Merge branch 'asoc-omap' into for-3.7
diff --git a/Documentation/devicetree/bindings/sound/cs4270.txt b/Documentation/devicetree/bindings/sound/cs4270.txt
new file mode 100644
index 0000000..6b222f9
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/cs4270.txt
@@ -0,0 +1,21 @@
+CS4270 audio CODEC
+
+The driver for this device currently only supports I2C.
+
+Required properties:
+
+ - compatible : "cirrus,cs4270"
+
+ - reg : the I2C address of the device for I2C
+
+Optional properties:
+
+ - reset-gpio : a GPIO spec for the reset pin. If specified, it will be
+ deasserted before communication to the codec starts.
+
+Example:
+
+codec: cs4270@48 {
+ compatible = "cirrus,cs4270";
+ reg = <0x48>;
+};
diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt
new file mode 100644
index 0000000..e6148ec
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt
@@ -0,0 +1,44 @@
+Texas Instruments McASP controller
+
+Required properties:
+- compatible :
+ "ti,dm646x-mcasp-audio" : for DM646x platforms
+ "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms
+
+- reg : Should contain McASP registers offset and length
+- interrupts : Interrupt number for McASP
+- op-mode : I2S/DIT ops mode.
+- tdm-slots : Slots for TDM operation.
+- num-serializer : Serializers used by McASP.
+- serial-dir : A list of serializer pin mode. The list number should be equal
+ to "num-serializer" parameter. Each entry is a number indication
+ serializer pin direction. (0 - INACTIVE, 1 - TX, 2 - RX)
+
+
+Optional properties:
+
+- ti,hwmods : Must be "mcasp<n>", n is controller instance starting 0
+- tx-num-evt : FIFO levels.
+- rx-num-evt : FIFO levels.
+- sram-size-playback : size of sram to be allocated during playback
+- sram-size-capture : size of sram to be allocated during capture
+
+Example:
+
+mcasp0: mcasp0@1d00000 {
+ compatible = "ti,da830-mcasp-audio";
+ #address-cells = <1>;
+ #size-cells = <0>;
+ reg = <0x100000 0x3000>;
+ interrupts = <82 83>;
+ op-mode = <0>; /* MCASP_IIS_MODE */
+ tdm-slots = <2>;
+ num-serializer = <16>;
+ serial-dir = <
+ 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */
+ 0 0 0 0
+ 0 0 0 1
+ 2 0 0 0 >;
+ tx-num-evt = <1>;
+ rx-num-evt = <1>;
+};
diff --git a/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt b/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt
new file mode 100644
index 0000000..65dec87
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt
@@ -0,0 +1,91 @@
+* Texas Instruments OMAP4+ and twl6040 based audio setups
+
+Required properties:
+- compatible: "ti,abe-twl6040"
+- ti,model: Name of the sound card ( for example "SDP4430")
+- ti,mclk-freq: MCLK frequency for HPPLL operation
+- ti,mcpdm: phandle for the McPDM node
+- ti,twl6040: phandle for the twl6040 core node
+- ti,audio-routing: List of connections between audio components.
+ Each entry is a pair of strings, the first being the connection's sink,
+ the second being the connection's source.
+
+Optional properties:
+- ti,dmic: phandle for the OMAP dmic node if the machine have it connected
+- ti,jack_detection: Need to be set to <1> if the board capable to detect jack
+ insertion, removal.
+
+Available audio endpoints for the audio-routing table:
+
+Board connectors:
+ * Headset Stereophone
+ * Earphone Spk
+ * Ext Spk
+ * Line Out
+ * Vibrator
+ * Headset Mic
+ * Main Handset Mic
+ * Sub Handset Mic
+ * Line In
+ * Digital Mic
+
+twl6040 pins:
+ * HSOL
+ * HSOR
+ * EP
+ * HFL
+ * HFR
+ * AUXL
+ * AUXR
+ * VIBRAL
+ * VIBRAR
+ * HSMIC
+ * MAINMIC
+ * SUBMIC
+ * AFML
+ * AFMR
+
+ * Headset Mic Bias
+ * Main Mic Bias
+ * Digital Mic1 Bias
+ * Digital Mic2 Bias
+
+Digital mic pins:
+ * DMic
+
+Example:
+
+sound {
+ compatible = "ti,abe-twl6040";
+ ti,model = "SDP4430";
+
+ ti,jack-detection = <1>;
+ ti,mclk-freq = <38400000>;
+
+ ti,mcpdm = <&mcpdm>;
+ ti,dmic = <&dmic>;
+
+ ti,twl6040 = <&twl6040>;
+
+ /* Audio routing */
+ ti,audio-routing =
+ "Headset Stereophone", "HSOL",
+ "Headset Stereophone", "HSOR",
+ "Earphone Spk", "EP",
+ "Ext Spk", "HFL",
+ "Ext Spk", "HFR",
+ "Line Out", "AUXL",
+ "Line Out", "AUXR",
+ "Vibrator", "VIBRAL",
+ "Vibrator", "VIBRAR",
+ "HSMIC", "Headset Mic",
+ "Headset Mic", "Headset Mic Bias",
+ "MAINMIC", "Main Handset Mic",
+ "Main Handset Mic", "Main Mic Bias",
+ "SUBMIC", "Sub Handset Mic",
+ "Sub Handset Mic", "Main Mic Bias",
+ "AFML", "Line In",
+ "AFMR", "Line In",
+ "DMic", "Digital Mic",
+ "Digital Mic", "Digital Mic1 Bias";
+};
diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
new file mode 100644
index 0000000..e7b98f4
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
@@ -0,0 +1,20 @@
+Texas Instruments - tlv320aic3x Codec module
+
+The tlv320aic3x serial control bus communicates through I2C protocols
+
+Required properties:
+- compatible - "string" - "ti,tlv320aic3x"
+- reg - <int> - I2C slave address
+
+
+Optional properties:
+
+- gpio-reset - gpio pin number used for codec reset
+- ai3x-gpio-func - <array of 2 int> - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality
+
+Example:
+
+tlv320aic3x: tlv320aic3x@1b {
+ compatible = "ti,tlv320aic3x";
+ reg = <0x1b>;
+};
diff --git a/arch/arm/mach-davinci/asp.h b/arch/arm/mach-davinci/asp.h
new file mode 100644
index 0000000..d9b2acd
--- /dev/null
+++ b/arch/arm/mach-davinci/asp.h
@@ -0,0 +1,49 @@
+/*
+ * TI DaVinci Audio definitions
+ */
+#ifndef __ASM_ARCH_DAVINCI_ASP_H
+#define __ASM_ARCH_DAVINCI_ASP_H
+
+/* Bases of dm644x and dm355 register banks */
+#define DAVINCI_ASP0_BASE 0x01E02000
+#define DAVINCI_ASP1_BASE 0x01E04000
+
+/* Bases of dm365 register banks */
+#define DAVINCI_DM365_ASP0_BASE 0x01D02000
+
+/* Bases of dm646x register banks */
+#define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000
+#define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800
+
+/* Bases of da850/da830 McASP0 register banks */
+#define DAVINCI_DA8XX_MCASP0_REG_BASE 0x01D00000
+
+/* Bases of da830 McASP1 register banks */
+#define DAVINCI_DA830_MCASP1_REG_BASE 0x01D04000
+
+/* EDMA channels of dm644x and dm355 */
+#define DAVINCI_DMA_ASP0_TX 2
+#define DAVINCI_DMA_ASP0_RX 3
+#define DAVINCI_DMA_ASP1_TX 8
+#define DAVINCI_DMA_ASP1_RX 9
+
+/* EDMA channels of dm646x */
+#define DAVINCI_DM646X_DMA_MCASP0_AXEVT0 6
+#define DAVINCI_DM646X_DMA_MCASP0_AREVT0 9
+#define DAVINCI_DM646X_DMA_MCASP1_AXEVT1 12
+
+/* EDMA channels of da850/da830 McASP0 */
+#define DAVINCI_DA8XX_DMA_MCASP0_AREVT 0
+#define DAVINCI_DA8XX_DMA_MCASP0_AXEVT 1
+
+/* EDMA channels of da830 McASP1 */
+#define DAVINCI_DA830_DMA_MCASP1_AREVT 2
+#define DAVINCI_DA830_DMA_MCASP1_AXEVT 3
+
+/* Interrupts */
+#define DAVINCI_ASP0_RX_INT IRQ_MBRINT
+#define DAVINCI_ASP0_TX_INT IRQ_MBXINT
+#define DAVINCI_ASP1_RX_INT IRQ_MBRINT
+#define DAVINCI_ASP1_TX_INT IRQ_MBXINT
+
+#endif /* __ASM_ARCH_DAVINCI_ASP_H */
diff --git a/arch/arm/mach-davinci/davinci.h b/arch/arm/mach-davinci/davinci.h
index 8db0fc6..8661b20 100644
--- a/arch/arm/mach-davinci/davinci.h
+++ b/arch/arm/mach-davinci/davinci.h
@@ -22,10 +22,11 @@
#include <linux/davinci_emac.h>
#include <linux/platform_device.h>
#include <linux/spi/spi.h>
+#include <linux/platform_data/davinci_asp.h>
-#include <mach/asp.h>
#include <mach/keyscan.h>
#include <mach/hardware.h>
+#include <mach/edma.h>
#include <media/davinci/vpfe_capture.h>
#include <media/davinci/vpif_types.h>
diff --git a/arch/arm/mach-davinci/devices-da8xx.c b/arch/arm/mach-davinci/devices-da8xx.c
index 783eab6..bd2f72b 100644
--- a/arch/arm/mach-davinci/devices-da8xx.c
+++ b/arch/arm/mach-davinci/devices-da8xx.c
@@ -24,6 +24,7 @@
#include <mach/cpuidle.h>
#include "clock.h"
+#include "asp.h"
#define DA8XX_TPCC_BASE 0x01c00000
#define DA8XX_TPTC0_BASE 0x01c08000
@@ -505,15 +506,8 @@
.resource = da850_mcasp_resources,
};
-static struct platform_device davinci_pcm_device = {
- .name = "davinci-pcm-audio",
- .id = -1,
-};
-
void __init da8xx_register_mcasp(int id, struct snd_platform_data *pdata)
{
- platform_device_register(&davinci_pcm_device);
-
/* DA830/OMAP-L137 has 3 instances of McASP */
if (cpu_is_davinci_da830() && id == 1) {
da830_mcasp1_device.dev.platform_data = pdata;
diff --git a/arch/arm/mach-davinci/devices.c b/arch/arm/mach-davinci/devices.c
index d2f96662..70056fb 100644
--- a/arch/arm/mach-davinci/devices.c
+++ b/arch/arm/mach-davinci/devices.c
@@ -313,16 +313,6 @@
/*-------------------------------------------------------------------------*/
-static struct platform_device davinci_pcm_device = {
- .name = "davinci-pcm-audio",
- .id = -1,
-};
-
-static void davinci_init_pcm(void)
-{
- platform_device_register(&davinci_pcm_device);
-}
-
/*-------------------------------------------------------------------------*/
struct davinci_timer_instance davinci_timer_instance[2] = {
@@ -345,7 +335,6 @@
/* please keep these calls, and their implementations above,
* in alphabetical order so they're easier to sort through.
*/
- davinci_init_pcm();
davinci_init_wdt();
return 0;
diff --git a/arch/arm/mach-davinci/dm355.c b/arch/arm/mach-davinci/dm355.c
index 678cd99..e47a3f0 100644
--- a/arch/arm/mach-davinci/dm355.c
+++ b/arch/arm/mach-davinci/dm355.c
@@ -26,13 +26,13 @@
#include <mach/time.h>
#include <mach/serial.h>
#include <mach/common.h>
-#include <mach/asp.h>
#include <mach/spi.h>
#include <mach/gpio-davinci.h>
#include "davinci.h"
#include "clock.h"
#include "mux.h"
+#include "asp.h"
#define DM355_UART2_BASE (IO_PHYS + 0x206000)
diff --git a/arch/arm/mach-davinci/dm365.c b/arch/arm/mach-davinci/dm365.c
index a50d49de..f473745 100644
--- a/arch/arm/mach-davinci/dm365.c
+++ b/arch/arm/mach-davinci/dm365.c
@@ -29,7 +29,6 @@
#include <mach/time.h>
#include <mach/serial.h>
#include <mach/common.h>
-#include <mach/asp.h>
#include <mach/keyscan.h>
#include <mach/spi.h>
#include <mach/gpio-davinci.h>
@@ -37,6 +36,7 @@
#include "davinci.h"
#include "clock.h"
#include "mux.h"
+#include "asp.h"
#define DM365_REF_FREQ 24000000 /* 24 MHz on the DM365 EVM */
diff --git a/arch/arm/mach-davinci/dm644x.c b/arch/arm/mach-davinci/dm644x.c
index c8b8666..0755d46 100644
--- a/arch/arm/mach-davinci/dm644x.c
+++ b/arch/arm/mach-davinci/dm644x.c
@@ -23,12 +23,12 @@
#include <mach/time.h>
#include <mach/serial.h>
#include <mach/common.h>
-#include <mach/asp.h>
#include <mach/gpio-davinci.h>
#include "davinci.h"
#include "clock.h"
#include "mux.h"
+#include "asp.h"
/*
* Device specific clocks
diff --git a/arch/arm/mach-davinci/dm646x.c b/arch/arm/mach-davinci/dm646x.c
index 9eb87c1..97c0f8e 100644
--- a/arch/arm/mach-davinci/dm646x.c
+++ b/arch/arm/mach-davinci/dm646x.c
@@ -24,12 +24,12 @@
#include <mach/time.h>
#include <mach/serial.h>
#include <mach/common.h>
-#include <mach/asp.h>
#include <mach/gpio-davinci.h>
#include "davinci.h"
#include "clock.h"
#include "mux.h"
+#include "asp.h"
#define DAVINCI_VPIF_BASE (0x01C12000)
diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h
deleted file mode 100644
index 9aa2409..0000000
--- a/arch/arm/mach-davinci/include/mach/asp.h
+++ /dev/null
@@ -1,137 +0,0 @@
-/*
- * <mach/asp.h> - DaVinci Audio Serial Port support
- */
-#ifndef __ASM_ARCH_DAVINCI_ASP_H
-#define __ASM_ARCH_DAVINCI_ASP_H
-
-#include <mach/irqs.h>
-#include <mach/edma.h>
-
-/* Bases of dm644x and dm355 register banks */
-#define DAVINCI_ASP0_BASE 0x01E02000
-#define DAVINCI_ASP1_BASE 0x01E04000
-
-/* Bases of dm365 register banks */
-#define DAVINCI_DM365_ASP0_BASE 0x01D02000
-
-/* Bases of dm646x register banks */
-#define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000
-#define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800
-
-/* Bases of da850/da830 McASP0 register banks */
-#define DAVINCI_DA8XX_MCASP0_REG_BASE 0x01D00000
-
-/* Bases of da830 McASP1 register banks */
-#define DAVINCI_DA830_MCASP1_REG_BASE 0x01D04000
-
-/* EDMA channels of dm644x and dm355 */
-#define DAVINCI_DMA_ASP0_TX 2
-#define DAVINCI_DMA_ASP0_RX 3
-#define DAVINCI_DMA_ASP1_TX 8
-#define DAVINCI_DMA_ASP1_RX 9
-
-/* EDMA channels of dm646x */
-#define DAVINCI_DM646X_DMA_MCASP0_AXEVT0 6
-#define DAVINCI_DM646X_DMA_MCASP0_AREVT0 9
-#define DAVINCI_DM646X_DMA_MCASP1_AXEVT1 12
-
-/* EDMA channels of da850/da830 McASP0 */
-#define DAVINCI_DA8XX_DMA_MCASP0_AREVT 0
-#define DAVINCI_DA8XX_DMA_MCASP0_AXEVT 1
-
-/* EDMA channels of da830 McASP1 */
-#define DAVINCI_DA830_DMA_MCASP1_AREVT 2
-#define DAVINCI_DA830_DMA_MCASP1_AXEVT 3
-
-/* Interrupts */
-#define DAVINCI_ASP0_RX_INT IRQ_MBRINT
-#define DAVINCI_ASP0_TX_INT IRQ_MBXINT
-#define DAVINCI_ASP1_RX_INT IRQ_MBRINT
-#define DAVINCI_ASP1_TX_INT IRQ_MBXINT
-
-struct snd_platform_data {
- u32 tx_dma_offset;
- u32 rx_dma_offset;
- enum dma_event_q asp_chan_q; /* event queue number for ASP channel */
- enum dma_event_q ram_chan_q; /* event queue number for RAM channel */
- unsigned int codec_fmt;
- /*
- * Allowing this is more efficient and eliminates left and right swaps
- * caused by underruns, but will swap the left and right channels
- * when compared to previous behavior.
- */
- unsigned enable_channel_combine:1;
- unsigned sram_size_playback;
- unsigned sram_size_capture;
-
- /*
- * If McBSP peripheral gets the clock from an external pin,
- * there are three chooses, that are MCBSP_CLKX, MCBSP_CLKR
- * and MCBSP_CLKS.
- * Depending on different hardware connections it is possible
- * to use this setting to change the behaviour of McBSP
- * driver. The dm365_clk_input_pin enum is available for dm365
- */
- int clk_input_pin;
-
- /*
- * This flag works when both clock and FS are outputs for the cpu
- * and makes clock more accurate (FS is not symmetrical and the
- * clock is very fast.
- * The clock becoming faster is named
- * i2s continuous serial clock (I2S_SCK) and it is an externally
- * visible bit clock.
