIDTH 3 /* AIF2DAC_NG_THR - [3:1] */
+#define WM8958_AIF2DAC_NG_ENA 0x0001 /* AIF2DAC_NG_ENA */
+#define WM8958_AIF2DAC_NG_ENA_MASK 0x0001 /* AIF2DAC_NG_ENA */
+#define WM8958_AIF2DAC_NG_ENA_SHIFT 0 /* AIF2DAC_NG_ENA */
+#define WM8958_AIF2DAC_NG_ENA_WIDTH 1 /* AIF2DAC_NG_ENA */
+
+/*
* R1344 (0x540) - AIF2 DRC (1)
*/
#define WM8994_AIF2DRC_SIG_DET_RMS_MASK 0xF800 /* AIF2DRC_SIG_DET_RMS - [15:11] */
diff --git a/include/sound/adau1373.h b/include/sound/adau1373.h
new file mode 100644
index 0000000..1b19c76
--- /dev/null
+++ b/include/sound/adau1373.h
@@ -0,0 +1,34 @@
+/*
+ * Analog Devices ADAU1373 Audio Codec drive
+ *
+ * Copyright 2011 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#ifndef __SOUND_ADAU1373_H__
+#define __SOUND_ADAU1373_H__
+
+enum adau1373_micbias_voltage {
+ ADAU1373_MICBIAS_2_9V = 0,
+ ADAU1373_MICBIAS_2_2V = 1,
+ ADAU1373_MICBIAS_2_6V = 2,
+ ADAU1373_MICBIAS_1_8V = 3,
+};
+
+#define ADAU1373_DRC_SIZE 13
+
+struct adau1373_platform_data {
+ bool input_differential[4];
+ bool lineout_differential;
+ bool lineout_ground_sense;
+
+ unsigned int num_drc;
+ uint8_t drc_setting[3][ADAU1373_DRC_SIZE];
+
+ enum adau1373_micbias_voltage micbias1;
+ enum adau1373_micbias_voltage micbias2;
+};
+
+#endif
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index e0583b7..350b1b3 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -524,6 +524,8 @@
enum snd_soc_bias_level target_bias_level;
struct list_head list;
+ int (*stream_event)(struct snd_soc_dapm_context *dapm, int event);
+
#ifdef CONFIG_DEBUG_FS
struct dentry *debugfs_dapm;
#endif
diff --git a/include/sound/soc.h b/include/sound/soc.h
index aa19f5a..6da55a1 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -19,6 +19,7 @@
#include <linux/workqueue.h>
#include <linux/interrupt.h>
#include <linux/kernel.h>
+#include <linux/regmap.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/control.h>
@@ -260,6 +261,7 @@
enum snd_soc_control_type {
SND_SOC_I2C = 1,
SND_SOC_SPI,
+ SND_SOC_REGMAP,
};
enum snd_soc_compress_type {
@@ -576,6 +578,7 @@
const void *reg_def_copy;
const struct snd_soc_cache_ops *cache_ops;
struct mutex cache_rw_mutex;
+ int val_bytes;
/* dapm */
struct snd_soc_dapm_context dapm;
@@ -630,10 +633,14 @@
/* codec bias level */
int (*set_bias_level)(struct snd_soc_codec *,
enum snd_soc_bias_level level);
+ bool idle_bias_off;
void (*seq_notifier)(struct snd_soc_dapm_context *,
enum snd_soc_dapm_type, int);
+ /* codec stream completion event */
+ int (*stream_event)(struct snd_soc_dapm_context *dapm, int event);
+
/* probe ordering - for components with runtime dependencies */
int probe_order;
int remove_order;
@@ -669,6 +676,9 @@
/* platform stream ops */
struct snd_pcm_ops *ops;
+ /* platform stream completion event */
+ int (*stream_event)(struct snd_soc_dapm_context *dapm, int event);
+
/* probe ordering - for components with runtime dependencies */
int probe_order;
int remove_order;
diff --git a/sound/mips/Kconfig b/sound/mips/Kconfig
index a9823fa..77dd0a1 100644
--- a/sound/mips/Kconfig
+++ b/sound/mips/Kconfig
@@ -23,12 +23,15 @@
config SND_AU1X00
- tristate "Au1x00 AC97 Port Driver"
+ tristate "Au1x00 AC97 Port Driver (DEPRECATED)"
depends on SOC_AU1000 || SOC_AU1100 || SOC_AU1500
select SND_PCM
select SND_AC97_CODEC
help
ALSA Sound driver for the Au1x00's AC97 port.
+ Newer drivers for ASoC are available, please do not use
+ this driver as it will be removed in the future.
+
endif # SND_MIPS
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 8224db5..1381db8 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -7,6 +7,8 @@
select SND_PCM
select AC97_BUS if SND_SOC_AC97_BUS
select SND_JACK if INPUT=y || INPUT=SND
+ select REGMAP_I2C if I2C
+ select REGMAP_SPI if SPI_MASTER
---help---
If you want ASoC support, you should say Y here and also to the
@@ -51,6 +53,7 @@
source "sound/soc/omap/Kconfig"
source "sound/soc/kirkwood/Kconfig"
source "sound/soc/mid-x86/Kconfig"
+source "sound/soc/mxs/Kconfig"
source "sound/soc/pxa/Kconfig"
source "sound/soc/samsung/Kconfig"
source "sound/soc/s6000/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 4f91387..9ea8ac8 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -12,6 +12,7 @@
obj-$(CONFIG_SND_SOC) += imx/
obj-$(CONFIG_SND_SOC) += jz4740/
obj-$(CONFIG_SND_SOC) += mid-x86/
+obj-$(CONFIG_SND_SOC) += mxs/
obj-$(CONFIG_SND_SOC) += nuc900/
obj-$(CONFIG_SND_SOC) += omap/
obj-$(CONFIG_SND_SOC) += kirkwood/
diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig
index 4b67140..6d59254 100644
--- a/sound/soc/au1x/Kconfig
+++ b/sound/soc/au1x/Kconfig
@@ -18,10 +18,38 @@
select SND_AC97_CODEC
select SND_SOC_AC97_BUS
+##
+## Au1000/1500/1100 DMA + AC97C/I2SC
+##
+config SND_SOC_AU1XAUDIO
+ tristate "SoC Audio for Au1000/Au1500/Au1100"
+ depends on MIPS_ALCHEMY
+ help
+ This is a driver set for the AC97 unit and the
+ old DMA controller as found on the Au1000/Au1500/Au1100 chips.
+
+config SND_SOC_AU1XAC97C
+ tristate
+ select AC97_BUS
+ select SND_AC97_CODEC
+ select SND_SOC_AC97_BUS
+
+config SND_SOC_AU1XI2SC
+ tristate
+
##
## Boards
##
+config SND_SOC_DB1000
+ tristate "DB1000 Audio support"
+ depends on SND_SOC_AU1XAUDIO
+ select SND_SOC_AU1XAC97C
+ select SND_SOC_AC97_CODEC
+ help
+ Select this option to enable AC97 audio on the early DB1x00 series
+ of boards (DB1000/DB1500/DB1100).
+
config SND_SOC_DB1200
tristate "DB1200 AC97+I2S audio support"
depends on SND_SOC_AU1XPSC
diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile
index 1687307..9207105 100644
--- a/sound/soc/au1x/Makefile
+++ b/sound/soc/au1x/Makefile
@@ -3,11 +3,21 @@
snd-soc-au1xpsc-i2s-objs := psc-i2s.o
snd-soc-au1xpsc-ac97-objs := psc-ac97.o
+# Au1000/1500/1100 Audio units
+snd-soc-au1x-dma-objs := dma.o
+snd-soc-au1x-ac97c-objs := ac97c.o
+snd-soc-au1x-i2sc-objs := i2sc.o
+
obj-$(CONFIG_SND_SOC_AU1XPSC) += snd-soc-au1xpsc-dbdma.o
obj-$(CONFIG_SND_SOC_AU1XPSC_I2S) += snd-soc-au1xpsc-i2s.o
obj-$(CONFIG_SND_SOC_AU1XPSC_AC97) += snd-soc-au1xpsc-ac97.o
+obj-$(CONFIG_SND_SOC_AU1XAUDIO) += snd-soc-au1x-dma.o
+obj-$(CONFIG_SND_SOC_AU1XAC97C) += snd-soc-au1x-ac97c.o
+obj-$(CONFIG_SND_SOC_AU1XI2SC) += snd-soc-au1x-i2sc.o
# Boards
+snd-soc-db1000-objs := db1000.o
snd-soc-db1200-objs := db1200.o
+obj-$(CONFIG_SND_SOC_DB1000) += snd-soc-db1000.o
obj-$(CONFIG_SND_SOC_DB1200) += snd-soc-db1200.o
diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c
new file mode 100644
index 0000000..13802ff
--- /dev/null
+++ b/sound/soc/au1x/ac97c.c
@@ -0,0 +1,363 @@
+/*
+ * Au1000/Au1500/Au1100 AC97C controller driver for ASoC
+ *
+ * (c) 2011 Manuel Lauss <manuel.lauss@googlemail.com>
+ *
+ * based on the old ALSA driver originally written by
+ * Charles Eidsness <charles@cooper-street.com>
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/mutex.h>
+#include <linux/platform_device.h>
+#include <linux/suspend.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <asm/mach-au1x00/au1000.h>
+
+#include "psc.h"
+
+/* register offsets and bits */
+#define AC97_CONFIG 0x00
+#define AC97_STATUS 0x04
+#define AC97_DATA 0x08
+#define AC97_CMDRESP 0x0c
+#define AC97_ENABLE 0x10
+
+#define CFG_RC(x) (((x) & 0x3ff) << 13) /* valid rx slots mask */
+#define CFG_XS(x) (((x) & 0x3ff) << 3) /* valid tx slots mask */
+#define CFG_SG (1 << 2) /* sync gate */
+#define CFG_SN (1 << 1) /* sync control */
+#define CFG_RS (1 << 0) /* acrst# control */
+#define STAT_XU (1 << 11) /* tx underflow */
+#define STAT_XO (1 << 10) /* tx overflow */
+#define STAT_RU (1 << 9) /* rx underflow */
+#define STAT_RO (1 << 8) /* rx overflow */
+#define STAT_RD (1 << 7) /* codec ready */
+#define STAT_CP (1 << 6) /* command pending */
+#define STAT_TE (1 << 4) /* tx fifo empty */
+#define STAT_TF (1 << 3) /* tx fifo full */
+#define STAT_RE (1 << 1) /* rx fifo empty */
+#define STAT_RF (1 << 0) /* rx fifo full */
+#define CMD_SET_DATA(x) (((x) & 0xffff) << 16)
+#define CMD_GET_DATA(x) ((x) & 0xffff)
+#define CMD_READ (1 << 7)
+#define CMD_WRITE (0 << 7)
+#define CMD_IDX(x) ((x) & 0x7f)
+#define EN_D (1 << 1) /* DISable bit */
+#define EN_CE (1 << 0) /* clock enable bit */
+
+/* how often to retry failed codec register reads/writes */
+#define AC97_RW_RETRIES 5
+
+#define AC97_RATES \
+ SNDRV_PCM_RATE_CONTINUOUS
+
+#define AC97_FMTS \
+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE)
+
+/* instance data. There can be only one, MacLeod!!!!, fortunately there IS only
+ * once AC97C on early Alchemy chips. The newer ones aren't so lucky.
+ */
+static struct au1xpsc_audio_data *ac97c_workdata;
+#define ac97_to_ctx(x) ac97c_workdata
+
+static inline unsigned long RD(struct au1xpsc_audio_data *ctx, int reg)
+{
+ return __raw_readl(ctx->mmio + reg);
+}
+
+static inline void WR(struct au1xpsc_audio_data *ctx, int reg, unsigned long v)
+{
+ __raw_writel(v, ctx->mmio + reg);
+ wmb();
+}
+
+static unsigned short au1xac97c_ac97_read(struct snd_ac97 *ac97,
+ unsigned short r)
+{
+ struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97);
+ unsigned int tmo, retry;
+ unsigned long data;
+
+ data = ~0;
+ retry = AC97_RW_RETRIES;
+ do {
+ mutex_lock(&ctx->lock);
+
+ tmo = 5;
+ while ((RD(ctx, AC97_STATUS) & STAT_CP) && tmo--)
+ udelay(21); /* wait an ac97 frame time */
+ if (!tmo) {
+ pr_debug("ac97rd timeout #1\n");
+ goto next;
+ }
+
+ WR(ctx, AC97_CMDRESP, CMD_IDX(r) | CMD_READ);
+
+ /* stupid errata: data is only valid for 21us, so
+ * poll, Forrest, poll...
+ */
+ tmo = 0x10000;
+ while ((RD(ctx, AC97_STATUS) & STAT_CP) && tmo--)
+ asm volatile ("nop");
+ data = RD(ctx, AC97_CMDRESP);
+
+ if (!tmo)
+ pr_debug("ac97rd timeout #2\n");
+
+next:
+ mutex_unlock(&ctx->lock);
+ } while (--retry && !tmo);
+
+ pr_debug("AC97RD %04x %04lx %d\n", r, data, retry);
+
+ return retry ? data & 0xffff : 0xffff;
+}
+
+static void au1xac97c_ac97_write(struct snd_ac97 *ac97, unsigned short r,
+ unsigned short v)
+{
+ struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97);
+ unsigned int tmo, retry;
+
+ retry = AC97_RW_RETRIES;
+ do {
+ mutex_lock(&ctx->lock);
+
+ for (tmo = 5; (RD(ctx, AC97_STATUS) & STAT_CP) && tmo; tmo--)
+ udelay(21);
+ if (!tmo) {
+ pr_debug("ac97wr timeout #1\n");
+ goto next;
+ }
+
+ WR(ctx, AC97_CMDRESP, CMD_WRITE | CMD_IDX(r) | CMD_SET_DATA(v));
+
+ for (tmo = 10; (RD(ctx, AC97_STATUS) & STAT_CP) && tmo; tmo--)
+ udelay(21);
+ if (!tmo)
+ pr_debug("ac97wr timeout #2\n");
+next:
+ mutex_unlock(&ctx->lock);
+ } while (--retry && !tmo);
+
+ pr_debug("AC97WR %04x %04x %d\n", r, v, retry);
+}
+
+static void au1xac97c_ac97_warm_reset(struct snd_ac97 *ac97)
+{
+ struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97);
+
+ WR(ctx, AC97_CONFIG, ctx->cfg | CFG_SG | CFG_SN);
+ msleep(20);
+ WR(ctx, AC97_CONFIG, ctx->cfg | CFG_SG);
+ WR(ctx, AC97_CONFIG, ctx->cfg);
+}
+
+static void au1xac97c_ac97_cold_reset(struct snd_ac97 *ac97)
+{
+ struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97);
+ int i;
+
+ WR(ctx, AC97_CONFIG, ctx->cfg | CFG_RS);
+ msleep(500);
+ WR(ctx, AC97_CONFIG, ctx->cfg);
+
+ /* wait for codec ready */
+ i = 50;
+ while (((RD(ctx, AC97_STATUS) & STAT_RD) == 0) && --i)
+ msleep(20);
+ if (!i)
+ printk(KERN_ERR "ac97c: codec not ready after cold reset\n");
+}
+
+/* AC97 controller operations */
+struct snd_ac97_bus_ops soc_ac97_ops = {
+ .read = au1xac97c_ac97_read,
+ .write = au1xac97c_ac97_write,
+ .reset = au1xac97c_ac97_cold_reset,
+ .warm_reset = au1xac97c_ac97_warm_reset,
+};
+EXPORT_SYMBOL_GPL(soc_ac97_ops); /* globals be gone! */
+
+static int alchemy_ac97c_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai);
+ snd_soc_dai_set_dma_data(dai, substream, &ctx->dmaids[0]);
+ return 0;
+}
+
+static struct snd_soc_dai_ops alchemy_ac97c_ops = {
+ .startup = alchemy_ac97c_startup,
+};
+
+static int au1xac97c_dai_probe(struct snd_soc_dai *dai)
+{
+ return ac97c_workdata ? 0 : -ENODEV;
+}
+
+static struct snd_soc_dai_driver au1xac97c_dai_driver = {
+ .name = "alchemy-ac97c",
+ .ac97_control = 1,
+ .probe = au1xac97c_dai_probe,
+ .playback = {
+ .rates = AC97_RATES,
+ .formats = AC97_FMTS,
+ .channels_min = 2,
+ .channels_max = 2,
+ },
+ .capture = {
+ .rates = AC97_RATES,
+ .formats = AC97_FMTS,
+ .channels_min = 2,
+ .channels_max = 2,
+ },
+ .ops = &alchemy_ac97c_ops,
+};
+
+static int __devinit au1xac97c_drvprobe(struct platform_device *pdev)
+{
+ int ret;
+ struct resource *r;
+ struct au1xpsc_audio_data *ctx;
+
+ ctx = kzalloc(sizeof(*ctx), GFP_KERNEL);
+ if (!ctx)
+ return -ENOMEM;
+
+ mutex_init(&ctx->lock);
+
+ r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!r) {
+ ret = -ENODEV;
+ goto out0;
+ }
+
+ ret = -EBUSY;
+ if (!request_mem_region(r->start, resource_size(r), pdev->name))
+ goto out0;
+
+ ctx->mmio = ioremap_nocache(r->start, resource_size(r));
+ if (!ctx->mmio)
+ goto out1;
+
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+ if (!r)
+ goto out1;
+ ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start;
+
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+ if (!r)
+ goto out1;
+ ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start;
+
+ /* switch it on */
+ WR(ctx, AC97_ENABLE, EN_D | EN_CE);
+ WR(ctx, AC97_ENABLE, EN_CE);
+
+ ctx->cfg = CFG_RC(3) | CFG_XS(3);
+ WR(ctx, AC97_CONFIG, ctx->cfg);
+
+ platform_set_drvdata(pdev, ctx);
+
+ ret = snd_soc_register_dai(&pdev->dev, &au1xac97c_dai_driver);
+ if (ret)
+ goto out1;
+
+ ac97c_workdata = ctx;
+ return 0;
+
+out1:
+ release_mem_region(r->start, resource_size(r));
+out0:
+ kfree(ctx);
+ return ret;
+}
+
+static int __devexit au1xac97c_drvremove(struct platform_device *pdev)
+{
+ struct au1xpsc_audio_data *ctx = platform_get_drvdata(pdev);
+ struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+
+ snd_soc_unregister_dai(&pdev->dev);
+
+ WR(ctx, AC97_ENABLE, EN_D); /* clock off, disable */
+
+ iounmap(ctx->mmio);
+ release_mem_region(r->start, resource_size(r));
+ kfree(ctx);
+
+ ac97c_workdata = NULL; /* MDEV */
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int au1xac97c_drvsuspend(struct device *dev)
+{
+ struct au1xpsc_audio_data *ctx = dev_get_drvdata(dev);
+
+ WR(ctx, AC97_ENABLE, EN_D); /* clock off, disable */
+
+ return 0;
+}
+
+static int au1xac97c_drvresume(struct device *dev)
+{
+ struct au1xpsc_audio_data *ctx = dev_get_drvdata(dev);
+
+ WR(ctx, AC97_ENABLE, EN_D | EN_CE);
+ WR(ctx, AC97_ENABLE, EN_CE);
+ WR(ctx, AC97_CONFIG, ctx->cfg);
+
+ return 0;
+}
+
+static const struct dev_pm_ops au1xpscac97_pmops = {
+ .suspend = au1xac97c_drvsuspend,
+ .resume = au1xac97c_drvresume,
+};
+
+#define AU1XPSCAC97_PMOPS (&au1xpscac97_pmops)
+
+#else
+
+#define AU1XPSCAC97_PMOPS NULL
+
+#endif
+
+static struct platform_driver au1xac97c_driver = {
+ .driver = {
+ .name = "alchemy-ac97c",
+ .owner = THIS_MODULE,
+ .pm = AU1XPSCAC97_PMOPS,
+ },
+ .probe = au1xac97c_drvprobe,
+ .remove = __devexit_p(au1xac97c_drvremove),
+};
+
+static int __init au1xac97c_load(void)
+{
+ ac97c_workdata = NULL;
+ return platform_driver_register(&au1xac97c_driver);
+}
+
+static void __exit au1xac97c_unload(void)
+{
+ platform_driver_unregister(&au1xac97c_driver);
+}
+
+module_init(au1xac97c_load);
+module_exit(au1xac97c_unload);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au1000/1500/1100 AC97C ASoC driver");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/db1000.c b/sound/soc/au1x/db1000.c
new file mode 100644
index 0000000..127477a
--- /dev/null
+++ b/sound/soc/au1x/db1000.c
@@ -0,0 +1,75 @@
+/*
+ * DB1000/DB1500/DB1100 ASoC audio fabric support code.
+ *
+ * (c) 2011 Manuel Lauss <manuel.lauss@googlemail.com>
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <asm/mach-au1x00/au1000.h>
+#include <asm/mach-db1x00/bcsr.h>
+
+#include "psc.h"
+
+static struct snd_soc_dai_link db1000_ac97_dai = {
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .codec_dai_name = "ac97-hifi",
+ .cpu_dai_name = "alchemy-ac97c",
+ .platform_name = "alchemy-pcm-dma.0",
+ .codec_name = "ac97-codec",
+};
+
+static struct snd_soc_card db1000_ac97 = {
+ .name = "DB1000_AC97",
+ .dai_link = &db1000_ac97_dai,
+ .num_links = 1,
+};
+
+static int __devinit db1000_audio_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &db1000_ac97;
+ card->dev = &pdev->dev;
+ return snd_soc_register_card(card);
+}
+
+static int __devexit db1000_audio_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+ snd_soc_unregister_card(card);
+ return 0;
+}
+
+static struct platform_driver db1000_audio_driver = {
+ .driver = {
+ .name = "db1000-audio",
+ .owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = db1000_audio_probe,
+ .remove = __devexit_p(db1000_audio_remove),
+};
+
+static int __init db1000_audio_load(void)
+{
+ return platform_driver_register(&db1000_audio_driver);
+}
+
+static void __exit db1000_audio_unload(void)
+{
+ platform_driver_unregister(&db1000_audio_driver);
+}
+
+module_init(db1000_audio_load);
+module_exit(db1000_audio_unload);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("DB1000/DB1500/DB1100 ASoC audio");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c
index 1d3e258..289312c 100644
--- a/sound/soc/au1x/db1200.c
+++ b/sound/soc/au1x/db1200.c
@@ -1,7 +1,7 @@
/*
* DB1200 ASoC audio fabric support code.
*
- * (c) 2008-9 Manuel Lauss <manuel.lauss@gmail.com>
+ * (c) 2008-2011 Manuel Lauss <manuel.lauss@googlemail.com>
*
*/
@@ -21,6 +21,17 @@
#include "../codecs/wm8731.h"
#include "psc.h"
+static struct platform_device_id db1200_pids[] = {
+ {
+ .name = "db1200-ac97",
+ .driver_data = 0,
+ }, {
+ .name = "db1200-i2s",
+ .driver_data = 1,
+ },
+ {},
+};
+
/*------------------------- AC97 PART ---------------------------*/
static struct snd_soc_dai_link db1200_ac97_dai = {
@@ -89,36 +100,47 @@
/*------------------------- COMMON PART ---------------------------*/
-static struct platform_device *db1200_asoc_dev;
+static struct snd_soc_card *db1200_cards[] __devinitdata = {
+ &db1200_ac97_machine,
+ &db1200_i2s_machine,
+};
+
+static int __devinit db1200_audio_probe(struct platform_device *pdev)
+{
+ const struct platform_device_id *pid = platform_get_device_id(pdev);
+ struct snd_soc_card *card;
+
+ card = db1200_cards[pid->driver_data];
+ card->dev = &pdev->dev;
+ return snd_soc_register_card(card);
+}
+
+static int __devexit db1200_audio_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+ snd_soc_unregister_card(card);
+ return 0;
+}
+
+static struct platform_driver db1200_audio_driver = {
+ .driver = {
+ .name = "db1200-ac97",
+ .owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
+ },
+ .id_table = db1200_pids,
+ .probe = db1200_audio_probe,
+ .remove = __devexit_p(db1200_audio_remove),
+};
static int __init db1200_audio_load(void)
{
- int ret;
-
- ret = -ENOMEM;
- db1200_asoc_dev = platform_device_alloc("soc-audio", 1); /* PSC1 */
- if (!db1200_asoc_dev)
- goto out;
-
- /* DB1200 board setup set PSC1MUX to preferred audio device */
- if (bcsr_read(BCSR_RESETS) & BCSR_RESETS_PSC1MUX)
- platform_set_drvdata(db1200_asoc_dev, &db1200_i2s_machine);
- else
- platform_set_drvdata(db1200_asoc_dev, &db1200_ac97_machine);
-
- ret = platform_device_add(db1200_asoc_dev);
-
- if (ret) {
- platform_device_put(db1200_asoc_dev);
- db1200_asoc_dev = NULL;
- }
-out:
- return ret;
+ return platform_driver_register(&db1200_audio_driver);
}
static void __exit db1200_audio_unload(void)
{
- platform_device_unregister(db1200_asoc_dev);
+ platform_driver_unregister(&db1200_audio_driver);
}
module_init(db1200_audio_load);
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 20bb53a..d7d04e2 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -169,7 +169,7 @@
au1x_pcm_dbdma_free(pcd);
- if (stype == PCM_RX)
+ if (stype == SNDRV_PCM_STREAM_CAPTURE)
pcd->ddma_chan = au1xxx_dbdma_chan_alloc(pcd->ddma_id,
DSCR_CMD0_ALWAYS,
au1x_pcm_dmarx_cb, (void *)pcd);
@@ -198,7 +198,7 @@
struct snd_soc_pcm_runtime *rtd = ss->private_data;
struct au1xpsc_audio_dmadata *pcd =
snd_soc_platform_get_drvdata(rtd->platform);
- return &pcd[SUBSTREAM_TYPE(ss)];
+ return &pcd[ss->stream];
}
static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream,
@@ -212,7 +212,7 @@
if (ret < 0)
goto out;
- stype = SUBSTREAM_TYPE(substream);
+ stype = substream->stream;
pcd = to_dmadata(substream);
DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d "
@@ -255,7 +255,7 @@
au1xxx_dbdma_reset(pcd->ddma_chan);
- if (SUBSTREAM_TYPE(substream) == PCM_RX) {
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
au1x_pcm_queue_rx(pcd);
au1x_pcm_queue_rx(pcd);
} else {
@@ -293,6 +293,16 @@
static int au1xpsc_pcm_open(struct snd_pcm_substream *substream)
{
+ struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream);
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ int stype = substream->stream, *dmaids;
+
+ dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ if (!dmaids)
+ return -ENODEV; /* whoa, has ordering changed? */
+
+ pcd->ddma_id = dmaids[stype];
+
snd_soc_set_runtime_hwparams(substream, &au1xpsc_pcm_hardware);
return 0;
}
@@ -340,36 +350,18 @@
static int __devinit au1xpsc_pcm_drvprobe(struct platform_device *pdev)
{
struct au1xpsc_audio_dmadata *dmadata;
- struct resource *r;
int ret;
dmadata = kzalloc(2 * sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL);
if (!dmadata)
return -ENOMEM;
- r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
- if (!r) {
- ret = -ENODEV;
- goto out1;
- }
- dmadata[PCM_TX].ddma_id = r->start;
-
- /* RX DMA */
- r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
- if (!r) {
- ret = -ENODEV;
- goto out1;
- }
- dmadata[PCM_RX].ddma_id = r->start;
-
platform_set_drvdata(pdev, dmadata);
ret = snd_soc_register_platform(&pdev->dev, &au1xpsc_soc_platform);
- if (!ret)
- return ret;
+ if (ret)
+ kfree(dmadata);
-out1:
- kfree(dmadata);
return ret;
}
@@ -405,57 +397,6 @@
module_init(au1xpsc_audio_dbdma_load);
module_exit(au1xpsc_audio_dbdma_unload);
-
-struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev)
-{
- struct resource *res, *r;
- struct platform_device *pd;
- int id[2];
- int ret;
-
- r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
- if (!r)
- return NULL;
- id[0] = r->start;
-
- r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
- if (!r)
- return NULL;
- id[1] = r->start;
-
- res = kzalloc(sizeof(struct resource) * 2, GFP_KERNEL);
- if (!res)
- return NULL;
-
- res[0].start = res[0].end = id[0];
- res[1].start = res[1].end = id[1];
- res[0].flags = res[1].flags = IORESOURCE_DMA;
-
- pd = platform_device_alloc("au1xpsc-pcm", pdev->id);
- if (!pd)
- goto out;
-
- pd->resource = res;
- pd->num_resources = 2;
-
- ret = platform_device_add(pd);
- if (!ret)
- return pd;
-
- platform_device_put(pd);
-out:
- kfree(res);
- return NULL;
-}
-EXPORT_SYMBOL_GPL(au1xpsc_pcm_add);
-
-void au1xpsc_pcm_destroy(struct platform_device *dmapd)
-{
- if (dmapd)
- platform_device_unregister(dmapd);
-}
-EXPORT_SYMBOL_GPL(au1xpsc_pcm_destroy);
-
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Au12x0/Au1550 PSC Audio DMA driver");
MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c
new file mode 100644
index 0000000..7aa5b76
--- /dev/null
+++ b/sound/soc/au1x/dma.c
@@ -0,0 +1,377 @@
+/*
+ * Au1000/Au1500/Au1100 Audio DMA support.
+ *
+ * (c) 2011 Manuel Lauss <manuel.lauss@googlemail.com>
+ *
+ * copied almost verbatim from the old ALSA driver, written by
+ * Charles Eidsness <charles@cooper-street.com>
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <asm/mach-au1x00/au1000.h>
+#include <asm/mach-au1x00/au1000_dma.h>
+
+#include "psc.h"
+
+#define ALCHEMY_PCM_FMTS \
+ (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \
+ SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE | \
+ SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE | \
+ SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE | \
+ 0)
+
+struct pcm_period {
+ u32 start;
+ u32 relative_end; /* relative to start of buffer */
+ struct pcm_period *next;
+};
+
+struct audio_stream {
+ struct snd_pcm_substream *substream;
+ int dma;
+ struct pcm_period *buffer;
+ unsigned int period_size;
+ unsigned int periods;
+};
+
+struct alchemy_pcm_ctx {
+ struct audio_stream stream[2]; /* playback & capture */
+};
+
+static void au1000_release_dma_link(struct audio_stream *stream)
+{
+ struct pcm_period *pointer;
+ struct pcm_period *pointer_next;
+
+ stream->period_size = 0;
+ stream->periods = 0;
+ pointer = stream->buffer;
+ if (!pointer)
+ return;
+ do {
+ pointer_next = pointer->next;
+ kfree(pointer);
+ pointer = pointer_next;
+ } while (pointer != stream->buffer);
+ stream->buffer = NULL;
+}
+
+static int au1000_setup_dma_link(struct audio_stream *stream,
+ unsigned int period_bytes,
+ unsigned int periods)
+{
+ struct snd_pcm_substream *substream = stream->substream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pcm_period *pointer;
+ unsigned long dma_start;
+ int i;
+
+ dma_start = virt_to_phys(runtime->dma_area);
+
+ if (stream->period_size == period_bytes &&
+ stream->periods == periods)
+ return 0; /* not changed */
+
+ au1000_release_dma_link(stream);
+
+ stream->period_size = period_bytes;
+ stream->periods = periods;
+
+ stream->buffer = kmalloc(sizeof(struct pcm_period), GFP_KERNEL);
+ if (!stream->buffer)
+ return -ENOMEM;
+ pointer = stream->buffer;
+ for (i = 0; i < periods; i++) {
+ pointer->start = (u32)(dma_start + (i * period_bytes));
+ pointer->relative_end = (u32) (((i+1) * period_bytes) - 0x1);
+ if (i < periods - 1) {
+ pointer->next = kmalloc(sizeof(struct pcm_period),
+ GFP_KERNEL);
+ if (!pointer->next) {
+ au1000_release_dma_link(stream);
+ return -ENOMEM;
+ }
+ pointer = pointer->next;
+ }
+ }
+ pointer->next = stream->buffer;
+ return 0;
+}
+
+static void au1000_dma_stop(struct audio_stream *stream)
+{
+ if (stream->buffer)
+ disable_dma(stream->dma);
+}
+
+static void au1000_dma_start(struct audio_stream *stream)
+{
+ if (!stream->buffer)
+ return;
+
+ init_dma(stream->dma);
+ if (get_dma_active_buffer(stream->dma) == 0) {
+ clear_dma_done0(stream->dma);
+ set_dma_addr0(stream->dma, stream->buffer->start);
+ set_dma_count0(stream->dma, stream->period_size >> 1);
+ set_dma_addr1(stream->dma, stream->buffer->next->start);
+ set_dma_count1(stream->dma, stream->period_size >> 1);
+ } else {
+ clear_dma_done1(stream->dma);
+ set_dma_addr1(stream->dma, stream->buffer->start);
+ set_dma_count1(stream->dma, stream->period_size >> 1);
+ set_dma_addr0(stream->dma, stream->buffer->next->start);
+ set_dma_count0(stream->dma, stream->period_size >> 1);
+ }
+ enable_dma_buffers(stream->dma);
+ start_dma(stream->dma);
+}
+
+static irqreturn_t au1000_dma_interrupt(int irq, void *ptr)
+{
+ struct audio_stream *stream = (struct audio_stream *)ptr;
+ struct snd_pcm_substream *substream = stream->substream;
+
+ switch (get_dma_buffer_done(stream->dma)) {
+ case DMA_D0:
+ stream->buffer = stream->buffer->next;
+ clear_dma_done0(stream->dma);
+ set_dma_addr0(stream->dma, stream->buffer->next->start);
+ set_dma_count0(stream->dma, stream->period_size >> 1);
+ enable_dma_buffer0(stream->dma);
+ break;
+ case DMA_D1:
+ stream->buffer = stream->buffer->next;
+ clear_dma_done1(stream->dma);
+ set_dma_addr1(stream->dma, stream->buffer->next->start);
+ set_dma_count1(stream->dma, stream->period_size >> 1);
+ enable_dma_buffer1(stream->dma);
+ break;
+ case (DMA_D0 | DMA_D1):
+ pr_debug("DMA %d missed interrupt.\n", stream->dma);
+ au1000_dma_stop(stream);
+ au1000_dma_start(stream);
+ break;
+ case (~DMA_D0 & ~DMA_D1):
+ pr_debug("DMA %d empty irq.\n", stream->dma);
+ }
+ snd_pcm_period_elapsed(substream);
+ return IRQ_HANDLED;
+}
+
+static const struct snd_pcm_hardware alchemy_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH,
+ .formats = ALCHEMY_PCM_FMTS,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .rate_min = SNDRV_PCM_RATE_8000,
+ .rate_max = SNDRV_PCM_RATE_192000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .period_bytes_min = 1024,
+ .period_bytes_max = 16 * 1024 - 1,
+ .periods_min = 4,
+ .periods_max = 255,
+ .buffer_bytes_max = 128 * 1024,
+ .fifo_size = 16,
+};
+
+static inline struct alchemy_pcm_ctx *ss_to_ctx(struct snd_pcm_substream *ss)
+{
+ struct snd_soc_pcm_runtime *rtd = ss->private_data;
+ return snd_soc_platform_get_drvdata(rtd->platform);
+}
+
+static inline struct audio_stream *ss_to_as(struct snd_pcm_substream *ss)
+{
+ struct alchemy_pcm_ctx *ctx = ss_to_ctx(ss);
+ return &(ctx->stream[ss->stream]);
+}
+
+static int alchemy_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct alchemy_pcm_ctx *ctx = ss_to_ctx(substream);
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ int *dmaids, s = substream->stream;
+ char *name;
+
+ dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ if (!dmaids)
+ return -ENODEV; /* whoa, has ordering changed? */
+
+ /* DMA setup */
+ name = (s == SNDRV_PCM_STREAM_PLAYBACK) ? "audio-tx" : "audio-rx";
+ ctx->stream[s].dma = request_au1000_dma(dmaids[s], name,
+ au1000_dma_interrupt, IRQF_DISABLED,
+ &ctx->stream[s]);
+ set_dma_mode(ctx->stream[s].dma,
+ get_dma_mode(ctx->stream[s].dma) & ~DMA_NC);
+
+ ctx->stream[s].substream = substream;
+ ctx->stream[s].buffer = NULL;
+ snd_soc_set_runtime_hwparams(substream, &alchemy_pcm_hardware);
+
+ return 0;
+}
+
+static int alchemy_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct alchemy_pcm_ctx *ctx = ss_to_ctx(substream);
+ int stype = substream->stream;
+
+ ctx->stream[stype].substream = NULL;
+ free_au1000_dma(ctx->stream[stype].dma);
+
+ return 0;
+}
+
+static int alchemy_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct audio_stream *stream = ss_to_as(substream);
+ int err;
+
+ err = snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+ err = au1000_setup_dma_link(stream,
+ params_period_bytes(hw_params),
+ params_periods(hw_params));
+ if (err)
+ snd_pcm_lib_free_pages(substream);
+
+ return err;
+}
+
+static int alchemy_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct audio_stream *stream = ss_to_as(substream);
+ au1000_release_dma_link(stream);
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int alchemy_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct audio_stream *stream = ss_to_as(substream);
+ int err = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ au1000_dma_start(stream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ au1000_dma_stop(stream);
+ break;
+ default:
+ err = -EINVAL;
+ break;
+ }
+ return err;
+}
+
+static snd_pcm_uframes_t alchemy_pcm_pointer(struct snd_pcm_substream *ss)
+{
+ struct audio_stream *stream = ss_to_as(ss);
+ long location;
+
+ location = get_dma_residue(stream->dma);
+ location = stream->buffer->relative_end - location;
+ if (location == -1)
+ location = 0;
+ return bytes_to_frames(ss->runtime, location);
+}
+
+static struct snd_pcm_ops alchemy_pcm_ops = {
+ .open = alchemy_pcm_open,
+ .close = alchemy_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = alchemy_pcm_hw_params,
+ .hw_free = alchemy_pcm_hw_free,
+ .trigger = alchemy_pcm_trigger,
+ .pointer = alchemy_pcm_pointer,
+};
+
+static void alchemy_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+ snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static int alchemy_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_pcm *pcm = rtd->pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data(GFP_KERNEL), 65536, (4096 * 1024) - 1);
+
+ return 0;
+}
+
+struct snd_soc_platform_driver alchemy_pcm_soc_platform = {
+ .ops = &alchemy_pcm_ops,
+ .pcm_new = alchemy_pcm_new,
+ .pcm_free = alchemy_pcm_free_dma_buffers,
+};
+
+static int __devinit alchemy_pcm_drvprobe(struct platform_device *pdev)
+{
+ struct alchemy_pcm_ctx *ctx;
+ int ret;
+
+ ctx = kzalloc(sizeof(*ctx), GFP_KERNEL);
+ if (!ctx)
+ return -ENOMEM;
+
+ platform_set_drvdata(pdev, ctx);
+
+ ret = snd_soc_register_platform(&pdev->dev, &alchemy_pcm_soc_platform);
+ if (ret)
+ kfree(ctx);
+
+ return ret;
+}
+
+static int __devexit alchemy_pcm_drvremove(struct platform_device *pdev)
+{
+ struct alchemy_pcm_ctx *ctx = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_platform(&pdev->dev);
+ kfree(ctx);
+
+ return 0;
+}
+
+static struct platform_driver alchemy_pcmdma_driver = {
+ .driver = {
+ .name = "alchemy-pcm-dma",
+ .owner = THIS_MODULE,
+ },
+ .probe = alchemy_pcm_drvprobe,
+ .remove = __devexit_p(alchemy_pcm_drvremove),
+};
+
+static int __init alchemy_pcmdma_load(void)
+{
+ return platform_driver_register(&alchemy_pcmdma_driver);
+}
+
+static void __exit alchemy_pcmdma_unload(void)
+{
+ platform_driver_unregister(&alchemy_pcmdma_driver);
+}
+
+module_init(alchemy_pcmdma_load);
+module_exit(alchemy_pcmdma_unload);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au1000/Au1500/Au1100 Audio DMA driver");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c
new file mode 100644
index 0000000..19e0d2a
--- /dev/null
+++ b/sound/soc/au1x/i2sc.c
@@ -0,0 +1,346 @@
+/*
+ * Au1000/Au1500/Au1100 I2S controller driver for ASoC
+ *
+ * (c) 2011 Manuel Lauss <manuel.lauss@googlemail.com>
+ *
+ * Note: clock supplied to the I2S controller must be 256x samplerate.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/suspend.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <asm/mach-au1x00/au1000.h>
+
+#include "psc.h"
+
+#define I2S_RXTX 0x00
+#define I2S_CFG 0x04
+#define I2S_ENABLE 0x08
+
+#define CFG_XU (1 << 25) /* tx underflow */
+#define CFG_XO (1 << 24)
+#define CFG_RU (1 << 23)
+#define CFG_RO (1 << 22)
+#define CFG_TR (1 << 21)
+#define CFG_TE (1 << 20)
+#define CFG_TF (1 << 19)
+#define CFG_RR (1 << 18)
+#define CFG_RF (1 << 17)
+#define CFG_ICK (1 << 12) /* clock invert */
+#define CFG_PD (1 << 11) /* set to make I2SDIO INPUT */
+#define CFG_LB (1 << 10) /* loopback */
+#define CFG_IC (1 << 9) /* word select invert */
+#define CFG_FM_I2S (0 << 7) /* I2S format */
+#define CFG_FM_LJ (1 << 7) /* left-justified */
+#define CFG_FM_RJ (2 << 7) /* right-justified */
+#define CFG_FM_MASK (3 << 7)
+#define CFG_TN (1 << 6) /* tx fifo en */
+#define CFG_RN (1 << 5) /* rx fifo en */
+#define CFG_SZ_8 (0x08)
+#define CFG_SZ_16 (0x10)
+#define CFG_SZ_18 (0x12)
+#define CFG_SZ_20 (0x14)
+#define CFG_SZ_24 (0x18)
+#define CFG_SZ_MASK (0x1f)
+#define EN_D (1 << 1) /* DISable */
+#define EN_CE (1 << 0) /* clock enable */
+
+/* only limited by clock generator and board design */
+#define AU1XI2SC_RATES \
+ SNDRV_PCM_RATE_CONTINUOUS
+
+#define AU1XI2SC_FMTS \
+ (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \
+ SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE | \
+ SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_U18_3LE | \
+ SNDRV_PCM_FMTBIT_S18_3BE | SNDRV_PCM_FMTBIT_U18_3BE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \
+ SNDRV_PCM_FMTBIT_S20_3BE | SNDRV_PCM_FMTBIT_U20_3BE | \
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | \
+ SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_U24_BE | \
+ 0)
+
+static inline unsigned long RD(struct au1xpsc_audio_data *ctx, int reg)
+{
+ return __raw_readl(ctx->mmio + reg);
+}
+
+static inline void WR(struct au1xpsc_audio_data *ctx, int reg, unsigned long v)
+{
+ __raw_writel(v, ctx->mmio + reg);
+ wmb();
+}
+
+static int au1xi2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+ struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(cpu_dai);
+ unsigned long c;
+ int ret;
+
+ ret = -EINVAL;
+ c = ctx->cfg;
+
+ c &= ~CFG_FM_MASK;
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ c |= CFG_FM_I2S;
+ break;
+ case SND_SOC_DAIFMT_MSB:
+ c |= CFG_FM_RJ;
+ break;
+ case SND_SOC_DAIFMT_LSB:
+ c |= CFG_FM_LJ;
+ break;
+ default:
+ goto out;
+ }
+
+ c &= ~(CFG_IC | CFG_ICK); /* IB-IF */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ c |= CFG_IC | CFG_ICK;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ c |= CFG_IC;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ c |= CFG_ICK;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ break;
+ default:
+ goto out;
+ }
+
+ /* I2S controller only supports master */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS: /* CODEC slave */
+ break;
+ default:
+ goto out;
+ }
+
+ ret = 0;
+ ctx->cfg = c;
+out:
+ return ret;
+}
+
+static int au1xi2s_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *dai)
+{
+ struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai);
+ int stype = SUBSTREAM_TYPE(substream);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ /* power up */
+ WR(ctx, I2S_ENABLE, EN_D | EN_CE);
+ WR(ctx, I2S_ENABLE, EN_CE);
+ ctx->cfg |= (stype == PCM_TX) ? CFG_TN : CFG_RN;
+ WR(ctx, I2S_CFG, ctx->cfg);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ ctx->cfg &= ~((stype == PCM_TX) ? CFG_TN : CFG_RN);
+ WR(ctx, I2S_CFG, ctx->cfg);
+ WR(ctx, I2S_ENABLE, EN_D); /* power off */
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static unsigned long msbits_to_reg(int msbits)
+{
+ switch (msbits) {
+ case 8:
+ return CFG_SZ_8;
+ case 16:
+ return CFG_SZ_16;
+ case 18:
+ return CFG_SZ_18;
+ case 20:
+ return CFG_SZ_20;
+ case 24:
+ return CFG_SZ_24;
+ }
+ return 0;
+}
+
+static int au1xi2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai);
+ unsigned long v;
+
+ v = msbits_to_reg(params->msbits);
+ if (!v)
+ return -EINVAL;
+
+ ctx->cfg &= ~CFG_SZ_MASK;
+ ctx->cfg |= v;
+ return 0;
+}
+
+static int au1xi2s_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai);
+ snd_soc_dai_set_dma_data(dai, substream, &ctx->dmaids[0]);
+ return 0;
+}
+
+static const struct snd_soc_dai_ops au1xi2s_dai_ops = {
+ .startup = au1xi2s_startup,
+ .trigger = au1xi2s_trigger,
+ .hw_params = au1xi2s_hw_params,
+ .set_fmt = au1xi2s_set_fmt,
+};
+
+static struct snd_soc_dai_driver au1xi2s_dai_driver = {
+ .symmetric_rates = 1,
+ .playback = {
+ .rates = AU1XI2SC_RATES,
+ .formats = AU1XI2SC_FMTS,
+ .channels_min = 2,
+ .channels_max = 2,
+ },
+ .capture = {
+ .rates = AU1XI2SC_RATES,
+ .formats = AU1XI2SC_FMTS,
+ .channels_min = 2,
+ .channels_max = 2,
+ },
+ .ops = &au1xi2s_dai_ops,
+};
+
+static int __devinit au1xi2s_drvprobe(struct platform_device *pdev)
+{
+ int ret;
+ struct resource *r;
+ struct au1xpsc_audio_data *ctx;
+
+ ctx = kzalloc(sizeof(*ctx), GFP_KERNEL);
+ if (!ctx)
+ return -ENOMEM;
+
+ r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!r) {
+ ret = -ENODEV;
+ goto out0;
+ }
+
+ ret = -EBUSY;
+ if (!request_mem_region(r->start, resource_size(r), pdev->name))
+ goto out0;
+
+ ctx->mmio = ioremap_nocache(r->start, resource_size(r));
+ if (!ctx->mmio)
+ goto out1;
+
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+ if (!r)
+ goto out1;
+ ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start;
+
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+ if (!r)
+ goto out1;
+ ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start;
+
+ platform_set_drvdata(pdev, ctx);
+
+ ret = snd_soc_register_dai(&pdev->dev, &au1xi2s_dai_driver);
+ if (ret)
+ goto out1;
+
+ return 0;
+
+out1:
+ release_mem_region(r->start, resource_size(r));
+out0:
+ kfree(ctx);
+ return ret;
+}
+
+static int __devexit au1xi2s_drvremove(struct platform_device *pdev)
+{
+ struct au1xpsc_audio_data *ctx = platform_get_drvdata(pdev);
+ struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+
+ snd_soc_unregister_dai(&pdev->dev);
+
+ WR(ctx, I2S_ENABLE, EN_D); /* clock off, disable */
+
+ iounmap(ctx->mmio);
+ release_mem_region(r->start, resource_size(r));
+ kfree(ctx);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int au1xi2s_drvsuspend(struct device *dev)
+{
+ struct au1xpsc_audio_data *ctx = dev_get_drvdata(dev);
+
+ WR(ctx, I2S_ENABLE, EN_D); /* clock off, disable */
+
+ return 0;
+}
+
+static int au1xi2s_drvresume(struct device *dev)
+{
+ return 0;
+}
+
+static const struct dev_pm_ops au1xi2sc_pmops = {
+ .suspend = au1xi2s_drvsuspend,
+ .resume = au1xi2s_drvresume,
+};
+
+#define AU1XI2SC_PMOPS (&au1xi2sc_pmops)
+
+#else
+
+#define AU1XI2SC_PMOPS NULL
+
+#endif
+
+static struct platform_driver au1xi2s_driver = {
+ .driver = {
+ .name = "alchemy-i2sc",
+ .owner = THIS_MODULE,
+ .pm = AU1XI2SC_PMOPS,
+ },
+ .probe = au1xi2s_drvprobe,
+ .remove = __devexit_p(au1xi2s_drvremove),
+};
+
+static int __init au1xi2s_load(void)
+{
+ return platform_driver_register(&au1xi2s_driver);
+}
+
+static void __exit au1xi2s_unload(void)
+{
+ platform_driver_unregister(&au1xi2s_driver);
+}
+
+module_init(au1xi2s_load);
+module_exit(au1xi2s_unload);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au1000/1500/1100 I2S ASoC driver");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index d0db66f..172eefd 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -41,14 +41,14 @@
(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3BE)
#define AC97PCR_START(stype) \
- ((stype) == PCM_TX ? PSC_AC97PCR_TS : PSC_AC97PCR_RS)
+ ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TS : PSC_AC97PCR_RS)
#define AC97PCR_STOP(stype) \
- ((stype) == PCM_TX ? PSC_AC97PCR_TP : PSC_AC97PCR_RP)
+ ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TP : PSC_AC97PCR_RP)
#define AC97PCR_CLRFIFO(stype) \
- ((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC)
+ ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TC : PSC_AC97PCR_RC)
#define AC97STAT_BUSY(stype) \
- ((stype) == PCM_TX ? PSC_AC97STAT_TB : PSC_AC97STAT_RB)
+ ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97STAT_TB : PSC_AC97STAT_RB)
/* instance data. There can be only one, MacLeod!!!! */
static struct au1xpsc_audio_data *au1xpsc_ac97_workdata;
@@ -215,7 +215,7 @@
{
struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
unsigned long r, ro, stat;
- int chans, t, stype = SUBSTREAM_TYPE(substream);
+ int chans, t, stype = substream->stream;
chans = params_channels(params);
@@ -235,7 +235,7 @@
r |= PSC_AC97CFG_SET_LEN(params->msbits);
/* channels: enable slots for front L/R channel */
- if (stype == PCM_TX) {
+ if (stype == SNDRV_PCM_STREAM_PLAYBACK) {
r &= ~PSC_AC97CFG_TXSLOT_MASK;
r |= PSC_AC97CFG_TXSLOT_ENA(3);
r |= PSC_AC97CFG_TXSLOT_ENA(4);
@@ -294,7 +294,7 @@
int cmd, struct snd_soc_dai *dai)
{
struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
- int ret, stype = SUBSTREAM_TYPE(substream);
+ int ret, stype = substream->stream;
ret = 0;
@@ -324,12 +324,21 @@
return ret;
}
+static int au1xpsc_ac97_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
+ snd_soc_dai_set_dma_data(dai, substream, &pscdata->dmaids[0]);
+ return 0;
+}
+
static int au1xpsc_ac97_probe(struct snd_soc_dai *dai)
{
return au1xpsc_ac97_workdata ? 0 : -ENODEV;
}
static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = {
+ .startup = au1xpsc_ac97_startup,
.trigger = au1xpsc_ac97_trigger,
.hw_params = au1xpsc_ac97_hw_params,
};
@@ -379,6 +388,16 @@
if (!wd->mmio)
goto out1;
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+ if (!r)
+ goto out2;
+ wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start;
+
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+ if (!r)
+ goto out2;
+ wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start;
+
/* configuration: max dma trigger threshold, enable ac97 */
wd->cfg = PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8 |
PSC_AC97CFG_DE_ENABLE;
@@ -401,15 +420,13 @@
ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv);
if (ret)
- goto out1;
+ goto out2;
- wd->dmapd = au1xpsc_pcm_add(pdev);
- if (wd->dmapd) {
- au1xpsc_ac97_workdata = wd;
- return 0;
- }
+ au1xpsc_ac97_workdata = wd;
+ return 0;
- snd_soc_unregister_dai(&pdev->dev);
+out2:
+ iounmap(wd->mmio);
out1:
release_mem_region(r->start, resource_size(r));
out0:
@@ -422,9 +439,6 @@
struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev);
struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (wd->dmapd)
- au1xpsc_pcm_destroy(wd->dmapd);
-
snd_soc_unregister_dai(&pdev->dev);
/* disable PSC completely */
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index fca0912..7c5ae92 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -42,13 +42,13 @@
(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE)
#define I2SSTAT_BUSY(stype) \
- ((stype) == PCM_TX ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB)
+ ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB)
#define I2SPCR_START(stype) \
- ((stype) == PCM_TX ? PSC_I2SPCR_TS : PSC_I2SPCR_RS)
+ ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TS : PSC_I2SPCR_RS)
#define I2SPCR_STOP(stype) \
- ((stype) == PCM_TX ? PSC_I2SPCR_TP : PSC_I2SPCR_RP)
+ ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TP : PSC_I2SPCR_RP)
#define I2SPCR_CLRFIFO(stype) \
- ((stype) == PCM_TX ? PSC_I2SPCR_TC : PSC_I2SPCR_RC)
+ ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TC : PSC_I2SPCR_RC)
static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
@@ -240,7 +240,7 @@
struct snd_soc_dai *dai)
{
struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
- int ret, stype = SUBSTREAM_TYPE(substream);
+ int ret, stype = substream->stream;
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
@@ -257,7 +257,16 @@
return ret;
}
+static int au1xpsc_i2s_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
+ snd_soc_dai_set_dma_data(dai, substream, &pscdata->dmaids[0]);
+ return 0;
+}
+
static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = {
+ .startup = au1xpsc_i2s_startup,
.trigger = au1xpsc_i2s_trigger,
.hw_params = au1xpsc_i2s_hw_params,
.set_fmt = au1xpsc_i2s_set_fmt,
@@ -304,6 +313,16 @@
if (!wd->mmio)
goto out1;
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+ if (!r)
+ goto out2;
+ wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start;
+
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+ if (!r)
+ goto out2;
+ wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start;
+
/* preserve PSC clock source set up by platform (dev.platform_data
* is already occupied by soc layer)
*/
@@ -330,15 +349,11 @@
platform_set_drvdata(pdev, wd);
ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv);
- if (ret)
- goto out1;
-
- /* finally add the DMA device for this PSC */
- wd->dmapd = au1xpsc_pcm_add(pdev);
- if (wd->dmapd)
+ if (!ret)
return 0;
- snd_soc_unregister_dai(&pdev->dev);
+out2:
+ iounmap(wd->mmio);
out1:
release_mem_region(r->start, resource_size(r));
out0:
@@ -351,9 +366,6 @@
struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev);
struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (wd->dmapd)
- au1xpsc_pcm_destroy(wd->dmapd);
-
snd_soc_unregister_dai(&pdev->dev);
au_writel(0, I2S_CFG(wd));
diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h
index b30eadd..b16b2e0 100644
--- a/sound/soc/au1x/psc.h
+++ b/sound/soc/au1x/psc.h
@@ -1,7 +1,7 @@
/*
- * Au12x0/Au1550 PSC ALSA ASoC audio support.
+ * Alchemy ALSA ASoC audio support.
*
- * (c) 2007-2008 MSC Vertriebsges.m.b.H.,
+ * (c) 2007-2011 MSC Vertriebsges.m.b.H.,
* Manuel Lauss <manuel.lauss@gmail.com>
*
* This program is free software; you can redistribute it and/or modify
@@ -13,10 +13,6 @@
#ifndef _AU1X_PCM_H
#define _AU1X_PCM_H
-/* DBDMA helpers */
-extern struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev);
-extern void au1xpsc_pcm_destroy(struct platform_device *dmapd);
-
struct au1xpsc_audio_data {
void __iomem *mmio;
@@ -27,15 +23,9 @@
unsigned long pm[2];
struct mutex lock;
- struct platform_device *dmapd;
+ int dmaids[2];
};
-#define PCM_TX 0
-#define PCM_RX 1
-
-#define SUBSTREAM_TYPE(substream) \
- ((substream)->stream == SNDRV_PCM_STREAM_PLAYBACK ? PCM_TX : PCM_RX)
-
/* easy access macros */
#define PSC_CTRL(x) ((unsigned long)((x)->mmio) + PSC_CTRL_OFFSET)
#define PSC_SEL(x) ((unsigned long)((x)->mmio) + PSC_SEL_OFFSET)
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index fe9d548..9f6bc55 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -27,6 +27,19 @@
board connected to one of the Blackfin evaluation boards like the
BF5XX-STAMP or BF5XX-EZKIT.
+config SND_SOC_BFIN_EVAL_ADAU1373
+ tristate "Support for the EVAL-ADAU1373 board on Blackfin eval boards"
+ depends on SND_BF5XX_I2S && I2C
+ select SND_BF5XX_SOC_I2S
+ select SND_SOC_ADAU1373
+ help
+ Say Y if you want to add support for the Analog Devices EVAL-ADAU1373
+ board connected to one of the Blackfin evaluation boards like the
+ BF5XX-STAMP or BF5XX-EZKIT.
+
+ Note: This driver assumes that first ADAU1373 DAI is connected to the
+ first SPORT port on the BF5XX board.
+
config SND_SOC_BFIN_EVAL_ADAV80X
tristate "Support for the EVAL-ADAV80X boards on Blackfin eval boards"
depends on SND_BF5XX_I2S && (SPI_MASTER || I2C)
diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile
index 6018bf5..1bf86cc 100644
--- a/sound/soc/blackfin/Makefile
+++ b/sound/soc/blackfin/Makefile
@@ -21,6 +21,7 @@
snd-ssm2602-objs := bf5xx-ssm2602.o
snd-ad73311-objs := bf5xx-ad73311.o
snd-ad193x-objs := bf5xx-ad193x.o
+snd-soc-bfin-eval-adau1373-objs := bfin-eval-adau1373.o
snd-soc-bfin-eval-adau1701-objs := bfin-eval-adau1701.o
snd-soc-bfin-eval-adav80x-objs := bfin-eval-adav80x.o
@@ -29,5 +30,6 @@
obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o
obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o
obj-$(CONFIG_SND_BF5XX_SOC_AD193X) += snd-ad193x.o
+obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAU1373) += snd-soc-bfin-eval-adau1373.o
obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAU1701) += snd-soc-bfin-eval-adau1701.o
obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAV80X) += snd-soc-bfin-eval-adav80x.o
diff --git a/sound/soc/blackfin/bfin-eval-adau1373.c b/sound/soc/blackfin/bfin-eval-adau1373.c
new file mode 100644
index 0000000..8df2a3b
--- /dev/null
+++ b/sound/soc/blackfin/bfin-eval-adau1373.c
@@ -0,0 +1,202 @@
+/*
+ * Machine driver for EVAL-ADAU1373 on Analog Devices bfin
+ * evaluation boards.