- *
- * first line : WordSelect
- * second line : ContinuousSerialClock
- * third line: SerialData
- *
- * SYMMETRICAL APPROACH:
- * _______________________ LEFT
- * _| RIGHT |______________________|
- * _ _ _ _ _ _ _ _
- * _| |_| |_ x16 _| |_| |_| |_| |_ x16 _| |_| |_
- * _ _ _ _ _ _ _ _
- * _/ \_/ \_ ... _/ \_/ \_/ \_/ \_ ... _/ \_/ \_
- * \_/ \_/ \_/ \_/ \_/ \_/ \_/ \_/
- *
- * ACCURATE CLOCK APPROACH:
- * ______________ LEFT
- * _| RIGHT |_______________________________|
- * _ _ _ _ _ _ _ _ _
- * _| |_ x16 _| |_| |_ x16 _| |_| |_| |_| |_| |_| |
- * _ _ _ _ dummy cycles
- * _/ \_ ... _/ \_/ \_ ... _/ \__________________
- * \_/ \_/ \_/ \_/
- *
- */
- bool i2s_accurate_sck;
-
- /* McASP specific fields */
- int tdm_slots;
- u8 op_mode;
- u8 num_serializer;
- u8 *serial_dir;
- u8 version;
- u8 txnumevt;
- u8 rxnumevt;
-};
-
-enum {
- MCASP_VERSION_1 = 0, /* DM646x */
- MCASP_VERSION_2, /* DA8xx/OMAPL1x */
-};
-
-enum dm365_clk_input_pin {
- MCBSP_CLKR = 0, /* DM365 */
- MCBSP_CLKS,
-};
-
-#define INACTIVE_MODE 0
-#define TX_MODE 1
-#define RX_MODE 2
-
-#define DAVINCI_MCASP_IIS_MODE 0
-#define DAVINCI_MCASP_DIT_MODE 1
-
-#endif /* __ASM_ARCH_DAVINCI_ASP_H */
diff --git a/arch/arm/mach-davinci/include/mach/da8xx.h b/arch/arm/mach-davinci/include/mach/da8xx.h
index a2f1f27..c74a6ab 100644
--- a/arch/arm/mach-davinci/include/mach/da8xx.h
+++ b/arch/arm/mach-davinci/include/mach/da8xx.h
@@ -16,11 +16,11 @@
#include <linux/platform_device.h>
#include <linux/davinci_emac.h>
#include <linux/spi/spi.h>
+#include <linux/platform_data/davinci_asp.h>
#include <mach/serial.h>
#include <mach/edma.h>
#include <mach/i2c.h>
-#include <mach/asp.h>
#include <mach/mmc.h>
#include <mach/usb.h>
#include <mach/pm.h>
diff --git a/arch/blackfin/mach-bf527/boards/ezkit.c b/arch/blackfin/mach-bf527/boards/ezkit.c
index fc179ca..29f16e5 100644
--- a/arch/blackfin/mach-bf527/boards/ezkit.c
+++ b/arch/blackfin/mach-bf527/boards/ezkit.c
@@ -587,6 +587,21 @@
};
#endif
+#if defined(CONFIG_SND_BF5XX_SOC_AD1836) \
+ || defined(CONFIG_SND_BF5XX_SOC_AD1836_MODULE)
+static const char * const ad1836_link[] = {
+ "bfin-tdm.0",
+ "spi0.4",
+};
+static struct platform_device bfin_ad1836_machine = {
+ .name = "bfin-snd-ad1836",
+ .id = -1,
+ .dev = {
+ .platform_data = (void *)ad1836_link,
+ },
+};
+#endif
+
static struct spi_board_info bfin_spi_board_info[] __initdata = {
#if defined(CONFIG_MTD_M25P80) \
|| defined(CONFIG_MTD_M25P80_MODULE)
@@ -1269,6 +1284,11 @@
#if defined(CONFIG_SND_BF5XX_TDM) || defined(CONFIG_SND_BF5XX_TDM_MODULE)
&bfin_tdm,
#endif
+
+#if defined(CONFIG_SND_BF5XX_SOC_AD1836) || \
+ defined(CONFIG_SND_BF5XX_SOC_AD1836_MODULE)
+ &bfin_ad1836_machine,
+#endif
};
static int __init ezkit_init(void)
diff --git a/arch/blackfin/mach-bf533/boards/stamp.c b/arch/blackfin/mach-bf533/boards/stamp.c
index ce88a71..6fca869 100644
--- a/arch/blackfin/mach-bf533/boards/stamp.c
+++ b/arch/blackfin/mach-bf533/boards/stamp.c
@@ -617,6 +617,21 @@
};
#endif
+#if defined(CONFIG_SND_BF5XX_SOC_AD1836) \
+ || defined(CONFIG_SND_BF5XX_SOC_AD1836_MODULE)
+static const char * const ad1836_link[] = {
+ "bfin-tdm.0",
+ "spi0.4",
+};
+static struct platform_device bfin_ad1836_machine = {
+ .name = "bfin-snd-ad1836",
+ .id = -1,
+ .dev = {
+ .platform_data = (void *)ad1836_link,
+ },
+};
+#endif
+
#if defined(CONFIG_SND_BF5XX_SOC_AD73311) || \
defined(CONFIG_SND_BF5XX_SOC_AD73311_MODULE)
static const unsigned ad73311_gpio[] = {
@@ -754,6 +769,11 @@
&bfin_ac97_pcm,
#endif
+#if defined(CONFIG_SND_BF5XX_SOC_AD1836) || \
+ defined(CONFIG_SND_BF5XX_SOC_AD1836_MODULE)
+ &bfin_ad1836_machine,
+#endif
+
#if defined(CONFIG_SND_BF5XX_SOC_AD73311) || \
defined(CONFIG_SND_BF5XX_SOC_AD73311_MODULE)
&bfin_ad73311_machine,
diff --git a/arch/blackfin/mach-bf537/boards/stamp.c b/arch/blackfin/mach-bf537/boards/stamp.c
index 5ed654a..307bd7e 100644
--- a/arch/blackfin/mach-bf537/boards/stamp.c
+++ b/arch/blackfin/mach-bf537/boards/stamp.c
@@ -2641,6 +2641,21 @@
};
#endif
+#if defined(CONFIG_SND_BF5XX_SOC_AD1836) \
+ || defined(CONFIG_SND_BF5XX_SOC_AD1836_MODULE)
+static const char * const ad1836_link[] = {
+ "bfin-tdm.0",
+ "spi0.4",
+};
+static struct platform_device bfin_ad1836_machine = {
+ .name = "bfin-snd-ad1836",
+ .id = -1,
+ .dev = {
+ .platform_data = (void *)ad1836_link,
+ },
+};
+#endif
+
#if defined(CONFIG_SND_BF5XX_SOC_AD73311) || \
defined(CONFIG_SND_BF5XX_SOC_AD73311_MODULE)
static const unsigned ad73311_gpio[] = {
@@ -2927,6 +2942,11 @@
&bfin_ac97_pcm,
#endif
+#if defined(CONFIG_SND_BF5XX_SOC_AD1836) || \
+ defined(CONFIG_SND_BF5XX_SOC_AD1836_MODULE)
+ &bfin_ad1836_machine,
+#endif
+
#if defined(CONFIG_SND_BF5XX_SOC_AD73311) || \
defined(CONFIG_SND_BF5XX_SOC_AD73311_MODULE)
&bfin_ad73311_machine,
diff --git a/arch/blackfin/mach-bf561/boards/ezkit.c b/arch/blackfin/mach-bf561/boards/ezkit.c
index 7c36777..551f866 100644
--- a/arch/blackfin/mach-bf561/boards/ezkit.c
+++ b/arch/blackfin/mach-bf561/boards/ezkit.c
@@ -539,6 +539,21 @@
};
#endif
+#if defined(CONFIG_SND_BF5XX_SOC_AD1836) \
+ || defined(CONFIG_SND_BF5XX_SOC_AD1836_MODULE)
+static const char * const ad1836_link[] = {
+ "bfin-tdm.0",
+ "spi0.4",
+};
+static struct platform_device bfin_ad1836_machine = {
+ .name = "bfin-snd-ad1836",
+ .id = -1,
+ .dev = {
+ .platform_data = (void *)ad1836_link,
+ },
+};
+#endif
+
static struct platform_device *ezkit_devices[] __initdata = {
&bfin_dpmc,
@@ -603,6 +618,11 @@
#if defined(CONFIG_SND_BF5XX_AC97) || defined(CONFIG_SND_BF5XX_AC97_MODULE)
&bfin_ac97,
#endif
+
+#if defined(CONFIG_SND_BF5XX_SOC_AD1836) || \
+ defined(CONFIG_SND_BF5XX_SOC_AD1836_MODULE)
+ &bfin_ad1836_machine,
+#endif
};
static int __init net2272_init(void)
diff --git a/arch/blackfin/mach-bf609/boards/ezkit.c b/arch/blackfin/mach-bf609/boards/ezkit.c
index c2cf1ae..61c1f47 100644
--- a/arch/blackfin/mach-bf609/boards/ezkit.c
+++ b/arch/blackfin/mach-bf609/boards/ezkit.c
@@ -818,6 +818,21 @@
};
#endif
+#if defined(CONFIG_SND_BF5XX_SOC_AD1836) \
+ || defined(CONFIG_SND_BF5XX_SOC_AD1836_MODULE)
+static const char * const ad1836_link[] = {
+ "bfin-tdm.0",
+ "spi0.76",
+};
+static struct platform_device bfin_ad1836_machine = {
+ .name = "bfin-snd-ad1836",
+ .id = -1,
+ .dev = {
+ .platform_data = (void *)ad1836_link,
+ },
+};
+#endif
+
#if defined(CONFIG_SND_SOC_BFIN_EVAL_ADAU1X61) || \
defined(CONFIG_SND_SOC_BFIN_EVAL_ADAU1X61_MODULE)
static struct platform_device adau1761_device = {
@@ -1557,6 +1572,10 @@
defined(CONFIG_SND_BF6XX_SOC_I2S_MODULE)
&bfin_i2s,
#endif
+#if defined(CONFIG_SND_BF5XX_SOC_AD1836) || \
+ defined(CONFIG_SND_BF5XX_SOC_AD1836_MODULE)
+ &bfin_ad1836_machine,
+#endif
#if defined(CONFIG_SND_SOC_BFIN_EVAL_ADAU1X61) || \
defined(CONFIG_SND_SOC_BFIN_EVAL_ADAU1X61_MODULE)
&adau1761_device,
diff --git a/include/linux/mfd/wm8994/pdata.h b/include/linux/mfd/wm8994/pdata.h
index f0361c0..fc87be4 100644
--- a/include/linux/mfd/wm8994/pdata.h
+++ b/include/linux/mfd/wm8994/pdata.h
@@ -164,6 +164,10 @@
int num_micd_rates;
struct wm8958_micd_rate *micd_rates;
+ /* Power up delays to add after microphone bias power up (ms) */
+ int micb1_delay;
+ int micb2_delay;
+
/* LINEOUT can be differential or single ended */
unsigned int lineout1_diff:1;
unsigned int lineout2_diff:1;
diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h
new file mode 100644
index 0000000..79c26aa
--- /dev/null
+++ b/include/linux/platform_data/davinci_asp.h
@@ -0,0 +1,104 @@
+/*
+ * TI DaVinci Audio Serial Port support
+ *
+ * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com/
+ *
+ * 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 version 2.
+ *
+ * This program is distributed "as is" WITHOUT ANY WARRANTY of any
+ * kind, whether express or implied; without even the implied warranty
+ * of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#ifndef __DAVINCI_ASP_H
+#define __DAVINCI_ASP_H
+
+struct snd_platform_data {
+ u32 tx_dma_offset;
+ u32 rx_dma_offset;
+ int asp_chan_q; /* event queue number for ASP channel */
+ int ram_chan_q; /* event queue number for RAM channel */
+ unsigned int codec_fmt;
+ /*
+ * Allowing this is more efficient and eliminates left and right swaps
+ * caused by underruns, but will swap the left and right channels
+ * when compared to previous behavior.
+ */
+ unsigned enable_channel_combine:1;
+ unsigned sram_size_playback;
+ unsigned sram_size_capture;
+
+ /*
+ * If McBSP peripheral gets the clock from an external pin,
+ * there are three chooses, that are MCBSP_CLKX, MCBSP_CLKR
+ * and MCBSP_CLKS.
+ * Depending on different hardware connections it is possible
+ * to use this setting to change the behaviour of McBSP
+ * driver.
+ */
+ int clk_input_pin;
+
+ /*
+ * This flag works when both clock and FS are outputs for the cpu
+ * and makes clock more accurate (FS is not symmetrical and the
+ * clock is very fast.
+ * The clock becoming faster is named
+ * i2s continuous serial clock (I2S_SCK) and it is an externally
+ * visible bit clock.
+ *
+ * first line : WordSelect
+ * second line : ContinuousSerialClock
+ * third line: SerialData
+ *
+ * SYMMETRICAL APPROACH:
+ * _______________________ LEFT
+ * _| RIGHT |______________________|
+ * _ _ _ _ _ _ _ _
+ * _| |_| |_ x16 _| |_| |_| |_| |_ x16 _| |_| |_
+ * _ _ _ _ _ _ _ _
+ * _/ \_/ \_ ... _/ \_/ \_/ \_/ \_ ... _/ \_/ \_
+ * \_/ \_/ \_/ \_/ \_/ \_/ \_/ \_/
+ *
+ * ACCURATE CLOCK APPROACH:
+ * ______________ LEFT
+ * _| RIGHT |_______________________________|
+ * _ _ _ _ _ _ _ _ _
+ * _| |_ x16 _| |_| |_ x16 _| |_| |_| |_| |_| |_| |
+ * _ _ _ _ dummy cycles
+ * _/ \_ ... _/ \_/ \_ ... _/ \__________________
+ * \_/ \_/ \_/ \_/
+ *
+ */
+ bool i2s_accurate_sck;
+
+ /* McASP specific fields */
+ int tdm_slots;
+ u8 op_mode;
+ u8 num_serializer;
+ u8 *serial_dir;
+ u8 version;
+ u8 txnumevt;
+ u8 rxnumevt;
+};
+
+enum {
+ MCASP_VERSION_1 = 0, /* DM646x */
+ MCASP_VERSION_2, /* DA8xx/OMAPL1x */
+};
+
+enum mcbsp_clk_input_pin {
+ MCBSP_CLKR = 0, /* as in DM365 */
+ MCBSP_CLKS,
+};
+
+#define INACTIVE_MODE 0
+#define TX_MODE 1
+#define RX_MODE 2
+
+#define DAVINCI_MCASP_IIS_MODE 0
+#define DAVINCI_MCASP_DIT_MODE 1
+
+#endif
diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h
index 48f2a1f..f2912ab 100644
--- a/include/sound/compress_driver.h
+++ b/include/sound/compress_driver.h
@@ -61,6 +61,7 @@
u64 total_bytes_available;
u64 total_bytes_transferred;
wait_queue_head_t sleep;
+ void *private_data;
};
/**
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 1f69e0a..628db7b 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -18,6 +18,7 @@
struct snd_pcm_substream;
struct snd_soc_dapm_widget;
+struct snd_compr_stream;
/*
* DAI hardware audio formats.
@@ -205,6 +206,8 @@
int (*remove)(struct snd_soc_dai *dai);
int (*suspend)(struct snd_soc_dai *dai);
int (*resume)(struct snd_soc_dai *dai);
+ /* compress dai */
+ bool compress_dai;
/* ops */
const struct snd_soc_dai_ops *ops;
diff --git a/include/sound/soc.h b/include/sound/soc.h
index e063380..313b766 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -22,6 +22,7 @@
#include <linux/regmap.h>
#include <sound/core.h>
#include <sound/pcm.h>
+#include <sound/compress_driver.h>
#include <sound/control.h>
#include <sound/ac97_codec.h>
@@ -399,6 +400,7 @@
int snd_soc_platform_write(struct snd_soc_platform *platform,
unsigned int reg, unsigned int val);
int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num);
+int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num);
struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card,
const char *dai_link, int stream);
@@ -632,6 +634,13 @@
int (*trigger)(struct snd_pcm_substream *, int);
};
+struct snd_soc_compr_ops {
+ int (*startup)(struct snd_compr_stream *);
+ void (*shutdown)(struct snd_compr_stream *);
+ int (*set_params)(struct snd_compr_stream *);
+ int (*trigger)(struct snd_compr_stream *);
+};
+
/* SoC cache ops */
struct snd_soc_cache_ops {
const char *name;
@@ -787,9 +796,12 @@
snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
struct snd_soc_dai *);
- /* platform stream ops */
+ /* platform stream pcm ops */
struct snd_pcm_ops *ops;
+ /* platform stream compress ops */
+ struct snd_compr_ops *compr_ops;
+
/* platform stream completion event */
int (*stream_event)(struct snd_soc_dapm_context *dapm, int event);
@@ -891,6 +903,7 @@
/* machine stream operations */
struct snd_soc_ops *ops;
+ struct snd_soc_compr_ops *compr_ops;
};
struct snd_soc_codec_conf {
@@ -1027,6 +1040,7 @@
/* runtime devices */
struct snd_pcm *pcm;
+ struct snd_compr *compr;
struct snd_soc_codec *codec;
struct snd_soc_platform *platform;
struct snd_soc_dai *codec_dai;
diff --git a/include/sound/wm0010.h b/include/sound/wm0010.h
new file mode 100644
index 0000000..3261e90
--- /dev/null
+++ b/include/sound/wm0010.h
@@ -0,0 +1,27 @@
+/*
+ * wm0010.h -- Platform data for WM0010 DSP Driver
+ *
+ * Copyright 2012 Wolfson Microelectronics PLC.