+ *
+ * Copyright 2011 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+
+#include "../codecs/adau1373.h"
+
+static const struct snd_soc_dapm_widget bfin_eval_adau1373_dapm_widgets[] = {
+ SND_SOC_DAPM_LINE("Line In1", NULL),
+ SND_SOC_DAPM_LINE("Line In2", NULL),
+ SND_SOC_DAPM_LINE("Line In3", NULL),
+ SND_SOC_DAPM_LINE("Line In4", NULL),
+
+ SND_SOC_DAPM_LINE("Line Out1", NULL),
+ SND_SOC_DAPM_LINE("Line Out2", NULL),
+ SND_SOC_DAPM_LINE("Stereo Out", NULL),
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_HP("Earpiece", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+static const struct snd_soc_dapm_route bfin_eval_adau1373_dapm_routes[] = {
+ { "AIN1L", NULL, "Line In1" },
+ { "AIN1R", NULL, "Line In1" },
+ { "AIN2L", NULL, "Line In2" },
+ { "AIN2R", NULL, "Line In2" },
+ { "AIN3L", NULL, "Line In3" },
+ { "AIN3R", NULL, "Line In3" },
+ { "AIN4L", NULL, "Line In4" },
+ { "AIN4R", NULL, "Line In4" },
+
+ /* MICBIAS can be connected via a jumper to the line-in jack, since w
+ don't know which one is going to be used, just power both. */
+ { "Line In1", NULL, "MICBIAS1" },
+ { "Line In2", NULL, "MICBIAS1" },
+ { "Line In3", NULL, "MICBIAS1" },
+ { "Line In4", NULL, "MICBIAS1" },
+ { "Line In1", NULL, "MICBIAS2" },
+ { "Line In2", NULL, "MICBIAS2" },
+ { "Line In3", NULL, "MICBIAS2" },
+ { "Line In4", NULL, "MICBIAS2" },
+
+ { "Line Out1", NULL, "LOUT1L" },
+ { "Line Out1", NULL, "LOUT1R" },
+ { "Line Out2", NULL, "LOUT2L" },
+ { "Line Out2", NULL, "LOUT2R" },
+ { "Headphone", NULL, "HPL" },
+ { "Headphone", NULL, "HPR" },
+ { "Earpiece", NULL, "EP" },
+ { "Speaker", NULL, "SPKL" },
+ { "Stereo Out", NULL, "SPKR" },
+};
+
+static int bfin_eval_adau1373_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 *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+ int pll_rate;
+
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret)
+ return ret;
+
+ switch (params_rate(params)) {
+ case 48000:
+ case 8000:
+ case 12000:
+ case 16000:
+ case 24000:
+ case 32000:
+ pll_rate = 48000 * 1024;
+ break;
+ case 44100:
+ case 7350:
+ case 11025:
+ case 14700:
+ case 22050:
+ case 29400:
+ pll_rate = 44100 * 1024;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ ret = snd_soc_dai_set_pll(codec_dai, ADAU1373_PLL1,
+ ADAU1373_PLL_SRC_MCLK1, 12288000, pll_rate);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1373_CLK_SRC_PLL1, pll_rate,
+ SND_SOC_CLOCK_IN);
+
+ return ret;
+}
+
+static int bfin_eval_adau1373_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ unsigned int pll_rate = 48000 * 1024;
+ int ret;
+
+ ret = snd_soc_dai_set_pll(codec_dai, ADAU1373_PLL1,
+ ADAU1373_PLL_SRC_MCLK1, 12288000, pll_rate);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1373_CLK_SRC_PLL1, pll_rate,
+ SND_SOC_CLOCK_IN);
+
+ return ret;
+}
+static struct snd_soc_ops bfin_eval_adau1373_ops = {
+ .hw_params = bfin_eval_adau1373_hw_params,
+};
+
+static struct snd_soc_dai_link bfin_eval_adau1373_dai = {
+ .name = "adau1373",
+ .stream_name = "adau1373",
+ .cpu_dai_name = "bfin-i2s.0",
+ .codec_dai_name = "adau1373-aif1",
+ .platform_name = "bfin-i2s-pcm-audio",
+ .codec_name = "adau1373.0-001a",
+ .ops = &bfin_eval_adau1373_ops,
+ .init = bfin_eval_adau1373_codec_init,
+};
+
+static struct snd_soc_card bfin_eval_adau1373 = {
+ .name = "bfin-eval-adau1373",
+ .dai_link = &bfin_eval_adau1373_dai,
+ .num_links = 1,
+
+ .dapm_widgets = bfin_eval_adau1373_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(bfin_eval_adau1373_dapm_widgets),
+ .dapm_routes = bfin_eval_adau1373_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(bfin_eval_adau1373_dapm_routes),
+};
+
+static int bfin_eval_adau1373_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &bfin_eval_adau1373;
+
+ card->dev = &pdev->dev;
+
+ return snd_soc_register_card(&bfin_eval_adau1373);
+}
+
+static int __devexit bfin_eval_adau1373_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
+
+ return 0;
+}
+
+static struct platform_driver bfin_eval_adau1373_driver = {
+ .driver = {
+ .name = "bfin-eval-adau1373",
+ .owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = bfin_eval_adau1373_probe,
+ .remove = __devexit_p(bfin_eval_adau1373_remove),
+};
+
+static int __init bfin_eval_adau1373_init(void)
+{
+ return platform_driver_register(&bfin_eval_adau1373_driver);
+}
+module_init(bfin_eval_adau1373_init);
+
+static void __exit bfin_eval_adau1373_exit(void)
+{
+ platform_driver_unregister(&bfin_eval_adau1373_driver);
+}
+module_exit(bfin_eval_adau1373_exit);
+
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_DESCRIPTION("ALSA SoC bfin adau1373 driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:bfin-eval-adau1373");
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 665d924..71b46c8 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -17,6 +17,7 @@
select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI
select SND_SOC_AD1980 if SND_SOC_AC97_BUS
select SND_SOC_AD73311
+ select SND_SOC_ADAU1373 if I2C
select SND_SOC_ADAV80X
select SND_SOC_ADS117X
select SND_SOC_AK4104 if SPI_MASTER
@@ -139,6 +140,9 @@
select SIGMA
tristate
+config SND_SOC_ADAU1373
+ tristate
+
config SND_SOC_ADAV80X
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 5119a7e..70c1769 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -5,6 +5,7 @@
snd-soc-ad1980-objs := ad1980.o
snd-soc-ad73311-objs := ad73311.o
snd-soc-adau1701-objs := adau1701.o
+snd-soc-adau1373-objs := adau1373.o
snd-soc-adav80x-objs := adav80x.o
snd-soc-ads117x-objs := ads117x.o
snd-soc-ak4104-objs := ak4104.o
@@ -100,6 +101,7 @@
obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o
obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o
obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
+obj-$(CONFIG_SND_SOC_ADAU1373) += snd-soc-adau1373.o
obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o
obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o
obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o
diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
new file mode 100644
index 0000000..2aa40c3
--- /dev/null
+++ b/sound/soc/codecs/adau1373.c
@@ -0,0 +1,1414 @@
+/*
+ * Analog Devices ADAU1373 Audio Codec drive
+ *
+ * Copyright 2011 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+#include <linux/gcd.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+#include <sound/soc.h>
+#include <sound/adau1373.h>
+
+#include "adau1373.h"
+
+struct adau1373_dai {
+ unsigned int clk_src;
+ unsigned int sysclk;
+ bool enable_src;
+ bool master;
+};
+
+struct adau1373 {
+ struct adau1373_dai dais[3];
+};
+
+#define ADAU1373_INPUT_MODE 0x00
+#define ADAU1373_AINL_CTRL(x) (0x01 + (x) * 2)
+#define ADAU1373_AINR_CTRL(x) (0x02 + (x) * 2)
+#define ADAU1373_LLINE_OUT(x) (0x9 + (x) * 2)
+#define ADAU1373_RLINE_OUT(x) (0xa + (x) * 2)
+#define ADAU1373_LSPK_OUT 0x0d
+#define ADAU1373_RSPK_OUT 0x0e
+#define ADAU1373_LHP_OUT 0x0f
+#define ADAU1373_RHP_OUT 0x10
+#define ADAU1373_ADC_GAIN 0x11
+#define ADAU1373_LADC_MIXER 0x12
+#define ADAU1373_RADC_MIXER 0x13
+#define ADAU1373_LLINE1_MIX 0x14
+#define ADAU1373_RLINE1_MIX 0x15
+#define ADAU1373_LLINE2_MIX 0x16
+#define ADAU1373_RLINE2_MIX 0x17
+#define ADAU1373_LSPK_MIX 0x18
+#define ADAU1373_RSPK_MIX 0x19
+#define ADAU1373_LHP_MIX 0x1a
+#define ADAU1373_RHP_MIX 0x1b
+#define ADAU1373_EP_MIX 0x1c
+#define ADAU1373_HP_CTRL 0x1d
+#define ADAU1373_HP_CTRL2 0x1e
+#define ADAU1373_LS_CTRL 0x1f
+#define ADAU1373_EP_CTRL 0x21
+#define ADAU1373_MICBIAS_CTRL1 0x22
+#define ADAU1373_MICBIAS_CTRL2 0x23
+#define ADAU1373_OUTPUT_CTRL 0x24
+#define ADAU1373_PWDN_CTRL1 0x25
+#define ADAU1373_PWDN_CTRL2 0x26
+#define ADAU1373_PWDN_CTRL3 0x27
+#define ADAU1373_DPLL_CTRL(x) (0x28 + (x) * 7)
+#define ADAU1373_PLL_CTRL1(x) (0x29 + (x) * 7)
+#define ADAU1373_PLL_CTRL2(x) (0x2a + (x) * 7)
+#define ADAU1373_PLL_CTRL3(x) (0x2b + (x) * 7)
+#define ADAU1373_PLL_CTRL4(x) (0x2c + (x) * 7)
+#define ADAU1373_PLL_CTRL5(x) (0x2d + (x) * 7)
+#define ADAU1373_PLL_CTRL6(x) (0x2e + (x) * 7)
+#define ADAU1373_PLL_CTRL7(x) (0x2f + (x) * 7)
+#define ADAU1373_HEADDECT 0x36
+#define ADAU1373_ADC_DAC_STATUS 0x37
+#define ADAU1373_ADC_CTRL 0x3c
+#define ADAU1373_DAI(x) (0x44 + (x))
+#define ADAU1373_CLK_SRC_DIV(x) (0x40 + (x) * 2)
+#define ADAU1373_BCLKDIV(x) (0x47 + (x))
+#define ADAU1373_SRC_RATIOA(x) (0x4a + (x) * 2)
+#define ADAU1373_SRC_RATIOB(x) (0x4b + (x) * 2)
+#define ADAU1373_DEEMP_CTRL 0x50
+#define ADAU1373_SRC_DAI_CTRL(x) (0x51 + (x))
+#define ADAU1373_DIN_MIX_CTRL(x) (0x56 + (x))
+#define ADAU1373_DOUT_MIX_CTRL(x) (0x5b + (x))
+#define ADAU1373_DAI_PBL_VOL(x) (0x62 + (x) * 2)
+#define ADAU1373_DAI_PBR_VOL(x) (0x63 + (x) * 2)
+#define ADAU1373_DAI_RECL_VOL(x) (0x68 + (x) * 2)
+#define ADAU1373_DAI_RECR_VOL(x) (0x69 + (x) * 2)
+#define ADAU1373_DAC1_PBL_VOL 0x6e
+#define ADAU1373_DAC1_PBR_VOL 0x6f
+#define ADAU1373_DAC2_PBL_VOL 0x70
+#define ADAU1373_DAC2_PBR_VOL 0x71
+#define ADAU1373_ADC_RECL_VOL 0x72
+#define ADAU1373_ADC_RECR_VOL 0x73
+#define ADAU1373_DMIC_RECL_VOL 0x74
+#define ADAU1373_DMIC_RECR_VOL 0x75
+#define ADAU1373_VOL_GAIN1 0x76
+#define ADAU1373_VOL_GAIN2 0x77
+#define ADAU1373_VOL_GAIN3 0x78
+#define ADAU1373_HPF_CTRL 0x7d
+#define ADAU1373_BASS1 0x7e
+#define ADAU1373_BASS2 0x7f
+#define ADAU1373_DRC(x) (0x80 + (x) * 0x10)
+#define ADAU1373_3D_CTRL1 0xc0
+#define ADAU1373_3D_CTRL2 0xc1
+#define ADAU1373_FDSP_SEL1 0xdc
+#define ADAU1373_FDSP_SEL2 0xdd
+#define ADAU1373_FDSP_SEL3 0xde
+#define ADAU1373_FDSP_SEL4 0xdf
+#define ADAU1373_DIGMICCTRL 0xe2
+#define ADAU1373_DIGEN 0xeb
+#define ADAU1373_SOFT_RESET 0xff
+
+
+#define ADAU1373_PLL_CTRL6_DPLL_BYPASS BIT(1)
+#define ADAU1373_PLL_CTRL6_PLL_EN BIT(0)
+
+#define ADAU1373_DAI_INVERT_BCLK BIT(7)
+#define ADAU1373_DAI_MASTER BIT(6)
+#define ADAU1373_DAI_INVERT_LRCLK BIT(4)
+#define ADAU1373_DAI_WLEN_16 0x0
+#define ADAU1373_DAI_WLEN_20 0x4
+#define ADAU1373_DAI_WLEN_24 0x8
+#define ADAU1373_DAI_WLEN_32 0xc
+#define ADAU1373_DAI_WLEN_MASK 0xc
+#define ADAU1373_DAI_FORMAT_RIGHT_J 0x0
+#define ADAU1373_DAI_FORMAT_LEFT_J 0x1
+#define ADAU1373_DAI_FORMAT_I2S 0x2
+#define ADAU1373_DAI_FORMAT_DSP 0x3
+
+#define ADAU1373_BCLKDIV_SOURCE BIT(5)
+#define ADAU1373_BCLKDIV_32 0x03
+#define ADAU1373_BCLKDIV_64 0x02
+#define ADAU1373_BCLKDIV_128 0x01
+#define ADAU1373_BCLKDIV_256 0x00
+
+#define ADAU1373_ADC_CTRL_PEAK_DETECT BIT(0)
+#define ADAU1373_ADC_CTRL_RESET BIT(1)
+#define ADAU1373_ADC_CTRL_RESET_FORCE BIT(2)
+
+#define ADAU1373_OUTPUT_CTRL_LDIFF BIT(3)
+#define ADAU1373_OUTPUT_CTRL_LNFBEN BIT(2)
+
+#define ADAU1373_PWDN_CTRL3_PWR_EN BIT(0)
+
+#define ADAU1373_EP_CTRL_MICBIAS1_OFFSET 4
+#define ADAU1373_EP_CTRL_MICBIAS2_OFFSET 2
+
+static const uint8_t adau1373_default_regs[] = {
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x00 */
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x10 */
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x20 */
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00,
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, 0x00, /* 0x30 */
+ 0x00, 0x00, 0x00, 0x80, 0x00, 0x01, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00, 0x0a, 0x0a, 0x0a, 0x00, /* 0x40 */
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x08, 0x08, 0x08, 0x00, 0x00, 0x00, 0x00, /* 0x50 */
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x60 */
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x70 */
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+ 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0x80 */
+ 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00,
+ 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0x90 */
+ 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00,
+ 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0xa0 */
+ 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0xff, 0xff, 0xff, 0xff, 0xff, /* 0xb0 */
+ 0xff, 0xff, 0xff, 0xff, 0xff, 0x1f, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0xc0 */
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0xd0 */
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, /* 0xe0 */
+ 0x00, 0x1f, 0x0f, 0x00, 0x00,
+};
+
+static const unsigned int adau1373_out_tlv[] = {
+ TLV_DB_RANGE_HEAD(4),
+ 0, 7, TLV_DB_SCALE_ITEM(-7900, 400, 1),
+ 8, 15, TLV_DB_SCALE_ITEM(-4700, 300, 0),
+ 16, 23, TLV_DB_SCALE_ITEM(-2300, 200, 0),
+ 24, 31, TLV_DB_SCALE_ITEM(-700, 100, 0),
+};
+
+static const DECLARE_TLV_DB_MINMAX(adau1373_digital_tlv, -9563, 0);
+static const DECLARE_TLV_DB_SCALE(adau1373_in_pga_tlv, -1300, 100, 1);
+static const DECLARE_TLV_DB_SCALE(adau1373_ep_tlv, -600, 600, 1);
+
+static const DECLARE_TLV_DB_SCALE(adau1373_input_boost_tlv, 0, 2000, 0);
+static const DECLARE_TLV_DB_SCALE(adau1373_gain_boost_tlv, 0, 600, 0);
+static const DECLARE_TLV_DB_SCALE(adau1373_speaker_boost_tlv, 1200, 600, 0);
+
+static const char *adau1373_fdsp_sel_text[] = {
+ "None",
+ "Channel 1",
+ "Channel 2",
+ "Channel 3",
+ "Channel 4",
+ "Channel 5",
+};
+
+static const SOC_ENUM_SINGLE_DECL(adau1373_drc1_channel_enum,
+ ADAU1373_FDSP_SEL1, 4, adau1373_fdsp_sel_text);
+static const SOC_ENUM_SINGLE_DECL(adau1373_drc2_channel_enum,
+ ADAU1373_FDSP_SEL1, 0, adau1373_fdsp_sel_text);
+static const SOC_ENUM_SINGLE_DECL(adau1373_drc3_channel_enum,
+ ADAU1373_FDSP_SEL2, 0, adau1373_fdsp_sel_text);
+static const SOC_ENUM_SINGLE_DECL(adau1373_hpf_channel_enum,
+ ADAU1373_FDSP_SEL3, 0, adau1373_fdsp_sel_text);
+static const SOC_ENUM_SINGLE_DECL(adau1373_bass_channel_enum,
+ ADAU1373_FDSP_SEL4, 4, adau1373_fdsp_sel_text);
+
+static const char *adau1373_hpf_cutoff_text[] = {
+ "3.7Hz", "50Hz", "100Hz", "150Hz", "200Hz", "250Hz", "300Hz", "350Hz",
+ "400Hz", "450Hz", "500Hz", "550Hz", "600Hz", "650Hz", "700Hz", "750Hz",
+ "800Hz",
+};
+
+static const SOC_ENUM_SINGLE_DECL(adau1373_hpf_cutoff_enum,
+ ADAU1373_HPF_CTRL, 3, adau1373_hpf_cutoff_text);
+
+static const char *adau1373_bass_lpf_cutoff_text[] = {
+ "801Hz", "1001Hz",
+};
+
+static const char *adau1373_bass_clip_level_text[] = {
+ "0.125", "0.250", "0.370", "0.500", "0.625", "0.750", "0.875",
+};
+
+static const unsigned int adau1373_bass_clip_level_values[] = {
+ 1, 2, 3, 4, 5, 6, 7,
+};
+
+static const char *adau1373_bass_hpf_cutoff_text[] = {
+ "158Hz", "232Hz", "347Hz", "520Hz",
+};
+
+static const unsigned int adau1373_bass_tlv[] = {
+ TLV_DB_RANGE_HEAD(4),
+ 0, 2, TLV_DB_SCALE_ITEM(-600, 600, 1),
+ 3, 4, TLV_DB_SCALE_ITEM(950, 250, 0),
+ 5, 7, TLV_DB_SCALE_ITEM(1400, 150, 0),
+};
+
+static const SOC_ENUM_SINGLE_DECL(adau1373_bass_lpf_cutoff_enum,
+ ADAU1373_BASS1, 5, adau1373_bass_lpf_cutoff_text);
+
+static const SOC_VALUE_ENUM_SINGLE_DECL(adau1373_bass_clip_level_enum,
+ ADAU1373_BASS1, 2, 7, adau1373_bass_clip_level_text,
+ adau1373_bass_clip_level_values);
+
+static const SOC_ENUM_SINGLE_DECL(adau1373_bass_hpf_cutoff_enum,
+ ADAU1373_BASS1, 0, adau1373_bass_hpf_cutoff_text);
+
+static const char *adau1373_3d_level_text[] = {
+ "0%", "6.67%", "13.33%", "20%", "26.67%", "33.33%",
+ "40%", "46.67%", "53.33%", "60%", "66.67%", "73.33%",
+ "80%", "86.67", "99.33%", "100%"
+};
+
+static const char *adau1373_3d_cutoff_text[] = {
+ "No 3D", "0.03125 fs", "0.04583 fs", "0.075 fs", "0.11458 fs",
+ "0.16875 fs", "0.27083 fs"
+};
+
+static const SOC_ENUM_SINGLE_DECL(adau1373_3d_level_enum,
+ ADAU1373_3D_CTRL1, 4, adau1373_3d_level_text);
+static const SOC_ENUM_SINGLE_DECL(adau1373_3d_cutoff_enum,
+ ADAU1373_3D_CTRL1, 0, adau1373_3d_cutoff_text);
+
+static const unsigned int adau1373_3d_tlv[] = {
+ TLV_DB_RANGE_HEAD(2),
+ 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
+ 1, 7, TLV_DB_LINEAR_ITEM(-1800, -120),
+};
+
+static const char *adau1373_lr_mux_text[] = {
+ "Mute",
+ "Right Channel (L+R)",
+ "Left Channel (L+R)",
+ "Stereo",
+};
+
+static const SOC_ENUM_SINGLE_DECL(adau1373_lineout1_lr_mux_enum,
+ ADAU1373_OUTPUT_CTRL, 4, adau1373_lr_mux_text);
+static const SOC_ENUM_SINGLE_DECL(adau1373_lineout2_lr_mux_enum,
+ ADAU1373_OUTPUT_CTRL, 6, adau1373_lr_mux_text);
+static const SOC_ENUM_SINGLE_DECL(adau1373_speaker_lr_mux_enum,
+ ADAU1373_LS_CTRL, 4, adau1373_lr_mux_text);
+
+static const struct snd_kcontrol_new adau1373_controls[] = {
+ SOC_DOUBLE_R_TLV("AIF1 Capture Volume", ADAU1373_DAI_RECL_VOL(0),
+ ADAU1373_DAI_RECR_VOL(0), 0, 0xff, 1, adau1373_digital_tlv),
+ SOC_DOUBLE_R_TLV("AIF2 Capture Volume", ADAU1373_DAI_RECL_VOL(1),
+ ADAU1373_DAI_RECR_VOL(1), 0, 0xff, 1, adau1373_digital_tlv),
+ SOC_DOUBLE_R_TLV("AIF3 Capture Volume", ADAU1373_DAI_RECL_VOL(2),
+ ADAU1373_DAI_RECR_VOL(2), 0, 0xff, 1, adau1373_digital_tlv),
+
+ SOC_DOUBLE_R_TLV("ADC Capture Volume", ADAU1373_ADC_RECL_VOL,
+ ADAU1373_ADC_RECR_VOL, 0, 0xff, 1, adau1373_digital_tlv),
+ SOC_DOUBLE_R_TLV("DMIC Capture Volume", ADAU1373_DMIC_RECL_VOL,
+ ADAU1373_DMIC_RECR_VOL, 0, 0xff, 1, adau1373_digital_tlv),
+
+ SOC_DOUBLE_R_TLV("AIF1 Playback Volume", ADAU1373_DAI_PBL_VOL(0),
+ ADAU1373_DAI_PBR_VOL(0), 0, 0xff, 1, adau1373_digital_tlv),
+ SOC_DOUBLE_R_TLV("AIF2 Playback Volume", ADAU1373_DAI_PBL_VOL(1),
+ ADAU1373_DAI_PBR_VOL(1), 0, 0xff, 1, adau1373_digital_tlv),
+ SOC_DOUBLE_R_TLV("AIF3 Playback Volume", ADAU1373_DAI_PBL_VOL(2),
+ ADAU1373_DAI_PBR_VOL(2), 0, 0xff, 1, adau1373_digital_tlv),
+
+ SOC_DOUBLE_R_TLV("DAC1 Playback Volume", ADAU1373_DAC1_PBL_VOL,
+ ADAU1373_DAC1_PBR_VOL, 0, 0xff, 1, adau1373_digital_tlv),
+ SOC_DOUBLE_R_TLV("DAC2 Playback Volume", ADAU1373_DAC2_PBL_VOL,
+ ADAU1373_DAC2_PBR_VOL, 0, 0xff, 1, adau1373_digital_tlv),
+
+ SOC_DOUBLE_R_TLV("Lineout1 Playback Volume", ADAU1373_LLINE_OUT(0),
+ ADAU1373_RLINE_OUT(0), 0, 0x1f, 0, adau1373_out_tlv),
+ SOC_DOUBLE_R_TLV("Speaker Playback Volume", ADAU1373_LSPK_OUT,
+ ADAU1373_RSPK_OUT, 0, 0x1f, 0, adau1373_out_tlv),
+ SOC_DOUBLE_R_TLV("Headphone Playback Volume", ADAU1373_LHP_OUT,
+ ADAU1373_RHP_OUT, 0, 0x1f, 0, adau1373_out_tlv),
+
+ SOC_DOUBLE_R_TLV("Input 1 Capture Volume", ADAU1373_AINL_CTRL(0),
+ ADAU1373_AINR_CTRL(0), 0, 0x1f, 0, adau1373_in_pga_tlv),
+ SOC_DOUBLE_R_TLV("Input 2 Capture Volume", ADAU1373_AINL_CTRL(1),
+ ADAU1373_AINR_CTRL(1), 0, 0x1f, 0, adau1373_in_pga_tlv),
+ SOC_DOUBLE_R_TLV("Input 3 Capture Volume", ADAU1373_AINL_CTRL(2),
+ ADAU1373_AINR_CTRL(2), 0, 0x1f, 0, adau1373_in_pga_tlv),
+ SOC_DOUBLE_R_TLV("Input 4 Capture Volume", ADAU1373_AINL_CTRL(3),
+ ADAU1373_AINR_CTRL(3), 0, 0x1f, 0, adau1373_in_pga_tlv),
+
+ SOC_SINGLE_TLV("Earpiece Playback Volume", ADAU1373_EP_CTRL, 0, 3, 0,
+ adau1373_ep_tlv),
+
+ SOC_DOUBLE_TLV("AIF3 Boost Playback Volume", ADAU1373_VOL_GAIN1, 4, 5,
+ 1, 0, adau1373_gain_boost_tlv),
+ SOC_DOUBLE_TLV("AIF2 Boost Playback Volume", ADAU1373_VOL_GAIN1, 2, 3,
+ 1, 0, adau1373_gain_boost_tlv),
+ SOC_DOUBLE_TLV("AIF1 Boost Playback Volume", ADAU1373_VOL_GAIN1, 0, 1,
+ 1, 0, adau1373_gain_boost_tlv),
+ SOC_DOUBLE_TLV("AIF3 Boost Capture Volume", ADAU1373_VOL_GAIN2, 4, 5,
+ 1, 0, adau1373_gain_boost_tlv),
+ SOC_DOUBLE_TLV("AIF2 Boost Capture Volume", ADAU1373_VOL_GAIN2, 2, 3,
+ 1, 0, adau1373_gain_boost_tlv),
+ SOC_DOUBLE_TLV("AIF1 Boost Capture Volume", ADAU1373_VOL_GAIN2, 0, 1,
+ 1, 0, adau1373_gain_boost_tlv),
+ SOC_DOUBLE_TLV("DMIC Boost Capture Volume", ADAU1373_VOL_GAIN3, 6, 7,
+ 1, 0, adau1373_gain_boost_tlv),
+ SOC_DOUBLE_TLV("ADC Boost Capture Volume", ADAU1373_VOL_GAIN3, 4, 5,
+ 1, 0, adau1373_gain_boost_tlv),
+ SOC_DOUBLE_TLV("DAC2 Boost Playback Volume", ADAU1373_VOL_GAIN3, 2, 3,
+ 1, 0, adau1373_gain_boost_tlv),
+ SOC_DOUBLE_TLV("DAC1 Boost Playback Volume", ADAU1373_VOL_GAIN3, 0, 1,
+ 1, 0, adau1373_gain_boost_tlv),
+
+ SOC_DOUBLE_TLV("Input 1 Boost Capture Volume", ADAU1373_ADC_GAIN, 0, 4,
+ 1, 0, adau1373_input_boost_tlv),
+ SOC_DOUBLE_TLV("Input 2 Boost Capture Volume", ADAU1373_ADC_GAIN, 1, 5,
+ 1, 0, adau1373_input_boost_tlv),
+ SOC_DOUBLE_TLV("Input 3 Boost Capture Volume", ADAU1373_ADC_GAIN, 2, 6,
+ 1, 0, adau1373_input_boost_tlv),
+ SOC_DOUBLE_TLV("Input 4 Boost Capture Volume", ADAU1373_ADC_GAIN, 3, 7,
+ 1, 0, adau1373_input_boost_tlv),
+
+ SOC_DOUBLE_TLV("Speaker Boost Playback Volume", ADAU1373_LS_CTRL, 2, 3,
+ 1, 0, adau1373_speaker_boost_tlv),
+
+ SOC_ENUM("Lineout1 LR Mux", adau1373_lineout1_lr_mux_enum),
+ SOC_ENUM("Speaker LR Mux", adau1373_speaker_lr_mux_enum),
+
+ SOC_ENUM("HPF Cutoff", adau1373_hpf_cutoff_enum),
+ SOC_DOUBLE("HPF Switch", ADAU1373_HPF_CTRL, 1, 0, 1, 0),
+ SOC_ENUM("HPF Channel", adau1373_hpf_channel_enum),
+
+ SOC_ENUM("Bass HPF Cutoff", adau1373_bass_hpf_cutoff_enum),
+ SOC_VALUE_ENUM("Bass Clip Level Threshold",
+ adau1373_bass_clip_level_enum),
+ SOC_ENUM("Bass LPF Cutoff", adau1373_bass_lpf_cutoff_enum),
+ SOC_DOUBLE("Bass Playback Switch", ADAU1373_BASS2, 0, 1, 1, 0),
+ SOC_SINGLE_TLV("Bass Playback Volume", ADAU1373_BASS2, 2, 7, 0,
+ adau1373_bass_tlv),
+ SOC_ENUM("Bass Channel", adau1373_bass_channel_enum),
+
+ SOC_ENUM("3D Freq", adau1373_3d_cutoff_enum),
+ SOC_ENUM("3D Level", adau1373_3d_level_enum),
+ SOC_SINGLE("3D Playback Switch", ADAU1373_3D_CTRL2, 0, 1, 0),
+ SOC_SINGLE_TLV("3D Playback Volume", ADAU1373_3D_CTRL2, 2, 7, 0,
+ adau1373_3d_tlv),
+ SOC_ENUM("3D Channel", adau1373_bass_channel_enum),
+
+ SOC_SINGLE("Zero Cross Switch", ADAU1373_PWDN_CTRL3, 7, 1, 0),
+};
+
+static const struct snd_kcontrol_new adau1373_lineout2_controls[] = {
+ SOC_DOUBLE_R_TLV("Lineout2 Playback Volume", ADAU1373_LLINE_OUT(1),
+ ADAU1373_RLINE_OUT(1), 0, 0x1f, 0, adau1373_out_tlv),
+ SOC_ENUM("Lineout2 LR Mux", adau1373_lineout2_lr_mux_enum),
+};
+
+static const struct snd_kcontrol_new adau1373_drc_controls[] = {
+ SOC_ENUM("DRC1 Channel", adau1373_drc1_channel_enum),
+ SOC_ENUM("DRC2 Channel", adau1373_drc2_channel_enum),
+ SOC_ENUM("DRC3 Channel", adau1373_drc3_channel_enum),
+};
+
+static int adau1373_pll_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ unsigned int pll_id = w->name[3] - '1';
+ unsigned int val;
+
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ val = ADAU1373_PLL_CTRL6_PLL_EN;
+ else
+ val = 0;
+
+ snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id),
+ ADAU1373_PLL_CTRL6_PLL_EN, val);
+
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ mdelay(5);
+
+ return 0;
+}
+
+static const char *adau1373_decimator_text[] = {
+ "ADC",
+ "DMIC1",
+};
+
+static const struct soc_enum adau1373_decimator_enum =
+ SOC_ENUM_SINGLE(0, 0, 2, adau1373_decimator_text);
+
+static const struct snd_kcontrol_new adau1373_decimator_mux =
+ SOC_DAPM_ENUM_VIRT("Decimator Mux", adau1373_decimator_enum);
+
+static const struct snd_kcontrol_new adau1373_left_adc_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DAC1 Switch", ADAU1373_LADC_MIXER, 4, 1, 0),
+ SOC_DAPM_SINGLE("Input 4 Switch", ADAU1373_LADC_MIXER, 3, 1, 0),
+ SOC_DAPM_SINGLE("Input 3 Switch", ADAU1373_LADC_MIXER, 2, 1, 0),
+ SOC_DAPM_SINGLE("Input 2 Switch", ADAU1373_LADC_MIXER, 1, 1, 0),
+ SOC_DAPM_SINGLE("Input 1 Switch", ADAU1373_LADC_MIXER, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new adau1373_right_adc_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DAC1 Switch", ADAU1373_RADC_MIXER, 4, 1, 0),
+ SOC_DAPM_SINGLE("Input 4 Switch", ADAU1373_RADC_MIXER, 3, 1, 0),
+ SOC_DAPM_SINGLE("Input 3 Switch", ADAU1373_RADC_MIXER, 2, 1, 0),
+ SOC_DAPM_SINGLE("Input 2 Switch", ADAU1373_RADC_MIXER, 1, 1, 0),
+ SOC_DAPM_SINGLE("Input 1 Switch", ADAU1373_RADC_MIXER, 0, 1, 0),
+};
+
+#define DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(_name, _reg) \
+const struct snd_kcontrol_new _name[] = { \
+ SOC_DAPM_SINGLE("Left DAC2 Switch", _reg, 7, 1, 0), \
+ SOC_DAPM_SINGLE("Right DAC2 Switch", _reg, 6, 1, 0), \
+ SOC_DAPM_SINGLE("Left DAC1 Switch", _reg, 5, 1, 0), \
+ SOC_DAPM_SINGLE("Right DAC1 Switch", _reg, 4, 1, 0), \
+ SOC_DAPM_SINGLE("Input 4 Bypass Switch", _reg, 3, 1, 0), \
+ SOC_DAPM_SINGLE("Input 3 Bypass Switch", _reg, 2, 1, 0), \
+ SOC_DAPM_SINGLE("Input 2 Bypass Switch", _reg, 1, 1, 0), \
+ SOC_DAPM_SINGLE("Input 1 Bypass Switch", _reg, 0, 1, 0), \
+}
+
+static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_left_line1_mixer_controls,
+ ADAU1373_LLINE1_MIX);
+static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_right_line1_mixer_controls,
+ ADAU1373_RLINE1_MIX);
+static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_left_line2_mixer_controls,
+ ADAU1373_LLINE2_MIX);
+static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_right_line2_mixer_controls,
+ ADAU1373_RLINE2_MIX);
+static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_left_spk_mixer_controls,
+ ADAU1373_LSPK_MIX);
+static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_right_spk_mixer_controls,
+ ADAU1373_RSPK_MIX);
+static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_ep_mixer_controls,
+ ADAU1373_EP_MIX);
+
+static const struct snd_kcontrol_new adau1373_left_hp_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Left DAC1 Switch", ADAU1373_LHP_MIX, 5, 1, 0),
+ SOC_DAPM_SINGLE("Left DAC2 Switch", ADAU1373_LHP_MIX, 4, 1, 0),
+ SOC_DAPM_SINGLE("Input 4 Bypass Switch", ADAU1373_LHP_MIX, 3, 1, 0),
+ SOC_DAPM_SINGLE("Input 3 Bypass Switch", ADAU1373_LHP_MIX, 2, 1, 0),
+ SOC_DAPM_SINGLE("Input 2 Bypass Switch", ADAU1373_LHP_MIX, 1, 1, 0),
+ SOC_DAPM_SINGLE("Input 1 Bypass Switch", ADAU1373_LHP_MIX, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new adau1373_right_hp_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Right DAC1 Switch", ADAU1373_RHP_MIX, 5, 1, 0),
+ SOC_DAPM_SINGLE("Right DAC2 Switch", ADAU1373_RHP_MIX, 4, 1, 0),
+ SOC_DAPM_SINGLE("Input 4 Bypass Switch", ADAU1373_RHP_MIX, 3, 1, 0),
+ SOC_DAPM_SINGLE("Input 3 Bypass Switch", ADAU1373_RHP_MIX, 2, 1, 0),
+ SOC_DAPM_SINGLE("Input 2 Bypass Switch", ADAU1373_RHP_MIX, 1, 1, 0),
+ SOC_DAPM_SINGLE("Input 1 Bypass Switch", ADAU1373_RHP_MIX, 0, 1, 0),
+};
+
+#define DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(_name, _reg) \
+const struct snd_kcontrol_new _name[] = { \
+ SOC_DAPM_SINGLE("DMIC2 Swapped Switch", _reg, 6, 1, 0), \
+ SOC_DAPM_SINGLE("DMIC2 Switch", _reg, 5, 1, 0), \
+ SOC_DAPM_SINGLE("ADC/DMIC1 Swapped Switch", _reg, 4, 1, 0), \
+ SOC_DAPM_SINGLE("ADC/DMIC1 Switch", _reg, 3, 1, 0), \
+ SOC_DAPM_SINGLE("AIF3 Switch", _reg, 2, 1, 0), \
+ SOC_DAPM_SINGLE("AIF2 Switch", _reg, 1, 1, 0), \
+ SOC_DAPM_SINGLE("AIF1 Switch", _reg, 0, 1, 0), \
+}
+
+static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel1_mixer_controls,
+ ADAU1373_DIN_MIX_CTRL(0));
+static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel2_mixer_controls,
+ ADAU1373_DIN_MIX_CTRL(1));
+static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel3_mixer_controls,
+ ADAU1373_DIN_MIX_CTRL(2));
+static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel4_mixer_controls,
+ ADAU1373_DIN_MIX_CTRL(3));
+static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel5_mixer_controls,
+ ADAU1373_DIN_MIX_CTRL(4));
+
+#define DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(_name, _reg) \
+const struct snd_kcontrol_new _name[] = { \
+ SOC_DAPM_SINGLE("DSP Channel5 Switch", _reg, 4, 1, 0), \
+ SOC_DAPM_SINGLE("DSP Channel4 Switch", _reg, 3, 1, 0), \
+ SOC_DAPM_SINGLE("DSP Channel3 Switch", _reg, 2, 1, 0), \
+ SOC_DAPM_SINGLE("DSP Channel2 Switch", _reg, 1, 1, 0), \
+ SOC_DAPM_SINGLE("DSP Channel1 Switch", _reg, 0, 1, 0), \
+}
+
+static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_aif1_mixer_controls,
+ ADAU1373_DOUT_MIX_CTRL(0));
+static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_aif2_mixer_controls,
+ ADAU1373_DOUT_MIX_CTRL(1));
+static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_aif3_mixer_controls,
+ ADAU1373_DOUT_MIX_CTRL(2));
+static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_dac1_mixer_controls,
+ ADAU1373_DOUT_MIX_CTRL(3));
+static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_dac2_mixer_controls,
+ ADAU1373_DOUT_MIX_CTRL(4));
+
+static const struct snd_soc_dapm_widget adau1373_dapm_widgets[] = {
+ /* Datasheet claims Left ADC is bit 6 and Right ADC is bit 7, but that
+ * doesn't seem to be the case. */
+ SND_SOC_DAPM_ADC("Left ADC", NULL, ADAU1373_PWDN_CTRL1, 7, 0),
+ SND_SOC_DAPM_ADC("Right ADC", NULL, ADAU1373_PWDN_CTRL1, 6, 0),
+
+ SND_SOC_DAPM_ADC("DMIC1", NULL, ADAU1373_DIGMICCTRL, 0, 0),
+ SND_SOC_DAPM_ADC("DMIC2", NULL, ADAU1373_DIGMICCTRL, 2, 0),
+
+ SND_SOC_DAPM_VIRT_MUX("Decimator Mux", SND_SOC_NOPM, 0, 0,
+ &adau1373_decimator_mux),
+
+ SND_SOC_DAPM_SUPPLY("MICBIAS2", ADAU1373_PWDN_CTRL1, 5, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("MICBIAS1", ADAU1373_PWDN_CTRL1, 4, 0, NULL, 0),
+
+ SND_SOC_DAPM_PGA("IN4PGA", ADAU1373_PWDN_CTRL1, 3, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IN3PGA", ADAU1373_PWDN_CTRL1, 2, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IN2PGA", ADAU1373_PWDN_CTRL1, 1, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IN1PGA", ADAU1373_PWDN_CTRL1, 0, 0, NULL, 0),
+
+ SND_SOC_DAPM_DAC("Left DAC2", NULL, ADAU1373_PWDN_CTRL2, 7, 0),
+ SND_SOC_DAPM_DAC("Right DAC2", NULL, ADAU1373_PWDN_CTRL2, 6, 0),
+ SND_SOC_DAPM_DAC("Left DAC1", NULL, ADAU1373_PWDN_CTRL2, 5, 0),
+ SND_SOC_DAPM_DAC("Right DAC1", NULL, ADAU1373_PWDN_CTRL2, 4, 0),
+
+ SOC_MIXER_ARRAY("Left ADC Mixer", SND_SOC_NOPM, 0, 0,
+ adau1373_left_adc_mixer_controls),
+ SOC_MIXER_ARRAY("Right ADC Mixer", SND_SOC_NOPM, 0, 0,
+ adau1373_right_adc_mixer_controls),
+
+ SOC_MIXER_ARRAY("Left Lineout2 Mixer", ADAU1373_PWDN_CTRL2, 3, 0,
+ adau1373_left_line2_mixer_controls),
+ SOC_MIXER_ARRAY("Right Lineout2 Mixer", ADAU1373_PWDN_CTRL2, 2, 0,
+ adau1373_right_line2_mixer_controls),
+ SOC_MIXER_ARRAY("Left Lineout1 Mixer", ADAU1373_PWDN_CTRL2, 1, 0,
+ adau1373_left_line1_mixer_controls),
+ SOC_MIXER_ARRAY("Right Lineout1 Mixer", ADAU1373_PWDN_CTRL2, 0, 0,
+ adau1373_right_line1_mixer_controls),
+
+ SOC_MIXER_ARRAY("Earpiece Mixer", ADAU1373_PWDN_CTRL3, 4, 0,
+ adau1373_ep_mixer_controls),
+ SOC_MIXER_ARRAY("Left Speaker Mixer", ADAU1373_PWDN_CTRL3, 3, 0,
+ adau1373_left_spk_mixer_controls),
+ SOC_MIXER_ARRAY("Right Speaker Mixer", ADAU1373_PWDN_CTRL3, 2, 0,
+ adau1373_right_spk_mixer_controls),
+ SOC_MIXER_ARRAY("Left Headphone Mixer", SND_SOC_NOPM, 0, 0,
+ adau1373_left_hp_mixer_controls),
+ SOC_MIXER_ARRAY("Right Headphone Mixer", SND_SOC_NOPM, 0, 0,
+ adau1373_right_hp_mixer_controls),
+ SND_SOC_DAPM_SUPPLY("Headphone Enable", ADAU1373_PWDN_CTRL3, 1, 0,
+ NULL, 0),
+
+ SND_SOC_DAPM_SUPPLY("AIF1 CLK", ADAU1373_SRC_DAI_CTRL(0), 0, 0,
+ NULL, 0),
+ SND_SOC_DAPM_SUPPLY("AIF2 CLK", ADAU1373_SRC_DAI_CTRL(1), 0, 0,
+ NULL, 0),
+ SND_SOC_DAPM_SUPPLY("AIF3 CLK", ADAU1373_SRC_DAI_CTRL(2), 0, 0,
+ NULL, 0),
+ SND_SOC_DAPM_SUPPLY("AIF1 IN SRC", ADAU1373_SRC_DAI_CTRL(0), 2, 0,
+ NULL, 0),
+ SND_SOC_DAPM_SUPPLY("AIF1 OUT SRC", ADAU1373_SRC_DAI_CTRL(0), 1, 0,
+ NULL, 0),
+ SND_SOC_DAPM_SUPPLY("AIF2 IN SRC", ADAU1373_SRC_DAI_CTRL(1), 2, 0,
+ NULL, 0),
+ SND_SOC_DAPM_SUPPLY("AIF2 OUT SRC", ADAU1373_SRC_DAI_CTRL(1), 1, 0,
+ NULL, 0),
+ SND_SOC_DAPM_SUPPLY("AIF3 IN SRC", ADAU1373_SRC_DAI_CTRL(2), 2, 0,
+ NULL, 0),
+ SND_SOC_DAPM_SUPPLY("AIF3 OUT SRC", ADAU1373_SRC_DAI_CTRL(2), 1, 0,
+ NULL, 0),
+
+ SND_SOC_DAPM_AIF_IN("AIF1 IN", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("AIF1 OUT", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("AIF2 IN", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("AIF2 OUT", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("AIF3 IN", "AIF3 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("AIF3 OUT", "AIF3 Capture", 0, SND_SOC_NOPM, 0, 0),
+
+ SOC_MIXER_ARRAY("DSP Channel1 Mixer", SND_SOC_NOPM, 0, 0,
+ adau1373_dsp_channel1_mixer_controls),
+ SOC_MIXER_ARRAY("DSP Channel2 Mixer", SND_SOC_NOPM, 0, 0,
+ adau1373_dsp_channel2_mixer_controls),
+ SOC_MIXER_ARRAY("DSP Channel3 Mixer", SND_SOC_NOPM, 0, 0,
+ adau1373_dsp_channel3_mixer_controls),
+ SOC_MIXER_ARRAY("DSP Channel4 Mixer", SND_SOC_NOPM, 0, 0,
+ adau1373_dsp_channel4_mixer_controls),
+ SOC_MIXER_ARRAY("DSP Channel5 Mixer", SND_SOC_NOPM, 0, 0,
+ adau1373_dsp_channel5_mixer_controls),
+
+ SOC_MIXER_ARRAY("AIF1 Mixer", SND_SOC_NOPM, 0, 0,
+ adau1373_aif1_mixer_controls),
+ SOC_MIXER_ARRAY("AIF2 Mixer", SND_SOC_NOPM, 0, 0,
+ adau1373_aif2_mixer_controls),
+ SOC_MIXER_ARRAY("AIF3 Mixer", SND_SOC_NOPM, 0, 0,
+ adau1373_aif3_mixer_controls),
+ SOC_MIXER_ARRAY("DAC1 Mixer", SND_SOC_NOPM, 0, 0,
+ adau1373_dac1_mixer_controls),
+ SOC_MIXER_ARRAY("DAC2 Mixer", SND_SOC_NOPM, 0, 0,
+ adau1373_dac2_mixer_controls),
+
+ SND_SOC_DAPM_SUPPLY("DSP", ADAU1373_DIGEN, 4, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("Recording Engine B", ADAU1373_DIGEN, 3, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("Recording Engine A", ADAU1373_DIGEN, 2, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("Playback Engine B", ADAU1373_DIGEN, 1, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("Playback Engine A", ADAU1373_DIGEN, 0, 0, NULL, 0),
+
+ SND_SOC_DAPM_SUPPLY("PLL1", SND_SOC_NOPM, 0, 0, adau1373_pll_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SUPPLY("PLL2", SND_SOC_NOPM, 0, 0, adau1373_pll_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SUPPLY("SYSCLK1", ADAU1373_CLK_SRC_DIV(0), 7, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("SYSCLK2", ADAU1373_CLK_SRC_DIV(1), 7, 0, NULL, 0),
+
+ SND_SOC_DAPM_INPUT("AIN1L"),
+ SND_SOC_DAPM_INPUT("AIN1R"),
+ SND_SOC_DAPM_INPUT("AIN2L"),
+ SND_SOC_DAPM_INPUT("AIN2R"),
+ SND_SOC_DAPM_INPUT("AIN3L"),
+ SND_SOC_DAPM_INPUT("AIN3R"),
+ SND_SOC_DAPM_INPUT("AIN4L"),
+ SND_SOC_DAPM_INPUT("AIN4R"),
+
+ SND_SOC_DAPM_INPUT("DMIC1DAT"),
+ SND_SOC_DAPM_INPUT("DMIC2DAT"),
+
+ SND_SOC_DAPM_OUTPUT("LOUT1L"),
+ SND_SOC_DAPM_OUTPUT("LOUT1R"),
+ SND_SOC_DAPM_OUTPUT("LOUT2L"),
+ SND_SOC_DAPM_OUTPUT("LOUT2R"),
+ SND_SOC_DAPM_OUTPUT("HPL"),
+ SND_SOC_DAPM_OUTPUT("HPR"),
+ SND_SOC_DAPM_OUTPUT("SPKL"),
+ SND_SOC_DAPM_OUTPUT("SPKR"),
+ SND_SOC_DAPM_OUTPUT("EP"),
+};
+
+static int adau1373_check_aif_clk(struct snd_soc_dapm_widget *source,
+ struct snd_soc_dapm_widget *sink)
+{
+ struct snd_soc_codec *codec = source->codec;
+ struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec);
+ unsigned int dai;
+ const char *clk;
+
+ dai = sink->name[3] - '1';
+
+ if (!adau1373->dais[dai].master)
+ return 0;
+
+ if (adau1373->dais[dai].clk_src == ADAU1373_CLK_SRC_PLL1)
+ clk = "SYSCLK1";
+ else
+ clk = "SYSCLK2";
+
+ return strcmp(source->name, clk) == 0;
+}
+
+static int adau1373_check_src(struct snd_soc_dapm_widget *source,
+ struct snd_soc_dapm_widget *sink)
+{
+ struct snd_soc_codec *codec = source->codec;
+ struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec);
+ unsigned int dai;
+
+ dai = sink->name[3] - '1';
+
+ return adau1373->dais[dai].enable_src;
+}
+
+#define DSP_CHANNEL_MIXER_ROUTES(_sink) \
+ { _sink, "DMIC2 Swapped Switch", "DMIC2" }, \
+ { _sink, "DMIC2 Switch", "DMIC2" }, \
+ { _sink, "ADC/DMIC1 Swapped Switch", "Decimator Mux" }, \
+ { _sink, "ADC/DMIC1 Switch", "Decimator Mux" }, \
+ { _sink, "AIF1 Switch", "AIF1 IN" }, \
+ { _sink, "AIF2 Switch", "AIF2 IN" }, \
+ { _sink, "AIF3 Switch", "AIF3 IN" }
+
+#define DSP_OUTPUT_MIXER_ROUTES(_sink) \
+ { _sink, "DSP Channel1 Switch", "DSP Channel1 Mixer" }, \
+ { _sink, "DSP Channel2 Switch", "DSP Channel2 Mixer" }, \
+ { _sink, "DSP Channel3 Switch", "DSP Channel3 Mixer" }, \
+ { _sink, "DSP Channel4 Switch", "DSP Channel4 Mixer" }, \
+ { _sink, "DSP Channel5 Switch", "DSP Channel5 Mixer" }
+
+#define LEFT_OUTPUT_MIXER_ROUTES(_sink) \
+ { _sink, "Right DAC2 Switch", "Right DAC2" }, \
+ { _sink, "Left DAC2 Switch", "Left DAC2" }, \
+ { _sink, "Right DAC1 Switch", "Right DAC1" }, \
+ { _sink, "Left DAC1 Switch", "Left DAC1" }, \
+ { _sink, "Input 1 Bypass Switch", "IN1PGA" }, \
+ { _sink, "Input 2 Bypass Switch", "IN2PGA" }, \
+ { _sink, "Input 3 Bypass Switch", "IN3PGA" }, \
+ { _sink, "Input 4 Bypass Switch", "IN4PGA" }
+
+#define RIGHT_OUTPUT_MIXER_ROUTES(_sink) \
+ { _sink, "Right DAC2 Switch", "Right DAC2" }, \
+ { _sink, "Left DAC2 Switch", "Left DAC2" }, \
+ { _sink, "Right DAC1 Switch", "Right DAC1" }, \
+ { _sink, "Left DAC1 Switch", "Left DAC1" }, \
+ { _sink, "Input 1 Bypass Switch", "IN1PGA" }, \
+ { _sink, "Input 2 Bypass Switch", "IN2PGA" }, \
+ { _sink, "Input 3 Bypass Switch", "IN3PGA" }, \
+ { _sink, "Input 4 Bypass Switch", "IN4PGA" }
+
+static const struct snd_soc_dapm_route adau1373_dapm_routes[] = {
+ { "Left ADC Mixer", "DAC1 Switch", "Left DAC1" },
+ { "Left ADC Mixer", "Input 1 Switch", "IN1PGA" },
+ { "Left ADC Mixer", "Input 2 Switch", "IN2PGA" },
+ { "Left ADC Mixer", "Input 3 Switch", "IN3PGA" },
+ { "Left ADC Mixer", "Input 4 Switch", "IN4PGA" },
+
+ { "Right ADC Mixer", "DAC1 Switch", "Right DAC1" },
+ { "Right ADC Mixer", "Input 1 Switch", "IN1PGA" },
+ { "Right ADC Mixer", "Input 2 Switch", "IN2PGA" },
+ { "Right ADC Mixer", "Input 3 Switch", "IN3PGA" },
+ { "Right ADC Mixer", "Input 4 Switch", "IN4PGA" },
+
+ { "Left ADC", NULL, "Left ADC Mixer" },
+ { "Right ADC", NULL, "Right ADC Mixer" },
+
+ { "Decimator Mux", "ADC", "Left ADC" },
+ { "Decimator Mux", "ADC", "Right ADC" },
+ { "Decimator Mux", "DMIC1", "DMIC1" },
+
+ DSP_CHANNEL_MIXER_ROUTES("DSP Channel1 Mixer"),
+ DSP_CHANNEL_MIXER_ROUTES("DSP Channel2 Mixer"),
+ DSP_CHANNEL_MIXER_ROUTES("DSP Channel3 Mixer"),
+ DSP_CHANNEL_MIXER_ROUTES("DSP Channel4 Mixer"),
+ DSP_CHANNEL_MIXER_ROUTES("DSP Channel5 Mixer"),
+
+ DSP_OUTPUT_MIXER_ROUTES("AIF1 Mixer"),
+ DSP_OUTPUT_MIXER_ROUTES("AIF2 Mixer"),
+ DSP_OUTPUT_MIXER_ROUTES("AIF3 Mixer"),
+ DSP_OUTPUT_MIXER_ROUTES("DAC1 Mixer"),
+ DSP_OUTPUT_MIXER_ROUTES("DAC2 Mixer"),
+
+ { "AIF1 OUT", NULL, "AIF1 Mixer" },
+ { "AIF2 OUT", NULL, "AIF2 Mixer" },
+ { "AIF3 OUT", NULL, "AIF3 Mixer" },
+ { "Left DAC1", NULL, "DAC1 Mixer" },
+ { "Right DAC1", NULL, "DAC1 Mixer" },
+ { "Left DAC2", NULL, "DAC2 Mixer" },
+ { "Right DAC2", NULL, "DAC2 Mixer" },
+
+ LEFT_OUTPUT_MIXER_ROUTES("Left Lineout1 Mixer"),
+ RIGHT_OUTPUT_MIXER_ROUTES("Right Lineout1 Mixer"),
+ LEFT_OUTPUT_MIXER_ROUTES("Left Lineout2 Mixer"),
+ RIGHT_OUTPUT_MIXER_ROUTES("Right Lineout2 Mixer"),
+ LEFT_OUTPUT_MIXER_ROUTES("Left Speaker Mixer"),
+ RIGHT_OUTPUT_MIXER_ROUTES("Right Speaker Mixer"),
+
+ { "Left Headphone Mixer", "Left DAC2 Switch", "Left DAC2" },
+ { "Left Headphone Mixer", "Left DAC1 Switch", "Left DAC1" },
+ { "Left Headphone Mixer", "Input 1 Bypass Switch", "IN1PGA" },
+ { "Left Headphone Mixer", "Input 2 Bypass Switch", "IN2PGA" },
+ { "Left Headphone Mixer", "Input 3 Bypass Switch", "IN3PGA" },
+ { "Left Headphone Mixer", "Input 4 Bypass Switch", "IN4PGA" },
+ { "Right Headphone Mixer", "Right DAC2 Switch", "Right DAC2" },
+ { "Right Headphone Mixer", "Right DAC1 Switch", "Right DAC1" },
+ { "Right Headphone Mixer", "Input 1 Bypass Switch", "IN1PGA" },
+ { "Right Headphone Mixer", "Input 2 Bypass Switch", "IN2PGA" },
+ { "Right Headphone Mixer", "Input 3 Bypass Switch", "IN3PGA" },
+ { "Right Headphone Mixer", "Input 4 Bypass Switch", "IN4PGA" },
+
+ { "Left Headphone Mixer", NULL, "Headphone Enable" },
+ { "Right Headphone Mixer", NULL, "Headphone Enable" },
+
+ { "Earpiece Mixer", "Right DAC2 Switch", "Right DAC2" },
+ { "Earpiece Mixer", "Left DAC2 Switch", "Left DAC2" },
+ { "Earpiece Mixer", "Right DAC1 Switch", "Right DAC1" },
+ { "Earpiece Mixer", "Left DAC1 Switch", "Left DAC1" },
+ { "Earpiece Mixer", "Input 1 Bypass Switch", "IN1PGA" },
+ { "Earpiece Mixer", "Input 2 Bypass Switch", "IN2PGA" },
+ { "Earpiece Mixer", "Input 3 Bypass Switch", "IN3PGA" },
+ { "Earpiece Mixer", "Input 4 Bypass Switch", "IN4PGA" },
+
+ { "LOUT1L", NULL, "Left Lineout1 Mixer" },
+ { "LOUT1R", NULL, "Right Lineout1 Mixer" },
+ { "LOUT2L", NULL, "Left Lineout2 Mixer" },
+ { "LOUT2R", NULL, "Right Lineout2 Mixer" },
+ { "SPKL", NULL, "Left Speaker Mixer" },
+ { "SPKR", NULL, "Right Speaker Mixer" },
+ { "HPL", NULL, "Left Headphone Mixer" },
+ { "HPR", NULL, "Right Headphone Mixer" },
+ { "EP", NULL, "Earpiece Mixer" },
+
+ { "IN1PGA", NULL, "AIN1L" },
+ { "IN2PGA", NULL, "AIN2L" },
+ { "IN3PGA", NULL, "AIN3L" },
+ { "IN4PGA", NULL, "AIN4L" },
+ { "IN1PGA", NULL, "AIN1R" },
+ { "IN2PGA", NULL, "AIN2R" },
+ { "IN3PGA", NULL, "AIN3R" },
+ { "IN4PGA", NULL, "AIN4R" },
+
+ { "SYSCLK1", NULL, "PLL1" },
+ { "SYSCLK2", NULL, "PLL2" },
+
+ { "Left DAC1", NULL, "SYSCLK1" },
+ { "Right DAC1", NULL, "SYSCLK1" },
+ { "Left DAC2", NULL, "SYSCLK1" },
+ { "Right DAC2", NULL, "SYSCLK1" },
+ { "Left ADC", NULL, "SYSCLK1" },
+ { "Right ADC", NULL, "SYSCLK1" },
+
+ { "DSP", NULL, "SYSCLK1" },
+
+ { "AIF1 Mixer", NULL, "DSP" },
+ { "AIF2 Mixer", NULL, "DSP" },
+ { "AIF3 Mixer", NULL, "DSP" },
+ { "DAC1 Mixer", NULL, "DSP" },
+ { "DAC2 Mixer", NULL, "DSP" },
+ { "DAC1 Mixer", NULL, "Playback Engine A" },
+ { "DAC2 Mixer", NULL, "Playback Engine B" },
+ { "Left ADC Mixer", NULL, "Recording Engine A" },
+ { "Right ADC Mixer", NULL, "Recording Engine A" },
+
+ { "AIF1 CLK", NULL, "SYSCLK1", adau1373_check_aif_clk },
+ { "AIF2 CLK", NULL, "SYSCLK1", adau1373_check_aif_clk },
+ { "AIF3 CLK", NULL, "SYSCLK1", adau1373_check_aif_clk },
+ { "AIF1 CLK", NULL, "SYSCLK2", adau1373_check_aif_clk },
+ { "AIF2 CLK", NULL, "SYSCLK2", adau1373_check_aif_clk },
+ { "AIF3 CLK", NULL, "SYSCLK2", adau1373_check_aif_clk },
+
+ { "AIF1 IN", NULL, "AIF1 CLK" },
+ { "AIF1 OUT", NULL, "AIF1 CLK" },
+ { "AIF2 IN", NULL, "AIF2 CLK" },
+ { "AIF2 OUT", NULL, "AIF2 CLK" },
+ { "AIF3 IN", NULL, "AIF3 CLK" },
+ { "AIF3 OUT", NULL, "AIF3 CLK" },
+ { "AIF1 IN", NULL, "AIF1 IN SRC", adau1373_check_src },
+ { "AIF1 OUT", NULL, "AIF1 OUT SRC", adau1373_check_src },
+ { "AIF2 IN", NULL, "AIF2 IN SRC", adau1373_check_src },
+ { "AIF2 OUT", NULL, "AIF2 OUT SRC", adau1373_check_src },
+ { "AIF3 IN", NULL, "AIF3 IN SRC", adau1373_check_src },
+ { "AIF3 OUT", NULL, "AIF3 OUT SRC", adau1373_check_src },
+
+ { "DMIC1", NULL, "DMIC1DAT" },
+ { "DMIC1", NULL, "SYSCLK1" },
+ { "DMIC1", NULL, "Recording Engine A" },
+ { "DMIC2", NULL, "DMIC2DAT" },
+ { "DMIC2", NULL, "SYSCLK1" },
+ { "DMIC2", NULL, "Recording Engine B" },
+};
+
+static int adau1373_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec);
+ struct adau1373_dai *adau1373_dai = &adau1373->dais[dai->id];
+ unsigned int div;
+ unsigned int freq;
+ unsigned int ctrl;
+
+ freq = adau1373_dai->sysclk;
+
+ if (freq % params_rate(params) != 0)
+ return -EINVAL;
+
+ switch (freq / params_rate(params)) {
+ case 1024: /* sysclk / 256 */
+ div = 0;
+ break;
+ case 1536: /* 2/3 sysclk / 256 */
+ div = 1;
+ break;
+ case 2048: /* 1/2 sysclk / 256 */
+ div = 2;
+ break;
+ case 3072: /* 1/3 sysclk / 256 */
+ div = 3;
+ break;
+ case 4096: /* 1/4 sysclk / 256 */
+ div = 4;
+ break;
+ case 6144: /* 1/6 sysclk / 256 */
+ div = 5;
+ break;
+ case 5632: /* 2/11 sysclk / 256 */
+ div = 6;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ adau1373_dai->enable_src = (div != 0);
+
+ snd_soc_update_bits(codec, ADAU1373_BCLKDIV(dai->id),
+ ~ADAU1373_BCLKDIV_SOURCE, (div << 2) | ADAU1373_BCLKDIV_64);
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ ctrl = ADAU1373_DAI_WLEN_16;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ ctrl = ADAU1373_DAI_WLEN_20;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ ctrl = ADAU1373_DAI_WLEN_24;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ ctrl = ADAU1373_DAI_WLEN_32;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return snd_soc_update_bits(codec, ADAU1373_DAI(dai->id),
+ ADAU1373_DAI_WLEN_MASK, ctrl);
+}
+
+static int adau1373_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec);
+ struct adau1373_dai *adau1373_dai = &adau1373->dais[dai->id];
+ unsigned int ctrl;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ ctrl = ADAU1373_DAI_MASTER;
+ adau1373_dai->master = true;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ ctrl = 0;
+ adau1373_dai->master = true;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ ctrl |= ADAU1373_DAI_FORMAT_I2S;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ ctrl |= ADAU1373_DAI_FORMAT_LEFT_J;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ ctrl |= ADAU1373_DAI_FORMAT_RIGHT_J;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ ctrl |= ADAU1373_DAI_FORMAT_DSP;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ ctrl |= ADAU1373_DAI_INVERT_BCLK;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ ctrl |= ADAU1373_DAI_INVERT_LRCLK;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ ctrl |= ADAU1373_DAI_INVERT_LRCLK | ADAU1373_DAI_INVERT_BCLK;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, ADAU1373_DAI(dai->id),
+ ~ADAU1373_DAI_WLEN_MASK, ctrl);
+
+ return 0;
+}
+
+static int adau1373_set_dai_sysclk(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(dai->codec);
+ struct adau1373_dai *adau1373_dai = &adau1373->dais[dai->id];
+
+ switch (clk_id) {
+ case ADAU1373_CLK_SRC_PLL1:
+ case ADAU1373_CLK_SRC_PLL2:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ adau1373_dai->sysclk = freq;
+ adau1373_dai->clk_src = clk_id;
+
+ snd_soc_update_bits(dai->codec, ADAU1373_BCLKDIV(dai->id),
+ ADAU1373_BCLKDIV_SOURCE, clk_id << 5);
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops adau1373_dai_ops = {
+ .hw_params = adau1373_hw_params,
+ .set_sysclk = adau1373_set_dai_sysclk,
+ .set_fmt = adau1373_set_dai_fmt,
+};
+
+#define ADAU1373_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_driver adau1373_dai_driver[] = {
+ {
+ .id = 0,
+ .name = "adau1373-aif1",
+ .playback = {
+ .stream_name = "AIF1 Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = ADAU1373_FORMATS,
+ },
+ .capture = {
+ .stream_name = "AIF1 Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = ADAU1373_FORMATS,
+ },
+ .ops = &adau1373_dai_ops,
+ .symmetric_rates = 1,
+ },
+ {
+ .id = 1,
+ .name = "adau1373-aif2",
+ .playback = {
+ .stream_name = "AIF2 Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = ADAU1373_FORMATS,
+ },
+ .capture = {
+ .stream_name = "AIF2 Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = ADAU1373_FORMATS,
+ },
+ .ops = &adau1373_dai_ops,
+ .symmetric_rates = 1,
+ },
+ {
+ .id = 2,
+ .name = "adau1373-aif3",
+ .playback = {
+ .stream_name = "AIF3 Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = ADAU1373_FORMATS,
+ },
+ .capture = {
+ .stream_name = "AIF3 Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = ADAU1373_FORMATS,
+ },
+ .ops = &adau1373_dai_ops,
+ .symmetric_rates = 1,
+ },
+};
+
+static int adau1373_set_pll(struct snd_soc_codec *codec, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
+{
+ unsigned int dpll_div = 0;
+ unsigned int x, r, n, m, i, j, mode;
+
+ switch (pll_id) {
+ case ADAU1373_PLL1:
+ case ADAU1373_PLL2:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (source) {
+ case ADAU1373_PLL_SRC_BCLK1:
+ case ADAU1373_PLL_SRC_BCLK2:
+ case ADAU1373_PLL_SRC_BCLK3:
+ case ADAU1373_PLL_SRC_LRCLK1:
+ case ADAU1373_PLL_SRC_LRCLK2:
+ case ADAU1373_PLL_SRC_LRCLK3:
+ case ADAU1373_PLL_SRC_MCLK1:
+ case ADAU1373_PLL_SRC_MCLK2:
+ case ADAU1373_PLL_SRC_GPIO1:
+ case ADAU1373_PLL_SRC_GPIO2:
+ case ADAU1373_PLL_SRC_GPIO3:
+ case ADAU1373_PLL_SRC_GPIO4:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (freq_in < 7813 || freq_in > 27000000)
+ return -EINVAL;
+
+ if (freq_out < 45158000 || freq_out > 49152000)
+ return -EINVAL;
+
+ /* APLL input needs to be >= 8Mhz, so in case freq_in is less we use the
+ * DPLL to get it there. DPLL_out = (DPLL_in / div) * 1024 */
+ while (freq_in < 8000000) {
+ freq_in *= 2;
+ dpll_div++;
+ }
+
+ if (freq_out % freq_in != 0) {
+ /* fout = fin * (r + (n/m)) / x */
+ x = DIV_ROUND_UP(freq_in, 13500000);
+ freq_in /= x;
+ r = freq_out / freq_in;
+ i = freq_out % freq_in;
+ j = gcd(i, freq_in);
+ n = i / j;
+ m = freq_in / j;
+ x--;
+ mode = 1;
+ } else {
+ /* fout = fin / r */
+ r = freq_out / freq_in;
+ n = 0;
+ m = 0;
+ x = 0;
+ mode = 0;
+ }
+
+ if (r < 2 || r > 8 || x > 3 || m > 0xffff || n > 0xffff)
+ return -EINVAL;
+
+ if (dpll_div) {
+ dpll_div = 11 - dpll_div;
+ snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id),
+ ADAU1373_PLL_CTRL6_DPLL_BYPASS, 0);
+ } else {
+ snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id),
+ ADAU1373_PLL_CTRL6_DPLL_BYPASS,
+ ADAU1373_PLL_CTRL6_DPLL_BYPASS);
+ }
+
+ snd_soc_write(codec, ADAU1373_DPLL_CTRL(pll_id),
+ (source << 4) | dpll_div);
+ snd_soc_write(codec, ADAU1373_PLL_CTRL1(pll_id), (m >> 8) & 0xff);
+ snd_soc_write(codec, ADAU1373_PLL_CTRL2(pll_id), m & 0xff);
+ snd_soc_write(codec, ADAU1373_PLL_CTRL3(pll_id), (n >> 8) & 0xff);
+ snd_soc_write(codec, ADAU1373_PLL_CTRL4(pll_id), n & 0xff);
+ snd_soc_write(codec, ADAU1373_PLL_CTRL5(pll_id),
+ (r << 3) | (x << 1) | mode);
+
+ /* Set sysclk to pll_rate / 4 */
+ snd_soc_update_bits(codec, ADAU1373_CLK_SRC_DIV(pll_id), 0x3f, 0x09);
+
+ return 0;
+}
+
+static void adau1373_load_drc_settings(struct snd_soc_codec *codec,
+ unsigned int nr, uint8_t *drc)
+{
+ unsigned int i;
+
+ for (i = 0; i < ADAU1373_DRC_SIZE; ++i)
+ snd_soc_write(codec, ADAU1373_DRC(nr) + i, drc[i]);
+}
+
+static bool adau1373_valid_micbias(enum adau1373_micbias_voltage micbias)
+{
+ switch (micbias) {
+ case ADAU1373_MICBIAS_2_9V:
+ case ADAU1373_MICBIAS_2_2V:
+ case ADAU1373_MICBIAS_2_6V:
+ case ADAU1373_MICBIAS_1_8V:
+ return true;
+ default:
+ break;
+ }
+ return false;
+}
+
+static int adau1373_probe(struct snd_soc_codec *codec)
+{
+ struct adau1373_platform_data *pdata = codec->dev->platform_data;
+ bool lineout_differential = false;
+ unsigned int val;
+ int ret;
+ int i;
+
+ ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
+ if (ret) {
+ dev_err(codec->dev, "failed to set cache I/O: %d\n", ret);
+ return ret;
+ }
+
+ codec->dapm.idle_bias_off = true;
+
+ if (pdata) {
+ if (pdata->num_drc > ARRAY_SIZE(pdata->drc_setting))
+ return -EINVAL;
+
+ if (!adau1373_valid_micbias(pdata->micbias1) ||
+ !adau1373_valid_micbias(pdata->micbias2))
+ return -EINVAL;
+
+ for (i = 0; i < pdata->num_drc; ++i) {
+ adau1373_load_drc_settings(codec, i,
+ pdata->drc_setting[i]);
+ }
+
+ snd_soc_add_controls(codec, adau1373_drc_controls,
+ pdata->num_drc);
+
+ val = 0;
+ for (i = 0; i < 4; ++i) {
+ if (pdata->input_differential[i])
+ val |= BIT(i);
+ }
+ snd_soc_write(codec, ADAU1373_INPUT_MODE, val);
+
+ val = 0;
+ if (pdata->lineout_differential)
+ val |= ADAU1373_OUTPUT_CTRL_LDIFF;
+ if (pdata->lineout_ground_sense)
+ val |= ADAU1373_OUTPUT_CTRL_LNFBEN;
+ snd_soc_write(codec, ADAU1373_OUTPUT_CTRL, val);
+
+ lineout_differential = pdata->lineout_differential;
+
+ snd_soc_write(codec, ADAU1373_EP_CTRL,
+ (pdata->micbias1 << ADAU1373_EP_CTRL_MICBIAS1_OFFSET) |
+ (pdata->micbias2 << ADAU1373_EP_CTRL_MICBIAS2_OFFSET));
+ }
+
+ if (!lineout_differential) {
+ snd_soc_add_controls(codec, adau1373_lineout2_controls,
+ ARRAY_SIZE(adau1373_lineout2_controls));
+ }
+
+ snd_soc_write(codec, ADAU1373_ADC_CTRL,
+ ADAU1373_ADC_CTRL_RESET_FORCE | ADAU1373_ADC_CTRL_PEAK_DETECT);
+
+ return 0;
+}
+
+static int adau1373_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ snd_soc_update_bits(codec, ADAU1373_PWDN_CTRL3,
+ ADAU1373_PWDN_CTRL3_PWR_EN, ADAU1373_PWDN_CTRL3_PWR_EN);
+ break;
+ case SND_SOC_BIAS_OFF:
+ snd_soc_update_bits(codec, ADAU1373_PWDN_CTRL3,
+ ADAU1373_PWDN_CTRL3_PWR_EN, 0);
+ break;
+ }
+ codec->dapm.bias_level = level;
+ return 0;
+}
+
+static int adau1373_remove(struct snd_soc_codec *codec)
+{
+ adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int adau1373_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+ return adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF);
+}
+
+static int adau1373_resume(struct snd_soc_codec *codec)
+{
+ adau1373_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ snd_soc_cache_sync(codec);
+
+ return 0;
+}
+
+static struct snd_soc_codec_driver adau1373_codec_driver = {
+ .probe = adau1373_probe,
+ .remove = adau1373_remove,
+ .suspend = adau1373_suspend,
+ .resume = adau1373_resume,
+ .set_bias_level = adau1373_set_bias_level,
+ .reg_cache_size = ARRAY_SIZE(adau1373_default_regs),
+ .reg_cache_default = adau1373_default_regs,
+ .reg_word_size = sizeof(uint8_t),
+
+ .set_pll = adau1373_set_pll,
+
+ .controls = adau1373_controls,