+ *
+ * Author: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
+ *
+ * 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.
+ */
+
+#ifndef WM0010_PDATA_H
+#define WM0010_PDATA_H
+
+struct wm0010_pdata {
+ int gpio_reset;
+
+ /* Set if there is an inverter between the GPIO controlling
+ * the reset signal and the device.
+ */
+ int reset_active_high;
+ int irq_flags;
+};
+
+#endif
diff --git a/include/sound/wm8993.h b/include/sound/wm8993.h
index eee19f6..8016fd8 100644
--- a/include/sound/wm8993.h
+++ b/include/sound/wm8993.h
@@ -32,6 +32,10 @@
unsigned int lineout1fb:1;
unsigned int lineout2fb:1;
+ /* Delay to add for microphones to stabalise after power up */
+ int micbias1_delay;
+ int micbias2_delay;
+
/* Microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */
unsigned int micbias1_lvl:1;
unsigned int micbias2_lvl:1;
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index c5de0a8..5da8ca7 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -9,6 +9,7 @@
select SND_JACK if INPUT=y || INPUT=SND
select REGMAP_I2C if I2C
select REGMAP_SPI if SPI_MASTER
+ select SND_COMPRESS_OFFLOAD
---help---
If you want ASoC support, you should say Y here and also to the
@@ -32,9 +33,9 @@
source "sound/soc/atmel/Kconfig"
source "sound/soc/au1x/Kconfig"
source "sound/soc/blackfin/Kconfig"
+source "sound/soc/cirrus/Kconfig"
source "sound/soc/davinci/Kconfig"
source "sound/soc/dwc/Kconfig"
-source "sound/soc/ep93xx/Kconfig"
source "sound/soc/fsl/Kconfig"
source "sound/soc/jz4740/Kconfig"
source "sound/soc/nuc900/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 00a555a..bcbf1d0 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,5 +1,5 @@
snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o
-snd-soc-core-objs += soc-pcm.o soc-io.o
+snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o
snd-soc-dmaengine-pcm-objs := soc-dmaengine-pcm.o
obj-$(CONFIG_SND_SOC_DMAENGINE_PCM) += snd-soc-dmaengine-pcm.o
@@ -10,9 +10,9 @@
obj-$(CONFIG_SND_SOC) += atmel/
obj-$(CONFIG_SND_SOC) += au1x/
obj-$(CONFIG_SND_SOC) += blackfin/
+obj-$(CONFIG_SND_SOC) += cirrus/
obj-$(CONFIG_SND_SOC) += davinci/
obj-$(CONFIG_SND_SOC) += dwc/
-obj-$(CONFIG_SND_SOC) += ep93xx/
obj-$(CONFIG_SND_SOC) += fsl/
obj-$(CONFIG_SND_SOC) += jz4740/
obj-$(CONFIG_SND_SOC) += mid-x86/
diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c
index d542d40..16b9c9e 100644
--- a/sound/soc/blackfin/bf5xx-ad1836.c
+++ b/sound/soc/blackfin/bf5xx-ad1836.c
@@ -59,62 +59,63 @@
#define BF5XX_AD1836_DAIFMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | \
SND_SOC_DAIFMT_CBM_CFM)
-static struct snd_soc_dai_link bf5xx_ad1836_dai[] = {
- {
- .name = "ad1836",
- .stream_name = "AD1836",
- .cpu_dai_name = "bfin-tdm.0",
- .codec_dai_name = "ad1836-hifi",
- .platform_name = "bfin-tdm-pcm-audio",
- .codec_name = "spi0.4",
- .ops = &bf5xx_ad1836_ops,
- .dai_fmt = BF5XX_AD1836_DAIFMT,
- },
- {
- .name = "ad1836",
- .stream_name = "AD1836",
- .cpu_dai_name = "bfin-tdm.1",
- .codec_dai_name = "ad1836-hifi",
- .platform_name = "bfin-tdm-pcm-audio",
- .codec_name = "spi0.4",
- .ops = &bf5xx_ad1836_ops,
- .dai_fmt = BF5XX_AD1836_DAIFMT,
- },
+static struct snd_soc_dai_link bf5xx_ad1836_dai = {
+ .name = "ad1836",
+ .stream_name = "AD1836",
+ .codec_dai_name = "ad1836-hifi",
+ .platform_name = "bfin-tdm-pcm-audio",
+ .ops = &bf5xx_ad1836_ops,
+ .dai_fmt = BF5XX_AD1836_DAIFMT,
};
static struct snd_soc_card bf5xx_ad1836 = {
.name = "bfin-ad1836",
.owner = THIS_MODULE,
- .dai_link = &bf5xx_ad1836_dai[CONFIG_SND_BF5XX_SPORT_NUM],
+ .dai_link = &bf5xx_ad1836_dai,
.num_links = 1,
};
-static struct platform_device *bfxx_ad1836_snd_device;
-
-static int __init bf5xx_ad1836_init(void)
+static __devinit int bf5xx_ad1836_driver_probe(struct platform_device *pdev)
{
+ struct snd_soc_card *card = &bf5xx_ad1836;
+ const char **link_name;
int ret;
- bfxx_ad1836_snd_device = platform_device_alloc("soc-audio", -1);
- if (!bfxx_ad1836_snd_device)
- return -ENOMEM;
+ link_name = pdev->dev.platform_data;
+ if (!link_name) {
+ dev_err(&pdev->dev, "No platform data supplied\n");
+ return -EINVAL;
+ }
+ bf5xx_ad1836_dai.cpu_dai_name = link_name[0];
+ bf5xx_ad1836_dai.codec_name = link_name[1];
- platform_set_drvdata(bfxx_ad1836_snd_device, &bf5xx_ad1836);
- ret = platform_device_add(bfxx_ad1836_snd_device);
+ card->dev = &pdev->dev;
+ platform_set_drvdata(pdev, card);
+ ret = snd_soc_register_card(card);
if (ret)
- platform_device_put(bfxx_ad1836_snd_device);
-
+ dev_err(&pdev->dev, "Failed to register card\n");
return ret;
}
-static void __exit bf5xx_ad1836_exit(void)
+static int __devexit bf5xx_ad1836_driver_remove(struct platform_device *pdev)
{
- platform_device_unregister(bfxx_ad1836_snd_device);
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
+ return 0;
}
-module_init(bf5xx_ad1836_init);
-module_exit(bf5xx_ad1836_exit);
+static struct platform_driver bf5xx_ad1836_driver = {
+ .driver = {
+ .name = "bfin-snd-ad1836",
+ .owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = bf5xx_ad1836_driver_probe,
+ .remove = __devexit_p(bf5xx_ad1836_driver_remove),
+};
+module_platform_driver(bf5xx_ad1836_driver);
/* Module information */
MODULE_AUTHOR("Barry Song");
diff --git a/sound/soc/ep93xx/Kconfig b/sound/soc/cirrus/Kconfig
similarity index 100%
rename from sound/soc/ep93xx/Kconfig
rename to sound/soc/cirrus/Kconfig
diff --git a/sound/soc/ep93xx/Makefile b/sound/soc/cirrus/Makefile
similarity index 100%
rename from sound/soc/ep93xx/Makefile
rename to sound/soc/cirrus/Makefile
diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/cirrus/edb93xx.c
similarity index 100%
rename from sound/soc/ep93xx/edb93xx.c
rename to sound/soc/cirrus/edb93xx.c
diff --git a/sound/soc/ep93xx/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c
similarity index 100%
rename from sound/soc/ep93xx/ep93xx-ac97.c
rename to sound/soc/cirrus/ep93xx-ac97.c
diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c
similarity index 100%
rename from sound/soc/ep93xx/ep93xx-i2s.c
rename to sound/soc/cirrus/ep93xx-i2s.c
diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c
similarity index 100%
rename from sound/soc/ep93xx/ep93xx-pcm.c
rename to sound/soc/cirrus/ep93xx-pcm.c
diff --git a/sound/soc/ep93xx/ep93xx-pcm.h b/sound/soc/cirrus/ep93xx-pcm.h
similarity index 100%
rename from sound/soc/ep93xx/ep93xx-pcm.h
rename to sound/soc/cirrus/ep93xx-pcm.h
diff --git a/sound/soc/ep93xx/simone.c b/sound/soc/cirrus/simone.c
similarity index 100%
rename from sound/soc/ep93xx/simone.c
rename to sound/soc/cirrus/simone.c
diff --git a/sound/soc/ep93xx/snappercl15.c b/sound/soc/cirrus/snappercl15.c
similarity index 100%
rename from sound/soc/ep93xx/snappercl15.c
rename to sound/soc/cirrus/snappercl15.c
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 9f8e859..3684255 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -70,6 +70,7 @@
select SND_SOC_UDA134X
select SND_SOC_UDA1380 if I2C
select SND_SOC_WL1273 if MFD_WL1273_CORE
+ select SND_SOC_WM0010 if SPI_MASTER
select SND_SOC_WM1250_EV1 if I2C
select SND_SOC_WM2000 if I2C
select SND_SOC_WM2200 if I2C
@@ -326,6 +327,9 @@
config SND_SOC_WL1273
tristate
+config SND_SOC_WM0010
+ tristate
+
config SND_SOC_WM1250_EV1
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 34148bb..ca508b2 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -61,6 +61,7 @@
snd-soc-uda134x-objs := uda134x.o
snd-soc-uda1380-objs := uda1380.o
snd-soc-wl1273-objs := wl1273.o
+snd-soc-wm0010-objs := wm0010.o
snd-soc-wm1250-ev1-objs := wm1250-ev1.o
snd-soc-wm2000-objs := wm2000.o
snd-soc-wm2200-objs := wm2200.o
@@ -177,6 +178,7 @@
obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o
obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o
obj-$(CONFIG_SND_SOC_WL1273) += snd-soc-wl1273.o
+obj-$(CONFIG_SND_SOC_WM0010) += snd-soc-wm0010.o
obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o
obj-$(CONFIG_SND_SOC_WM2000) += snd-soc-wm2000.o
obj-$(CONFIG_SND_SOC_WM2200) += snd-soc-wm2200.o
diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c
index 23b4018..b783650 100644
--- a/sound/soc/codecs/ab8500-codec.c
+++ b/sound/soc/codecs/ab8500-codec.c
@@ -2404,12 +2404,12 @@
dev_dbg(dev, "%s: Enter.\n", __func__);
- /* Setup AB8500 according to board-settings */
- pdata = (struct ab8500_platform_data *)dev_get_platdata(dev->parent);
-
/* Inform SoC Core that we have our own I/O arrangements. */
codec->control_data = (void *)true;
+ /* Setup AB8500 according to board-settings */
+ pdata = dev_get_platdata(dev->parent);
+
status = ab8500_audio_setup_mics(codec, &pdata->codec->amics);
if (status < 0) {
pr_err("%s: Failed to setup mics (%d)!\n", __func__, status);
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index c67b50d..ae1eb51 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -379,17 +379,7 @@
.id_table = ad1836_ids,
};
-static int __init ad1836_init(void)
-{
- return spi_register_driver(&ad1836_spi_driver);
-}
-module_init(ad1836_init);
-
-static void __exit ad1836_exit(void)
-{
- spi_unregister_driver(&ad1836_spi_driver);
-}
-module_exit(ad1836_exit);
+module_spi_driver(ad1836_spi_driver);
MODULE_DESCRIPTION("ASoC ad1836 driver");
MODULE_AUTHOR("Barry Song <21cnbao@gmail.com>");
diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
index 44f5906..704544b 100644
--- a/sound/soc/codecs/adau1373.c
+++ b/sound/soc/codecs/adau1373.c
@@ -1392,17 +1392,7 @@
.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_i2c_driver(adau1373_i2c_driver);
MODULE_DESCRIPTION("ASoC ADAU1373 driver");
MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
index 3d50fc8..51f2f3c 100644
--- a/sound/soc/codecs/adau1701.c
+++ b/sound/soc/codecs/adau1701.c
@@ -527,17 +527,7 @@
.id_table = adau1701_i2c_id,
};
-static int __init adau1701_init(void)
-{
- return i2c_add_driver(&adau1701_i2c_driver);
-}
-module_init(adau1701_init);
-
-static void __exit adau1701_exit(void)
-{
- i2c_del_driver(&adau1701_i2c_driver);
-}
-module_exit(adau1701_exit);
+module_i2c_driver(adau1701_i2c_driver);
MODULE_DESCRIPTION("ASoC ADAU1701 SigmaDSP driver");
MODULE_AUTHOR("Cliff Cai <cliff.cai@analog.com>");
diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c
index 5fb7c2a..2b45797 100644
--- a/sound/soc/codecs/ak4671.c
+++ b/sound/soc/codecs/ak4671.c
@@ -696,17 +696,7 @@
.id_table = ak4671_i2c_id,
};
-static int __init ak4671_modinit(void)
-{
- return i2c_add_driver(&ak4671_i2c_driver);
-}
-module_init(ak4671_modinit);
-
-static void __exit ak4671_exit(void)
-{
- i2c_del_driver(&ak4671_i2c_driver);
-}
-module_exit(ak4671_exit);
+module_i2c_driver(ak4671_i2c_driver);
MODULE_DESCRIPTION("ASoC AK4671 codec driver");
MODULE_AUTHOR("Joonyoung Shim <jy0922.shim@samsung.com>");
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 5c9caca..5e96a0a 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -229,6 +229,69 @@
}
EXPORT_SYMBOL_GPL(arizona_out_ev);
+static unsigned int arizona_sysclk_48k_rates[] = {
+ 6144000,
+ 12288000,
+ 22579200,
+ 49152000,
+};
+
+static unsigned int arizona_sysclk_44k1_rates[] = {
+ 5644800,
+ 11289600,
+ 24576000,
+ 45158400,
+};
+
+static int arizona_set_opclk(struct snd_soc_codec *codec, unsigned int clk,
+ unsigned int freq)
+{
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+ unsigned int reg;
+ unsigned int *rates;
+ int ref, div, refclk;
+
+ switch (clk) {
+ case ARIZONA_CLK_OPCLK:
+ reg = ARIZONA_OUTPUT_SYSTEM_CLOCK;
+ refclk = priv->sysclk;
+ break;
+ case ARIZONA_CLK_ASYNC_OPCLK:
+ reg = ARIZONA_OUTPUT_ASYNC_CLOCK;
+ refclk = priv->asyncclk;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (refclk % 8000)
+ rates = arizona_sysclk_44k1_rates;
+ else
+ rates = arizona_sysclk_48k_rates;
+
+ for (ref = 0; ref < ARRAY_SIZE(arizona_sysclk_48k_rates) &&
+ rates[ref] <= refclk; ref++) {
+ div = 1;
+ while (rates[ref] / div >= freq && div < 32) {
+ if (rates[ref] / div == freq) {
+ dev_dbg(codec->dev, "Configured %dHz OPCLK\n",
+ freq);
+ snd_soc_update_bits(codec, reg,
+ ARIZONA_OPCLK_DIV_MASK |
+ ARIZONA_OPCLK_SEL_MASK,
+ (div <<
+ ARIZONA_OPCLK_DIV_SHIFT) |
+ ref);
+ return 0;
+ }
+ div++;
+ }
+ }
+
+ dev_err(codec->dev, "Unable to generate %dHz OPCLK\n", freq);
+ return -EINVAL;
+}
+
int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id,
int source, unsigned int freq, int dir)
{
@@ -252,6 +315,9 @@
reg = ARIZONA_ASYNC_CLOCK_1;
clk = &priv->asyncclk;
break;
+ case ARIZONA_CLK_OPCLK:
+ case ARIZONA_CLK_ASYNC_OPCLK:
+ return arizona_set_opclk(codec, clk_id, freq);
default:
return -EINVAL;
}
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 59caca8..eb66b52 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -17,8 +17,10 @@
#include <sound/soc.h>
-#define ARIZONA_CLK_SYSCLK 1
-#define ARIZONA_CLK_ASYNCCLK 2
+#define ARIZONA_CLK_SYSCLK 1
+#define ARIZONA_CLK_ASYNCCLK 2
+#define ARIZONA_CLK_OPCLK 3
+#define ARIZONA_CLK_ASYNC_OPCLK 4
#define ARIZONA_CLK_SRC_MCLK1 0x0
#define ARIZONA_CLK_SRC_MCLK2 0x1
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 047917f..44a176f 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -29,6 +29,8 @@
#include <linux/i2c.h>
#include <linux/delay.h>
#include <linux/regulator/consumer.h>
+#include <linux/of_device.h>
+#include <linux/of_gpio.h>
/*
* The codec isn't really big-endian or little-endian, since the I2S
@@ -639,6 +641,15 @@
.reg_cache_default = cs4270_default_reg_cache,
};
+/*
+ * cs4270_of_match - the device tree bindings
+ */
+static const struct of_device_id cs4270_of_match[] = {
+ { .compatible = "cirrus,cs4270", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, cs4270_of_match);
+
/**
* cs4270_i2c_probe - initialize the I2C interface of the CS4270
* @i2c_client: the I2C client object
@@ -650,9 +661,25 @@
static int cs4270_i2c_probe(struct i2c_client *i2c_client,
const struct i2c_device_id *id)
{
+ struct device_node *np = i2c_client->dev.of_node;
struct cs4270_private *cs4270;
int ret;
+ /* See if we have a way to bring the codec out of reset */
+ if (np) {
+ enum of_gpio_flags flags;
+ int gpio = of_get_named_gpio_flags(np, "reset-gpio", 0, &flags);
+
+ if (gpio_is_valid(gpio)) {
+ ret = devm_gpio_request_one(&i2c_client->dev, gpio,
+ flags & OF_GPIO_ACTIVE_LOW ?