+ .num_controls = ARRAY_SIZE(adau1373_controls),
+ .dapm_widgets = adau1373_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(adau1373_dapm_widgets),
+ .dapm_routes = adau1373_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(adau1373_dapm_routes),
+};
+
+static int __devinit adau1373_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ struct adau1373 *adau1373;
+ int ret;
+
+ adau1373 = kzalloc(sizeof(*adau1373), GFP_KERNEL);
+ if (!adau1373)
+ return -ENOMEM;
+
+ dev_set_drvdata(&client->dev, adau1373);
+
+ ret = snd_soc_register_codec(&client->dev, &adau1373_codec_driver,
+ adau1373_dai_driver, ARRAY_SIZE(adau1373_dai_driver));
+ if (ret < 0)
+ kfree(adau1373);
+
+ return ret;
+}
+
+static int __devexit adau1373_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ kfree(dev_get_drvdata(&client->dev));
+ return 0;
+}
+
+static const struct i2c_device_id adau1373_i2c_id[] = {
+ { "adau1373", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, adau1373_i2c_id);
+
+static struct i2c_driver adau1373_i2c_driver = {
+ .driver = {
+ .name = "adau1373",
+ .owner = THIS_MODULE,
+ },
+ .probe = adau1373_i2c_probe,
+ .remove = __devexit_p(adau1373_i2c_remove),
+ .id_table = adau1373_i2c_id,
+};
+
+static int __init adau1373_init(void)
+{
+ return i2c_add_driver(&adau1373_i2c_driver);
+}
+module_init(adau1373_init);
+
+static void __exit adau1373_exit(void)
+{
+ i2c_del_driver(&adau1373_i2c_driver);
+}
+module_exit(adau1373_exit);
+
+MODULE_DESCRIPTION("ASoC ADAU1373 driver");
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/adau1373.h b/sound/soc/codecs/adau1373.h
new file mode 100644
index 0000000..c6ab553
--- /dev/null
+++ b/sound/soc/codecs/adau1373.h
@@ -0,0 +1,29 @@
+#ifndef __ADAU1373_H__
+#define __ADAU1373_H__
+
+enum adau1373_pll_src {
+ ADAU1373_PLL_SRC_MCLK1 = 0,
+ ADAU1373_PLL_SRC_BCLK1 = 1,
+ ADAU1373_PLL_SRC_BCLK2 = 2,
+ ADAU1373_PLL_SRC_BCLK3 = 3,
+ ADAU1373_PLL_SRC_LRCLK1 = 4,
+ ADAU1373_PLL_SRC_LRCLK2 = 5,
+ ADAU1373_PLL_SRC_LRCLK3 = 6,
+ ADAU1373_PLL_SRC_GPIO1 = 7,
+ ADAU1373_PLL_SRC_GPIO2 = 8,
+ ADAU1373_PLL_SRC_GPIO3 = 9,
+ ADAU1373_PLL_SRC_GPIO4 = 10,
+ ADAU1373_PLL_SRC_MCLK2 = 11,
+};
+
+enum adau1373_pll {
+ ADAU1373_PLL1 = 0,
+ ADAU1373_PLL2 = 1,
+};
+
+enum adau1373_clk_src {
+ ADAU1373_CLK_SRC_PLL1 = 0,
+ ADAU1373_CLK_SRC_PLL2 = 1,
+};
+
+#endif
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 7e4066e..91130fb 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -20,6 +20,7 @@
#include <linux/regulator/driver.h>
#include <linux/regulator/machine.h>
#include <linux/regulator/consumer.h>
+#include <linux/of_device.h>
#include <sound/core.h>
#include <sound/tlv.h>
#include <sound/pcm.h>
@@ -1436,10 +1437,17 @@
MODULE_DEVICE_TABLE(i2c, sgtl5000_id);
+static const struct of_device_id sgtl5000_dt_ids[] = {
+ { .compatible = "fsl,sgtl5000", },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, sgtl5000_dt_ids);
+
static struct i2c_driver sgtl5000_i2c_driver = {
.driver = {
.name = "sgtl5000",
.owner = THIS_MODULE,
+ .of_match_table = sgtl5000_dt_ids,
},
.probe = sgtl5000_i2c_probe,
.remove = __devexit_p(sgtl5000_i2c_remove),
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index fbd7eb9..5c7def3 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -524,13 +524,17 @@
rate = params_rate(params);
pr_debug("rate: %u\n", rate);
for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++)
- if (interpolation_ratios[i].fs == rate)
+ if (interpolation_ratios[i].fs == rate) {
ir = interpolation_ratios[i].ir;
+ break;
+ }
if (ir < 0)
return -EINVAL;
for (i = 0; mclk_ratios[ir][i].ratio; i++)
- if (mclk_ratios[ir][i].ratio * rate == sta32x->mclk)
+ if (mclk_ratios[ir][i].ratio * rate == sta32x->mclk) {
mcs = mclk_ratios[ir][i].mcs;
+ break;
+ }
if (mcs < 0)
return -EINVAL;
@@ -808,6 +812,7 @@
{
struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
+ sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF);
regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
@@ -867,18 +872,8 @@
static __devexit int sta32x_i2c_remove(struct i2c_client *client)
{
struct sta32x_priv *sta32x = i2c_get_clientdata(client);
- struct snd_soc_codec *codec = sta32x->codec;
- if (codec)
- sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
- regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
-
- if (codec) {
- snd_soc_unregister_codec(&client->dev);
- snd_soc_codec_set_drvdata(codec, NULL);
- }
-
+ snd_soc_unregister_codec(&client->dev);
kfree(sta32x);
return 0;
}
diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c
index bcc2089..4523c4c 100644
--- a/sound/soc/codecs/wm1250-ev1.c
+++ b/sound/soc/codecs/wm1250-ev1.c
@@ -56,8 +56,26 @@
};
static int __devinit wm1250_ev1_probe(struct i2c_client *i2c,
- const struct i2c_device_id *id)
+ const struct i2c_device_id *i2c_id)
{
+ int id, board, rev;
+
+ board = i2c_smbus_read_byte_data(i2c, 0);
+ if (board < 0) {
+ dev_err(&i2c->dev, "Failed to read ID: %d\n", board);
+ return board;
+ }
+
+ id = (board & 0xfe) >> 2;
+ rev = board & 0x3;
+
+ if (id != 1) {
+ dev_err(&i2c->dev, "Unknown board ID %d\n", id);
+ return -ENODEV;
+ }
+
+ dev_info(&i2c->dev, "revision %d\n", rev + 1);
+
return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm1250_ev1,
&wm1250_ev1_dai, 1);
}
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index db0dced..55a4c83 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -20,6 +20,7 @@
#include <linux/platform_device.h>
#include <linux/spi/spi.h>
#include <linux/slab.h>
+#include <linux/of_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -598,6 +599,11 @@
.reg_cache_default =wm8510_reg,
};
+static const struct of_device_id wm8510_of_match[] = {
+ { .compatible = "wlf,wm8510" },
+ { },
+};
+
#if defined(CONFIG_SPI_MASTER)
static int __devinit wm8510_spi_probe(struct spi_device *spi)
{
@@ -628,6 +634,7 @@
.driver = {
.name = "wm8510",
.owner = THIS_MODULE,
+ .of_match_table = wm8510_of_match,
},
.probe = wm8510_spi_probe,
.remove = __devexit_p(wm8510_spi_remove),
@@ -671,6 +678,7 @@
.driver = {
.name = "wm8510-codec",
.owner = THIS_MODULE,
+ .of_match_table = wm8510_of_match,
},
.probe = wm8510_i2c_probe,
.remove = __devexit_p(wm8510_i2c_remove),
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index 4fd4d8d..5355a7a 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -20,6 +20,7 @@
#include <linux/platform_device.h>
#include <linux/regulator/consumer.h>
#include <linux/slab.h>
+#include <linux/of_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -84,7 +85,7 @@
static const struct soc_enum wm8523_zc_count =
SOC_ENUM_SINGLE(WM8523_ZERO_DETECT, 0, 2, wm8523_zd_count_text);
-static const struct snd_kcontrol_new wm8523_snd_controls[] = {
+static const struct snd_kcontrol_new wm8523_controls[] = {
SOC_DOUBLE_R_TLV("Playback Volume", WM8523_DAC_GAINL, WM8523_DAC_GAINR,
0, 448, 0, dac_tlv),
SOC_SINGLE("ZC Switch", WM8523_DAC_CTRL3, 4, 1, 0),
@@ -101,22 +102,11 @@
SND_SOC_DAPM_OUTPUT("LINEVOUTR"),
};
-static const struct snd_soc_dapm_route intercon[] = {
+static const struct snd_soc_dapm_route wm8523_dapm_routes[] = {
{ "LINEVOUTL", NULL, "DAC" },
{ "LINEVOUTR", NULL, "DAC" },
};
-static int wm8523_add_widgets(struct snd_soc_codec *codec)
-{
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
- snd_soc_dapm_new_controls(dapm, wm8523_dapm_widgets,
- ARRAY_SIZE(wm8523_dapm_widgets));
- snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
-
- return 0;
-}
-
static struct {
int value;
int ratio;
@@ -479,10 +469,6 @@
/* Bias level configuration will have done an extra enable */
regulator_bulk_disable(ARRAY_SIZE(wm8523->supplies), wm8523->supplies);
- snd_soc_add_controls(codec, wm8523_snd_controls,
- ARRAY_SIZE(wm8523_snd_controls));
- wm8523_add_widgets(codec);
-
return 0;
err_enable:
@@ -512,6 +498,18 @@
.reg_word_size = sizeof(u16),
.reg_cache_default = wm8523_reg,
.volatile_register = wm8523_volatile_register,
+
+ .controls = wm8523_controls,
+ .num_controls = ARRAY_SIZE(wm8523_controls),
+ .dapm_widgets = wm8523_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(wm8523_dapm_widgets),
+ .dapm_routes = wm8523_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(wm8523_dapm_routes),
+};
+
+static const struct of_device_id wm8523_of_match[] = {
+ { .compatible = "wlf,wm8523" },
+ { },
};
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
@@ -551,8 +549,9 @@
static struct i2c_driver wm8523_i2c_driver = {
.driver = {
- .name = "wm8523-codec",
+ .name = "wm8523",
.owner = THIS_MODULE,
+ .of_match_table = wm8523_of_match,
},
.probe = wm8523_i2c_probe,
.remove = __devexit_p(wm8523_i2c_remove),
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 4bbc0a7..4664c3a 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -26,6 +26,7 @@
#include <linux/platform_device.h>
#include <linux/regulator/consumer.h>
#include <linux/slab.h>
+#include <linux/of_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -907,6 +908,11 @@
.reg_cache_default = wm8580_reg,
};
+static const struct of_device_id wm8580_of_match[] = {
+ { .compatible = "wlf,wm8580" },
+ { },
+};
+
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
static int wm8580_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
@@ -943,8 +949,9 @@
static struct i2c_driver wm8580_i2c_driver = {
.driver = {
- .name = "wm8580-codec",
+ .name = "wm8580",
.owner = THIS_MODULE,
+ .of_match_table = wm8580_of_match,
},
.probe = wm8580_i2c_probe,
.remove = wm8580_i2c_remove,
diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c
index a537e4a..8457d3c 100644
--- a/sound/soc/codecs/wm8711.c
+++ b/sound/soc/codecs/wm8711.c
@@ -21,6 +21,7 @@
#include <linux/platform_device.h>
#include <linux/spi/spi.h>
#include <linux/slab.h>
+#include <linux/of_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -414,6 +415,12 @@
.num_dapm_routes = ARRAY_SIZE(wm8711_intercon),
};
+static const struct of_device_id wm8711_of_match[] = {
+ { .compatible = "wlf,wm8711", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, wm8711_of_match);
+
#if defined(CONFIG_SPI_MASTER)
static int __devinit wm8711_spi_probe(struct spi_device *spi)
{
@@ -443,8 +450,9 @@
static struct spi_driver wm8711_spi_driver = {
.driver = {
- .name = "wm8711-codec",
+ .name = "wm8711",
.owner = THIS_MODULE,
+ .of_match_table = wm8711_of_match,
},
.probe = wm8711_spi_probe,
.remove = __devexit_p(wm8711_spi_remove),
@@ -487,8 +495,9 @@
static struct i2c_driver wm8711_i2c_driver = {
.driver = {
- .name = "wm8711-codec",
+ .name = "wm8711",
.owner = THIS_MODULE,
+ .of_match_table = wm8711_of_match,
},
.probe = wm8711_i2c_probe,
.remove = __devexit_p(wm8711_i2c_remove),
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index 86d4718..04b027e 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -19,6 +19,7 @@
#include <linux/platform_device.h>
#include <linux/spi/spi.h>
#include <linux/slab.h>
+#include <linux/of_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -269,6 +270,12 @@
.num_dapm_routes = ARRAY_SIZE(wm8728_intercon),
};
+static const struct of_device_id wm8728_of_match[] = {
+ { .compatible = "wlf,wm8728", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, wm8728_of_match);
+
#if defined(CONFIG_SPI_MASTER)
static int __devinit wm8728_spi_probe(struct spi_device *spi)
{
@@ -298,8 +305,9 @@
static struct spi_driver wm8728_spi_driver = {
.driver = {
- .name = "wm8728-codec",
+ .name = "wm8728",
.owner = THIS_MODULE,
+ .of_match_table = wm8728_of_match,
},
.probe = wm8728_spi_probe,
.remove = __devexit_p(wm8728_spi_remove),
@@ -342,8 +350,9 @@
static struct i2c_driver wm8728_i2c_driver = {
.driver = {
- .name = "wm8728-codec",
+ .name = "wm8728",
.owner = THIS_MODULE,
+ .of_match_table = wm8728_of_match,
},
.probe = wm8728_i2c_probe,
.remove = __devexit_p(wm8728_i2c_remove),
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 76b4361..f76b6fc 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -22,6 +22,7 @@
#include <linux/platform_device.h>
#include <linux/regulator/consumer.h>
#include <linux/spi/spi.h>
+#include <linux/of_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -607,6 +608,13 @@
.num_dapm_routes = ARRAY_SIZE(wm8731_intercon),
};
+static const struct of_device_id wm8731_of_match[] = {
+ { .compatible = "wlf,wm8731", },
+ { }
+};
+
+MODULE_DEVICE_TABLE(of, wm8731_of_match);
+
#if defined(CONFIG_SPI_MASTER)
static int __devinit wm8731_spi_probe(struct spi_device *spi)
{
@@ -638,6 +646,7 @@
.driver = {
.name = "wm8731",
.owner = THIS_MODULE,
+ .of_match_table = wm8731_of_match,
},
.probe = wm8731_spi_probe,
.remove = __devexit_p(wm8731_spi_remove),
@@ -682,6 +691,7 @@
.driver = {
.name = "wm8731",
.owner = THIS_MODULE,
+ .of_match_table = wm8731_of_match,
},
.probe = wm8731_i2c_probe,
.remove = __devexit_p(wm8731_i2c_remove),
diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c
index 30c67d0..f6aef58 100644
--- a/sound/soc/codecs/wm8737.c
+++ b/sound/soc/codecs/wm8737.c
@@ -20,6 +20,7 @@
#include <linux/regulator/consumer.h>
#include <linux/spi/spi.h>
#include <linux/slab.h>
+#include <linux/of_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -634,6 +635,13 @@
.reg_cache_default = wm8737_reg,
};
+static const struct of_device_id wm8737_of_match[] = {
+ { .compatible = "wlf,wm8737", },
+ { }
+};
+
+MODULE_DEVICE_TABLE(of, wm8737_of_match);
+
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
static __devinit int wm8737_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
@@ -673,6 +681,7 @@
.driver = {
.name = "wm8737",
.owner = THIS_MODULE,
+ .of_match_table = wm8737_of_match,
},
.probe = wm8737_i2c_probe,
.remove = __devexit_p(wm8737_i2c_remove),
@@ -711,6 +720,7 @@
.driver = {
.name = "wm8737",
.owner = THIS_MODULE,
+ .of_match_table = wm8737_of_match,
},
.probe = wm8737_spi_probe,
.remove = __devexit_p(wm8737_spi_remove),
diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c
index 25af901..78c9e5a 100644
--- a/sound/soc/codecs/wm8741.c
+++ b/sound/soc/codecs/wm8741.c
@@ -17,9 +17,11 @@
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/i2c.h>
+#include <linux/spi/spi.h>
#include <linux/platform_device.h>
#include <linux/regulator/consumer.h>
#include <linux/slab.h>
+#include <linux/of_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -422,17 +424,35 @@
{
struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec);
int ret = 0;
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++)
+ wm8741->supplies[i].supply = wm8741_supply_names[i];
+
+ ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8741->supplies),
+ wm8741->supplies);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to request supplies: %d\n", ret);
+ goto err;
+ }
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(wm8741->supplies),
+ wm8741->supplies);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to enable supplies: %d\n", ret);
+ goto err_get;
+ }
ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8741->control_type);
if (ret != 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
- return ret;
+ goto err_enable;
}
ret = wm8741_reset(codec);
if (ret < 0) {
dev_err(codec->dev, "Failed to issue reset\n");
- return ret;
+ goto err_enable;
}
/* Change some default settings - latch VU */
@@ -451,58 +471,61 @@
dev_dbg(codec->dev, "Successful registration\n");
return ret;
+
+err_enable:
+ regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies);
+err_get:
+ regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies);
+err:
+ return ret;
+}
+
+static int wm8741_remove(struct snd_soc_codec *codec)
+{
+ struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec);
+
+ regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies);
+ regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies);
+
+ return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_wm8741 = {
.probe = wm8741_probe,
+ .remove = wm8741_remove,
.resume = wm8741_resume,
.reg_cache_size = ARRAY_SIZE(wm8741_reg_defaults),
.reg_word_size = sizeof(u16),
.reg_cache_default = wm8741_reg_defaults,
};
+static const struct of_device_id wm8741_of_match[] = {
+ { .compatible = "wlf,wm8741", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, wm8741_of_match);
+
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
static int wm8741_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct wm8741_priv *wm8741;
- int ret, i;
+ int ret;
wm8741 = kzalloc(sizeof(struct wm8741_priv), GFP_KERNEL);
if (wm8741 == NULL)
return -ENOMEM;
- for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++)
- wm8741->supplies[i].supply = wm8741_supply_names[i];
-
- ret = regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8741->supplies),
- wm8741->supplies);
- if (ret != 0) {
- dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret);
- goto err;
- }
-
- ret = regulator_bulk_enable(ARRAY_SIZE(wm8741->supplies),
- wm8741->supplies);
- if (ret != 0) {
- dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret);
- goto err_get;
- }
-
i2c_set_clientdata(i2c, wm8741);
wm8741->control_type = SND_SOC_I2C;
- ret = snd_soc_register_codec(&i2c->dev,
- &soc_codec_dev_wm8741, &wm8741_dai, 1);
- if (ret < 0)
- goto err_enable;
+ ret = snd_soc_register_codec(&i2c->dev,
+ &soc_codec_dev_wm8741, &wm8741_dai, 1);
+ if (ret != 0)
+ goto err;
+
return ret;
-err_enable:
- regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies);
-
-err_get:
- regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies);
err:
kfree(wm8741);
return ret;
@@ -510,10 +533,7 @@
static int wm8741_i2c_remove(struct i2c_client *client)
{
- struct wm8741_priv *wm8741 = i2c_get_clientdata(client);
-
snd_soc_unregister_codec(&client->dev);
- regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies);
kfree(i2c_get_clientdata(client));
return 0;
}
@@ -526,8 +546,9 @@
static struct i2c_driver wm8741_i2c_driver = {
.driver = {
- .name = "wm8741-codec",
+ .name = "wm8741",
.owner = THIS_MODULE,
+ .of_match_table = wm8741_of_match,
},
.probe = wm8741_i2c_probe,
.remove = wm8741_i2c_remove,
@@ -535,6 +556,44 @@
};
#endif
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit wm8741_spi_probe(struct spi_device *spi)
+{
+ struct wm8741_priv *wm8741;
+ int ret;
+
+ wm8741 = kzalloc(sizeof(struct wm8741_priv), GFP_KERNEL);
+ if (wm8741 == NULL)
+ return -ENOMEM;
+
+ wm8741->control_type = SND_SOC_SPI;
+ spi_set_drvdata(spi, wm8741);
+
+ ret = snd_soc_register_codec(&spi->dev,
+ &soc_codec_dev_wm8741, &wm8741_dai, 1);
+ if (ret < 0)
+ kfree(wm8741);
+ return ret;
+}
+
+static int __devexit wm8741_spi_remove(struct spi_device *spi)
+{
+ snd_soc_unregister_codec(&spi->dev);
+ kfree(spi_get_drvdata(spi));
+ return 0;
+}
+
+static struct spi_driver wm8741_spi_driver = {
+ .driver = {
+ .name = "wm8741",
+ .owner = THIS_MODULE,
+ .of_match_table = wm8741_of_match,
+ },
+ .probe = wm8741_spi_probe,
+ .remove = __devexit_p(wm8741_spi_remove),
+};
+#endif /* CONFIG_SPI_MASTER */
+
static int __init wm8741_modinit(void)
{
int ret = 0;
@@ -544,6 +603,13 @@
if (ret != 0)
pr_err("Failed to register WM8741 I2C driver: %d\n", ret);
#endif
+#if defined(CONFIG_SPI_MASTER)
+ ret = spi_register_driver(&wm8741_spi_driver);
+ if (ret != 0) {
+ printk(KERN_ERR "Failed to register wm8741 SPI driver: %d\n",
+ ret);
+ }
+#endif
return ret;
}
@@ -551,6 +617,9 @@
static void __exit wm8741_exit(void)
{
+#if defined(CONFIG_SPI_MASTER)
+ spi_unregister_driver(&wm8741_spi_driver);
+#endif
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
i2c_del_driver(&wm8741_i2c_driver);
#endif
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index d0003cc..15f0372 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -21,6 +21,7 @@
#include <linux/platform_device.h>
#include <linux/spi/spi.h>
#include <linux/slab.h>
+#include <linux/of_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -751,6 +752,13 @@
.reg_cache_default = wm8750_reg,
};
+static const struct of_device_id wm8750_of_match[] = {
+ { .compatible = "wlf,wm8750", },
+ { .compatible = "wlf,wm8987", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, wm8750_of_match);
+
#if defined(CONFIG_SPI_MASTER)
static int __devinit wm8750_spi_probe(struct spi_device *spi)
{
@@ -787,8 +795,9 @@
static struct spi_driver wm8750_spi_driver = {
.driver = {
- .name = "wm8750-codec",
+ .name = "wm8750",
.owner = THIS_MODULE,
+ .of_match_table = wm8750_of_match,
},
.id_table = wm8750_spi_ids,
.probe = wm8750_spi_probe,
@@ -833,8 +842,9 @@
static struct i2c_driver wm8750_i2c_driver = {
.driver = {
- .name = "wm8750-codec",
+ .name = "wm8750",
.owner = THIS_MODULE,
+ .of_match_table = wm8750_of_match,
},
.probe = wm8750_i2c_probe,
.remove = __devexit_p(wm8750_i2c_remove),
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index ffa2ffe..fe04a10 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -38,6 +38,7 @@
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/i2c.h>
+#include <linux/of_device.h>
#include <linux/platform_device.h>
#include <linux/spi/spi.h>
#include <linux/slab.h>
@@ -1490,6 +1491,12 @@
.reg_cache_default = wm8753_reg,
};
+static const struct of_device_id wm8753_of_match[] = {
+ { .compatible = "wlf,wm8753", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, wm8753_of_match);
+
#if defined(CONFIG_SPI_MASTER)
static int __devinit wm8753_spi_probe(struct spi_device *spi)
{
@@ -1519,8 +1526,9 @@
static struct spi_driver wm8753_spi_driver = {
.driver = {
- .name = "wm8753-codec",
+ .name = "wm8753",
.owner = THIS_MODULE,
+ .of_match_table = wm8753_of_match,
},
.probe = wm8753_spi_probe,
.remove = __devexit_p(wm8753_spi_remove),
@@ -1563,8 +1571,9 @@
static struct i2c_driver wm8753_i2c_driver = {
.driver = {
- .name = "wm8753-codec",
+ .name = "wm8753",
.owner = THIS_MODULE,
+ .of_match_table = wm8753_of_match,
},
.probe = wm8753_i2c_probe,
.remove = __devexit_p(wm8753_i2c_remove),
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 1725550..8e397b2 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -63,6 +63,8 @@
int fll_fref;
int fll_fout;
+ u16 dsp2_ena;
+
struct delayed_work mic_work;
struct snd_soc_jack *jack;
@@ -837,7 +839,7 @@
[40] = { 0x00FF, 0x01FF, 0x0000 }, /* R40 - SPKOUTL volume */
[41] = { 0x00FF, 0x01FF, 0x0000 }, /* R41 - SPKOUTR volume */
- [47] = { 0x000F, 0x0000, 0x0000 }, /* R47 - Thermal Shutdown Status */
+ [47] = { 0x000F, 0x0000, 0xFFFF }, /* R47 - Thermal Shutdown Status */
[48] = { 0x7EC7, 0x7E07, 0xFFFF }, /* R48 - Additional Control (4) */
[49] = { 0x00D3, 0x00D7, 0xFFFF }, /* R49 - Class D Control 1 */
[51] = { 0x0047, 0x0047, 0x0000 }, /* R51 - Class D Control 2 */
@@ -965,7 +967,7 @@
[584] = { 0x002D, 0x002D, 0x0000 }, /* R584 - IRQ Debounce */
[586] = { 0xC000, 0xC000, 0x0000 }, /* R586 - MICINT Source Pol */
[768] = { 0x0001, 0x0001, 0x0000 }, /* R768 - DSP2 Power Management */
- [1037] = { 0x0000, 0x003F, 0x0000 }, /* R1037 - DSP2_ExecControl */
+ [1037] = { 0x0000, 0x003F, 0xFFFF }, /* R1037 - DSP2_ExecControl */
[4096] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4096 - Write Sequencer 0 */
[4097] = { 0x00FF, 0x00FF, 0x0000 }, /* R4097 - Write Sequencer 1 */
[4098] = { 0x070F, 0x070F, 0x0000 }, /* R4098 - Write Sequencer 2 */
@@ -1986,6 +1988,122 @@
};
static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
+static int wm8962_dsp2_write_config(struct snd_soc_codec *codec)
+{
+ return 0;
+}
+
+static int wm8962_dsp2_set_enable(struct snd_soc_codec *codec, u16 val)
+{
+ u16 adcl = snd_soc_read(codec, WM8962_LEFT_ADC_VOLUME);
+ u16 adcr = snd_soc_read(codec, WM8962_RIGHT_ADC_VOLUME);
+ u16 dac = snd_soc_read(codec, WM8962_ADC_DAC_CONTROL_1);
+
+ /* Mute the ADCs and DACs */
+ snd_soc_write(codec, WM8962_LEFT_ADC_VOLUME, 0);
+ snd_soc_write(codec, WM8962_RIGHT_ADC_VOLUME, WM8962_ADC_VU);
+ snd_soc_update_bits(codec, WM8962_ADC_DAC_CONTROL_1,
+ WM8962_DAC_MUTE, WM8962_DAC_MUTE);
+
+ snd_soc_write(codec, WM8962_SOUNDSTAGE_ENABLES_0, val);
+
+ /* Restore the ADCs and DACs */
+ snd_soc_write(codec, WM8962_LEFT_ADC_VOLUME, adcl);
+ snd_soc_write(codec, WM8962_RIGHT_ADC_VOLUME, adcr);
+ snd_soc_update_bits(codec, WM8962_ADC_DAC_CONTROL_1,
+ WM8962_DAC_MUTE, dac);
+
+ return 0;
+}
+
+static int wm8962_dsp2_start(struct snd_soc_codec *codec)
+{
+ struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
+
+ wm8962_dsp2_write_config(codec);
+
+ snd_soc_write(codec, WM8962_DSP2_EXECCONTROL, WM8962_DSP2_RUNR);
+
+ wm8962_dsp2_set_enable(codec, wm8962->dsp2_ena);
+
+ return 0;
+}
+
+static int wm8962_dsp2_stop(struct snd_soc_codec *codec)
+{
+ wm8962_dsp2_set_enable(codec, 0);
+
+ snd_soc_write(codec, WM8962_DSP2_EXECCONTROL, WM8962_DSP2_STOP);
+
+ return 0;
+}
+
+#define WM8962_DSP2_ENABLE(xname, xshift) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = wm8962_dsp2_ena_info, \
+ .get = wm8962_dsp2_ena_get, .put = wm8962_dsp2_ena_put, \
+ .private_value = xshift }
+
+static int wm8962_dsp2_ena_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+
+ return 0;
+}
+
+static int wm8962_dsp2_ena_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int shift = kcontrol->private_value;
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
+
+ ucontrol->value.integer.value[0] = !!(wm8962->dsp2_ena & 1 << shift);
+
+ return 0;
+}
+
+static int wm8962_dsp2_ena_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int shift = kcontrol->private_value;
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
+ int old = wm8962->dsp2_ena;
+ int ret = 0;
+ int dsp2_running = snd_soc_read(codec, WM8962_DSP2_POWER_MANAGEMENT) &
+ WM8962_DSP2_ENA;
+
+ mutex_lock(&codec->mutex);
+
+ if (ucontrol->value.integer.value[0])
+ wm8962->dsp2_ena |= 1 << shift;
+ else
+ wm8962->dsp2_ena &= ~(1 << shift);
+
+ if (wm8962->dsp2_ena == old)
+ goto out;
+
+ ret = 1;
+
+ if (dsp2_running) {
+ if (wm8962->dsp2_ena)
+ wm8962_dsp2_set_enable(codec, wm8962->dsp2_ena);
+ else
+ wm8962_dsp2_stop(codec);
+ }
+
+out:
+ mutex_unlock(&codec->mutex);
+
+ return ret;
+}
+
/* The VU bits for the headphones are in a different register to the mute
* bits and only take effect on the PGA if it is actually powered.