+ GPIOF_OUT_INIT_LOW : GPIOF_OUT_INIT_HIGH,
+ "cs4270 reset");
+ if (ret < 0)
+ return ret;
+ }
+ }
+
/* Verify that we have a CS4270 */
ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID);
@@ -718,23 +745,14 @@
.driver = {
.name = "cs4270",
.owner = THIS_MODULE,
+ .of_match_table = cs4270_of_match,
},
.id_table = cs4270_id,
.probe = cs4270_i2c_probe,
.remove = cs4270_i2c_remove,
};
-static int __init cs4270_init(void)
-{
- return i2c_add_driver(&cs4270_i2c_driver);
-}
-module_init(cs4270_init);
-
-static void __exit cs4270_exit(void)
-{
- i2c_del_driver(&cs4270_i2c_driver);
-}
-module_exit(cs4270_exit);
+module_i2c_driver(cs4270_i2c_driver);
MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
MODULE_DESCRIPTION("Cirrus Logic CS4270 ALSA SoC Codec Driver");
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index 091d019..1e0fa3b 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -614,24 +614,7 @@
.remove = cs42l51_i2c_remove,
};
-static int __init cs42l51_init(void)
-{
- int ret;
-
- ret = i2c_add_driver(&cs42l51_i2c_driver);
- if (ret != 0) {
- printk(KERN_ERR "%s: can't add i2c driver\n", __func__);
- return ret;
- }
- return 0;
-}
-module_init(cs42l51_init);
-
-static void __exit cs42l51_exit(void)
-{
- i2c_del_driver(&cs42l51_i2c_driver);
-}
-module_exit(cs42l51_exit);
+module_i2c_driver(cs42l51_i2c_driver);
MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
MODULE_DESCRIPTION("Cirrus Logic CS42L51 ALSA SoC Codec Driver");
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 628daf6..6159929 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -24,7 +24,6 @@
#include <linux/slab.h>
#include <linux/workqueue.h>
#include <linux/platform_device.h>
-#include <linux/slab.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c
index 5d8f39e..1bf5560 100644
--- a/sound/soc/codecs/isabelle.c
+++ b/sound/soc/codecs/isabelle.c
@@ -13,7 +13,6 @@
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
-#include <linux/version.h>
#include <linux/kernel.h>
#include <linux/init.h>
#include <linux/delay.h>
diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c
index ba4fafb..81a328c 100644
--- a/sound/soc/codecs/lm4857.c
+++ b/sound/soc/codecs/lm4857.c
@@ -250,17 +250,7 @@
.id_table = lm4857_i2c_id,
};
-static int __init lm4857_init(void)
-{
- return i2c_add_driver(&lm4857_i2c_driver);
-}
-module_init(lm4857_init);
-
-static void __exit lm4857_exit(void)
-{
- i2c_del_driver(&lm4857_i2c_driver);
-}
-module_exit(lm4857_exit);
+module_i2c_driver(lm4857_i2c_driver);
MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
MODULE_DESCRIPTION("LM4857 amplifier driver");
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index af7324b..3264a51 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -2107,23 +2107,7 @@
.id_table = max98088_i2c_id,
};
-static int __init max98088_init(void)
-{
- int ret;
-
- ret = i2c_add_driver(&max98088_i2c_driver);
- if (ret)
- pr_err("Failed to register max98088 I2C driver: %d\n", ret);
-
- return ret;
-}
-module_init(max98088_init);
-
-static void __exit max98088_exit(void)
-{
- i2c_del_driver(&max98088_i2c_driver);
-}
-module_exit(max98088_exit);
+module_i2c_driver(max98088_i2c_driver);
MODULE_DESCRIPTION("ALSA SoC MAX98088 driver");
MODULE_AUTHOR("Peter Hsiang, Jesse Marroquin");
diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index 7cd508e..38d43c5 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -2533,23 +2533,7 @@
.id_table = max98095_i2c_id,
};
-static int __init max98095_init(void)
-{
- int ret;
-
- ret = i2c_add_driver(&max98095_i2c_driver);
- if (ret)
- pr_err("Failed to register max98095 I2C driver: %d\n", ret);
-
- return ret;
-}
-module_init(max98095_init);
-
-static void __exit max98095_exit(void)
-{
- i2c_del_driver(&max98095_i2c_driver);
-}
-module_exit(max98095_exit);
+module_i2c_driver(max98095_i2c_driver);
MODULE_DESCRIPTION("ALSA SoC MAX98095 driver");
MODULE_AUTHOR("Peter Hsiang");
diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c
index a191309..efe535c 100644
--- a/sound/soc/codecs/max9850.c
+++ b/sound/soc/codecs/max9850.c
@@ -369,17 +369,7 @@
.id_table = max9850_i2c_id,
};
-static int __init max9850_init(void)
-{
- return i2c_add_driver(&max9850_i2c_driver);
-}
-module_init(max9850_init);
-
-static void __exit max9850_exit(void)
-{
- i2c_del_driver(&max9850_i2c_driver);
-}
-module_exit(max9850_exit);
+module_i2c_driver(max9850_i2c_driver);
MODULE_AUTHOR("Christian Glindkamp <christian.glindkamp@taskit.de>");
MODULE_DESCRIPTION("ASoC MAX9850 codec driver");
diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c
index 3a2ba3d..d15e594 100644
--- a/sound/soc/codecs/max9877.c
+++ b/sound/soc/codecs/max9877.c
@@ -291,17 +291,7 @@
.id_table = max9877_i2c_id,
};
-static int __init max9877_init(void)
-{
- return i2c_add_driver(&max9877_i2c_driver);
-}
-module_init(max9877_init);
-
-static void __exit max9877_exit(void)
-{
- i2c_del_driver(&max9877_i2c_driver);
-}
-module_exit(max9877_exit);
+module_i2c_driver(max9877_i2c_driver);
MODULE_DESCRIPTION("ASoC MAX9877 amp driver");
MODULE_AUTHOR("Joonyoung Shim <jy0922.shim@samsung.com>");
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 8d717f4..51b7313 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -1006,17 +1006,7 @@
.id_table = sta32x_i2c_id,
};
-static int __init sta32x_init(void)
-{
- return i2c_add_driver(&sta32x_i2c_driver);
-}
-module_init(sta32x_init);
-
-static void __exit sta32x_exit(void)
-{
- i2c_del_driver(&sta32x_i2c_driver);
-}
-module_exit(sta32x_exit);
+module_i2c_driver(sta32x_i2c_driver);
MODULE_DESCRIPTION("ASoC STA32X driver");
MODULE_AUTHOR("Johannes Stezenbach <js@sig21.net>");
diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c
index 0c225cd..9e31448 100644
--- a/sound/soc/codecs/sta529.c
+++ b/sound/soc/codecs/sta529.c
@@ -358,7 +358,7 @@
return 0;
}
-struct snd_soc_codec_driver sta529_codec_driver = {
+static const struct snd_soc_codec_driver sta529_codec_driver = {
.probe = sta529_probe,
.remove = sta529_remove,
.set_bias_level = sta529_set_bias_level,
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index 85944e9..b1f6982 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -444,14 +444,4 @@
.remove = aic26_spi_remove,
};
-static int __init aic26_init(void)
-{
- return spi_register_driver(&aic26_spi);
-}
-module_init(aic26_init);
-
-static void __exit aic26_exit(void)
-{
- spi_unregister_driver(&aic26_spi);
-}
-module_exit(aic26_exit);
+module_spi_driver(aic26_spi);
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index b0a73d3..f230292 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -746,24 +746,7 @@
.id_table = aic32x4_i2c_id,
};
-static int __init aic32x4_modinit(void)
-{
- int ret = 0;
-
- ret = i2c_add_driver(&aic32x4_i2c_driver);
- if (ret != 0) {
- printk(KERN_ERR "Failed to register aic32x4 I2C driver: %d\n",
- ret);
- }
- return ret;
-}
-module_init(aic32x4_modinit);
-
-static void __exit aic32x4_exit(void)
-{
- i2c_del_driver(&aic32x4_i2c_driver);
-}
-module_exit(aic32x4_exit);
+module_i2c_driver(aic32x4_i2c_driver);
MODULE_DESCRIPTION("ASoC tlv320aic32x4 codec driver");
MODULE_AUTHOR("Javier Martin <javier.martin@vista-silicon.com>");
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index dc78f5a..5708a97 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -40,6 +40,7 @@
#include <linux/i2c.h>
#include <linux/gpio.h>
#include <linux/regulator/consumer.h>
+#include <linux/of_gpio.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -1457,6 +1458,8 @@
{
struct aic3x_pdata *pdata = i2c->dev.platform_data;
struct aic3x_priv *aic3x;
+ struct aic3x_setup_data *ai3x_setup;
+ struct device_node *np = i2c->dev.of_node;
int ret;
aic3x = devm_kzalloc(&i2c->dev, sizeof(struct aic3x_priv), GFP_KERNEL);
@@ -1471,6 +1474,25 @@
if (pdata) {
aic3x->gpio_reset = pdata->gpio_reset;
aic3x->setup = pdata->setup;
+ } else if (np) {
+ ai3x_setup = devm_kzalloc(&i2c->dev, sizeof(*ai3x_setup),
+ GFP_KERNEL);
+ if (ai3x_setup == NULL) {
+ dev_err(&i2c->dev, "failed to create private data\n");
+ return -ENOMEM;
+ }
+
+ ret = of_get_named_gpio(np, "gpio-reset", 0);
+ if (ret >= 0)
+ aic3x->gpio_reset = ret;
+ else
+ aic3x->gpio_reset = -1;
+
+ if (of_property_read_u32_array(np, "ai3x-gpio-func",
+ ai3x_setup->gpio_func, 2) >= 0) {
+ aic3x->setup = ai3x_setup;
+ }
+
} else {
aic3x->gpio_reset = -1;
}
@@ -1488,34 +1510,27 @@
return 0;
}
+#if defined(CONFIG_OF)
+static const struct of_device_id tlv320aic3x_of_match[] = {
+ { .compatible = "ti,tlv320aic3x", },
+ {},
+};
+MODULE_DEVICE_TABLE(of, tlv320aic3x_of_match);
+#endif
+
/* machine i2c codec control layer */
static struct i2c_driver aic3x_i2c_driver = {
.driver = {
.name = "tlv320aic3x-codec",
.owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(tlv320aic3x_of_match),
},
.probe = aic3x_i2c_probe,
.remove = aic3x_i2c_remove,
.id_table = aic3x_i2c_id,
};
-static int __init aic3x_modinit(void)
-{
- int ret = 0;
- ret = i2c_add_driver(&aic3x_i2c_driver);
- if (ret != 0) {
- printk(KERN_ERR "Failed to register TLV320AIC3x I2C driver: %d\n",
- ret);
- }
- return ret;
-}
-module_init(aic3x_modinit);
-
-static void __exit aic3x_exit(void)
-{
- i2c_del_driver(&aic3x_i2c_driver);
-}
-module_exit(aic3x_exit);
+module_i2c_driver(aic3x_i2c_driver);
MODULE_DESCRIPTION("ASoC TLV320AIC3X codec driver");
MODULE_AUTHOR("Vladimir Barinov");
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 0dd4107..d2e16c5 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -1621,24 +1621,7 @@
.id_table = tlv320dac33_i2c_id,
};
-static int __init dac33_module_init(void)
-{
- int r;
- r = i2c_add_driver(&tlv320dac33_i2c_driver);
- if (r < 0) {
- printk(KERN_ERR "DAC33: driver registration failed\n");
- return r;
- }
- return 0;
-}
-module_init(dac33_module_init);
-
-static void __exit dac33_module_exit(void)
-{
- i2c_del_driver(&tlv320dac33_i2c_driver);
-}
-module_exit(dac33_module_exit);
-
+module_i2c_driver(tlv320dac33_i2c_driver);
MODULE_DESCRIPTION("ASoC TLV320DAC33 codec driver");
MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@ti.com>");
diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c
index 6fe4aa3..565ff39 100644
--- a/sound/soc/codecs/tpa6130a2.c
+++ b/sound/soc/codecs/tpa6130a2.c
@@ -487,19 +487,8 @@
.id_table = tpa6130a2_id,
};
-static int __init tpa6130a2_init(void)
-{
- return i2c_add_driver(&tpa6130a2_i2c_driver);
-}
-
-static void __exit tpa6130a2_exit(void)
-{
- i2c_del_driver(&tpa6130a2_i2c_driver);
-}
+module_i2c_driver(tpa6130a2_i2c_driver);
MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@ti.com>");
MODULE_DESCRIPTION("TPA6130A2 Headphone amplifier driver");
MODULE_LICENSE("GPL");
-
-module_init(tpa6130a2_init);
-module_exit(tpa6130a2_exit);
diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c
new file mode 100644
index 0000000..5f99148
--- /dev/null
+++ b/sound/soc/codecs/wm0010.c
@@ -0,0 +1,942 @@
+/*
+ * wm0010.c -- WM0010 DSP Driver
+ *
+ * Copyright 2012 Wolfson Microelectronics PLC.
+ *
+ * Authors: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ * Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
+ * Scott Ling <sl@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/irqreturn.h>
+#include <linux/init.h>
+#include <linux/spi/spi.h>
+#include <linux/firmware.h>
+#include <linux/delay.h>
+#include <linux/fs.h>
+#include <linux/miscdevice.h>
+#include <linux/gpio.h>
+#include <linux/regulator/consumer.h>
+#include <linux/mutex.h>
+#include <linux/workqueue.h>
+
+#include <sound/soc.h>
+#include <sound/wm0010.h>
+
+#define DEVICE_ID_WM0010 10
+
+enum dfw_cmd {
+ DFW_CMD_FUSE = 0x01,
+ DFW_CMD_CODE_HDR,
+ DFW_CMD_CODE_DATA,
+ DFW_CMD_PLL,
+ DFW_CMD_INFO = 0xff
+};
+
+struct dfw_binrec {
+ u8 command;
+ u32 length:24;
+ u32 address;
+ uint8_t data[0];
+} __packed;
+
+struct dfw_pllrec {
+ u8 command;
+ u32 length:24;
+ u32 address;
+ u32 clkctrl1;
+ u32 clkctrl2;
+ u32 clkctrl3;
+ u32 ldetctrl;
+ u32 uart_div;
+ u32 spi_div;
+} __packed;
+
+static struct pll_clock_map {
+ int max_sysclk;
+ int max_pll_spi_speed;
+ u32 pll_clkctrl1;
+} pll_clock_map[] = { /* Dividers */
+ { 22000000, 26000000, 0x00201f11 }, /* 2,32,2 */
+ { 18000000, 26000000, 0x00203f21 }, /* 2,64,4 */
+ { 14000000, 26000000, 0x00202620 }, /* 1,39,4 */
+ { 10000000, 22000000, 0x00203120 }, /* 1,50,4 */
+ { 6500000, 22000000, 0x00204520 }, /* 1,70,4 */
+ { 5500000, 22000000, 0x00103f10 }, /* 1,64,2 */
+};
+
+enum wm0010_state {
+ WM0010_POWER_OFF,
+ WM0010_OUT_OF_RESET,
+ WM0010_BOOTROM,
+ WM0010_STAGE2,
+ WM0010_FIRMWARE,
+};
+
+struct wm0010_priv {
+ struct snd_soc_codec *codec;
+
+ struct mutex lock;
+ struct device *dev;
+
+ struct wm0010_pdata pdata;
+
+ int gpio_reset;
+ int gpio_reset_value;
+
+ struct regulator_bulk_data core_supplies[2];
+ struct regulator *dbvdd;
+
+ int sysclk;
+
+ enum wm0010_state state;
+ bool boot_failed;
+ int boot_done;
+ bool ready;
+ bool pll_running;
+ int max_spi_freq;
+ int board_max_spi_speed;
+ u32 pll_clkctrl1;
+
+ spinlock_t irq_lock;
+ int irq;
+
+ struct completion boot_completion;
+};
+
+struct wm0010_spi_msg {
+ struct spi_message m;
+ struct spi_transfer t;
+ u8 *tx_buf;
+ u8 *rx_buf;
+ size_t len;
+};
+
+static const struct snd_soc_dapm_widget wm0010_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("CLKIN", SND_SOC_NOPM, 0, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_route wm0010_dapm_routes[] = {
+ { "SDI2 Capture", NULL, "SDI1 Playback" },
+ { "SDI1 Capture", NULL, "SDI2 Playback" },
+
+ { "SDI1 Capture", NULL, "CLKIN" },
+ { "SDI2 Capture", NULL, "CLKIN" },
+ { "SDI1 Playback", NULL, "CLKIN" },
+ { "SDI2 Playback", NULL, "CLKIN" },
+};
+
+static const char *wm0010_state_to_str(enum wm0010_state state)
+{
+ const char *state_to_str[] = {
+ "Power off",
+ "Out of reset",
+ "Boot ROM",
+ "Stage2",
+ "Firmware"
+ };
+
+ if (state < 0 || state >= ARRAY_SIZE(state_to_str))
+ return "null";
+ return state_to_str[state];
+}
+
+/* Called with wm0010->lock held */
+static void wm0010_halt(struct snd_soc_codec *codec)
+{
+ struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec);
+ unsigned long flags;
+ enum wm0010_state state;
+
+ /* Fetch the wm0010 state */
+ spin_lock_irqsave(&wm0010->irq_lock, flags);
+ state = wm0010->state;
+ spin_unlock_irqrestore(&wm0010->irq_lock, flags);
+
+ switch (state) {
+ case WM0010_POWER_OFF:
+ /* If there's nothing to do, bail out */
+ return;
+ case WM0010_OUT_OF_RESET:
+ case WM0010_BOOTROM:
+ case WM0010_STAGE2:
+ case WM0010_FIRMWARE:
+ /* Remember to put chip back into reset */
+ gpio_set_value(wm0010->gpio_reset, wm0010->gpio_reset_value);
+ /* Disable the regulators */
+ regulator_disable(wm0010->dbvdd);
+ regulator_bulk_disable(ARRAY_SIZE(wm0010->core_supplies),
+ wm0010->core_supplies);
+ break;
+ }
+
+ spin_lock_irqsave(&wm0010->irq_lock, flags);
+ wm0010->state = WM0010_POWER_OFF;
+ spin_unlock_irqrestore(&wm0010->irq_lock, flags);
+}
+
+struct wm0010_boot_xfer {
+ struct list_head list;
+ struct snd_soc_codec *codec;
+ struct completion *done;
+ struct spi_message m;
+ struct spi_transfer t;
+};
+
+/* Called with wm0010->lock held */
+static void wm0010_mark_boot_failure(struct wm0010_priv *wm0010)
+{
+ enum wm0010_state state;
+ unsigned long flags;
+
+ spin_lock_irqsave(&wm0010->irq_lock, flags);
+ state = wm0010->state;
+ spin_unlock_irqrestore(&wm0010->irq_lock, flags);
+
+ dev_err(wm0010->dev, "Failed to transition from `%s' state to `%s' state\n",
+ wm0010_state_to_str(state), wm0010_state_to_str(state + 1));
+
+ wm0010->boot_failed = true;
+}
+
+static void wm0010_boot_xfer_complete(void *data)
+{
+ struct wm0010_boot_xfer *xfer = data;
+ struct snd_soc_codec *codec = xfer->codec;
+ struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec);
+ u32 *out32 = xfer->t.rx_buf;
+ int i;
+
+ if (xfer->m.status != 0) {
+ dev_err(codec->dev, "SPI transfer failed: %d\n",
+ xfer->m.status);
+ wm0010_mark_boot_failure(wm0010);
+ if (xfer->done)
+ complete(xfer->done);
+ return;
+ }
+
+ for (i = 0; i < xfer->t.len / 4; i++) {
+ dev_dbg(codec->dev, "%d: %04x\n", i, out32[i]);
+
+ switch (be32_to_cpu(out32[i])) {
+ case 0xe0e0e0e0:
+ dev_err(codec->dev,
+ "%d: ROM error reported in stage 2\n", i);
+ wm0010_mark_boot_failure(wm0010);
+ break;
+
+ case 0x55555555:
+ if (wm0010->boot_done == 0)
+ break;
+ dev_err(codec->dev,
+ "%d: ROM bootloader running in stage 2\n", i);
+ wm0010_mark_boot_failure(wm0010);
+ break;
+
+ case 0x0fed0000:
+ dev_dbg(codec->dev, "Stage2 loader running\n");
+ break;
+
+ case 0x0fed0007:
+ dev_dbg(codec->dev, "CODE_HDR packet received\n");
+ break;
+
+ case 0x0fed0008:
+ dev_dbg(codec->dev, "CODE_DATA packet received\n");
+ break;
+
+ case 0x0fed0009:
+ dev_dbg(codec->dev, "Download complete\n");
+ break;
+
+ case 0x0fed000c:
+ dev_dbg(codec->dev, "Application start\n");
+ break;
+
+ case 0x0fed000e:
+ dev_dbg(codec->dev, "PLL packet received\n");
+ wm0010->pll_running = true;
+ break;
+
+ case 0x0fed0025:
+ dev_err(codec->dev, "Device reports image too long\n");
+ wm0010_mark_boot_failure(wm0010);
+ break;
+
+ case 0x0fed002c:
+ dev_err(codec->dev, "Device reports bad SPI packet\n");
+ wm0010_mark_boot_failure(wm0010);
+ break;
+
+ case 0x0fed0031:
+ dev_err(codec->dev, "Device reports SPI read overflow\n");
+ wm0010_mark_boot_failure(wm0010);
+ break;
+
+ case 0x0fed0032:
+ dev_err(codec->dev, "Device reports SPI underclock\n");
+ wm0010_mark_boot_failure(wm0010);
+ break;
+
+ case 0x0fed0033:
+ dev_err(codec->dev, "Device reports bad header packet\n");
+ wm0010_mark_boot_failure(wm0010);
+ break;
+
+ case 0x0fed0034:
+ dev_err(codec->dev, "Device reports invalid packet type\n");
+ wm0010_mark_boot_failure(wm0010);
+ break;
+
+ case 0x0fed0035:
+ dev_err(codec->dev, "Device reports data before header error\n");
+ wm0010_mark_boot_failure(wm0010);
+ break;
+
+ case 0x0fed0038:
+ dev_err(codec->dev, "Device reports invalid PLL packet\n");
+ break;
+
+ case 0x0fed003a:
+ dev_err(codec->dev, "Device reports packet alignment error\n");
+ wm0010_mark_boot_failure(wm0010);
+ break;
+
+ default:
+ dev_err(codec->dev, "Unrecognised return 0x%x\n",
+ be32_to_cpu(out32[i]));
+ wm0010_mark_boot_failure(wm0010);
+ break;
+ }
+
+ if (wm0010->boot_failed)
+ break;
+ }
+
+ wm0010->boot_done++;
+ if (xfer->done)
+ complete(xfer->done);
+}
+
+static void byte_swap_64(u64 *data_in, u64 *data_out, u32 len)
+{
+ int i;
+
+ for (i = 0; i < len / 8; i++)
+ data_out[i] = cpu_to_be64(le64_to_cpu(data_in[i]));
+}
+
+static int wm0010_boot(struct snd_soc_codec *codec)
+{
+ struct spi_device *spi = to_spi_device(codec->dev);
+ struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec);
+ unsigned long flags;
+ struct list_head xfer_list;
+ struct wm0010_boot_xfer *xfer;
+ int ret;
+ struct completion done;
+ const struct firmware *fw;
+ const struct dfw_binrec *rec;
+ struct spi_message m;
+ struct spi_transfer t;
+ struct dfw_pllrec pll_rec;
+ u32 *img, *p;
+ u64 *img_swap;
+ u8 *out;
+ u32 len, offset;
+ int i;
+
+ spin_lock_irqsave(&wm0010->irq_lock, flags);
+ if (wm0010->state != WM0010_POWER_OFF)
+ dev_warn(wm0010->dev, "DSP already powered up!\n");
+ spin_unlock_irqrestore(&wm0010->irq_lock, flags);
+
+ if (wm0010->sysclk > 26000000) {
+ dev_err(codec->dev, "Max DSP clock frequency is 26MHz\n");
+ ret = -ECANCELED;
+ goto err;
+ }
+
+ INIT_LIST_HEAD(&xfer_list);
+
+ mutex_lock(&wm0010->lock);
+ wm0010->pll_running = false;
+
+ dev_dbg(codec->dev, "max_spi_freq: %d\n", wm0010->max_spi_freq);
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(wm0010->core_supplies),
+ wm0010->core_supplies);
+ if (ret != 0) {
+ dev_err(&spi->dev, "Failed to enable core supplies: %d\n",
+ ret);
+ mutex_unlock(&wm0010->lock);
+ goto err;
+ }
+
+ ret = regulator_enable(wm0010->dbvdd);
+ if (ret != 0) {
+ dev_err(&spi->dev, "Failed to enable DBVDD: %d\n", ret);
+ goto err_core;
+ }
+
+ /* Release reset */
+ gpio_set_value(wm0010->gpio_reset, !wm0010->gpio_reset_value);
+ spin_lock_irqsave(&wm0010->irq_lock, flags);
+ wm0010->state = WM0010_OUT_OF_RESET;
+ spin_unlock_irqrestore(&wm0010->irq_lock, flags);
+
+ /* First the bootloader */
+ ret = request_firmware(&fw, "wm0010_stage2.bin", codec->dev);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to request stage2 loader: %d\n",
+ ret);
+ goto abort;
+ }
+
+ if (!wait_for_completion_timeout(&wm0010->boot_completion,
+ msecs_to_jiffies(10)))
+ dev_err(codec->dev, "Failed to get interrupt from DSP\n");
+
+ spin_lock_irqsave(&wm0010->irq_lock, flags);
+ wm0010->state = WM0010_BOOTROM;
+ spin_unlock_irqrestore(&wm0010->irq_lock, flags);
+
+ dev_dbg(codec->dev, "Downloading %d byte stage 2 loader\n", fw->size);
+
+ /* Copy to local buffer first as vmalloc causes problems for dma */
+ img = kzalloc(fw->size, GFP_KERNEL);
+ if (!img) {
+ dev_err(codec->dev, "Failed to allocate image buffer\n");
+ goto abort;
+ }
+
+ out = kzalloc(fw->size, GFP_KERNEL);
+ if (!out) {
+ dev_err(codec->dev, "Failed to allocate output buffer\n");
+ goto abort;
+ }
+
+ memcpy(img, &fw->data[0], fw->size);
+
+ spi_message_init(&m);
+ memset(&t, 0, sizeof(t));
+ t.rx_buf = out;
+ t.tx_buf = img;
+ t.len = fw->size;
+ t.bits_per_word = 8;
+ t.speed_hz = wm0010->sysclk / 10;
+ spi_message_add_tail(&t, &m);
+
+ dev_dbg(codec->dev, "Starting initial download at %dHz\n",
+ t.speed_hz);
+
+ ret = spi_sync(spi, &m);
+ if (ret != 0) {
+ dev_err(codec->dev, "Initial download failed: %d\n", ret);
+ goto abort;
+ }
+
+ /* Look for errors from the boot ROM */
+ for (i = 0; i < fw->size; i++) {
+ if (out[i] != 0x55) {
+ ret = -EBUSY;
+ dev_err(codec->dev, "Boot ROM error: %x in %d\n",
+ out[i], i);
+ wm0010_mark_boot_failure(wm0010);
+ goto abort;
+ }
+ }
+
+ release_firmware(fw);
+ kfree(img);
+ kfree(out);
+
+ if (!wait_for_completion_timeout(&wm0010->boot_completion,
+ msecs_to_jiffies(10)))
+ dev_err(codec->dev, "Failed to get interrupt from DSP loader.\n");
+
+ spin_lock_irqsave(&wm0010->irq_lock, flags);
+ wm0010->state = WM0010_STAGE2;
+ spin_unlock_irqrestore(&wm0010->irq_lock, flags);
+
+ /* Only initialise PLL if max_spi_freq initialised */
+ if (wm0010->max_spi_freq) {
+
+ /* Initialise a PLL record */
+ memset(&pll_rec, 0, sizeof(pll_rec));
+ pll_rec.command = DFW_CMD_PLL;
+ pll_rec.length = (sizeof(pll_rec) - 8);
+
+ /* On wm0010 only the CLKCTRL1 value is used */
+ pll_rec.clkctrl1 = wm0010->pll_clkctrl1;
+
+ len = pll_rec.length + 8;
+ out = kzalloc(len, GFP_KERNEL);
+ if (!out) {
+ dev_err(codec->dev,
+ "Failed to allocate RX buffer\n");
+ goto abort;
+ }
+
+ img_swap = kzalloc(len, GFP_KERNEL);
+ if (!img_swap) {
+ dev_err(codec->dev,
+ "Failed to allocate image buffer\n");
+ goto abort;
+ }
+
+ /* We need to re-order for 0010 */
+ byte_swap_64((u64 *)&pll_rec, img_swap, len);
+
+ spi_message_init(&m);
+ memset(&t, 0, sizeof(t));
+ t.rx_buf = out;
+ t.tx_buf = img_swap;
+ t.len = len;
+ t.bits_per_word = 8;
+ t.speed_hz = wm0010->sysclk / 6;
+ spi_message_add_tail(&t, &m);
+
+ ret = spi_sync(spi, &m);
+ if (ret != 0) {
+ dev_err(codec->dev, "First PLL write failed: %d\n", ret);
+ goto abort;
+ }
+
+ /* Use a second send of the message to get the return status */
+ ret = spi_sync(spi, &m);
+ if (ret != 0) {
+ dev_err(codec->dev, "Second PLL write failed: %d\n", ret);
+ goto abort;
+ }
+
+ p = (u32 *)out;
+
+ /* Look for PLL active code from the DSP */
+ for (i = 0; i < len / 4; i++) {
+ if (*p == 0x0e00ed0f) {
+ dev_dbg(codec->dev, "PLL packet received\n");
+ wm0010->pll_running = true;
+ break;
+ }
+ p++;
+ }
+
+ kfree(img_swap);
+ kfree(out);
+ } else
+ dev_dbg(codec->dev, "Not enabling DSP PLL.");
+
+ ret = request_firmware(&fw, "wm0010.dfw", codec->dev);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to request application: %d\n",
+ ret);
+ goto abort;
+ }
+
+ rec = (const struct dfw_binrec *)fw->data;
+ offset = 0;
+ wm0010->boot_done = 0;
+ wm0010->boot_failed = false;
+ BUG_ON(!list_empty(&xfer_list));
+ init_completion(&done);
+
+ /* First record should be INFO */
+ if (rec->command != DFW_CMD_INFO) {
+ dev_err(codec->dev, "First record not INFO\r\n");
+ goto abort;
+ }
+
+ /* Check it's a 0010 file */
+ if (rec->data[0] != DEVICE_ID_WM0010) {
+ dev_err(codec->dev, "Not a WM0010 firmware file.\r\n");
+ goto abort;
+ }
+
+ /* Skip the info record as we don't need to send it */
+ offset += ((rec->length) + 8);
+ rec = (void *)&rec->data[rec->length];
+
+ while (offset < fw->size) {
+ dev_dbg(codec->dev,
+ "Packet: command %d, data length = 0x%x\r\n",
+ rec->command, rec->length);
+ len = rec->length + 8;
+
+ out = kzalloc(len, GFP_KERNEL);
+ if (!out) {
+ dev_err(codec->dev,
+ "Failed to allocate RX buffer\n");
+ goto abort;
+ }
+
+ img_swap = kzalloc(len, GFP_KERNEL);
+ if (!img_swap) {
+ dev_err(codec->dev,
+ "Failed to allocate image buffer\n");
+ goto abort;
+ }
+
+ /* We need to re-order for 0010 */
+ byte_swap_64((u64 *)&rec->command, img_swap, len);
+
+ xfer = kzalloc(sizeof(*xfer), GFP_KERNEL);
+ if (!xfer) {
+ dev_err(codec->dev, "Failed to allocate xfer\n");
+ goto abort;
+ }
+
+ xfer->codec = codec;
+ list_add_tail(&xfer->list, &xfer_list);
+
+ spi_message_init(&xfer->m);
+ xfer->m.complete = wm0010_boot_xfer_complete;
+ xfer->m.context = xfer;
+ xfer->t.tx_buf = img_swap;
+ xfer->t.rx_buf = out;
+ xfer->t.len = len;
+ xfer->t.bits_per_word = 8;
+
+ if (!wm0010->pll_running) {
+ xfer->t.speed_hz = wm0010->sysclk / 6;