*/
@@ -2049,6 +2167,14 @@
static const struct soc_enum cap_hpf_mode =
SOC_ENUM_SINGLE(WM8962_ADC_DAC_CONTROL_2, 10, 2, cap_hpf_mode_text);
+
+static const char *cap_lhpf_mode_text[] = {
+ "LPF", "HPF"
+};
+
+static const struct soc_enum cap_lhpf_mode =
+ SOC_ENUM_SINGLE(WM8962_LHPF1, 1, 2, cap_lhpf_mode_text);
+
static const struct snd_kcontrol_new wm8962_snd_controls[] = {
SOC_DOUBLE("Input Mixer Switch", WM8962_INPUT_MIXER_CONTROL_1, 3, 2, 1, 1),
@@ -2077,6 +2203,8 @@
SOC_SINGLE("Capture HPF Switch", WM8962_ADC_DAC_CONTROL_1, 0, 1, 1),
SOC_ENUM("Capture HPF Mode", cap_hpf_mode),
SOC_SINGLE("Capture HPF Cutoff", WM8962_ADC_DAC_CONTROL_2, 7, 7, 0),
+SOC_SINGLE("Capture LHPF Switch", WM8962_LHPF1, 0, 1, 0),
+SOC_ENUM("Capture LHPF Mode", cap_lhpf_mode),
SOC_DOUBLE_R_TLV("Sidetone Volume", WM8962_DAC_DSP_MIXING_1,
WM8962_DAC_DSP_MIXING_2, 4, 12, 0, st_tlv),
@@ -2134,6 +2262,11 @@
WM8962_EQL_B4_GAIN_SHIFT, 31, 0, eq_tlv),
SOC_DOUBLE_R_TLV("EQ5 Volume", WM8962_EQ3, WM8962_EQ23,
WM8962_EQL_B5_GAIN_SHIFT, 31, 0, eq_tlv),
+
+WM8962_DSP2_ENABLE("VSS Switch", WM8962_VSS_ENA_SHIFT),
+WM8962_DSP2_ENABLE("HPF1 Switch", WM8962_HPF1_ENA_SHIFT),
+WM8962_DSP2_ENABLE("HPF2 Switch", WM8962_HPF2_ENA_SHIFT),
+WM8962_DSP2_ENABLE("HD Bass Switch", WM8962_HDBASS_ENA_SHIFT),
};
static const struct snd_kcontrol_new wm8962_spk_mono_controls[] = {
@@ -2395,6 +2528,31 @@
}
}
+static int dsp2_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ if (wm8962->dsp2_ena)
+ wm8962_dsp2_start(codec);
+ break;
+
+ case SND_SOC_DAPM_PRE_PMD:
+ if (wm8962->dsp2_ena)
+ wm8962_dsp2_stop(codec);
+ break;
+
+ default:
+ BUG();
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
static const char *st_text[] = { "None", "Right", "Left" };
static const struct soc_enum str_enum =
@@ -2517,6 +2675,9 @@
SND_SOC_DAPM_SUPPLY("Charge Pump", WM8962_CHARGE_PUMP_1, 0, 0, cp_event,
SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_SUPPLY("TOCLK", WM8962_ADDITIONAL_CONTROL_1, 0, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY_S("DSP2", 1, WM8962_DSP2_POWER_MANAGEMENT,
+ WM8962_DSP2_ENA_SHIFT, 0, dsp2_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_MIXER("INPGAL", WM8962_LEFT_INPUT_PGA_CONTROL, 4, 0,
inpgal, ARRAY_SIZE(inpgal)),
@@ -2612,11 +2773,13 @@
{ "ADCL", NULL, "TOCLK" },
{ "ADCL", NULL, "MIXINL" },
{ "ADCL", NULL, "DMIC" },
+ { "ADCL", NULL, "DSP2" },
{ "ADCR", NULL, "SYSCLK" },
{ "ADCR", NULL, "TOCLK" },
{ "ADCR", NULL, "MIXINR" },
{ "ADCR", NULL, "DMIC" },
+ { "ADCR", NULL, "DSP2" },
{ "STL", "Left", "ADCL" },
{ "STL", "Right", "ADCR" },
@@ -2628,11 +2791,13 @@
{ "DACL", NULL, "TOCLK" },
{ "DACL", NULL, "Beep" },
{ "DACL", NULL, "STL" },
+ { "DACL", NULL, "DSP2" },
{ "DACR", NULL, "SYSCLK" },
{ "DACR", NULL, "TOCLK" },
{ "DACR", NULL, "Beep" },
{ "DACR", NULL, "STR" },
+ { "DACR", NULL, "DSP2" },
{ "HPMIXL", "IN4L Switch", "IN4L" },
{ "HPMIXL", "IN4R Switch", "IN4R" },
@@ -3403,12 +3568,16 @@
struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
int mask;
int active;
+ int reg;
mask = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2_MASK);
active = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2);
active &= ~mask;
+ if (!active)
+ return IRQ_NONE;
+
/* Acknowledge the interrupts */
snd_soc_write(codec, WM8962_INTERRUPT_STATUS_2, active);
@@ -3420,9 +3589,21 @@
if (active & WM8962_FIFOS_ERR_EINT)
dev_err(codec->dev, "FIFO error\n");
- if (active & WM8962_TEMP_SHUT_EINT)
+ if (active & WM8962_TEMP_SHUT_EINT) {
dev_crit(codec->dev, "Thermal shutdown\n");
+ reg = snd_soc_read(codec, WM8962_THERMAL_SHUTDOWN_STATUS);
+
+ if (reg & WM8962_TEMP_ERR_HP)
+ dev_crit(codec->dev, "Headphone thermal error\n");
+ if (reg & WM8962_TEMP_WARN_HP)
+ dev_crit(codec->dev, "Headphone thermal warning\n");
+ if (reg & WM8962_TEMP_ERR_SPK)
+ dev_crit(codec->dev, "Speaker thermal error\n");
+ if (reg & WM8962_TEMP_WARN_SPK)
+ dev_crit(codec->dev, "Speaker thermal warning\n");
+ }
+
if (active & (WM8962_MICSCD_EINT | WM8962_MICD_EINT)) {
dev_dbg(codec->dev, "Microphone event detected\n");
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 6e85b88..eec8e14 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -847,6 +847,7 @@
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_SUPPLY("TOCLK", WM8993_CLOCKING_1, 14, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8993_CLOCKING_3, 0, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("VMID", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_ADC("ADCL", NULL, WM8993_POWER_MANAGEMENT_2, 1, 0),
SND_SOC_DAPM_ADC("ADCR", NULL, WM8993_POWER_MANAGEMENT_2, 0, 0),
@@ -880,6 +881,9 @@
};
static const struct snd_soc_dapm_route routes[] = {
+ { "MICBIAS1", NULL, "VMID" },
+ { "MICBIAS2", NULL, "VMID" },
+
{ "ADCL", NULL, "CLK_SYS" },
{ "ADCL", NULL, "CLK_DSP" },
{ "ADCR", NULL, "CLK_SYS" },
@@ -1433,7 +1437,8 @@
int ret, i, val;
wm8993->hubs_data.hp_startup_mode = 1;
- wm8993->hubs_data.dcs_codes = -2;
+ wm8993->hubs_data.dcs_codes_l = -2;
+ wm8993->hubs_data.dcs_codes_r = -2;
wm8993->hubs_data.series_startup = 1;
ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C);
diff --git a/sound/soc/codecs/wm8994-tables.c b/sound/soc/codecs/wm8994-tables.c
index a87adbd..df5a8b9 100644
--- a/sound/soc/codecs/wm8994-tables.c
+++ b/sound/soc/codecs/wm8994-tables.c
@@ -1073,8 +1073,8 @@
{ 0x0000, 0x0000 }, /* R1069 */
{ 0x0000, 0x0000 }, /* R1070 */
{ 0x0000, 0x0000 }, /* R1071 */
- { 0x0000, 0x0000 }, /* R1072 */
- { 0x0000, 0x0000 }, /* R1073 */
+ { 0x006F, 0x006F }, /* R1072 - AIF1 DAC1 Noise Gate */
+ { 0x006F, 0x006F }, /* R1073 - AIF1 DAC2 Noise Gate */
{ 0x0000, 0x0000 }, /* R1074 */
{ 0x0000, 0x0000 }, /* R1075 */
{ 0x0000, 0x0000 }, /* R1076 */
@@ -1329,7 +1329,7 @@
{ 0x0000, 0x0000 }, /* R1325 */
{ 0x0000, 0x0000 }, /* R1326 */
{ 0x0000, 0x0000 }, /* R1327 */
- { 0x0000, 0x0000 }, /* R1328 */
+ { 0x006F, 0x006F }, /* R1328 - AIF2 DAC Noise Gate */
{ 0x0000, 0x0000 }, /* R1329 */
{ 0x0000, 0x0000 }, /* R1330 */
{ 0x0000, 0x0000 }, /* R1331 */
@@ -1635,8 +1635,8 @@
0x0000, /* R58 - MICBIAS */
0x000D, /* R59 - LDO 1 */
0x0003, /* R60 - LDO 2 */
- 0x0000, /* R61 */
- 0x0000, /* R62 */
+ 0x0039, /* R61 - MICBIAS1 */
+ 0x0039, /* R62 - MICBIAS2 */
0x0000, /* R63 */
0x0000, /* R64 */
0x0000, /* R65 */
@@ -2646,8 +2646,8 @@
0x0000, /* R1069 */
0x0000, /* R1070 */
0x0000, /* R1071 */
- 0x0000, /* R1072 */
- 0x0000, /* R1073 */
+ 0x0068, /* R1072 - AIF1 DAC1 Noise Gate */
+ 0x0068, /* R1073 - AIF1 DAC2 Noise Gate */
0x0000, /* R1074 */
0x0000, /* R1075 */
0x0000, /* R1076 */
@@ -2902,7 +2902,7 @@
0x0000, /* R1325 */
0x0000, /* R1326 */
0x0000, /* R1327 */
- 0x0000, /* R1328 */
+ 0x0068, /* R1328 - AIF2 DAC Noise Gate */
0x0000, /* R1329 */
0x0000, /* R1330 */
0x0000, /* R1331 */
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index b393f9f..e537267 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -107,6 +107,7 @@
case WM8994_LDO_2:
case WM8958_DSP2_EXECCONTROL:
case WM8958_MIC_DETECT_3:
+ case WM8994_DC_SERVO_4E:
return 1;
default:
return 0;
@@ -281,6 +282,7 @@
static const DECLARE_TLV_DB_SCALE(st_tlv, -3600, 300, 0);
static const DECLARE_TLV_DB_SCALE(wm8994_3d_tlv, -1600, 183, 0);
static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
+static const DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0);
#define WM8994_DRC_SWITCH(xname, reg, shift) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
@@ -660,8 +662,45 @@
eq_tlv),
};
+static const char *wm8958_ng_text[] = {
+ "30ms", "125ms", "250ms", "500ms",
+};
+
+static const struct soc_enum wm8958_aif1dac1_ng_hold =
+ SOC_ENUM_SINGLE(WM8958_AIF1_DAC1_NOISE_GATE,
+ WM8958_AIF1DAC1_NG_THR_SHIFT, 4, wm8958_ng_text);
+
+static const struct soc_enum wm8958_aif1dac2_ng_hold =
+ SOC_ENUM_SINGLE(WM8958_AIF1_DAC2_NOISE_GATE,
+ WM8958_AIF1DAC2_NG_THR_SHIFT, 4, wm8958_ng_text);
+
+static const struct soc_enum wm8958_aif2dac_ng_hold =
+ SOC_ENUM_SINGLE(WM8958_AIF2_DAC_NOISE_GATE,
+ WM8958_AIF2DAC_NG_THR_SHIFT, 4, wm8958_ng_text);
+
static const struct snd_kcontrol_new wm8958_snd_controls[] = {
SOC_SINGLE_TLV("AIF3 Boost Volume", WM8958_AIF3_CONTROL_2, 10, 3, 0, aif_tlv),
+
+SOC_SINGLE("AIF1DAC1 Noise Gate Switch", WM8958_AIF1_DAC1_NOISE_GATE,
+ WM8958_AIF1DAC1_NG_ENA_SHIFT, 1, 0),
+SOC_ENUM("AIF1DAC1 Noise Gate Hold Time", wm8958_aif1dac1_ng_hold),
+SOC_SINGLE_TLV("AIF1DAC1 Noise Gate Threshold Volume",
+ WM8958_AIF1_DAC1_NOISE_GATE, WM8958_AIF1DAC1_NG_THR_SHIFT,
+ 7, 1, ng_tlv),
+
+SOC_SINGLE("AIF1DAC2 Noise Gate Switch", WM8958_AIF1_DAC2_NOISE_GATE,
+ WM8958_AIF1DAC2_NG_ENA_SHIFT, 1, 0),
+SOC_ENUM("AIF1DAC2 Noise Gate Hold Time", wm8958_aif1dac2_ng_hold),
+SOC_SINGLE_TLV("AIF1DAC2 Noise Gate Threshold Volume",
+ WM8958_AIF1_DAC2_NOISE_GATE, WM8958_AIF1DAC2_NG_THR_SHIFT,
+ 7, 1, ng_tlv),
+
+SOC_SINGLE("AIF2DAC Noise Gate Switch", WM8958_AIF2_DAC_NOISE_GATE,
+ WM8958_AIF2DAC_NG_ENA_SHIFT, 1, 0),
+SOC_ENUM("AIF2DAC Noise Gate Hold Time", wm8958_aif2dac_ng_hold),
+SOC_SINGLE_TLV("AIF2DAC Noise Gate Threshold Volume",
+ WM8958_AIF2_DAC_NOISE_GATE, WM8958_AIF2DAC_NG_THR_SHIFT,
+ 7, 1, ng_tlv),
};
static int clk_sys_event(struct snd_soc_dapm_widget *w,
@@ -681,6 +720,97 @@
return 0;
}
+static void vmid_reference(struct snd_soc_codec *codec)
+{
+ struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+
+ wm8994->vmid_refcount++;
+
+ dev_dbg(codec->dev, "Referencing VMID, refcount is now %d\n",
+ wm8994->vmid_refcount);
+
+ if (wm8994->vmid_refcount == 1) {
+ /* Startup bias, VMID ramp & buffer */
+ snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
+ WM8994_STARTUP_BIAS_ENA |
+ WM8994_VMID_BUF_ENA |
+ WM8994_VMID_RAMP_MASK,
+ WM8994_STARTUP_BIAS_ENA |
+ WM8994_VMID_BUF_ENA |
+ (0x11 << WM8994_VMID_RAMP_SHIFT));
+
+ /* Main bias enable, VMID=2x40k */
+ snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1,
+ WM8994_BIAS_ENA |
+ WM8994_VMID_SEL_MASK,
+ WM8994_BIAS_ENA | 0x2);
+
+ msleep(20);
+ }
+}
+
+static void vmid_dereference(struct snd_soc_codec *codec)
+{
+ struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+
+ wm8994->vmid_refcount--;
+
+ dev_dbg(codec->dev, "Dereferencing VMID, refcount is now %d\n",
+ wm8994->vmid_refcount);
+
+ if (wm8994->vmid_refcount == 0) {
+ /* Switch over to startup biases */
+ snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
+ WM8994_BIAS_SRC |
+ WM8994_STARTUP_BIAS_ENA |
+ WM8994_VMID_BUF_ENA |
+ WM8994_VMID_RAMP_MASK,
+ WM8994_BIAS_SRC |
+ WM8994_STARTUP_BIAS_ENA |
+ WM8994_VMID_BUF_ENA |
+ (1 << WM8994_VMID_RAMP_SHIFT));
+
+ /* Disable main biases */
+ snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1,
+ WM8994_BIAS_ENA |
+ WM8994_VMID_SEL_MASK, 0);
+
+ /* Discharge line */
+ snd_soc_update_bits(codec, WM8994_ANTIPOP_1,
+ WM8994_LINEOUT1_DISCH |
+ WM8994_LINEOUT2_DISCH,
+ WM8994_LINEOUT1_DISCH |
+ WM8994_LINEOUT2_DISCH);
+
+ msleep(5);
+
+ /* Switch off startup biases */
+ snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
+ WM8994_BIAS_SRC |
+ WM8994_STARTUP_BIAS_ENA |
+ WM8994_VMID_BUF_ENA |
+ WM8994_VMID_RAMP_MASK, 0);
+ }
+}
+
+static int vmid_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ vmid_reference(codec);
+ break;
+
+ case SND_SOC_DAPM_POST_PMD:
+ vmid_dereference(codec);
+ break;
+ }
+
+ return 0;
+}
+
static void wm8994_update_class_w(struct snd_soc_codec *codec)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
@@ -1208,6 +1338,8 @@
SND_SOC_DAPM_SUPPLY_S("MICBIAS Supply", 1, SND_SOC_NOPM, 0, 0, micbias_ev,
SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_SUPPLY("VMID", SND_SOC_NOPM, 0, 0, vmid_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
@@ -1525,6 +1657,8 @@
static const struct snd_soc_dapm_route wm8994_intercon[] = {
{ "AIF2DACL", NULL, "AIF2DAC Mux" },
{ "AIF2DACR", NULL, "AIF2DAC Mux" },
+ { "MICBIAS1", NULL, "VMID" },
+ { "MICBIAS2", NULL, "VMID" },
};
static const struct snd_soc_dapm_route wm8958_intercon[] = {
@@ -1629,10 +1763,12 @@
unsigned int freq_in, unsigned int freq_out)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+ struct wm8994 *control = codec->control_data;
int reg_offset, ret;
struct fll_div fll;
u16 reg, aif1, aif2;
unsigned long timeout;
+ bool was_enabled;
aif1 = snd_soc_read(codec, WM8994_AIF1_CLOCKING_1)
& WM8994_AIF1CLK_ENA;
@@ -1653,6 +1789,9 @@
return -EINVAL;
}
+ reg = snd_soc_read(codec, WM8994_FLL1_CONTROL_1 + reg_offset);
+ was_enabled = reg & WM8994_FLL1_ENA;
+
switch (src) {
case 0:
/* Allow no source specification when stopping */
@@ -1719,6 +1858,21 @@
/* Enable (with fractional mode if required) */
if (freq_out) {
+ /* Enable VMID if we need it */
+ if (!was_enabled) {
+ switch (control->type) {
+ case WM8994:
+ vmid_reference(codec);
+ break;
+ case WM8958:
+ if (wm8994->revision < 1)
+ vmid_reference(codec);
+ break;
+ default:
+ break;
+ }
+ }
+
if (fll.k)
reg = WM8994_FLL1_ENA | WM8994_FLL1_FRAC;
else
@@ -1736,6 +1890,20 @@
} else {
msleep(5);
}
+ } else {
+ if (was_enabled) {
+ switch (control->type) {
+ case WM8994:
+ vmid_dereference(codec);
+ break;
+ case WM8958:
+ if (wm8994->revision < 1)
+ vmid_dereference(codec);
+ break;
+ default:
+ break;
+ }
+ }
}
wm8994->fll[id].in = freq_in;
@@ -1852,9 +2020,6 @@
break;
case SND_SOC_BIAS_PREPARE:
- /* VMID=2x40k */
- snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1,
- WM8994_VMID_SEL_MASK, 0x2);
break;
case SND_SOC_BIAS_STANDBY:
@@ -1896,65 +2061,13 @@
WM8994_LINEOUT2_DISCH,
WM8994_LINEOUT1_DISCH |
WM8994_LINEOUT2_DISCH);
-
- /* Startup bias, VMID ramp & buffer */
- snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
- WM8994_STARTUP_BIAS_ENA |
- WM8994_VMID_BUF_ENA |
- WM8994_VMID_RAMP_MASK,
- WM8994_STARTUP_BIAS_ENA |
- WM8994_VMID_BUF_ENA |
- (0x11 << WM8994_VMID_RAMP_SHIFT));
-
- /* Main bias enable, VMID=2x40k */
- snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1,
- WM8994_BIAS_ENA |
- WM8994_VMID_SEL_MASK,
- WM8994_BIAS_ENA | 0x2);
-
- msleep(20);
}
- /* VMID=2x500k */
- snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1,
- WM8994_VMID_SEL_MASK, 0x4);
break;
case SND_SOC_BIAS_OFF:
if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) {
- /* Switch over to startup biases */
- snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
- WM8994_BIAS_SRC |
- WM8994_STARTUP_BIAS_ENA |
- WM8994_VMID_BUF_ENA |
- WM8994_VMID_RAMP_MASK,
- WM8994_BIAS_SRC |
- WM8994_STARTUP_BIAS_ENA |
- WM8994_VMID_BUF_ENA |
- (1 << WM8994_VMID_RAMP_SHIFT));
-
- /* Disable main biases */
- snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1,
- WM8994_BIAS_ENA |
- WM8994_VMID_SEL_MASK, 0);
-
- /* Discharge line */
- snd_soc_update_bits(codec, WM8994_ANTIPOP_1,
- WM8994_LINEOUT1_DISCH |
- WM8994_LINEOUT2_DISCH,
- WM8994_LINEOUT1_DISCH |
- WM8994_LINEOUT2_DISCH);
-
- msleep(5);
-
- /* Switch off startup biases */
- snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
- WM8994_BIAS_SRC |
- WM8994_STARTUP_BIAS_ENA |
- WM8994_VMID_BUF_ENA |
- WM8994_VMID_RAMP_MASK, 0);
-
wm8994->cur_fw = NULL;
pm_runtime_put(codec->dev);
@@ -2384,6 +2497,21 @@
return snd_soc_update_bits(codec, reg, mask, val);
}
+static int wm8994_aif2_probe(struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+
+ /* Disable the pulls on the AIF if we're using it to save power. */
+ snd_soc_update_bits(codec, WM8994_GPIO_3,
+ WM8994_GPN_PU | WM8994_GPN_PD, 0);
+ snd_soc_update_bits(codec, WM8994_GPIO_4,
+ WM8994_GPN_PU | WM8994_GPN_PD, 0);
+ snd_soc_update_bits(codec, WM8994_GPIO_5,
+ WM8994_GPN_PU | WM8994_GPN_PD, 0);
+
+ return 0;
+}
+
#define WM8994_RATES SNDRV_PCM_RATE_8000_96000
#define WM8994_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
@@ -2451,6 +2579,7 @@
.rates = WM8994_RATES,
.formats = WM8994_FORMATS,
},
+ .probe = wm8994_aif2_probe,
.ops = &wm8994_aif2_dai_ops,
},
{
@@ -2916,6 +3045,24 @@
return IRQ_HANDLED;
}
+static irqreturn_t wm8994_temp_warn(int irq, void *data)
+{
+ struct snd_soc_codec *codec = data;
+
+ dev_err(codec->dev, "Thermal warning\n");
+
+ return IRQ_HANDLED;
+}
+
+static irqreturn_t wm8994_temp_shut(int irq, void *data)
+{
+ struct snd_soc_codec *codec = data;
+
+ dev_crit(codec->dev, "Thermal shutdown\n");
+
+ return IRQ_HANDLED;
+}
+
static int wm8994_codec_probe(struct snd_soc_codec *codec)
{
struct wm8994 *control;
@@ -2972,13 +3119,14 @@
switch (wm8994->revision) {
case 2:
case 3:
- wm8994->hubs.dcs_codes = -5;
+ wm8994->hubs.dcs_codes_l = -5;
+ wm8994->hubs.dcs_codes_r = -5;
wm8994->hubs.hp_startup_mode = 1;
wm8994->hubs.dcs_readback_mode = 1;
wm8994->hubs.series_startup = 1;
break;
default:
- wm8994->hubs.dcs_readback_mode = 1;
+ wm8994->hubs.dcs_readback_mode = 2;
break;
}
break;
@@ -2993,6 +3141,10 @@
wm8994_request_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR,
wm8994_fifo_error, "FIFO error", codec);
+ wm8994_request_irq(wm8994->control_data, WM8994_IRQ_TEMP_WARN,
+ wm8994_temp_warn, "Thermal warning", codec);
+ wm8994_request_irq(wm8994->control_data, WM8994_IRQ_TEMP_SHUT,
+ wm8994_temp_shut, "Thermal shutdown", codec);
ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_DCS_DONE,
wm_hubs_dcs_done, "DC servo done",
@@ -3257,6 +3409,8 @@
wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE,
&wm8994->hubs);
wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec);
+ wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_SHUT, codec);
+ wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_WARN, codec);
err:
kfree(wm8994);
return ret;
@@ -3279,6 +3433,8 @@
wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE,
&wm8994->hubs);
wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec);
+ wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_SHUT, codec);
+ wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_WARN, codec);
switch (control->type) {
case WM8994:
diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h
index 1ab2266..f4f1355 100644
--- a/sound/soc/codecs/wm8994.h
+++ b/sound/soc/codecs/wm8994.h
@@ -83,6 +83,8 @@
struct completion fll_locked[2];
bool fll_locked_irq;
+ int vmid_refcount;
+
int dac_rates[2];
int lrclk_shared[2];
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index 0cdb9d1..e5e4607 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -414,6 +414,7 @@
static const DECLARE_TLV_DB_SCALE(out_tlv, -900, 75, 0);
static const DECLARE_TLV_DB_SCALE(spk_tlv, -900, 150, 0);
static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
+static const DECLARE_TLV_DB_SCALE(threedstereo_tlv, -1600, 183, 1);
static const char *sidetone_hpf_text[] = {
"2.9kHz", "1.5kHz", "735Hz", "403Hz", "196Hz", "98Hz", "49Hz"
@@ -608,6 +609,14 @@
SOC_SINGLE("DAC Soft Mute Switch", WM8996_DAC_SOFTMUTE, 1, 1, 0),
SOC_SINGLE("DAC Slow Soft Mute Switch", WM8996_DAC_SOFTMUTE, 0, 1, 0),
+SOC_SINGLE("DSP1 3D Stereo Switch", WM8996_DSP1_RX_FILTERS_2, 8, 1, 0),
+SOC_SINGLE("DSP2 3D Stereo Switch", WM8996_DSP2_RX_FILTERS_2, 8, 1, 0),
+
+SOC_SINGLE_TLV("DSP1 3D Stereo Volume", WM8996_DSP1_RX_FILTERS_2, 10, 15,
+ 0, threedstereo_tlv),
+SOC_SINGLE_TLV("DSP2 3D Stereo Volume", WM8996_DSP2_RX_FILTERS_2, 10, 15,
+ 0, threedstereo_tlv),
+
SOC_DOUBLE_TLV("Digital Output 1 Volume", WM8996_DAC1_HPOUT1_VOLUME, 0, 4,
8, 0, out_digital_tlv),
SOC_DOUBLE_TLV("Digital Output 2 Volume", WM8996_DAC2_HPOUT2_VOLUME, 0, 4,
@@ -982,6 +991,8 @@
SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_SUPPLY("LDO2", WM8996_POWER_MANAGEMENT_2, 1, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICB1 Audio", WM8996_MICBIAS_1, 4, 1, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICB2 Audio", WM8996_MICBIAS_2, 4, 1, NULL, 0),
SND_SOC_DAPM_MICBIAS("MICB2", WM8996_POWER_MANAGEMENT_1, 9, 0),
SND_SOC_DAPM_MICBIAS("MICB1", WM8996_POWER_MANAGEMENT_1, 8, 0),
@@ -1137,7 +1148,9 @@
{ "Charge Pump", NULL, "SYSCLK" },
{ "MICB1", NULL, "LDO2" },
+ { "MICB1", NULL, "MICB1 Audio" },
{ "MICB2", NULL, "LDO2" },
+ { "MICB2", NULL, "MICB2 Audio" },
{ "IN1L PGA", NULL, "IN2LN" },
{ "IN1L PGA", NULL, "IN2LP" },
@@ -2412,6 +2425,9 @@
}
irq_val &= ~snd_soc_read(codec, WM8996_INTERRUPT_STATUS_2_MASK);
+ if (!irq_val)
+ return IRQ_NONE;
+
snd_soc_write(codec, WM8996_INTERRUPT_STATUS_2, irq_val);
if (irq_val & (WM8996_DCS_DONE_01_EINT | WM8996_DCS_DONE_23_EINT)) {
@@ -2430,10 +2446,7 @@
if (irq_val & WM8996_MICD_EINT)
wm8996_micd(codec);
- if (irq_val)
- return IRQ_HANDLED;
- else
- return IRQ_NONE;
+ return IRQ_HANDLED;
}
static irqreturn_t wm8996_edge_irq(int irq, void *data)
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index e763c54..ca8ce03 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -18,6 +18,7 @@
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
+#include <linux/mfd/wm8994/registers.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -116,14 +117,23 @@
{
struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
s8 offset;
- u16 reg, reg_l, reg_r, dcs_cfg;
+ u16 reg, reg_l, reg_r, dcs_cfg, dcs_reg;
+
+ switch (hubs->dcs_readback_mode) {
+ case 2:
+ dcs_reg = WM8994_DC_SERVO_4E;
+ break;
+ default:
+ dcs_reg = WM8993_DC_SERVO_3;
+ break;
+ }
/* If we're using a digital only path and have a previously
* callibrated DC servo offset stored then use that. */
if (hubs->class_w && hubs->class_w_dcs) {
dev_dbg(codec->dev, "Using cached DC servo offset %x\n",
hubs->class_w_dcs);
- snd_soc_write(codec, WM8993_DC_SERVO_3, hubs->class_w_dcs);
+ snd_soc_write(codec, dcs_reg, hubs->class_w_dcs);
wait_for_dc_servo(codec,
WM8993_DCS_TRIG_DAC_WR_0 |
WM8993_DCS_TRIG_DAC_WR_1);
@@ -154,8 +164,9 @@
reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2)
& WM8993_DCS_INTEG_CHAN_1_MASK;
break;
+ case 2:
case 1:
- reg = snd_soc_read(codec, WM8993_DC_SERVO_3);
+ reg = snd_soc_read(codec, dcs_reg);
reg_r = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK)
>> WM8993_DCS_DAC_WR_VAL_1_SHIFT;
reg_l = reg & WM8993_DCS_DAC_WR_VAL_0_MASK;
@@ -168,24 +179,25 @@
dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r);
/* Apply correction to DC servo result */
- if (hubs->dcs_codes) {
- dev_dbg(codec->dev, "Applying %d code DC servo correction\n",
- hubs->dcs_codes);
+ if (hubs->dcs_codes_l || hubs->dcs_codes_r) {
+ dev_dbg(codec->dev,
+ "Applying %d/%d code DC servo correction\n",
+ hubs->dcs_codes_l, hubs->dcs_codes_r);
/* HPOUT1R */
offset = reg_r;
- offset += hubs->dcs_codes;
+ offset += hubs->dcs_codes_r;
dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
/* HPOUT1L */
offset = reg_l;
- offset += hubs->dcs_codes;
+ offset += hubs->dcs_codes_l;
dcs_cfg |= (u8)offset;
dev_dbg(codec->dev, "DCS result: %x\n", dcs_cfg);
/* Do it */
- snd_soc_write(codec, WM8993_DC_SERVO_3, dcs_cfg);
+ snd_soc_write(codec, dcs_reg, dcs_cfg);
wait_for_dc_servo(codec,
WM8993_DCS_TRIG_DAC_WR_0 |
WM8993_DCS_TRIG_DAC_WR_1);
@@ -217,7 +229,7 @@
/* If we're applying an offset correction then updating the
* callibration would be likely to introduce further offsets. */
- if (hubs->dcs_codes || hubs->no_series_update)
+ if (hubs->dcs_codes_l || hubs->dcs_codes_r || hubs->no_series_update)
return ret;
/* Only need to do this if the outputs are active */
@@ -699,6 +711,11 @@
{ "IN1L PGA", "IN1LP Switch", "IN1LP" },
{ "IN1L PGA", "IN1LN Switch", "IN1LN" },
+ { "IN1L PGA", NULL, "VMID" },
+ { "IN1R PGA", NULL, "VMID" },
+ { "IN2L PGA", NULL, "VMID" },
+ { "IN2R PGA", NULL, "VMID" },
+
{ "IN1R PGA", "IN1RP Switch", "IN1RP" },
{ "IN1R PGA", "IN1RN Switch", "IN1RN" },
@@ -716,12 +733,14 @@
{ "MIXINL", NULL, "Direct Voice" },
{ "MIXINL", NULL, "IN1LP" },
{ "MIXINL", NULL, "Left Output Mixer" },
+ { "MIXINL", NULL, "VMID" },
{ "MIXINR", "IN1R Switch", "IN1R PGA" },
{ "MIXINR", "IN2R Switch", "IN2R PGA" },
{ "MIXINR", NULL, "Direct Voice" },
{ "MIXINR", NULL, "IN1RP" },
{ "MIXINR", NULL, "Right Output Mixer" },
+ { "MIXINR", NULL, "VMID" },
{ "ADCL", NULL, "MIXINL" },
{ "ADCR", NULL, "MIXINR" },
@@ -752,6 +771,7 @@
{ "Earpiece Mixer", "Left Output Switch", "Left Output PGA" },
{ "Earpiece Mixer", "Right Output Switch", "Right Output PGA" },
+ { "Earpiece Driver", NULL, "VMID" },
{ "Earpiece Driver", NULL, "Earpiece Mixer" },
{ "HPOUT2N", NULL, "Earpiece Driver" },
{ "HPOUT2P", NULL, "Earpiece Driver" },
@@ -774,9 +794,11 @@
{ "SPKR Boost", "SPKR Switch", "SPKR" },
{ "SPKR Boost", "SPKL Switch", "SPKL" },
+ { "SPKL Driver", NULL, "VMID" },
{ "SPKL Driver", NULL, "SPKL Boost" },
{ "SPKL Driver", NULL, "CLK_SYS" },
+ { "SPKR Driver", NULL, "VMID" },
{ "SPKR Driver", NULL, "SPKR Boost" },
{ "SPKR Driver", NULL, "CLK_SYS" },
@@ -790,12 +812,18 @@
{ "Headphone PGA", NULL, "Left Headphone Mux" },
{ "Headphone PGA", NULL, "Right Headphone Mux" },
+ { "Headphone PGA", NULL, "VMID" },
{ "Headphone PGA", NULL, "CLK_SYS" },
{ "Headphone PGA", NULL, "Headphone Supply" },
{ "HPOUT1L", NULL, "Headphone PGA" },
{ "HPOUT1R", NULL, "Headphone PGA" },
+ { "LINEOUT1N Driver", NULL, "VMID" },
+ { "LINEOUT1P Driver", NULL, "VMID" },
+ { "LINEOUT2N Driver", NULL, "VMID" },
+ { "LINEOUT2P Driver", NULL, "VMID" },
+
{ "LINEOUT1N", NULL, "LINEOUT1N Driver" },
{ "LINEOUT1P", NULL, "LINEOUT1P Driver" },
{ "LINEOUT2N", NULL, "LINEOUT2N Driver" },
diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h
index 676b125..c674c7a 100644
--- a/sound/soc/codecs/wm_hubs.h
+++ b/sound/soc/codecs/wm_hubs.h
@@ -23,7 +23,8 @@
/* This *must* be the first element of the codec->private_data struct */
struct wm_hubs_data {
- int dcs_codes;
+ int dcs_codes_l;
+ int dcs_codes_r;
int dcs_readback_mode;
int hp_startup_mode;
int series_startup;
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 8566238..7173df25 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -732,16 +732,19 @@
davinci_hw_param(dev, substream->stream);
switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_U8:
case SNDRV_PCM_FORMAT_S8:
dma_params->data_type = 1;
word_length = DAVINCI_AUDIO_WORD_8;
break;
+ case SNDRV_PCM_FORMAT_U16_LE:
case SNDRV_PCM_FORMAT_S16_LE:
dma_params->data_type = 2;
word_length = DAVINCI_AUDIO_WORD_16;
break;
+ case SNDRV_PCM_FORMAT_U32_LE:
case SNDRV_PCM_FORMAT_S32_LE:
dma_params->data_type = 4;
word_length = DAVINCI_AUDIO_WORD_32;
@@ -818,6 +821,13 @@
};
+#define DAVINCI_MCASP_PCM_FMTS (SNDRV_PCM_FMTBIT_S8 | \
+ SNDRV_PCM_FMTBIT_U8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_U16_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE | \
+ SNDRV_PCM_FMTBIT_U32_LE)
+
static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
{
.name = "davinci-mcasp.0",
@@ -825,17 +835,13 @@
.channels_min = 2,
.channels_max = 2,
.rates = DAVINCI_MCASP_RATES,
- .formats = SNDRV_PCM_FMTBIT_S8 |
- SNDRV_PCM_FMTBIT_S16_LE |
- SNDRV_PCM_FMTBIT_S32_LE,
+ .formats = DAVINCI_MCASP_PCM_FMTS,
},
.capture = {
.channels_min = 2,
.channels_max = 2,
.rates = DAVINCI_MCASP_RATES,
- .formats = SNDRV_PCM_FMTBIT_S8 |
- SNDRV_PCM_FMTBIT_S16_LE |
- SNDRV_PCM_FMTBIT_S32_LE,
+ .formats = DAVINCI_MCASP_PCM_FMTS,
},
.ops = &davinci_mcasp_dai_ops,
@@ -846,7 +852,7 @@
.channels_min = 1,
.channels_max = 384,
.rates = DAVINCI_MCASP_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .formats = DAVINCI_MCASP_PCM_FMTS,
},
.ops = &davinci_mcasp_dai_ops,
},
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index cb50598..ef15402 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -297,7 +297,6 @@
static int fsl_dma_new(struct snd_soc_pcm_runtime *rtd)
{
struct snd_card *card = rtd->card->snd_card;
- struct snd_soc_dai *dai = rtd->cpu_dai;
struct snd_pcm *pcm = rtd->pcm;
static u64 fsl_dma_dmamask = DMA_BIT_MASK(36);
int ret;
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index d48afea..06ac2b9 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -289,16 +289,6 @@
*/
if (!ssi_private->playback && !ssi_private->capture) {
struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
- int ret;
-
- /* The 'name' should not have any slashes in it. */
- ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0,
- ssi_private->name, ssi_private);
- if (ret < 0) {
- dev_err(substream->pcm->card->dev,
- "could not claim irq %u\n", ssi_private->irq);
- return ret;
- }
/*
* Section 16.5 of the MPC8610 reference manual says that the
@@ -522,15 +512,12 @@
ssi_private->second_stream = NULL;
/*
- * If this is the last active substream, disable the SSI and release
- * the IRQ.