+ } else {
+ xfer->t.speed_hz = wm0010->max_spi_freq;
+
+ if (wm0010->board_max_spi_speed &&
+ (wm0010->board_max_spi_speed < wm0010->max_spi_freq))
+ xfer->t.speed_hz = wm0010->board_max_spi_speed;
+ }
+
+ /* Store max usable spi frequency for later use */
+ wm0010->max_spi_freq = xfer->t.speed_hz;
+
+ spi_message_add_tail(&xfer->t, &xfer->m);
+
+ offset += ((rec->length) + 8);
+ rec = (void *)&rec->data[rec->length];
+
+ if (offset >= fw->size) {
+ dev_dbg(codec->dev, "All transfers scheduled\n");
+ xfer->done = &done;
+ }
+
+ ret = spi_async(spi, &xfer->m);
+ if (ret != 0) {
+ dev_err(codec->dev, "Write failed: %d\n", ret);
+ goto abort;
+ }
+
+ if (wm0010->boot_failed)
+ goto abort;
+ }
+
+ wait_for_completion(&done);
+
+ spin_lock_irqsave(&wm0010->irq_lock, flags);
+ wm0010->state = WM0010_FIRMWARE;
+ spin_unlock_irqrestore(&wm0010->irq_lock, flags);
+
+ mutex_unlock(&wm0010->lock);
+
+ release_firmware(fw);
+
+ while (!list_empty(&xfer_list)) {
+ xfer = list_first_entry(&xfer_list, struct wm0010_boot_xfer,
+ list);
+ kfree(xfer->t.rx_buf);
+ kfree(xfer->t.tx_buf);
+ list_del(&xfer->list);
+ kfree(xfer);
+ }
+
+ return 0;
+
+abort:
+ /* Put the chip back into reset */
+ wm0010_halt(codec);
+ mutex_unlock(&wm0010->lock);
+ return ret;
+err_core:
+ regulator_bulk_disable(ARRAY_SIZE(wm0010->core_supplies),
+ wm0010->core_supplies);
+err:
+ return ret;
+}
+
+static int wm0010_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec);
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE)
+ wm0010_boot(codec);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE) {
+ mutex_lock(&wm0010->lock);
+ wm0010_halt(codec);
+ mutex_unlock(&wm0010->lock);
+ }
+ break;
+ case SND_SOC_BIAS_OFF:
+ break;
+ }
+
+ codec->dapm.bias_level = level;
+
+ return 0;
+}
+
+static int wm0010_set_sysclk(struct snd_soc_codec *codec, int source,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec);
+ unsigned int i;
+
+ wm0010->sysclk = freq;
+
+ if (freq < pll_clock_map[ARRAY_SIZE(pll_clock_map)-1].max_sysclk) {
+ wm0010->max_spi_freq = 0;
+ } else {
+ for (i = 0; i < ARRAY_SIZE(pll_clock_map); i++)
+ if (freq >= pll_clock_map[i].max_sysclk)
+ break;
+
+ wm0010->max_spi_freq = pll_clock_map[i].max_pll_spi_speed;
+ wm0010->pll_clkctrl1 = pll_clock_map[i].pll_clkctrl1;
+ }
+
+ return 0;
+}
+
+static int wm0010_probe(struct snd_soc_codec *codec);
+
+static struct snd_soc_codec_driver soc_codec_dev_wm0010 = {
+ .probe = wm0010_probe,
+ .set_bias_level = wm0010_set_bias_level,
+ .set_sysclk = wm0010_set_sysclk,
+ .idle_bias_off = true,
+
+ .dapm_widgets = wm0010_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(wm0010_dapm_widgets),
+ .dapm_routes = wm0010_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(wm0010_dapm_routes),
+};
+
+#define WM0010_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+#define WM0010_FORMATS (SNDRV_PCM_FMTBIT_S8 | 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 wm0010_dai[] = {
+ {
+ .name = "wm0010-sdi1",
+ .playback = {
+ .stream_name = "SDI1 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM0010_RATES,
+ .formats = WM0010_FORMATS,
+ },
+ .capture = {
+ .stream_name = "SDI1 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM0010_RATES,
+ .formats = WM0010_FORMATS,
+ },
+ },
+ {
+ .name = "wm0010-sdi2",
+ .playback = {
+ .stream_name = "SDI2 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM0010_RATES,
+ .formats = WM0010_FORMATS,
+ },
+ .capture = {
+ .stream_name = "SDI2 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM0010_RATES,
+ .formats = WM0010_FORMATS,
+ },
+ },
+};
+
+static irqreturn_t wm0010_irq(int irq, void *data)
+{
+ struct wm0010_priv *wm0010 = data;
+
+ switch (wm0010->state) {
+ case WM0010_POWER_OFF:
+ case WM0010_OUT_OF_RESET:
+ case WM0010_BOOTROM:
+ case WM0010_STAGE2:
+ spin_lock(&wm0010->irq_lock);
+ complete(&wm0010->boot_completion);
+ spin_unlock(&wm0010->irq_lock);
+ return IRQ_HANDLED;
+ default:
+ return IRQ_NONE;
+ }
+
+ return IRQ_NONE;
+}
+
+static int wm0010_probe(struct snd_soc_codec *codec)
+{
+ struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec);
+
+ wm0010->codec = codec;
+
+ return 0;
+}
+
+static int __devinit wm0010_spi_probe(struct spi_device *spi)
+{
+ unsigned long flags;
+ unsigned long gpio_flags;
+ int ret;
+ int trigger;
+ int irq;
+ struct wm0010_priv *wm0010;
+
+ wm0010 = devm_kzalloc(&spi->dev, sizeof(*wm0010),
+ GFP_KERNEL);
+ if (!wm0010)
+ return -ENOMEM;
+
+ mutex_init(&wm0010->lock);
+ spin_lock_init(&wm0010->irq_lock);
+
+ spi_set_drvdata(spi, wm0010);
+ wm0010->dev = &spi->dev;
+
+ if (dev_get_platdata(&spi->dev))
+ memcpy(&wm0010->pdata, dev_get_platdata(&spi->dev),
+ sizeof(wm0010->pdata));
+
+ init_completion(&wm0010->boot_completion);
+
+ wm0010->core_supplies[0].supply = "AVDD";
+ wm0010->core_supplies[1].supply = "DCVDD";
+ ret = devm_regulator_bulk_get(wm0010->dev, ARRAY_SIZE(wm0010->core_supplies),
+ wm0010->core_supplies);
+ if (ret != 0) {
+ dev_err(wm0010->dev, "Failed to obtain core supplies: %d\n",
+ ret);
+ return ret;
+ }
+
+ wm0010->dbvdd = devm_regulator_get(wm0010->dev, "DBVDD");
+ if (IS_ERR(wm0010->dbvdd)) {
+ ret = PTR_ERR(wm0010->dbvdd);
+ dev_err(wm0010->dev, "Failed to obtain DBVDD: %d\n", ret);
+ return ret;
+ }
+
+ if (wm0010->pdata.gpio_reset) {
+ wm0010->gpio_reset = wm0010->pdata.gpio_reset;
+
+ if (wm0010->pdata.reset_active_high)
+ wm0010->gpio_reset_value = 1;
+ else
+ wm0010->gpio_reset_value = 0;
+
+ if (wm0010->gpio_reset_value)
+ gpio_flags = GPIOF_OUT_INIT_HIGH;
+ else
+ gpio_flags = GPIOF_OUT_INIT_LOW;
+
+ ret = devm_gpio_request_one(wm0010->dev, wm0010->gpio_reset,
+ gpio_flags, "wm0010 reset");
+ if (ret < 0) {
+ dev_err(wm0010->dev,
+ "Failed to request GPIO for DSP reset: %d\n",
+ ret);
+ return ret;
+ }
+ } else {
+ dev_err(wm0010->dev, "No reset GPIO configured\n");
+ return -EINVAL;
+ }
+
+ irq = spi->irq;
+ if (wm0010->pdata.irq_flags)
+ trigger = wm0010->pdata.irq_flags;
+ else
+ trigger = IRQF_TRIGGER_FALLING;
+ trigger |= IRQF_ONESHOT;
+
+ ret = request_threaded_irq(irq, NULL, wm0010_irq, trigger,
+ "wm0010", wm0010);
+ if (ret) {
+ dev_err(wm0010->dev, "Failed to request IRQ %d: %d\n",
+ irq, ret);
+ return ret;
+ }
+ wm0010->irq = irq;
+
+ if (spi->max_speed_hz)
+ wm0010->board_max_spi_speed = spi->max_speed_hz;
+ else
+ wm0010->board_max_spi_speed = 0;
+
+ spin_lock_irqsave(&wm0010->irq_lock, flags);
+ wm0010->state = WM0010_POWER_OFF;
+ spin_unlock_irqrestore(&wm0010->irq_lock, flags);
+
+ ret = snd_soc_register_codec(&spi->dev,
+ &soc_codec_dev_wm0010, wm0010_dai,
+ ARRAY_SIZE(wm0010_dai));
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static int __devexit wm0010_spi_remove(struct spi_device *spi)
+{
+ struct wm0010_priv *wm0010 = spi_get_drvdata(spi);
+
+ snd_soc_unregister_codec(&spi->dev);
+
+ if (wm0010->gpio_reset) {
+ /* Remember to put chip back into reset */
+ gpio_set_value(wm0010->gpio_reset, wm0010->gpio_reset_value);
+ }
+
+ if (wm0010->irq)
+ free_irq(wm0010->irq, wm0010);
+
+ return 0;
+}
+
+static struct spi_driver wm0010_spi_driver = {
+ .driver = {
+ .name = "wm0010",
+ .bus = &spi_bus_type,
+ .owner = THIS_MODULE,
+ },
+ .probe = wm0010_spi_probe,
+ .remove = __devexit_p(wm0010_spi_remove),
+};
+
+module_spi_driver(wm0010_spi_driver);
+
+MODULE_DESCRIPTION("ASoC WM0010 driver");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 3fd5b29..89cd6fc 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -858,17 +858,7 @@
.id_table = wm2000_i2c_id,
};
-static int __init wm2000_init(void)
-{
- return i2c_add_driver(&wm2000_i2c_driver);
-}
-module_init(wm2000_init);
-
-static void __exit wm2000_exit(void)
-{
- i2c_del_driver(&wm2000_i2c_driver);
-}
-module_exit(wm2000_exit);
+module_i2c_driver(wm2000_i2c_driver);
MODULE_DESCRIPTION("ASoC WM2000 driver");
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfonmicro.com>");
diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c
index 32682c1..71debd0 100644
--- a/sound/soc/codecs/wm2200.c
+++ b/sound/soc/codecs/wm2200.c
@@ -2270,17 +2270,7 @@
.id_table = wm2200_i2c_id,
};
-static int __init wm2200_modinit(void)
-{
- return i2c_add_driver(&wm2200_i2c_driver);
-}
-module_init(wm2200_modinit);
-
-static void __exit wm2200_exit(void)
-{
- i2c_del_driver(&wm2200_i2c_driver);
-}
-module_exit(wm2200_exit);
+module_i2c_driver(wm2200_i2c_driver);
MODULE_DESCRIPTION("ASoC WM2200 driver");
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index e33d327..e2fb07e 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -274,11 +274,36 @@
ARIZONA_MIXER_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE);
ARIZONA_MIXER_ENUMS(ASRC2R, ARIZONA_ASRC2RMIX_INPUT_1_SOURCE);
+
+static const char *wm5102_aec_loopback_texts[] = {
+ "HPOUT1L", "HPOUT1R", "HPOUT2L", "HPOUT2R", "EPOUT",
+ "SPKOUTL", "SPKOUTR", "SPKDAT1L", "SPKDAT1R",
+};
+
+static const unsigned int wm5102_aec_loopback_values[] = {
+ 0, 1, 2, 3, 4, 6, 7, 8, 9,
+};
+
+static const struct soc_enum wm5102_aec_loopback =
+ SOC_VALUE_ENUM_SINGLE(ARIZONA_DAC_AEC_CONTROL_1,
+ ARIZONA_AEC_LOOPBACK_SRC_SHIFT,
+ ARIZONA_AEC_LOOPBACK_SRC_MASK,
+ ARRAY_SIZE(wm5102_aec_loopback_texts),
+ wm5102_aec_loopback_texts,
+ wm5102_aec_loopback_values);
+
+static const struct snd_kcontrol_new wm5102_aec_loopback_mux =
+ SOC_DAPM_VALUE_ENUM("AEC Loopback", wm5102_aec_loopback);
+
static const struct snd_soc_dapm_widget wm5102_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT,
0, NULL, 0),
SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1,
ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK,
+ ARIZONA_OPCLK_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", ARIZONA_OUTPUT_ASYNC_CLOCK,
+ ARIZONA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0),
SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0),
SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0),
@@ -421,6 +446,9 @@
SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0,
ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1,
+ ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5102_aec_loopback_mux),
+
SND_SOC_DAPM_PGA_E("OUT1L", ARIZONA_OUTPUT_ENABLES_1,
ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
@@ -516,6 +544,7 @@
{ name, "Noise Generator", "Noise Generator" }, \
{ name, "Tone Generator 1", "Tone Generator 1" }, \
{ name, "Tone Generator 2", "Tone Generator 2" }, \
+ { name, "AEC", "AEC Loopback" }, \
{ name, "IN1L", "IN1L PGA" }, \
{ name, "IN1R", "IN1R PGA" }, \
{ name, "IN2L", "IN2L PGA" }, \
@@ -681,21 +710,30 @@
ARIZONA_MIXER_ROUTES("ASRC2L", "ASRC2L"),
ARIZONA_MIXER_ROUTES("ASRC2R", "ASRC2R"),
+ { "AEC Loopback", "HPOUT1L", "OUT1L" },
+ { "AEC Loopback", "HPOUT1R", "OUT1R" },
{ "HPOUT1L", NULL, "OUT1L" },
{ "HPOUT1R", NULL, "OUT1R" },
+ { "AEC Loopback", "HPOUT2L", "OUT2L" },
+ { "AEC Loopback", "HPOUT2R", "OUT2R" },
{ "HPOUT2L", NULL, "OUT2L" },
{ "HPOUT2R", NULL, "OUT2R" },
+ { "AEC Loopback", "EPOUT", "OUT3L" },
{ "EPOUTN", NULL, "OUT3L" },
{ "EPOUTP", NULL, "OUT3L" },
+ { "AEC Loopback", "SPKOUTL", "OUT4L" },
{ "SPKOUTLN", NULL, "OUT4L" },
{ "SPKOUTLP", NULL, "OUT4L" },
+ { "AEC Loopback", "SPKOUTR", "OUT4R" },
{ "SPKOUTRN", NULL, "OUT4R" },
{ "SPKOUTRP", NULL, "OUT4R" },
+ { "AEC Loopback", "SPKDAT1L", "OUT5L" },
+ { "AEC Loopback", "SPKDAT1R", "OUT5R" },
{ "SPKDAT1L", NULL, "OUT5L" },
{ "SPKDAT1R", NULL, "OUT5R" },
};
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 01ebbcc..57c7d9c 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -305,6 +305,10 @@
0, NULL, 0),
SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1,
ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK,
+ ARIZONA_OPCLK_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", ARIZONA_OUTPUT_ASYNC_CLOCK,
+ ARIZONA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0),
SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0),
SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0),
diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c
index a5127b4..c7c0034 100644
--- a/sound/soc/codecs/wm8770.c
+++ b/sound/soc/codecs/wm8770.c
@@ -724,24 +724,7 @@
.remove = __devexit_p(wm8770_spi_remove)
};
-static int __init wm8770_modinit(void)
-{
- int ret = 0;
-
- ret = spi_register_driver(&wm8770_spi_driver);
- if (ret) {
- printk(KERN_ERR "Failed to register wm8770 SPI driver: %d\n",
- ret);
- }
- return ret;
-}
-module_init(wm8770_modinit);
-
-static void __exit wm8770_exit(void)
-{
- spi_unregister_driver(&wm8770_spi_driver);
-}
-module_exit(wm8770_exit);
+module_spi_driver(wm8770_spi_driver);
MODULE_DESCRIPTION("ASoC WM8770 driver");
MODULE_AUTHOR("Dimitris Papastamos <dp@opensource.wolfsonmicro.com>");
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 73f1c8d..839414f 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -2241,23 +2241,7 @@
.id_table = wm8903_i2c_id,
};
-static int __init wm8903_modinit(void)
-{
- int ret = 0;
- ret = i2c_add_driver(&wm8903_i2c_driver);
- if (ret != 0) {
- printk(KERN_ERR "Failed to register wm8903 I2C driver: %d\n",
- ret);
- }
- return ret;
-}
-module_init(wm8903_modinit);
-
-static void __exit wm8903_exit(void)
-{
- i2c_del_driver(&wm8903_i2c_driver);
-}
-module_exit(wm8903_exit);
+module_i2c_driver(wm8903_i2c_driver);
MODULE_DESCRIPTION("ASoC WM8903 driver");
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.cm>");
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index 481a3d9..b20aa4e 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -785,23 +785,7 @@
.id_table = wm8940_i2c_id,
};
-static int __init wm8940_modinit(void)
-{
- int ret = 0;
- ret = i2c_add_driver(&wm8940_i2c_driver);
- if (ret != 0) {
- printk(KERN_ERR "Failed to register wm8940 I2C driver: %d\n",
- ret);
- }
- return ret;
-}
-module_init(wm8940_modinit);
-
-static void __exit wm8940_exit(void)
-{
- i2c_del_driver(&wm8940_i2c_driver);
-}
-module_exit(wm8940_exit);
+module_i2c_driver(wm8940_i2c_driver);
MODULE_DESCRIPTION("ASoC WM8940 driver");
MODULE_AUTHOR("Jonathan Cameron");
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index 61fe974..2f1c075 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -1071,23 +1071,7 @@
.id_table = wm8955_i2c_id,
};
-static int __init wm8955_modinit(void)
-{
- int ret = 0;
- ret = i2c_add_driver(&wm8955_i2c_driver);
- if (ret != 0) {
- printk(KERN_ERR "Failed to register WM8955 I2C driver: %d\n",
- ret);
- }
- return ret;
-}
-module_init(wm8955_modinit);
-
-static void __exit wm8955_exit(void)
-{
- i2c_del_driver(&wm8955_i2c_driver);
-}
-module_exit(wm8955_exit);
+module_i2c_driver(wm8955_i2c_driver);
MODULE_DESCRIPTION("ASoC WM8955 driver");
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c
index 1332692..00121ba 100644
--- a/sound/soc/codecs/wm8958-dsp2.c
+++ b/sound/soc/codecs/wm8958-dsp2.c
@@ -946,7 +946,7 @@
wm8994->mbc_texts = kmalloc(sizeof(char *)
* pdata->num_mbc_cfgs, GFP_KERNEL);
if (!wm8994->mbc_texts) {
- dev_err(wm8994->codec->dev,
+ dev_err(wm8994->hubs.codec->dev,
"Failed to allocate %d MBC config texts\n",
pdata->num_mbc_cfgs);
return;
@@ -958,9 +958,10 @@
wm8994->mbc_enum.max = pdata->num_mbc_cfgs;
wm8994->mbc_enum.texts = wm8994->mbc_texts;
- ret = snd_soc_add_codec_controls(wm8994->codec, control, 1);
+ ret = snd_soc_add_codec_controls(wm8994->hubs.codec,
+ control, 1);
if (ret != 0)
- dev_err(wm8994->codec->dev,
+ dev_err(wm8994->hubs.codec->dev,
"Failed to add MBC mode controls: %d\n", ret);
}
@@ -974,7 +975,7 @@
wm8994->vss_texts = kmalloc(sizeof(char *)
* pdata->num_vss_cfgs, GFP_KERNEL);
if (!wm8994->vss_texts) {
- dev_err(wm8994->codec->dev,
+ dev_err(wm8994->hubs.codec->dev,
"Failed to allocate %d VSS config texts\n",
pdata->num_vss_cfgs);
return;
@@ -986,9 +987,10 @@
wm8994->vss_enum.max = pdata->num_vss_cfgs;
wm8994->vss_enum.texts = wm8994->vss_texts;
- ret = snd_soc_add_codec_controls(wm8994->codec, control, 1);
+ ret = snd_soc_add_codec_controls(wm8994->hubs.codec,
+ control, 1);
if (ret != 0)
- dev_err(wm8994->codec->dev,
+ dev_err(wm8994->hubs.codec->dev,
"Failed to add VSS mode controls: %d\n", ret);
}
@@ -1003,7 +1005,7 @@
wm8994->vss_hpf_texts = kmalloc(sizeof(char *)
* pdata->num_vss_hpf_cfgs, GFP_KERNEL);
if (!wm8994->vss_hpf_texts) {
- dev_err(wm8994->codec->dev,
+ dev_err(wm8994->hubs.codec->dev,
"Failed to allocate %d VSS HPF config texts\n",
pdata->num_vss_hpf_cfgs);
return;
@@ -1015,9 +1017,10 @@
wm8994->vss_hpf_enum.max = pdata->num_vss_hpf_cfgs;
wm8994->vss_hpf_enum.texts = wm8994->vss_hpf_texts;
- ret = snd_soc_add_codec_controls(wm8994->codec, control, 1);
+ ret = snd_soc_add_codec_controls(wm8994->hubs.codec,
+ control, 1);
if (ret != 0)
- dev_err(wm8994->codec->dev,
+ dev_err(wm8994->hubs.codec->dev,
"Failed to add VSS HPFmode controls: %d\n",
ret);
}
@@ -1033,7 +1036,7 @@
wm8994->enh_eq_texts = kmalloc(sizeof(char *)
* pdata->num_enh_eq_cfgs, GFP_KERNEL);
if (!wm8994->enh_eq_texts) {
- dev_err(wm8994->codec->dev,
+ dev_err(wm8994->hubs.codec->dev,
"Failed to allocate %d enhanced EQ config texts\n",
pdata->num_enh_eq_cfgs);
return;
@@ -1045,9 +1048,10 @@
wm8994->enh_eq_enum.max = pdata->num_enh_eq_cfgs;
wm8994->enh_eq_enum.texts = wm8994->enh_eq_texts;
- ret = snd_soc_add_codec_controls(wm8994->codec, control, 1);
+ ret = snd_soc_add_codec_controls(wm8994->hubs.codec,
+ control, 1);
if (ret != 0)
- dev_err(wm8994->codec->dev,
+ dev_err(wm8994->hubs.codec->dev,
"Failed to add enhanced EQ controls: %d\n",
ret);
}
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 96518ac..804f411 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -1010,23 +1010,7 @@
.id_table = wm8960_i2c_id,
};
-static int __init wm8960_modinit(void)
-{
- int ret = 0;
- ret = i2c_add_driver(&wm8960_i2c_driver);
- if (ret != 0) {
- printk(KERN_ERR "Failed to register WM8960 I2C driver: %d\n",
- ret);
- }
- return ret;
-}
-module_init(wm8960_modinit);
-
-static void __exit wm8960_exit(void)
-{
- i2c_del_driver(&wm8960_i2c_driver);
-}
-module_exit(wm8960_exit);
+module_i2c_driver(wm8960_i2c_driver);
MODULE_DESCRIPTION("ASoC WM8960 driver");
MODULE_AUTHOR("Liam Girdwood");
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index 01edbcc..719fb69 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -1114,23 +1114,7 @@
.id_table = wm8961_i2c_id,
};
-static int __init wm8961_modinit(void)
-{
- int ret = 0;
- ret = i2c_add_driver(&wm8961_i2c_driver);
- if (ret != 0) {
- printk(KERN_ERR "Failed to register wm8961 I2C driver: %d\n",
- ret);
- }
- return ret;
-}
-module_init(wm8961_modinit);
-
-static void __exit wm8961_exit(void)
-{
- i2c_del_driver(&wm8961_i2c_driver);
-}
-module_exit(wm8961_exit);
+module_i2c_driver(wm8961_i2c_driver);
MODULE_DESCRIPTION("ASoC WM8961 driver");
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index eef783f..5ce6477 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -721,23 +721,7 @@
.id_table = wm8971_i2c_id,
};
-static int __init wm8971_modinit(void)
-{
- int ret = 0;
- ret = i2c_add_driver(&wm8971_i2c_driver);
- if (ret != 0) {
- printk(KERN_ERR "Failed to register WM8971 I2C driver: %d\n",
- ret);
- }
- return ret;
-}
-module_init(wm8971_modinit);
-
-static void __exit wm8971_exit(void)
-{
- i2c_del_driver(&wm8971_i2c_driver);
-}
-module_exit(wm8971_exit);
+module_i2c_driver(wm8971_i2c_driver);
MODULE_DESCRIPTION("ASoC WM8971 driver");
MODULE_AUTHOR("Lab126");
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index d93c03f..9a39511 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -659,23 +659,7 @@
.id_table = wm8974_i2c_id,
};
-static int __init wm8974_modinit(void)
-{
- int ret = 0;
- ret = i2c_add_driver(&wm8974_i2c_driver);
- if (ret != 0) {
- printk(KERN_ERR "Failed to register wm8974 I2C driver: %d\n",
- ret);
- }
- return ret;
-}
-module_init(wm8974_modinit);
-
-static void __exit wm8974_exit(void)
-{
- i2c_del_driver(&wm8974_i2c_driver);
-}
-module_exit(wm8974_exit);
+module_i2c_driver(wm8974_i2c_driver);
MODULE_DESCRIPTION("ASoC WM8974 driver");
MODULE_AUTHOR("Liam Girdwood");
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index a5be3ad..5421fd9 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -1105,23 +1105,7 @@
.id_table = wm8978_i2c_id,
};
-static int __init wm8978_modinit(void)
-{
- int ret = 0;
- ret = i2c_add_driver(&wm8978_i2c_driver);
- if (ret != 0) {
- printk(KERN_ERR "Failed to register WM8978 I2C driver: %d\n",
- ret);
- }
- return ret;
-}
-module_init(wm8978_modinit);
-
-static void __exit wm8978_exit(void)
-{
- i2c_del_driver(&wm8978_i2c_driver);
-}
-module_exit(wm8978_exit);
+module_i2c_driver(wm8978_i2c_driver);
MODULE_DESCRIPTION("ASoC WM8978 codec driver");
MODULE_AUTHOR("Guennadi Liakhovetski <g.liakhovetski@gmx.de>");
diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index 9ac31ba..b9dbfeb 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -1400,23 +1400,7 @@
.id_table = wm8991_i2c_id,
};
-static int __init wm8991_modinit(void)
-{
- int ret;
- ret = i2c_add_driver(&wm8991_i2c_driver);
- if (ret != 0) {
- printk(KERN_ERR "Failed to register WM8991 I2C driver: %d\n",
- ret);
- }
- return 0;
-}
-module_init(wm8991_modinit);
-
-static void __exit wm8991_exit(void)
-{
- i2c_del_driver(&wm8991_i2c_driver);
-}
-module_exit(wm8991_exit);
+module_i2c_driver(wm8991_i2c_driver);
MODULE_DESCRIPTION("ASoC WM8991 driver");
MODULE_AUTHOR("Graeme Gregory");
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 9fd80d6..94737a3 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -1520,6 +1520,8 @@
wm8993->pdata.lineout2fb,
wm8993->pdata.jd_scthr,
wm8993->pdata.jd_thr,
+ wm8993->pdata.micbias1_delay,
+ wm8993->pdata.micbias2_delay,
wm8993->pdata.micbias1_lvl,
wm8993->pdata.micbias2_lvl);
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 6c9eeca..2b2dadc 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -671,6 +671,18 @@
eq_tlv),
};
+static const struct snd_kcontrol_new wm8994_drc_controls[] = {
+SND_SOC_BYTES_MASK("AIF1.1 DRC", WM8994_AIF1_DRC1_1, 5,
+ WM8994_AIF1DAC1_DRC_ENA | WM8994_AIF1ADC1L_DRC_ENA |
+ WM8994_AIF1ADC1R_DRC_ENA),
+SND_SOC_BYTES_MASK("AIF1.2 DRC", WM8994_AIF1_DRC2_1, 5,
+ WM8994_AIF1DAC2_DRC_ENA | WM8994_AIF1ADC2L_DRC_ENA |
+ WM8994_AIF1ADC2R_DRC_ENA),
+SND_SOC_BYTES_MASK("AIF2 DRC", WM8994_AIF2_DRC_1, 5,
+ WM8994_AIF2DAC_DRC_ENA | WM8994_AIF2ADCL_DRC_ENA |
+ WM8994_AIF2ADCR_DRC_ENA),
+};
+
static const char *wm8958_ng_text[] = {
"30ms", "125ms", "250ms", "500ms",
};
@@ -789,11 +801,27 @@
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
+ struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
return configure_clock(codec);
+ case SND_SOC_DAPM_POST_PMU:
+ /*
+ * JACKDET won't run until we start the clock and it
+ * only reports deltas, make sure we notify the state
+ * up the stack on startup. Use a *very* generous
+ * timeout for paranoia, there's no urgency and we
+ * don't want false reports.
+ */
+ if (wm8994->jackdet && !wm8994->clk_has_run) {
+ schedule_delayed_work(&wm8994->jackdet_bootstrap,
+ msecs_to_jiffies(1000));
+ wm8994->clk_has_run = true;
+ }
+ break;
+
case SND_SOC_DAPM_POST_PMD:
configure_clock(codec);
break;
@@ -1632,7 +1660,8 @@
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),
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_SUPPLY("DSP1CLK", SND_SOC_NOPM, 3, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("DSP2CLK", SND_SOC_NOPM, 2, 0, NULL, 0),
@@ -2102,6 +2131,10 @@
case WM8994_FLL_SRC_LRCLK:
case WM8994_FLL_SRC_BCLK:
break;
+ case WM8994_FLL_SRC_INTERNAL:
+ freq_in = 12000000;
+ freq_out = 12000000;
+ break;
default:
return -EINVAL;
}
@@ -2161,12 +2194,14 @@
snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_4 + reg_offset,
WM8994_FLL1_N_MASK,
- fll.n << WM8994_FLL1_N_SHIFT);
+ fll.n << WM8994_FLL1_N_SHIFT);
snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset,
- WM8958_FLL1_BYP |
+ WM8994_FLL1_FRC_NCO | WM8958_FLL1_BYP |
WM8994_FLL1_REFCLK_DIV_MASK |
WM8994_FLL1_REFCLK_SRC_MASK,
+ ((src == WM8994_FLL_SRC_INTERNAL)
+ << WM8994_FLL1_FRC_NCO_SHIFT) |
(fll.clk_ref_div << WM8994_FLL1_REFCLK_DIV_SHIFT) |
(src - 1));
@@ -2192,13 +2227,16 @@
}
}
+ reg = WM8994_FLL1_ENA;
+
if (fll.k)
- reg = WM8994_FLL1_ENA | WM8994_FLL1_FRAC;
- else
- reg = WM8994_FLL1_ENA;
+ reg |= WM8994_FLL1_FRAC;
+ if (src == WM8994_FLL_SRC_INTERNAL)
+ reg |= WM8994_FLL1_OSC_ENA;
+
snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset,
- WM8994_FLL1_ENA | WM8994_FLL1_FRAC,
- reg);
+ WM8994_FLL1_ENA | WM8994_FLL1_OSC_ENA |
+ WM8994_FLL1_FRAC, reg);
if (wm8994->fll_locked_irq) {
timeout = wait_for_completion_timeout(&wm8994->fll_locked[id],
@@ -3027,7 +3065,7 @@
static void wm8994_handle_retune_mobile_pdata(struct wm8994_priv *wm8994)
{
- struct snd_soc_codec *codec = wm8994->codec;
+ struct snd_soc_codec *codec = wm8994->hubs.codec;
struct wm8994_pdata *pdata = wm8994->pdata;
struct snd_kcontrol_new controls[] = {
SOC_ENUM_EXT("AIF1.1 EQ Mode",
@@ -3085,16 +3123,16 @@
wm8994->retune_mobile_enum.max = wm8994->num_retune_mobile_texts;
wm8994->retune_mobile_enum.texts = wm8994->retune_mobile_texts;
- ret = snd_soc_add_codec_controls(wm8994->codec, controls,
+ ret = snd_soc_add_codec_controls(wm8994->hubs.codec, controls,
ARRAY_SIZE(controls));
if (ret != 0)
- dev_err(wm8994->codec->dev,
+ dev_err(wm8994->hubs.codec->dev,
"Failed to add ReTune Mobile controls: %d\n", ret);
}
static void wm8994_handle_pdata(struct wm8994_priv *wm8994)
{
- struct snd_soc_codec *codec = wm8994->codec;
+ struct snd_soc_codec *codec = wm8994->hubs.codec;
struct wm8994_pdata *pdata = wm8994->pdata;
int ret, i;
@@ -3107,6 +3145,8 @@
pdata->lineout2fb,
pdata->jd_scthr,
pdata->jd_thr,
+ pdata->micb1_delay,
+ pdata->micb2_delay,
pdata->micbias1_lvl,
pdata->micbias2_lvl);
@@ -3123,10 +3163,10 @@
};
/* We need an array of texts for the enum API */
- wm8994->drc_texts = devm_kzalloc(wm8994->codec->dev,
+ wm8994->drc_texts = devm_kzalloc(wm8994->hubs.codec->dev,
sizeof(char *) * pdata->num_drc_cfgs, GFP_KERNEL);
if (!wm8994->drc_texts) {
- dev_err(wm8994->codec->dev,
+ dev_err(wm8994->hubs.codec->dev,
"Failed to allocate %d DRC config texts\n",
pdata->num_drc_cfgs);
return;
@@ -3138,23 +3178,28 @@
wm8994->drc_enum.max = pdata->num_drc_cfgs;
wm8994->drc_enum.texts = wm8994->drc_texts;
- ret = snd_soc_add_codec_controls(wm8994->codec, controls,
+ ret = snd_soc_add_codec_controls(wm8994->hubs.codec, controls,
ARRAY_SIZE(controls));
- if (ret != 0)
- dev_err(wm8994->codec->dev,
- "Failed to add DRC mode controls: %d\n", ret);
-
for (i = 0; i < WM8994_NUM_DRC; i++)
wm8994_set_drc(codec, i);
+ } else {
+ ret = snd_soc_add_codec_controls(wm8994->hubs.codec,
+ wm8994_drc_controls,
+ ARRAY_SIZE(wm8994_drc_controls));
}
+ if (ret != 0)
+ dev_err(wm8994->hubs.codec->dev,
+ "Failed to add DRC mode controls: %d\n", ret);
+
+
dev_dbg(codec->dev, "%d ReTune Mobile configurations\n",
pdata->num_retune_mobile_cfgs);
if (pdata->num_retune_mobile_cfgs)
wm8994_handle_retune_mobile_pdata(wm8994);
else
- snd_soc_add_codec_controls(wm8994->codec, wm8994_eq_controls,
+ snd_soc_add_codec_controls(wm8994->hubs.codec, wm8994_eq_controls,
ARRAY_SIZE(wm8994_eq_controls));
for (i = 0; i < ARRAY_SIZE(pdata->micbias); i++) {
@@ -3236,6 +3281,12 @@
snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, reg);
+ /* enable MICDET and MICSHRT deboune */
+ snd_soc_update_bits(codec, WM8994_IRQ_DEBOUNCE,
+ WM8994_MIC1_DET_DB_MASK | WM8994_MIC1_SHRT_DB_MASK |
+ WM8994_MIC2_DET_DB_MASK | WM8994_MIC2_SHRT_DB_MASK,
+ WM8994_MIC1_DET_DB | WM8994_MIC1_SHRT_DB);
+
snd_soc_dapm_sync(&codec->dapm);
return 0;
@@ -3309,7 +3360,7 @@
static irqreturn_t wm8994_mic_irq(int irq, void *data)
{
struct wm8994_priv *priv = data;
- struct snd_soc_codec *codec = priv->codec;
+ struct snd_soc_codec *codec = priv->hubs.codec;
#ifndef CONFIG_SND_SOC_WM8994_MODULE
trace_snd_soc_jack_irq(dev_name(codec->dev));
@@ -3345,7 +3396,7 @@
snd_soc_jack_report(wm8994->micdet[0].jack, 0,
wm8994->btn_mask |
- SND_JACK_HEADSET);
+ SND_JACK_HEADSET);
}
return;
}
@@ -3422,7 +3473,7 @@
static irqreturn_t wm1811_jackdet_irq(int irq, void *data)
{
struct wm8994_priv *wm8994 = data;
- struct snd_soc_codec *codec = wm8994->codec;
+ struct snd_soc_codec *codec = wm8994->hubs.codec;
int reg;
bool present;
@@ -3499,10 +3550,22 @@
SND_JACK_MECHANICAL | SND_JACK_HEADSET |
wm8994->btn_mask);
+ /* Since we only report deltas force an update, ensures we
+ * avoid bootstrapping issues with the core. */
+ snd_soc_jack_report(wm8994->micdet[0].jack, 0, 0);
+
pm_runtime_put(codec->dev);
return IRQ_HANDLED;
}
+static void wm1811_jackdet_bootstrap(struct work_struct *work)
+{
+ struct wm8994_priv *wm8994 = container_of(work,
+ struct wm8994_priv,
+ jackdet_bootstrap.work);
+ wm1811_jackdet_irq(0, wm8994);
+}
+
/**
* wm8958_mic_detect - Enable microphone detection via the WM8958 IRQ
*
@@ -3573,6 +3636,10 @@
* otherwise jump straight to microphone detection.