+ * If this is the last active substream, disable the SSI.
*/
if (!ssi_private->playback && !ssi_private->capture) {
struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
-
- free_irq(ssi_private->irq, ssi_private);
}
}
@@ -675,17 +662,30 @@
ret = of_address_to_resource(np, 0, &res);
if (ret) {
dev_err(&pdev->dev, "could not determine device resources\n");
- kfree(ssi_private);
- return ret;
+ goto error_kmalloc;
}
ssi_private->ssi = of_iomap(np, 0);
if (!ssi_private->ssi) {
dev_err(&pdev->dev, "could not map device resources\n");
- kfree(ssi_private);
- return -ENOMEM;
+ ret = -ENOMEM;
+ goto error_kmalloc;
}
ssi_private->ssi_phys = res.start;
+
ssi_private->irq = irq_of_parse_and_map(np, 0);
+ if (ssi_private->irq == NO_IRQ) {
+ dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
+ ret = -ENXIO;
+ goto error_iomap;
+ }
+
+ /* The 'name' should not have any slashes in it. */
+ ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, ssi_private->name,
+ ssi_private);
+ if (ret < 0) {
+ dev_err(&pdev->dev, "could not claim irq %u\n", ssi_private->irq);
+ goto error_irqmap;
+ }
/* Are the RX and the TX clocks locked? */
if (of_find_property(np, "fsl,ssi-asynchronous", NULL))
@@ -711,7 +711,7 @@
if (ret) {
dev_err(&pdev->dev, "could not create sysfs %s file\n",
ssi_private->dev_attr.attr.name);
- goto error;
+ goto error_irq;
}
/* Register with ASoC */
@@ -720,7 +720,7 @@
ret = snd_soc_register_dai(&pdev->dev, &ssi_private->cpu_dai_drv);
if (ret) {
dev_err(&pdev->dev, "failed to register DAI: %d\n", ret);
- goto error;
+ goto error_dev;
}
/* Trigger the machine driver's probe function. The platform driver
@@ -741,18 +741,28 @@
if (IS_ERR(ssi_private->pdev)) {
ret = PTR_ERR(ssi_private->pdev);
dev_err(&pdev->dev, "failed to register platform: %d\n", ret);
- goto error;
+ goto error_dai;
}
return 0;
-error:
+error_dai:
snd_soc_unregister_dai(&pdev->dev);
+
+error_dev:
dev_set_drvdata(&pdev->dev, NULL);
- if (dev_attr)
- device_remove_file(&pdev->dev, dev_attr);
+ device_remove_file(&pdev->dev, dev_attr);
+
+error_irq:
+ free_irq(ssi_private->irq, ssi_private);
+
+error_irqmap:
irq_dispose_mapping(ssi_private->irq);
+
+error_iomap:
iounmap(ssi_private->ssi);
+
+error_kmalloc:
kfree(ssi_private);
return ret;
@@ -766,6 +776,9 @@
snd_soc_unregister_dai(&pdev->dev);
device_remove_file(&pdev->dev, &ssi_private->dev_attr);
+ free_irq(ssi_private->irq, ssi_private);
+ irq_dispose_mapping(ssi_private->irq);
+
kfree(ssi_private);
dev_set_drvdata(&pdev->dev, NULL);
diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c
index 309c59e..ac790e8 100644
--- a/sound/soc/imx/imx-pcm-fiq.c
+++ b/sound/soc/imx/imx-pcm-fiq.c
@@ -243,23 +243,22 @@
struct snd_card *card = rtd->card->snd_card;
struct snd_soc_dai *dai = rtd->cpu_dai;
struct snd_pcm *pcm = rtd->pcm;
+ struct snd_pcm_substream *substream;
int ret;
ret = imx_pcm_new(rtd);
if (ret)
return ret;
- if (dai->driver->playback.channels_min) {
- struct snd_pcm_substream *substream =
- pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+ substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+ if (substream) {
struct snd_dma_buffer *buf = &substream->dma_buffer;
imx_ssi_fiq_tx_buffer = (unsigned long)buf->area;
}
- if (dai->driver->capture.channels_min) {
- struct snd_pcm_substream *substream =
- pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
+ substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
+ if (substream) {
struct snd_dma_buffer *buf = &substream->dma_buffer;
imx_ssi_fiq_rx_buffer = (unsigned long)buf->area;
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 10a8e27..4297cb6 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -357,8 +357,8 @@
struct snd_pcm_runtime *runtime = substream->runtime;
int ret;
- ret = dma_mmap_coherent(NULL, vma, runtime->dma_area,
- runtime->dma_addr, runtime->dma_bytes);
+ ret = dma_mmap_writecombine(substream->pcm->card->dev, vma,
+ runtime->dma_area, runtime->dma_addr, runtime->dma_bytes);
pr_debug("%s: ret: %d %p 0x%08x 0x%08x\n", __func__, ret,
runtime->dma_area,
@@ -399,14 +399,14 @@
card->dev->dma_mask = &imx_pcm_dmamask;
if (!card->dev->coherent_dma_mask)
card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
- if (dai->driver->playback.channels_min) {
+ if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
ret = imx_pcm_preallocate_dma_buffer(pcm,
SNDRV_PCM_STREAM_PLAYBACK);
if (ret)
goto out;
}
- if (dai->driver->capture.channels_min) {
+ if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
ret = imx_pcm_preallocate_dma_buffer(pcm,
SNDRV_PCM_STREAM_CAPTURE);
if (ret)
diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig
new file mode 100644
index 0000000..e4ba8d5
--- /dev/null
+++ b/sound/soc/mxs/Kconfig
@@ -0,0 +1,20 @@
+menuconfig SND_MXS_SOC
+ tristate "SoC Audio for Freescale MXS CPUs"
+ depends on ARCH_MXS
+ select SND_PCM
+ help
+ Say Y or M if you want to add support for codecs attached to
+ the MXS SAIF interface.
+
+
+if SND_MXS_SOC
+
+config SND_SOC_MXS_SGTL5000
+ tristate "SoC Audio support for i.MX boards with sgtl5000"
+ depends on I2C
+ select SND_SOC_SGTL5000
+ help
+ Say Y if you want to add support for SoC audio on an MXS board with
+ a sgtl5000 codec.
+
+endif # SND_MXS_SOC
diff --git a/sound/soc/mxs/Makefile b/sound/soc/mxs/Makefile
new file mode 100644
index 0000000..565b5b5
--- /dev/null
+++ b/sound/soc/mxs/Makefile
@@ -0,0 +1,10 @@
+# MXS Platform Support
+snd-soc-mxs-objs := mxs-saif.o
+snd-soc-mxs-pcm-objs := mxs-pcm.o
+
+obj-$(CONFIG_SND_MXS_SOC) += snd-soc-mxs.o snd-soc-mxs-pcm.o
+
+# i.MX Machine Support
+snd-soc-mxs-sgtl5000-objs := mxs-sgtl5000.o
+
+obj-$(CONFIG_SND_SOC_MXS_SGTL5000) += snd-soc-mxs-sgtl5000.o
diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c
new file mode 100644
index 0000000..dea5aa4
--- /dev/null
+++ b/sound/soc/mxs/mxs-pcm.c
@@ -0,0 +1,359 @@
+/*
+ * Copyright (C) 2011 Freescale Semiconductor, Inc. All Rights Reserved.
+ *
+ * Based on sound/soc/imx/imx-pcm-dma-mx2.c
+ *
+ * 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.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/device.h>
+#include <linux/dma-mapping.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dmaengine.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <mach/dma.h>
+#include "mxs-pcm.h"
+
+static struct snd_pcm_hardware snd_mxs_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_INTERLEAVED,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S20_3LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 2,
+ .channels_max = 2,
+ .period_bytes_min = 32,
+ .period_bytes_max = 8192,
+ .periods_min = 1,
+ .periods_max = 52,
+ .buffer_bytes_max = 64 * 1024,
+ .fifo_size = 32,
+
+};
+
+static void audio_dma_irq(void *data)
+{
+ struct snd_pcm_substream *substream = (struct snd_pcm_substream *)data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+
+ iprtd->offset += iprtd->period_bytes;
+ iprtd->offset %= iprtd->period_bytes * iprtd->periods;
+ snd_pcm_period_elapsed(substream);
+}
+
+static bool filter(struct dma_chan *chan, void *param)
+{
+ struct mxs_pcm_runtime_data *iprtd = param;
+ struct mxs_pcm_dma_params *dma_params = iprtd->dma_params;
+
+ if (!mxs_dma_is_apbx(chan))
+ return false;
+
+ if (chan->chan_id != dma_params->chan_num)
+ return false;
+
+ chan->private = &iprtd->dma_data;
+
+ return true;
+}
+
+static int mxs_dma_alloc(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+ dma_cap_mask_t mask;
+
+ iprtd->dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+
+ dma_cap_zero(mask);
+ dma_cap_set(DMA_SLAVE, mask);
+ iprtd->dma_data.chan_irq = iprtd->dma_params->chan_irq;
+ iprtd->dma_chan = dma_request_channel(mask, filter, iprtd);
+ if (!iprtd->dma_chan)
+ return -EINVAL;
+
+ return 0;
+}
+
+static int snd_mxs_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+ unsigned long dma_addr;
+ struct dma_chan *chan;
+ int ret;
+
+ ret = mxs_dma_alloc(substream, params);
+ if (ret)
+ return ret;
+ chan = iprtd->dma_chan;
+
+ iprtd->size = params_buffer_bytes(params);
+ iprtd->periods = params_periods(params);
+ iprtd->period_bytes = params_period_bytes(params);
+ iprtd->offset = 0;
+ iprtd->period_time = HZ / (params_rate(params) /
+ params_period_size(params));
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+ dma_addr = runtime->dma_addr;
+
+ iprtd->buf = substream->dma_buffer.area;
+
+ iprtd->desc = chan->device->device_prep_dma_cyclic(chan, dma_addr,
+ iprtd->period_bytes * iprtd->periods,
+ iprtd->period_bytes,
+ substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ DMA_TO_DEVICE : DMA_FROM_DEVICE);
+ if (!iprtd->desc) {
+ dev_err(&chan->dev->device, "cannot prepare slave dma\n");
+ return -EINVAL;
+ }
+
+ iprtd->desc->callback = audio_dma_irq;
+ iprtd->desc->callback_param = substream;
+
+ return 0;
+}
+
+static int snd_mxs_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+
+ if (iprtd->dma_chan) {
+ dma_release_channel(iprtd->dma_chan);
+ iprtd->dma_chan = NULL;
+ }
+
+ return 0;
+}
+
+static int snd_mxs_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ dmaengine_submit(iprtd->desc);
+
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ dmaengine_terminate_all(iprtd->dma_chan);
+
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static snd_pcm_uframes_t snd_mxs_pcm_pointer(
+ struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+
+ return bytes_to_frames(substream->runtime, iprtd->offset);
+}
+
+static int snd_mxs_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct mxs_pcm_runtime_data *iprtd;
+ int ret;
+
+ iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL);
+ if (iprtd == NULL)
+ return -ENOMEM;
+ runtime->private_data = iprtd;
+
+ ret = snd_pcm_hw_constraint_integer(substream->runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0) {
+ kfree(iprtd);
+ return ret;
+ }
+
+ snd_soc_set_runtime_hwparams(substream, &snd_mxs_hardware);
+
+ return 0;
+}
+
+static int snd_mxs_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+
+ kfree(iprtd);
+
+ return 0;
+}
+
+static int snd_mxs_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ return dma_mmap_writecombine(substream->pcm->card->dev, vma,
+ runtime->dma_area,
+ runtime->dma_addr,
+ runtime->dma_bytes);
+}
+
+static struct snd_pcm_ops mxs_pcm_ops = {
+ .open = snd_mxs_open,
+ .close = snd_mxs_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_mxs_pcm_hw_params,
+ .hw_free = snd_mxs_pcm_hw_free,
+ .trigger = snd_mxs_pcm_trigger,
+ .pointer = snd_mxs_pcm_pointer,
+ .mmap = snd_mxs_pcm_mmap,
+};
+
+static int mxs_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size = snd_mxs_hardware.buffer_bytes_max;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = pcm->card->dev;
+ buf->private_data = NULL;
+ buf->area = dma_alloc_writecombine(pcm->card->dev, size,
+ &buf->addr, GFP_KERNEL);
+ if (!buf->area)
+ return -ENOMEM;
+ buf->bytes = size;
+
+ return 0;
+}
+
+static u64 mxs_pcm_dmamask = DMA_BIT_MASK(32);
+static int mxs_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_pcm *pcm = rtd->pcm;
+ int ret = 0;
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &mxs_pcm_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+
+ if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
+ ret = mxs_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret)
+ goto out;
+ }
+
+ if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+ ret = mxs_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_CAPTURE);
+ if (ret)
+ goto out;
+ }
+
+out:
+ return ret;
+}
+
+static void mxs_pcm_free(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ continue;
+
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+
+ dma_free_writecombine(pcm->card->dev, buf->bytes,
+ buf->area, buf->addr);
+ buf->area = NULL;
+ }
+}
+
+static struct snd_soc_platform_driver mxs_soc_platform = {
+ .ops = &mxs_pcm_ops,
+ .pcm_new = mxs_pcm_new,
+ .pcm_free = mxs_pcm_free,
+};
+
+static int __devinit mxs_soc_platform_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_platform(&pdev->dev, &mxs_soc_platform);
+}
+
+static int __devexit mxs_soc_platform_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+
+ return 0;
+}
+
+static struct platform_driver mxs_pcm_driver = {
+ .driver = {
+ .name = "mxs-pcm-audio",
+ .owner = THIS_MODULE,
+ },
+ .probe = mxs_soc_platform_probe,
+ .remove = __devexit_p(mxs_soc_platform_remove),
+};
+
+static int __init snd_mxs_pcm_init(void)
+{
+ return platform_driver_register(&mxs_pcm_driver);
+}
+module_init(snd_mxs_pcm_init);
+
+static void __exit snd_mxs_pcm_exit(void)
+{
+ platform_driver_unregister(&mxs_pcm_driver);
+}
+module_exit(snd_mxs_pcm_exit);
diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h
new file mode 100644
index 0000000..f55ac4f
--- /dev/null
+++ b/sound/soc/mxs/mxs-pcm.h
@@ -0,0 +1,43 @@
+/*
+ * Copyright (C) 2011 Freescale Semiconductor, Inc. All Rights Reserved.
+ *
+ * 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.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef _MXS_PCM_H
+#define _MXS_PCM_H
+
+#include <mach/dma.h>
+
+struct mxs_pcm_dma_params {
+ int chan_irq;
+ int chan_num;
+};
+
+struct mxs_pcm_runtime_data {
+ int period_bytes;
+ int periods;
+ int dma;
+ unsigned long offset;
+ unsigned long size;
+ void *buf;
+ int period_time;
+ struct dma_async_tx_descriptor *desc;
+ struct dma_chan *dma_chan;
+ struct mxs_dma_data dma_data;
+ struct mxs_pcm_dma_params *dma_params;
+};
+
+#endif
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
new file mode 100644
index 0000000..af5734f
--- /dev/null
+++ b/sound/soc/mxs/mxs-saif.c
@@ -0,0 +1,680 @@
+/*
+ * Copyright 2011 Freescale Semiconductor, Inc.
+ *
+ * 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.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <mach/dma.h>
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/mxs.h>
+
+#include "mxs-saif.h"
+
+static struct mxs_saif *mxs_saif[2];
+
+static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+
+ switch (clk_id) {
+ case MXS_SAIF_MCLK:
+ saif->mclk = freq;
+ break;
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+/*
+ * Set SAIF clock and MCLK
+ */
+static int mxs_saif_set_clk(struct mxs_saif *saif,
+ unsigned int mclk,
+ unsigned int rate)
+{
+ u32 scr;
+ int ret;
+
+ scr = __raw_readl(saif->base + SAIF_CTRL);
+ scr &= ~BM_SAIF_CTRL_BITCLK_MULT_RATE;
+ scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE;
+
+ /*
+ * Set SAIF clock
+ *
+ * The SAIF clock should be either 384*fs or 512*fs.
+ * If MCLK is used, the SAIF clk ratio need to match mclk ratio.
+ * For 32x mclk, set saif clk as 512*fs.
+ * For 48x mclk, set saif clk as 384*fs.
+ *
+ * If MCLK is not used, we just set saif clk to 512*fs.
+ */
+ if (saif->mclk_in_use) {
+ if (mclk % 32 == 0) {
+ scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE;
+ ret = clk_set_rate(saif->clk, 512 * rate);
+ } else if (mclk % 48 == 0) {
+ scr |= BM_SAIF_CTRL_BITCLK_BASE_RATE;
+ ret = clk_set_rate(saif->clk, 384 * rate);
+ } else {
+ /* SAIF MCLK should be either 32x or 48x */
+ return -EINVAL;
+ }
+ } else {
+ ret = clk_set_rate(saif->clk, 512 * rate);
+ scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE;
+ }
+
+ if (ret)
+ return ret;
+
+ if (!saif->mclk_in_use) {
+ __raw_writel(scr, saif->base + SAIF_CTRL);
+ return 0;
+ }
+
+ /*
+ * Program the over-sample rate for MCLK output
+ *
+ * The available MCLK range is 32x, 48x... 512x. The rate
+ * could be from 8kHz to 192kH.
+ */
+ switch (mclk / rate) {
+ case 32:
+ scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(4);
+ break;
+ case 64:
+ scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(3);
+ break;
+ case 128:
+ scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(2);
+ break;
+ case 256:
+ scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(1);
+ break;
+ case 512:
+ scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(0);
+ break;
+ case 48:
+ scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(3);
+ break;
+ case 96:
+ scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(2);
+ break;
+ case 192:
+ scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(1);
+ break;
+ case 384:
+ scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(0);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ __raw_writel(scr, saif->base + SAIF_CTRL);
+
+ return 0;
+}
+
+/*
+ * Put and disable MCLK.
+ */
+int mxs_saif_put_mclk(unsigned int saif_id)
+{
+ struct mxs_saif *saif = mxs_saif[saif_id];
+ u32 stat;
+
+ if (!saif)
+ return -EINVAL;
+
+ stat = __raw_readl(saif->base + SAIF_STAT);
+ if (stat & BM_SAIF_STAT_BUSY) {
+ dev_err(saif->dev, "error: busy\n");
+ return -EBUSY;
+ }
+
+ clk_disable(saif->clk);
+
+ /* disable MCLK output */
+ __raw_writel(BM_SAIF_CTRL_CLKGATE,
+ saif->base + SAIF_CTRL + MXS_SET_ADDR);
+ __raw_writel(BM_SAIF_CTRL_RUN,
+ saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+
+ saif->mclk_in_use = 0;
+ return 0;
+}
+
+/*
+ * Get MCLK and set clock rate, then enable it
+ *
+ * This interface is used for codecs who are using MCLK provided
+ * by saif.
+ */
+int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk,
+ unsigned int rate)
+{
+ struct mxs_saif *saif = mxs_saif[saif_id];
+ u32 stat;
+ int ret;
+
+ if (!saif)
+ return -EINVAL;
+
+ /* Clear Reset */
+ __raw_writel(BM_SAIF_CTRL_SFTRST,
+ saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+
+ /* FIXME: need clear clk gate for register r/w */
+ __raw_writel(BM_SAIF_CTRL_CLKGATE,
+ saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+
+ stat = __raw_readl(saif->base + SAIF_STAT);
+ if (stat & BM_SAIF_STAT_BUSY) {
+ dev_err(saif->dev, "error: busy\n");
+ return -EBUSY;
+ }
+
+ saif->mclk_in_use = 1;
+ ret = mxs_saif_set_clk(saif, mclk, rate);
+ if (ret)
+ return ret;
+
+ ret = clk_enable(saif->clk);
+ if (ret)
+ return ret;
+
+ /* enable MCLK output */
+ __raw_writel(BM_SAIF_CTRL_RUN,
+ saif->base + SAIF_CTRL + MXS_SET_ADDR);
+
+ return 0;
+}
+
+/*
+ * SAIF DAI format configuration.
+ * Should only be called when port is inactive.
+ */
+static int mxs_saif_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+ u32 scr, stat;
+ u32 scr0;
+ struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+
+ stat = __raw_readl(saif->base + SAIF_STAT);
+ if (stat & BM_SAIF_STAT_BUSY) {
+ dev_err(cpu_dai->dev, "error: busy\n");
+ return -EBUSY;
+ }
+
+ scr0 = __raw_readl(saif->base + SAIF_CTRL);
+ scr0 = scr0 & ~BM_SAIF_CTRL_BITCLK_EDGE & ~BM_SAIF_CTRL_LRCLK_POLARITY \
+ & ~BM_SAIF_CTRL_JUSTIFY & ~BM_SAIF_CTRL_DELAY;
+ scr = 0;
+
+ /* DAI mode */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ /* data frame low 1clk before data */
+ scr |= BM_SAIF_CTRL_DELAY;
+ scr &= ~BM_SAIF_CTRL_LRCLK_POLARITY;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ /* data frame high with data */
+ scr &= ~BM_SAIF_CTRL_DELAY;
+ scr &= ~BM_SAIF_CTRL_LRCLK_POLARITY;
+ scr &= ~BM_SAIF_CTRL_JUSTIFY;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* DAI clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_IB_IF:
+ scr |= BM_SAIF_CTRL_BITCLK_EDGE;
+ scr |= BM_SAIF_CTRL_LRCLK_POLARITY;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ scr |= BM_SAIF_CTRL_BITCLK_EDGE;
+ scr &= ~BM_SAIF_CTRL_LRCLK_POLARITY;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ scr &= ~BM_SAIF_CTRL_BITCLK_EDGE;
+ scr |= BM_SAIF_CTRL_LRCLK_POLARITY;
+ break;
+ case SND_SOC_DAIFMT_NB_NF:
+ scr &= ~BM_SAIF_CTRL_BITCLK_EDGE;
+ scr &= ~BM_SAIF_CTRL_LRCLK_POLARITY;
+ break;
+ }
+
+ /*
+ * Note: We simply just support master mode since SAIF TX can only
+ * work as master.
+ */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ scr &= ~BM_SAIF_CTRL_SLAVE_MODE;
+ __raw_writel(scr | scr0, saif->base + SAIF_CTRL);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int mxs_saif_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+ snd_soc_dai_set_dma_data(cpu_dai, substream, &saif->dma_param);
+
+ /* clear error status to 0 for each re-open */
+ saif->fifo_underrun = 0;
+ saif->fifo_overrun = 0;
+
+ /* Clear Reset for normal operations */
+ __raw_writel(BM_SAIF_CTRL_SFTRST,
+ saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+
+ /* clear clock gate */
+ __raw_writel(BM_SAIF_CTRL_CLKGATE,
+ saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+
+ return 0;
+}
+
+/*
+ * Should only be called when port is inactive.
+ * although can be called multiple times by upper layers.
+ */
+static int mxs_saif_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+ u32 scr, stat;
+ int ret;
+
+ /* mclk should already be set */
+ if (!saif->mclk && saif->mclk_in_use) {
+ dev_err(cpu_dai->dev, "set mclk first\n");
+ return -EINVAL;
+ }
+
+ stat = __raw_readl(saif->base + SAIF_STAT);
+ if (stat & BM_SAIF_STAT_BUSY) {
+ dev_err(cpu_dai->dev, "error: busy\n");
+ return -EBUSY;
+ }
+
+ /*
+ * Set saif clk based on sample rate.
+ * If mclk is used, we also set mclk, if not, saif->mclk is
+ * default 0, means not used.
+ */
+ ret = mxs_saif_set_clk(saif, saif->mclk, params_rate(params));
+ if (ret) {
+ dev_err(cpu_dai->dev, "unable to get proper clk\n");
+ return ret;
+ }
+
+ scr = __raw_readl(saif->base + SAIF_CTRL);
+
+ scr &= ~BM_SAIF_CTRL_WORD_LENGTH;
+ scr &= ~BM_SAIF_CTRL_BITCLK_48XFS_ENABLE;
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ scr |= BF_SAIF_CTRL_WORD_LENGTH(0);
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ scr |= BF_SAIF_CTRL_WORD_LENGTH(4);
+ scr |= BM_SAIF_CTRL_BITCLK_48XFS_ENABLE;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ scr |= BF_SAIF_CTRL_WORD_LENGTH(8);
+ scr |= BM_SAIF_CTRL_BITCLK_48XFS_ENABLE;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* Tx/Rx config */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ /* enable TX mode */
+ scr &= ~BM_SAIF_CTRL_READ_MODE;
+ } else {
+ /* enable RX mode */
+ scr |= BM_SAIF_CTRL_READ_MODE;
+ }
+
+ __raw_writel(scr, saif->base + SAIF_CTRL);
+ return 0;
+}
+
+static int mxs_saif_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+
+ /* enable FIFO error irqs */
+ __raw_writel(BM_SAIF_CTRL_FIFO_ERROR_IRQ_EN,
+ saif->base + SAIF_CTRL + MXS_SET_ADDR);
+
+ return 0;
+}
+
+static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ dev_dbg(cpu_dai->dev, "start\n");
+
+ clk_enable(saif->clk);
+ if (!saif->mclk_in_use)
+ __raw_writel(BM_SAIF_CTRL_RUN,
+ saif->base + SAIF_CTRL + MXS_SET_ADDR);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ /*
+ * write a data to saif data register to trigger
+ * the transfer
+ */
+ __raw_writel(0, saif->base + SAIF_DATA);
+ } else {
+ /*
+ * read a data from saif data register to trigger
+ * the receive
+ */
+ __raw_readl(saif->base + SAIF_DATA);
+ }
+
+ dev_dbg(cpu_dai->dev, "CTRL 0x%x STAT 0x%x\n",
+ __raw_readl(saif->base + SAIF_CTRL),
+ __raw_readl(saif->base + SAIF_STAT));
+
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ dev_dbg(cpu_dai->dev, "stop\n");
+
+ clk_disable(saif->clk);
+ if (!saif->mclk_in_use)
+ __raw_writel(BM_SAIF_CTRL_RUN,
+ saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+#define MXS_SAIF_RATES SNDRV_PCM_RATE_8000_192000
+#define MXS_SAIF_FORMATS \
+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops mxs_saif_dai_ops = {
+ .startup = mxs_saif_startup,
+ .trigger = mxs_saif_trigger,
+ .prepare = mxs_saif_prepare,
+ .hw_params = mxs_saif_hw_params,
+ .set_sysclk = mxs_saif_set_dai_sysclk,
+ .set_fmt = mxs_saif_set_dai_fmt,
+};
+
+static int mxs_saif_dai_probe(struct snd_soc_dai *dai)
+{
+ struct mxs_saif *saif = dev_get_drvdata(dai->dev);
+
+ snd_soc_dai_set_drvdata(dai, saif);
+
+ return 0;
+}
+
+static struct snd_soc_dai_driver mxs_saif_dai = {
+ .name = "mxs-saif",
+ .probe = mxs_saif_dai_probe,
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = MXS_SAIF_RATES,
+ .formats = MXS_SAIF_FORMATS,
+ },
+ .capture = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = MXS_SAIF_RATES,
+ .formats = MXS_SAIF_FORMATS,
+ },
+ .ops = &mxs_saif_dai_ops,
+};
+
+static irqreturn_t mxs_saif_irq(int irq, void *dev_id)
+{
+ struct mxs_saif *saif = dev_id;
+ unsigned int stat;
+
+ stat = __raw_readl(saif->base + SAIF_STAT);
+ if (!(stat & (BM_SAIF_STAT_FIFO_UNDERFLOW_IRQ |
+ BM_SAIF_STAT_FIFO_OVERFLOW_IRQ)))
+ return IRQ_NONE;
+
+ if (stat & BM_SAIF_STAT_FIFO_UNDERFLOW_IRQ) {
+ dev_dbg(saif->dev, "underrun!!! %d\n", ++saif->fifo_underrun);
+ __raw_writel(BM_SAIF_STAT_FIFO_UNDERFLOW_IRQ,
+ saif->base + SAIF_STAT + MXS_CLR_ADDR);
+ }
+
+ if (stat & BM_SAIF_STAT_FIFO_OVERFLOW_IRQ) {
+ dev_dbg(saif->dev, "overrun!!! %d\n", ++saif->fifo_overrun);
+ __raw_writel(BM_SAIF_STAT_FIFO_OVERFLOW_IRQ,
+ saif->base + SAIF_STAT + MXS_CLR_ADDR);
+ }
+
+ dev_dbg(saif->dev, "SAIF_CTRL %x SAIF_STAT %x\n",
+ __raw_readl(saif->base + SAIF_CTRL),
+ __raw_readl(saif->base + SAIF_STAT));
+
+ return IRQ_HANDLED;
+}
+
+static int mxs_saif_probe(struct platform_device *pdev)
+{
+ struct resource *res;
+ struct mxs_saif *saif;
+ int ret = 0;
+
+ if (pdev->id >= ARRAY_SIZE(mxs_saif))
+ return -EINVAL;
+
+ saif = kzalloc(sizeof(*saif), GFP_KERNEL);
+ if (!saif)
+ return -ENOMEM;
+
+ mxs_saif[pdev->id] = saif;
+
+ saif->clk = clk_get(&pdev->dev, NULL);
+ if (IS_ERR(saif->clk)) {
+ ret = PTR_ERR(saif->clk);
+ dev_err(&pdev->dev, "Cannot get the clock: %d\n",
+ ret);
+ goto failed_clk;
+ }
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!res) {
+ ret = -ENODEV;
+ dev_err(&pdev->dev, "failed to get io resource: %d\n",
+ ret);
+ goto failed_get_resource;
+ }
+
+ if (!request_mem_region(res->start, resource_size(res), "mxs-saif")) {
+ dev_err(&pdev->dev, "request_mem_region failed\n");
+ ret = -EBUSY;
+ goto failed_get_resource;
+ }
+
+ saif->base = ioremap(res->start, resource_size(res));
+ if (!saif->base) {
+ dev_err(&pdev->dev, "ioremap failed\n");
+ ret = -ENODEV;
+ goto failed_ioremap;
+ }
+
+ res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+ if (!res) {
+ ret = -ENODEV;
+ dev_err(&pdev->dev, "failed to get dma resource: %d\n",
+ ret);
+ goto failed_ioremap;
+ }
+ saif->dma_param.chan_num = res->start;
+
+ saif->irq = platform_get_irq(pdev, 0);
+ if (saif->irq < 0) {
+ ret = saif->irq;
+ dev_err(&pdev->dev, "failed to get irq resource: %d\n",
+ ret);
+ goto failed_get_irq1;
+ }
+
+ saif->dev = &pdev->dev;
+ ret = request_irq(saif->irq, mxs_saif_irq, 0, "mxs-saif", saif);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to request irq\n");
+ goto failed_get_irq1;
+ }
+
+ saif->dma_param.chan_irq = platform_get_irq(pdev, 1);
+ if (saif->dma_param.chan_irq < 0) {
+ ret = saif->dma_param.chan_irq;
+ dev_err(&pdev->dev, "failed to get dma irq resource: %d\n",
+ ret);
+ goto failed_get_irq2;
+ }
+
+ platform_set_drvdata(pdev, saif);
+
+ ret = snd_soc_register_dai(&pdev->dev, &mxs_saif_dai);
+ if (ret) {
+ dev_err(&pdev->dev, "register DAI failed\n");
+ goto failed_register;
+ }
+
+ saif->soc_platform_pdev = platform_device_alloc(
+ "mxs-pcm-audio", pdev->id);
+ if (!saif->soc_platform_pdev) {
+ ret = -ENOMEM;
+ goto failed_pdev_alloc;
+ }
+
+ platform_set_drvdata(saif->soc_platform_pdev, saif);
+ ret = platform_device_add(saif->soc_platform_pdev);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to add soc platform device\n");
+ goto failed_pdev_add;
+ }
+
+ return 0;
+
+failed_pdev_add:
+ platform_device_put(saif->soc_platform_pdev);
+failed_pdev_alloc:
+ snd_soc_unregister_dai(&pdev->dev);
+failed_register:
+failed_get_irq2:
+ free_irq(saif->irq, saif);
+failed_get_irq1:
+ iounmap(saif->base);
+failed_ioremap:
+ release_mem_region(res->start, resource_size(res));
+failed_get_resource:
+ clk_put(saif->clk);
+failed_clk:
+ kfree(saif);
+
+ return ret;
+}
+
+static int __devexit mxs_saif_remove(struct platform_device *pdev)
+{
+ struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ struct mxs_saif *saif = platform_get_drvdata(pdev);
+
+ platform_device_unregister(saif->soc_platform_pdev);
+
+ snd_soc_unregister_dai(&pdev->dev);
+
+ iounmap(saif->base);
+ release_mem_region(res->start, resource_size(res));
+ free_irq(saif->irq, saif);
+
+ clk_put(saif->clk);
+ kfree(saif);
+
+ return 0;
+}
+
+static struct platform_driver mxs_saif_driver = {
+ .probe = mxs_saif_probe,
+ .remove = __devexit_p(mxs_saif_remove),
+
+ .driver = {
+ .name = "mxs-saif",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init mxs_saif_init(void)
+{
+ return platform_driver_register(&mxs_saif_driver);
+}
+
+static void __exit mxs_saif_exit(void)
+{
+ platform_driver_unregister(&mxs_saif_driver);
+}
+
+module_init(mxs_saif_init);
+module_exit(mxs_saif_exit);
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("MXS ASoC SAIF driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h
new file mode 100644
index 0000000..0e2ff8c
--- /dev/null
+++ b/sound/soc/mxs/mxs-saif.h
@@ -0,0 +1,130 @@
+/*
+ * Copyright (C) 2011 Freescale Semiconductor, Inc. All Rights Reserved.
+ *
+ * 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.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+
+#ifndef _MXS_SAIF_H
+#define _MXS_SAIF_H
+
+#define SAIF_CTRL 0x0
+#define SAIF_STAT 0x10
+#define SAIF_DATA 0x20
+#define SAIF_VERSION 0X30
+
+/* SAIF_CTRL */
+#define BM_SAIF_CTRL_SFTRST 0x80000000
+#define BM_SAIF_CTRL_CLKGATE 0x40000000
+#define BP_SAIF_CTRL_BITCLK_MULT_RATE 27
+#define BM_SAIF_CTRL_BITCLK_MULT_RATE 0x38000000
+#define BF_SAIF_CTRL_BITCLK_MULT_RATE(v) \
+ (((v) << 27) & BM_SAIF_CTRL_BITCLK_MULT_RATE)
+#define BM_SAIF_CTRL_BITCLK_BASE_RATE 0x04000000
+#define BM_SAIF_CTRL_FIFO_ERROR_IRQ_EN 0x02000000
+#define BM_SAIF_CTRL_FIFO_SERVICE_IRQ_EN 0x01000000
+#define BP_SAIF_CTRL_RSRVD2 21
+#define BM_SAIF_CTRL_RSRVD2 0x00E00000
+
+#define BP_SAIF_CTRL_DMAWAIT_COUNT 16
+#define BM_SAIF_CTRL_DMAWAIT_COUNT 0x001F0000
+#define BF_SAIF_CTRL_DMAWAIT_COUNT(v) \
+ (((v) << 16) & BM_SAIF_CTRL_DMAWAIT_COUNT)
+#define BP_SAIF_CTRL_CHANNEL_NUM_SELECT 14
+#define BM_SAIF_CTRL_CHANNEL_NUM_SELECT 0x0000C000
+#define BF_SAIF_CTRL_CHANNEL_NUM_SELECT(v) \
+ (((v) << 14) & BM_SAIF_CTRL_CHANNEL_NUM_SELECT)
+#define BM_SAIF_CTRL_LRCLK_PULSE 0x00002000
+#define BM_SAIF_CTRL_BIT_ORDER 0x00001000
+#define BM_SAIF_CTRL_DELAY 0x00000800
+#define BM_SAIF_CTRL_JUSTIFY 0x00000400
+#define BM_SAIF_CTRL_LRCLK_POLARITY 0x00000200
+#define BM_SAIF_CTRL_BITCLK_EDGE 0x00000100
+#define BP_SAIF_CTRL_WORD_LENGTH 4
+#define BM_SAIF_CTRL_WORD_LENGTH 0x000000F0
+#define BF_SAIF_CTRL_WORD_LENGTH(v) \
+ (((v) << 4) & BM_SAIF_CTRL_WORD_LENGTH)
+#define BM_SAIF_CTRL_BITCLK_48XFS_ENABLE 0x00000008
+#define BM_SAIF_CTRL_SLAVE_MODE 0x00000004
+#define BM_SAIF_CTRL_READ_MODE 0x00000002
+#define BM_SAIF_CTRL_RUN 0x00000001
+
+/* SAIF_STAT */
+#define BM_SAIF_STAT_PRESENT 0x80000000
+#define BP_SAIF_STAT_RSRVD2 17
+#define BM_SAIF_STAT_RSRVD2 0x7FFE0000
+#define BF_SAIF_STAT_RSRVD2(v) \
+ (((v) << 17) & BM_SAIF_STAT_RSRVD2)
+#define BM_SAIF_STAT_DMA_PREQ 0x00010000
+#define BP_SAIF_STAT_RSRVD1 7
+#define BM_SAIF_STAT_RSRVD1 0x0000FF80
+#define BF_SAIF_STAT_RSRVD1(v) \
+ (((v) << 7) & BM_SAIF_STAT_RSRVD1)
+
+#define BM_SAIF_STAT_FIFO_UNDERFLOW_IRQ 0x00000040
+#define BM_SAIF_STAT_FIFO_OVERFLOW_IRQ 0x00000020
+#define BM_SAIF_STAT_FIFO_SERVICE_IRQ 0x00000010
+#define BP_SAIF_STAT_RSRVD0 1
+#define BM_SAIF_STAT_RSRVD0 0x0000000E
+#define BF_SAIF_STAT_RSRVD0(v) \
+ (((v) << 1) & BM_SAIF_STAT_RSRVD0)
+#define BM_SAIF_STAT_BUSY 0x00000001
+
+/* SAFI_DATA */
+#define BP_SAIF_DATA_PCM_RIGHT 16
+#define BM_SAIF_DATA_PCM_RIGHT 0xFFFF0000
+#define BF_SAIF_DATA_PCM_RIGHT(v) \
+ (((v) << 16) & BM_SAIF_DATA_PCM_RIGHT)
+#define BP_SAIF_DATA_PCM_LEFT 0
+#define BM_SAIF_DATA_PCM_LEFT 0x0000FFFF
+#define BF_SAIF_DATA_PCM_LEFT(v) \
+ (((v) << 0) & BM_SAIF_DATA_PCM_LEFT)
+
+/* SAIF_VERSION */
+#define BP_SAIF_VERSION_MAJOR 24
+#define BM_SAIF_VERSION_MAJOR 0xFF000000
+#define BF_SAIF_VERSION_MAJOR(v) \
+ (((v) << 24) & BM_SAIF_VERSION_MAJOR)
+#define BP_SAIF_VERSION_MINOR 16
+#define BM_SAIF_VERSION_MINOR 0x00FF0000
+#define BF_SAIF_VERSION_MINOR(v) \
+ (((v) << 16) & BM_SAIF_VERSION_MINOR)
+#define BP_SAIF_VERSION_STEP 0
+#define BM_SAIF_VERSION_STEP 0x0000FFFF
+#define BF_SAIF_VERSION_STEP(v) \
+ (((v) << 0) & BM_SAIF_VERSION_STEP)
+
+#define MXS_SAIF_MCLK 0
+
+#include "mxs-pcm.h"
+
+struct mxs_saif {
+ struct device *dev;
+ struct clk *clk;
+ unsigned int mclk;
+ unsigned int mclk_in_use;
+ void __iomem *base;
+ int irq;
+ struct mxs_pcm_dma_params dma_param;
+
+ struct platform_device *soc_platform_pdev;
+ u32 fifo_underrun;
+ u32 fifo_overrun;
+};
+
+extern int mxs_saif_put_mclk(unsigned int saif_id);
+extern int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk,
+ unsigned int rate);
+#endif
diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
new file mode 100644
index 0000000..7fbeaec
--- /dev/null
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -0,0 +1,173 @@
+/*
+ * Copyright 2011 Freescale Semiconductor, Inc.
+ *
+ * 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.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/soc-dapm.h>
+#include <asm/mach-types.h>
+
+#include "../codecs/sgtl5000.h"
+#include "mxs-saif.h"
+
+static int mxs_sgtl5000_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;
+ unsigned int rate = params_rate(params);
+ u32 dai_format, mclk;
+ int ret;
+
+ /* sgtl5000 does not support 512*rate when in 96000 fs */
+ switch (rate) {
+ case 96000:
+ mclk = 256 * rate;
+ break;
+ default:
+ mclk = 512 * rate;
+ break;
+ }
+
+ /* Sgtl5000 sysclk should be >= 8MHz and <= 27M */
+ if (mclk < 8000000 || mclk > 27000000)
+ return -EINVAL;
+
+ /* Set SGTL5000's SYSCLK (provided by SAIF MCLK) */
+ ret = snd_soc_dai_set_sysclk(codec_dai, SGTL5000_SYSCLK, mclk, 0);
+ if (ret)
+ return ret;
+
+ /* The SAIF MCLK should be the same as SGTL5000_SYSCLK */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, MXS_SAIF_MCLK, mclk, 0);
+ if (ret)
+ return ret;
+
+ /* set codec to slave mode */
+ dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS;
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, dai_format);
+ if (ret)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai, dai_format);
+ if (ret)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops mxs_sgtl5000_hifi_ops = {
+ .hw_params = mxs_sgtl5000_hw_params,
+};
+
+static struct snd_soc_dai_link mxs_sgtl5000_dai[] = {
+ {
+ .name = "HiFi Tx",
+ .stream_name = "HiFi Playback",
+ .codec_dai_name = "sgtl5000",
+ .codec_name = "sgtl5000.0-000a",
+ .cpu_dai_name = "mxs-saif.0",
+ .platform_name = "mxs-pcm-audio.0",
+ .ops = &mxs_sgtl5000_hifi_ops,
+ }, {
+ .name = "HiFi Rx",
+ .stream_name = "HiFi Capture",
+ .codec_dai_name = "sgtl5000",
+ .codec_name = "sgtl5000.0-000a",
+ .cpu_dai_name = "mxs-saif.1",
+ .platform_name = "mxs-pcm-audio.1",
+ .ops = &mxs_sgtl5000_hifi_ops,
+ },
+};
+
+static struct snd_soc_card mxs_sgtl5000 = {
+ .name = "mxs_sgtl5000",
+ .dai_link = mxs_sgtl5000_dai,
+ .num_links = ARRAY_SIZE(mxs_sgtl5000_dai),
+};
+
+static int __devinit mxs_sgtl5000_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &mxs_sgtl5000;
+ int ret;
+
+ /*
+ * Set an init clock(11.28Mhz) for sgtl5000 initialization(i2c r/w).
+ * The Sgtl5000 sysclk is derived from saif0 mclk and it's range
+ * should be >= 8MHz and <= 27M.
+ */
+ ret = mxs_saif_get_mclk(0, 44100 * 256, 44100);
+ if (ret)
+ return ret;
+
+ card->dev = &pdev->dev;
+ platform_set_drvdata(pdev, card);
+
+ ret = snd_soc_register_card(card);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n",
+ ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int __devexit mxs_sgtl5000_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ mxs_saif_put_mclk(0);
+
+ snd_soc_unregister_card(card);
+
+ return 0;
+}
+
+static struct platform_driver mxs_sgtl5000_audio_driver = {
+ .driver = {
+ .name = "mxs-sgtl5000",
+ .owner = THIS_MODULE,
+ },
+ .probe = mxs_sgtl5000_probe,
+ .remove = __devexit_p(mxs_sgtl5000_remove),
+};
+
+static int __init mxs_sgtl5000_init(void)
+{
+ return platform_driver_register(&mxs_sgtl5000_audio_driver);
+}
+module_init(mxs_sgtl5000_init);
+
+static void __exit mxs_sgtl5000_exit(void)
+{
+ platform_driver_unregister(&mxs_sgtl5000_audio_driver);
+}
+module_exit(mxs_sgtl5000_exit);
+
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("MXS ALSA SoC Machine driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c
index d589ef1..e46d551 100644
--- a/sound/soc/nuc900/nuc900-pcm.c
+++ b/sound/soc/nuc900/nuc900-pcm.c
@@ -318,7 +318,6 @@
static int nuc900_dma_new(struct snd_soc_pcm_runtime *rtd)
{
struct snd_card *card = rtd->card->snd_card;
- struct snd_soc_dai *dai = rtd->cpu_dai;
struct snd_pcm *pcm = rtd->pcm;
if (!card->dev->dma_mask)
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index b253d86..ce920e3 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -312,7 +312,7 @@
.cpu_dai_name = "pxa2xx-i2s",
.codec_dai_name = "wm8750-hifi",
.platform_name = "pxa-pcm-audio",
- .codec_name = "wm8750-codec.0-001b",
+ .codec_name = "wm8750.0-001b",
.init = spitz_wm8750_init,
.ops = &spitz_ops,
};
diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c
index d69d9fc..4b81ffd 100644
--- a/sound/soc/pxa/z2.c
+++ b/sound/soc/pxa/z2.c
@@ -198,7 +198,7 @@
.cpu_dai_name = "pxa2xx-i2s",
.codec_dai_name = "wm8750-hifi",
.platform_name = "pxa-pcm-audio",
- .codec_name = "wm8750-codec.0-001b",
+ .codec_name = "wm8750.0-001b",
.init = z2_wm8750_init,
.ops = &z2_ops,
};
diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c
index 80c85fd6..55efc2b 100644
--- a/sound/soc/s6000/s6000-pcm.c
+++ b/sound/soc/s6000/s6000-pcm.c
@@ -446,7 +446,6 @@
static int s6000_pcm_new(struct snd_soc_pcm_runtime *runtime)
{
struct snd_card *card = runtime->card->snd_card;
- struct snd_soc_dai *dai = runtime->cpu_dai;
struct snd_pcm *pcm = runtime->pcm;
struct s6000_pcm_dma_params *params;
int res;
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index 65f980e..dd3b3ea 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -63,7 +63,7 @@
config SND_SOC_SAMSUNG_SMDK_WM8994
tristate "SoC I2S Audio support for WM8994 on SMDK"
- depends on SND_SOC_SAMSUNG && (MACH_SMDKV310 || MACH_SMDKC210)
+ depends on SND_SOC_SAMSUNG && (MACH_SMDKV310 || MACH_SMDKC210 || MACH_SMDK4212)
select SND_SOC_WM8994
select SND_SAMSUNG_I2S
help
@@ -158,7 +158,7 @@
config SND_SOC_SAMSUNG_SMDK_SPDIF
tristate "SoC S/PDIF Audio support for SMDK"
- depends on SND_SOC_SAMSUNG && (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210 || MACH_SMDKV310)
+ depends on SND_SOC_SAMSUNG && (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210 || MACH_SMDKV310 || MACH_SMDK4212)
select SND_SAMSUNG_SPDIF
help
Say Y if you want to add support for SoC S/PDIF audio on the SMDK.
@@ -173,7 +173,7 @@
config SND_SOC_SMDK_WM8994_PCM
tristate "SoC PCM Audio support for WM8994 on SMDK"
- depends on SND_SOC_SAMSUNG && (MACH_SMDKC210 || MACH_SMDKV310)
+ depends on SND_SOC_SAMSUNG && (MACH_SMDKC210 || MACH_SMDKV310 || MACH_SMDK4212)
select SND_SOC_WM8994
select SND_SAMSUNG_PCM
help
diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c
index 14eb6ea..ed8f13a 100644
--- a/sound/soc/samsung/jive_wm8750.c
+++ b/sound/soc/samsung/jive_wm8750.c
@@ -131,7 +131,7 @@
.cpu_dai_name = "s3c2412-i2s",
.codec_dai_name = "wm8750-hifi",
.platform_name = "samsung-audio",
- .codec_name = "wm8750-codec.0-001a",
+ .codec_name = "wm8750.0-001a",
.init = jive_wm8750_init,
.ops = &jive_ops,
};
diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c
index 0a2c4f2..bbd1476 100644
--- a/sound/soc/samsung/smartq_wm8987.c
+++ b/sound/soc/samsung/smartq_wm8987.c
@@ -207,7 +207,7 @@
.cpu_dai_name = "samsung-i2s.0",
.codec_dai_name = "wm8750-hifi",
.platform_name = "samsung-audio",
- .codec_name = "wm8750-codec.0-0x1a",
+ .codec_name = "wm8750.0-0x1a",
.init = smartq_wm8987_init,
.ops = &smartq_hifi_ops,
},
diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c
index 3d26f66..20deecf 100644
--- a/sound/soc/samsung/smdk_wm8580.c
+++ b/sound/soc/samsung/smdk_wm8580.c
@@ -210,7 +210,7 @@
.cpu_dai_name = "samsung-i2s.0",
.codec_dai_name = "wm8580-hifi-playback",
.platform_name = "samsung-audio",
- .codec_name = "wm8580-codec.0-001b",
+ .codec_name = "wm8580.0-001b",
.init = smdk_wm8580_init_paifrx,
.ops = &smdk_ops,
},
@@ -220,7 +220,7 @@
.cpu_dai_name = "samsung-i2s.0",
.codec_dai_name = "wm8580-hifi-capture",
.platform_name = "samsung-audio",
- .codec_name = "wm8580-codec.0-001b",
+ .codec_name = "wm8580.0-001b",
.init = smdk_wm8580_init_paiftx,
.ops = &smdk_ops,
},
@@ -230,7 +230,7 @@
.cpu_dai_name = "samsung-i2s.x",
.codec_dai_name = "wm8580-hifi-playback",
.platform_name = "samsung-audio",
- .codec_name = "wm8580-codec.0-001b",
+ .codec_name = "wm8580.0-001b",
.init = smdk_wm8580_init_paifrx,
.ops = &smdk_ops,
},
diff --git a/sound/soc/samsung/smdk_wm8580pcm.c b/sound/soc/samsung/smdk_wm8580pcm.c
index 0d12092..4b9c734 100644
--- a/sound/soc/samsung/smdk_wm8580pcm.c
+++ b/sound/soc/samsung/smdk_wm8580pcm.c
@@ -127,7 +127,7 @@
.cpu_dai_name = "samsung-pcm.0",
.codec_dai_name = "wm8580-hifi-playback",
.platform_name = "samsung-audio",
- .codec_name = "wm8580-codec.0-001b",
+ .codec_name = "wm8580.0-001b",
.ops = &smdk_wm8580_pcm_ops,
}, {
.name = "WM8580 PAIF PCM TX",
@@ -135,7 +135,7 @@
.cpu_dai_name = "samsung-pcm.0",
.codec_dai_name = "wm8580-hifi-capture",
.platform_name = "samsung-audio",
- .codec_name = "wm8580-codec.0-001b",
+ .codec_name = "wm8580.0-001b",
.ops = &smdk_wm8580_pcm_ops,
},
};
diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c
index 590e927..bfed1ff 100644
--- a/sound/soc/samsung/speyside.c
+++ b/sound/soc/samsung/speyside.c
@@ -125,10 +125,6 @@
.pin = "Headset Mic",
.mask = SND_JACK_MICROPHONE,
},
- {
- .pin = "Headphone",
- .mask = SND_JACK_HEADPHONE,
- },
};
/* Default the headphone selection to active high */
@@ -252,6 +248,7 @@
SOC_DAPM_PIN_SWITCH("Main AMIC"),
SOC_DAPM_PIN_SWITCH("WM1250 Input"),
SOC_DAPM_PIN_SWITCH("WM1250 Output"),
+ SOC_DAPM_PIN_SWITCH("Headphone"),
};
static struct snd_soc_dapm_widget widgets[] = {
diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c
index 72535f2..3820a6b 100644
--- a/sound/soc/samsung/speyside_wm8962.c
+++ b/sound/soc/samsung/speyside_wm8962.c
@@ -31,13 +31,13 @@
if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
WM8962_FLL_MCLK, 32768,
- 44100 * 256);
+ 44100 * 512);
if (ret < 0)
pr_err("Failed to start FLL: %d\n", ret);
ret = snd_soc_dai_set_sysclk(codec_dai,
WM8962_SYSCLK_FLL,
- 44100 * 256,
+ 44100 * 512,
SND_SOC_CLOCK_IN);
if (ret < 0) {
pr_err("Failed to set SYSCLK: %d\n", ret);
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index 20b7f3b..143c705 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -548,9 +548,6 @@
static inline int snd_soc_lzo_get_blksize(struct snd_soc_codec *codec)
{
- const struct snd_soc_codec_driver *codec_drv;
-
- codec_drv = codec->driver;
return DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count());
}
@@ -868,10 +865,6 @@
static int snd_soc_flat_cache_init(struct snd_soc_codec *codec)
{
- const struct snd_soc_codec_driver *codec_drv;
-
- codec_drv = codec->driver;
-
if (codec->reg_def_copy)
codec->reg_cache = kmemdup(codec->reg_def_copy,
codec->reg_size, GFP_KERNEL);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d2ef014..4065d4e 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -105,7 +105,7 @@
if (wordsize + regsize + 2 + 1 != len)
return -EINVAL;
- ret = snd_soc_read(codec , reg);
+ ret = snd_soc_read(codec, reg);
if (ret < 0) {
memset(regbuf, 'X', regsize);
regbuf[regsize] = '\0';
@@ -956,6 +956,8 @@
snd_soc_dapm_new_controls(&codec->dapm, driver->dapm_widgets,
driver->num_dapm_widgets);
+ codec->dapm.idle_bias_off = driver->idle_bias_off;
+
if (driver->probe) {
ret = driver->probe(codec);
if (ret < 0) {
@@ -3141,6 +3143,7 @@
platform->driver = platform_drv;
platform->dapm.dev = dev;
platform->dapm.platform = platform;
+ platform->dapm.stream_event = platform_drv->stream_event;
mutex_lock(&client_mutex);
list_add(&platform->list, &platform_list);
@@ -3253,6 +3256,7 @@
codec->dapm.dev = dev;
codec->dapm.codec = codec;
codec->dapm.seq_notifier = codec_drv->seq_notifier;
+ codec->dapm.stream_event = codec_drv->stream_event;
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 d67c637..4859ad7 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -443,6 +443,11 @@
if (path->name != (char *)w->kcontrol_news[i].name)
continue;
+ if (w->kcontrols[i]) {
+ path->kcontrol = w->kcontrols[i];
+ continue;
+ }
+
wlistsize = sizeof(struct snd_soc_dapm_widget_list) +
sizeof(struct snd_soc_dapm_widget *),
wlist = kzalloc(wlistsize, GFP_KERNEL);
@@ -1556,7 +1561,6 @@
/* found, now check type */
found = 1;
path->connect = connect;
- break;
}
if (found)
@@ -2584,7 +2588,7 @@
{
if (!w->sname || w->dapm != dapm)
continue;
- dev_dbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n",
+ dev_vdbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n",
w->name, w->sname, stream, event);
if (strstr(w->sname, stream)) {
switch(event) {
@@ -2604,6 +2608,10 @@
}
dapm_power_widgets(dapm, event);
+
+ /* do we need to notify any clients that DAPM stream is complete */
+ if (dapm->stream_event)
+ dapm->stream_event(dapm, event);
}
/**
diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c
index a62f7dd..66fcccd 100644
--- a/sound/soc/soc-io.c
+++ b/sound/soc/soc-io.c
@@ -13,26 +13,14 @@
#include <linux/i2c.h>
#include <linux/spi/spi.h>
+#include <linux/regmap.h>
#include <sound/soc.h>
#include <trace/events/asoc.h>
-#ifdef CONFIG_SPI_MASTER
-static int do_spi_write(void *control, const char *data, int len)
-{
- struct spi_device *spi = control;
- int ret;
-
- ret = spi_write(spi, data, len);
- if (ret < 0)
- return ret;
-
- return len;
-}
-#endif
-
-static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value, const void *data, int len)
+#ifdef CONFIG_REGMAP
+static int hw_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
{
int ret;
@@ -49,13 +37,7 @@
return 0;
}
- ret = codec->hw_write(codec->control_data, data, len);
- if (ret == len)
- return 0;
- if (ret < 0)
- return ret;
- else
- return -EIO;
+ return regmap_write(codec->control_data, reg, value);
}
static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg)
@@ -69,8 +51,11 @@
if (codec->cache_only)
return -1;
- BUG_ON(!codec->hw_read);
- return codec->hw_read(codec, reg);
+ ret = regmap_read(codec->control_data, reg, &val);
+ if (ret == 0)
+ return val;
+ else
+ return ret;
}
ret = snd_soc_cache_read(codec, reg, &val);
@@ -79,202 +64,18 @@
return val;
}
-static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u16 data;
-
- data = cpu_to_be16((reg << 12) | (value & 0xffffff));
-
- return do_hw_write(codec, reg, value, &data, 2);
-}
-
-static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u16 data;
-
- data = cpu_to_be16((reg << 9) | (value & 0x1ff));
-
- return do_hw_write(codec, reg, value, &data, 2);
-}
-
-static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[2];
-
- reg &= 0xff;
- data[0] = reg;
- data[1] = value & 0xff;
-
- return do_hw_write(codec, reg, value, data, 2);
-}
-
-static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[3];
- u16 val = cpu_to_be16(value);
-
- data[0] = reg;
- memcpy(&data[1], &val, sizeof(val));
-
- return do_hw_write(codec, reg, value, data, 3);
-}
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int do_i2c_read(struct snd_soc_codec *codec,
- void *reg, int reglen,
- void *data, int datalen)
-{
- struct i2c_msg xfer[2];
- int ret;
- struct i2c_client *client = codec->control_data;
-
- /* Write register */
- xfer[0].addr = client->addr;
- xfer[0].flags = 0;
- xfer[0].len = reglen;
- xfer[0].buf = reg;
-
- /* Read data */
- xfer[1].addr = client->addr;
- xfer[1].flags = I2C_M_RD;
- xfer[1].len = datalen;
- xfer[1].buf = data;
-
- ret = i2c_transfer(client->adapter, xfer, 2);
- if (ret == 2)
- return 0;
- else if (ret < 0)
- return ret;
- else
- return -EIO;
-}
-#endif
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int snd_soc_8_8_read_i2c(struct snd_soc_codec *codec,
- unsigned int r)
-{
- u8 reg = r;
- u8 data;
- int ret;
-
- ret = do_i2c_read(codec, ®, 1, &data, 1);
- if (ret < 0)
- return 0;
- return data;
-}
-#else
-#define snd_soc_8_8_read_i2c NULL
-#endif
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int snd_soc_8_16_read_i2c(struct snd_soc_codec *codec,
- unsigned int r)
-{
- u8 reg = r;
- u16 data;
- int ret;
-
- ret = do_i2c_read(codec, ®, 1, &data, 2);
- if (ret < 0)
- return 0;
- return (data >> 8) | ((data & 0xff) << 8);
-}
-#else
-#define snd_soc_8_16_read_i2c NULL
-#endif
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec,
- unsigned int r)
-{
- u16 reg = r;
- u8 data;
- int ret;
-
- ret = do_i2c_read(codec, ®, 2, &data, 1);
- if (ret < 0)
- return 0;
- return data;
-}
-#else
-#define snd_soc_16_8_read_i2c NULL
-#endif
-
-#if defined(CONFIG_SPI_MASTER)
-static unsigned int snd_soc_16_8_read_spi(struct snd_soc_codec *codec,
- unsigned int r)
-{
- struct spi_device *spi = codec->control_data;
-
- const u16 reg = cpu_to_be16(r | 0x100);
- u8 data;
- int ret;
-
- ret = spi_write_then_read(spi, ®, 2, &data, 1);
- if (ret < 0)
- return 0;
- return data;
-}
-#else
-#define snd_soc_16_8_read_spi NULL
-#endif
-
-static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[3];
- u16 rval = cpu_to_be16(reg);
-
- memcpy(data, &rval, sizeof(rval));
- data[2] = value;
-
- return do_hw_write(codec, reg, value, data, 3);
-}
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec,
- unsigned int r)
-{
- u16 reg = cpu_to_be16(r);
- u16 data;
- int ret;
-
- ret = do_i2c_read(codec, ®, 2, &data, 2);
- if (ret < 0)
- return 0;
- return be16_to_cpu(data);
-}
-#else
-#define snd_soc_16_16_read_i2c NULL
-#endif
-
-static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u16 data[2];
-
- data[0] = cpu_to_be16(reg);
- data[1] = cpu_to_be16(value);
-
- return do_hw_write(codec, reg, value, data, sizeof(data));
-}
-
/* Primitive bulk write support for soc-cache. The data pointed to by
- * `data' needs to already be in the form the hardware expects
- * including any leading register specific data. Any data written
- * through this function will not go through the cache as it only
- * handles writing to volatile or out of bounds registers.
+ * `data' needs to already be in the form the hardware expects. Any
+ * data written through this function will not go through the cache as
+ * it only handles writing to volatile or out of bounds registers.
+ *
+ * This is currently only supported for devices using the regmap API
+ * wrappers.
*/
-static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int reg,
+static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec,
+ unsigned int reg,
const void *data, size_t len)
{
- int ret;
-
/* To ensure that we don't get out of sync with the cache, check
* whether the base register is volatile or if we've directly asked
* to bypass the cache. Out of bounds registers are considered
@@ -285,68 +86,9 @@
&& reg < codec->driver->reg_cache_size)
return -EINVAL;
- switch (codec->control_type) {
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
- case SND_SOC_I2C:
- ret = i2c_master_send(to_i2c_client(codec->dev), data, len);
- break;
-#endif
-#if defined(CONFIG_SPI_MASTER)
- case SND_SOC_SPI:
- ret = spi_write(to_spi_device(codec->dev), data, len);
- break;
-#endif
- default:
- BUG();
- }
-
- if (ret == len)
- return 0;
- if (ret < 0)
- return ret;
- else
- return -EIO;
+ return regmap_raw_write(codec->control_data, reg, data, len);
}
-static struct {
- int addr_bits;
- int data_bits;
- int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int);
- unsigned int (*read)(struct snd_soc_codec *, unsigned int);
- unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int);
- unsigned int (*spi_read)(struct snd_soc_codec *, unsigned int);
-} io_types[] = {
- {
- .addr_bits = 4, .data_bits = 12,
- .write = snd_soc_4_12_write,
- },
- {
- .addr_bits = 7, .data_bits = 9,
- .write = snd_soc_7_9_write,
- },
- {
- .addr_bits = 8, .data_bits = 8,
- .write = snd_soc_8_8_write,
- .i2c_read = snd_soc_8_8_read_i2c,
- },
- {
- .addr_bits = 8, .data_bits = 16,
- .write = snd_soc_8_16_write,
- .i2c_read = snd_soc_8_16_read_i2c,
- },
- {
- .addr_bits = 16, .data_bits = 8,
- .write = snd_soc_16_8_write,
- .i2c_read = snd_soc_16_8_read_i2c,
- .spi_read = snd_soc_16_8_read_spi,
- },
- {
- .addr_bits = 16, .data_bits = 16,
- .write = snd_soc_16_16_write,
- .i2c_read = snd_soc_16_16_read_i2c,
- },
-};
-
/**
* snd_soc_codec_set_cache_io: Set up standard I/O functions.
*
@@ -370,50 +112,51 @@
int addr_bits, int data_bits,
enum snd_soc_control_type control)
{
- int i;
+ struct regmap_config config;
- for (i = 0; i < ARRAY_SIZE(io_types); i++)
- if (io_types[i].addr_bits == addr_bits &&
- io_types[i].data_bits == data_bits)
- break;
- if (i == ARRAY_SIZE(io_types)) {
- printk(KERN_ERR
- "No I/O functions for %d bit address %d bit data\n",
- addr_bits, data_bits);
- return -EINVAL;
- }
-
- codec->write = io_types[i].write;
+ memset(&config, 0, sizeof(config));
+ codec->write = hw_write;
codec->read = hw_read;
codec->bulk_write_raw = snd_soc_hw_bulk_write_raw;
+ config.reg_bits = addr_bits;
+ config.val_bits = data_bits;
+
switch (control) {
+#if defined(CONFIG_REGMAP_I2C) || defined(CONFIG_REGMAP_I2C_MODULE)
case SND_SOC_I2C:
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
- codec->hw_write = (hw_write_t)i2c_master_send;
-#endif
- if (io_types[i].i2c_read)
- codec->hw_read = io_types[i].i2c_read;
-
- codec->control_data = container_of(codec->dev,
- struct i2c_client,
- dev);
+ codec->control_data = regmap_init_i2c(to_i2c_client(codec->dev),
+ &config);
break;
+#endif
+#if defined(CONFIG_REGMAP_SPI) || defined(CONFIG_REGMAP_SPI_MODULE)
case SND_SOC_SPI:
-#ifdef CONFIG_SPI_MASTER
- codec->hw_write = do_spi_write;
-#endif
- if (io_types[i].spi_read)
- codec->hw_read = io_types[i].spi_read;
-
- codec->control_data = container_of(codec->dev,
- struct spi_device,
- dev);
+ codec->control_data = regmap_init_spi(to_spi_device(codec->dev),
+ &config);
break;
+#endif
+
+ case SND_SOC_REGMAP:
+ /* Device has made its own regmap arrangements */
+ break;
+
+ default:
+ return -EINVAL;
}
+ if (IS_ERR(codec->control_data))
+ return PTR_ERR(codec->control_data);
+
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io);
-
+#else
+int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
+ int addr_bits, int data_bits,
+ enum snd_soc_control_type control)
+{
+ return -ENOTSUPP;
+}
+EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io);
+#endif
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 2879c88..1aee9fc 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -27,8 +27,6 @@
#include <sound/soc.h>
#include <sound/initval.h>
-static DEFINE_MUTEX(pcm_mutex);
-
static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;