*/
if (wm8994->jackdet) {
+ /* Disable debounce for the initial detect */
+ snd_soc_update_bits(codec, WM1811_JACKDET_CTRL,
+ WM1811_JACKDET_DB, 0);
+
snd_soc_update_bits(codec, WM8958_MICBIAS2,
WM8958_MICB2_DISCH,
WM8958_MICB2_DISCH);
@@ -3600,7 +3667,7 @@
static irqreturn_t wm8958_mic_irq(int irq, void *data)
{
struct wm8994_priv *wm8994 = data;
- struct snd_soc_codec *codec = wm8994->codec;
+ struct snd_soc_codec *codec = wm8994->hubs.codec;
int reg, count;
/*
@@ -3690,15 +3757,15 @@
unsigned int reg;
int ret, i;
- wm8994->codec = codec;
+ wm8994->hubs.codec = codec;
codec->control_data = control->regmap;
snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP);
- wm8994->codec = codec;
-
mutex_init(&wm8994->accdet_lock);
INIT_DELAYED_WORK(&wm8994->mic_work, wm8994_mic_work);
+ INIT_DELAYED_WORK(&wm8994->jackdet_bootstrap,
+ wm1811_jackdet_bootstrap);
for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++)
init_completion(&wm8994->fll_locked[i]);
@@ -3756,14 +3823,17 @@
wm8994->hubs.no_cache_dac_hp_direct = true;
wm8994->fll_byp = true;
- switch (wm8994->revision) {
+ switch (control->cust_id) {
case 0:
- case 1:
case 2:
- case 3:
wm8994->hubs.dcs_codes_l = -9;
wm8994->hubs.dcs_codes_r = -7;
break;
+ case 1:
+ case 3:
+ wm8994->hubs.dcs_codes_l = -8;
+ wm8994->hubs.dcs_codes_r = -7;
+ break;
default:
break;
}
@@ -3852,7 +3922,7 @@
switch (control->type) {
case WM1811:
- if (wm8994->revision > 1) {
+ if (control->cust_id > 1 || wm8994->revision > 1) {
ret = wm8994_request_irq(wm8994->wm8994,
WM8994_IRQ_GPIO(6),
wm1811_jackdet_irq, "JACKDET",
diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h
index d77e06f..f142ec1 100644
--- a/sound/soc/codecs/wm8994.h
+++ b/sound/soc/codecs/wm8994.h
@@ -28,10 +28,11 @@
#define WM8994_FLL1 1
#define WM8994_FLL2 2
-#define WM8994_FLL_SRC_MCLK1 1
-#define WM8994_FLL_SRC_MCLK2 2
-#define WM8994_FLL_SRC_LRCLK 3
-#define WM8994_FLL_SRC_BCLK 4
+#define WM8994_FLL_SRC_MCLK1 1
+#define WM8994_FLL_SRC_MCLK2 2
+#define WM8994_FLL_SRC_LRCLK 3
+#define WM8994_FLL_SRC_BCLK 4
+#define WM8994_FLL_SRC_INTERNAL 5
enum wm8994_vmid_mode {
WM8994_VMID_NORMAL,
@@ -72,7 +73,6 @@
struct wm8994_priv {
struct wm_hubs_data hubs;
struct wm8994 *wm8994;
- struct snd_soc_codec *codec;
int sysclk[2];
int sysclk_rate[2];
int mclk[2];
@@ -81,6 +81,7 @@
struct completion fll_locked[2];
bool fll_locked_irq;
bool fll_byp;
+ bool clk_has_run;
int vmid_refcount;
int active_refcount;
@@ -134,6 +135,7 @@
int btn_mask;
bool jackdet;
int jackdet_mode;
+ struct delayed_work jackdet_bootstrap;
wm8958_micdet_cb jack_cb;
void *jack_cb_data;
diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c
index 2c2346f..c7ddc56 100644
--- a/sound/soc/codecs/wm9090.c
+++ b/sound/soc/codecs/wm9090.c
@@ -695,17 +695,7 @@
.id_table = wm9090_id,
};
-static int __init wm9090_init(void)
-{
- return i2c_add_driver(&wm9090_i2c_driver);
-}
-module_init(wm9090_init);
-
-static void __exit wm9090_exit(void)
-{
- i2c_del_driver(&wm9090_i2c_driver);
-}
-module_exit(wm9090_exit);
+module_i2c_driver(wm9090_i2c_driver);
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
MODULE_DESCRIPTION("WM9090 ASoC driver");
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index c6d2076..1992a62 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -132,8 +132,9 @@
SOC_SINGLE("Phone Volume", AC97_PHONE, 0, 15, 1),
SOC_DOUBLE("Line Capture Volume", AC97_LINE, 8, 0, 31, 1),
-SOC_SINGLE("Capture 20dB Boost Switch", AC97_REC_SEL, 14, 1, 0),
-SOC_SINGLE("Capture to Phone 20dB Boost Switch", AC97_REC_SEL, 11, 1, 1),
+SOC_SINGLE_TLV("Capture Boost Switch", AC97_REC_SEL, 14, 1, 0, boost_tlv),
+SOC_SINGLE_TLV("Capture to Phone Boost Switch", AC97_REC_SEL, 11, 1, 1,
+ boost_tlv),
SOC_SINGLE("3D Upper Cut-off Switch", AC97_3D_CONTROL, 5, 1, 1),
SOC_SINGLE("3D Lower Cut-off Switch", AC97_3D_CONTROL, 4, 1, 1),
@@ -146,7 +147,7 @@
SOC_SINGLE("Bass Volume", AC97_MASTER_TONE, 8, 15, 1),
SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 1),
-SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1),
+SOC_SINGLE("Capture Switch", AC97_REC_GAIN, 15, 1, 1),
SOC_ENUM("Capture Volume Steps", wm9712_enum[6]),
SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 0),
SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0),
@@ -699,8 +700,8 @@
static struct platform_driver wm9712_codec_driver = {
.driver = {
- .name = "wm9712-codec",
- .owner = THIS_MODULE,
+ .name = "wm9712-codec",
+ .owner = THIS_MODULE,
},
.probe = wm9712_probe,
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 61baa48..7a773a8 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -199,15 +199,56 @@
list_add_tail(&cache->list, &hubs->dcs_cache);
}
+static void wm_hubs_read_dc_servo(struct snd_soc_codec *codec,
+ u16 *reg_l, u16 *reg_r)
+{
+ struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
+ u16 dcs_reg, reg;
+
+ switch (hubs->dcs_readback_mode) {
+ case 2:
+ dcs_reg = WM8994_DC_SERVO_4E;
+ break;
+ case 1:
+ dcs_reg = WM8994_DC_SERVO_READBACK;
+ break;
+ default:
+ dcs_reg = WM8993_DC_SERVO_3;
+ break;
+ }
+
+ /* Different chips in the family support different readback
+ * methods.
+ */
+ switch (hubs->dcs_readback_mode) {
+ case 0:
+ *reg_l = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1)
+ & WM8993_DCS_INTEG_CHAN_0_MASK;
+ *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, 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;
+ break;
+ default:
+ WARN(1, "Unknown DCS readback method\n");
+ return;
+ }
+}
+
/*
* Startup calibration of the DC servo
*/
-static void calibrate_dc_servo(struct snd_soc_codec *codec)
+static void enable_dc_servo(struct snd_soc_codec *codec)
{
struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
struct wm_hubs_dcs_cache *cache;
s8 offset;
- u16 reg, reg_l, reg_r, dcs_cfg, dcs_reg;
+ u16 reg_l, reg_r, dcs_cfg, dcs_reg;
switch (hubs->dcs_readback_mode) {
case 2:
@@ -245,27 +286,7 @@
WM8993_DCS_TRIG_STARTUP_1);
}
- /* Different chips in the family support different readback
- * methods.
- */
- switch (hubs->dcs_readback_mode) {
- case 0:
- reg_l = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1)
- & WM8993_DCS_INTEG_CHAN_0_MASK;
- 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, 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;
- break;
- default:
- WARN(1, "Unknown DCS readback method\n");
- return;
- }
+ wm_hubs_read_dc_servo(codec, ®_l, ®_r);
dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r);
@@ -276,12 +297,16 @@
hubs->dcs_codes_l, hubs->dcs_codes_r);
/* HPOUT1R */
- offset = reg_r;
+ offset = (s8)reg_r;
+ dev_dbg(codec->dev, "DCS right %d->%d\n", offset,
+ offset + hubs->dcs_codes_r);
offset += hubs->dcs_codes_r;
dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
/* HPOUT1L */
- offset = reg_l;
+ offset = (s8)reg_l;
+ dev_dbg(codec->dev, "DCS left %d->%d\n", offset,
+ offset + hubs->dcs_codes_l);
offset += hubs->dcs_codes_l;
dcs_cfg |= (u8)offset;
@@ -535,7 +560,7 @@
snd_soc_update_bits(codec, WM8993_DC_SERVO_1,
WM8993_DCS_TIMER_PERIOD_01_MASK, 0);
- calibrate_dc_servo(codec);
+ enable_dc_servo(codec);
reg |= WM8993_HPOUT1R_OUTP | WM8993_HPOUT1R_RMV_SHORT |
WM8993_HPOUT1L_OUTP | WM8993_HPOUT1L_RMV_SHORT;
@@ -619,6 +644,28 @@
return 0;
}
+static int micbias_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
+
+ switch (w->shift) {
+ case WM8993_MICB1_ENA_SHIFT:
+ if (hubs->micb1_delay)
+ msleep(hubs->micb1_delay);
+ break;
+ case WM8993_MICB2_ENA_SHIFT:
+ if (hubs->micb2_delay)
+ msleep(hubs->micb2_delay);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
void wm_hubs_update_class_w(struct snd_soc_codec *codec)
{
struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
@@ -809,8 +856,10 @@
SND_SOC_DAPM_INPUT("IN2RN"),
SND_SOC_DAPM_INPUT("IN2RP:VXRP"),
-SND_SOC_DAPM_SUPPLY("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0, NULL, 0),
-SND_SOC_DAPM_SUPPLY("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0,
+ micbias_event, SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_SUPPLY("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0,
+ micbias_event, SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_MIXER("IN1L PGA", WM8993_POWER_MANAGEMENT_2, 6, 0,
in1l_pga, ARRAY_SIZE(in1l_pga)),
@@ -1112,6 +1161,8 @@
struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
struct snd_soc_dapm_context *dapm = &codec->dapm;
+ hubs->codec = codec;
+
INIT_LIST_HEAD(&hubs->dcs_cache);
init_completion(&hubs->dcs_done);
@@ -1143,13 +1194,16 @@
int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec,
int lineout1_diff, int lineout2_diff,
int lineout1fb, int lineout2fb,
- int jd_scthr, int jd_thr, int micbias1_lvl,
- int micbias2_lvl)
+ int jd_scthr, int jd_thr,
+ int micbias1_delay, int micbias2_delay,
+ int micbias1_lvl, int micbias2_lvl)
{
struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
hubs->lineout1_se = !lineout1_diff;
hubs->lineout2_se = !lineout2_diff;
+ hubs->micb1_delay = micbias1_delay;
+ hubs->micb2_delay = micbias2_delay;
if (!lineout1_diff)
snd_soc_update_bits(codec, WM8993_LINE_MIXER1,
diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h
index da2dc89..24c763d 100644
--- a/sound/soc/codecs/wm_hubs.h
+++ b/sound/soc/codecs/wm_hubs.h
@@ -36,6 +36,9 @@
struct list_head dcs_cache;
bool (*check_class_w_digital)(struct snd_soc_codec *);
+ int micb1_delay;
+ int micb2_delay;
+
bool lineout1_se;
bool lineout1n_ena;
bool lineout1p_ena;
@@ -46,6 +49,8 @@
bool dcs_done_irq;
struct completion dcs_done;
+
+ struct snd_soc_codec *codec;
};
extern int wm_hubs_add_analogue_controls(struct snd_soc_codec *);
@@ -54,6 +59,7 @@
int lineout1_diff, int lineout2_diff,
int lineout1fb, int lineout2fb,
int jd_scthr, int jd_thr,
+ int micbias1_dly, int micbias2_dly,
int micbias1_lvl, int micbias2_lvl);
extern irqreturn_t wm_hubs_dcs_done(int irq, void *data);
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 10a2d8c..6fac5af 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -22,10 +22,6 @@
#include <asm/dma.h>
#include <asm/mach-types.h>
-#include <mach/asp.h>
-#include <mach/edma.h>
-#include <mach/mux.h>
-
#include "davinci-pcm.h"
#include "davinci-i2s.h"
#include "davinci-mcasp.h"
@@ -160,7 +156,7 @@
.cpu_dai_name = "davinci-mcbsp",
.codec_dai_name = "tlv320aic3x-hifi",
.codec_name = "tlv320aic3x-codec.1-001b",
- .platform_name = "davinci-pcm-audio",
+ .platform_name = "davinci-mcbsp",
.init = evm_aic3x_init,
.ops = &evm_ops,
};
@@ -171,7 +167,7 @@
.cpu_dai_name = "davinci-mcbsp.1",
.codec_dai_name = "tlv320aic3x-hifi",
.codec_name = "tlv320aic3x-codec.1-001b",
- .platform_name = "davinci-pcm-audio",
+ .platform_name = "davinci-mcbsp.1",
.init = evm_aic3x_init,
.ops = &evm_ops,
};
@@ -185,14 +181,15 @@
.init = evm_aic3x_init,
.codec_name = "tlv320aic3x-codec.1-0018",
.ops = &evm_ops,
+ .platform_name = "davinci-mcbsp",
#elif defined(CONFIG_SND_DM365_VOICE_CODEC)
.name = "Voice Codec - CQ93VC",
.stream_name = "CQ93",
.cpu_dai_name = "davinci-vcif",
.codec_dai_name = "cq93vc-hifi",
.codec_name = "cq93vc-codec",
+ .platform_name = "davinci-vcif",
#endif
- .platform_name = "davinci-pcm-audio",
};
static struct snd_soc_dai_link dm6467_evm_dai[] = {
@@ -201,7 +198,7 @@
.stream_name = "AIC3X",
.cpu_dai_name= "davinci-mcasp.0",
.codec_dai_name = "tlv320aic3x-hifi",
- .platform_name ="davinci-pcm-audio",
+ .platform_name = "davinci-mcasp.0",
.codec_name = "tlv320aic3x-codec.0-001a",
.init = evm_aic3x_init,
.ops = &evm_ops,
@@ -212,7 +209,7 @@
.cpu_dai_name= "davinci-mcasp.1",
.codec_dai_name = "dit-hifi",
.codec_name = "spdif_dit",
- .platform_name = "davinci-pcm-audio",
+ .platform_name = "davinci-mcasp.1",
.ops = &evm_spdif_ops,
},
};
@@ -223,7 +220,7 @@
.cpu_dai_name = "davinci-mcasp.1",
.codec_dai_name = "tlv320aic3x-hifi",
.codec_name = "tlv320aic3x-codec.1-0018",
- .platform_name = "davinci-pcm-audio",
+ .platform_name = "davinci-mcasp.1",
.init = evm_aic3x_init,
.ops = &evm_ops,
};
@@ -234,7 +231,7 @@
.cpu_dai_name= "davinci-mcasp.0",
.codec_dai_name = "tlv320aic3x-hifi",
.codec_name = "tlv320aic3x-codec.1-0018",
- .platform_name = "davinci-pcm-audio",
+ .platform_name = "davinci-mcasp.0",
.init = evm_aic3x_init,
.ops = &evm_ops,
};
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 0a74b95..8218312 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -16,6 +16,7 @@
#include <linux/delay.h>
#include <linux/io.h>
#include <linux/clk.h>
+#include <linux/platform_data/davinci_asp.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -23,8 +24,6 @@
#include <sound/initval.h>
#include <sound/soc.h>
-#include <mach/asp.h>
-
#include "davinci-pcm.h"
#include "davinci-i2s.h"
@@ -732,8 +731,16 @@
if (ret != 0)
goto err_release_clk;
+ ret = davinci_soc_platform_register(&pdev->dev);
+ if (ret) {
+ dev_err(&pdev->dev, "register PCM failed: %d\n", ret);
+ goto err_unregister_dai;
+ }
+
return 0;
+err_unregister_dai:
+ snd_soc_unregister_dai(&pdev->dev);
err_release_clk:
clk_disable(dev->clk);
clk_put(dev->clk);
@@ -745,6 +752,8 @@
struct davinci_mcbsp_dev *dev = dev_get_drvdata(&pdev->dev);
snd_soc_unregister_dai(&pdev->dev);
+ davinci_soc_platform_unregister(&pdev->dev);
+
clk_disable(dev->clk);
clk_put(dev->clk);
dev->clk = NULL;
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index ce5e5cd..c3eae1d 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -21,7 +21,10 @@
#include <linux/slab.h>
#include <linux/delay.h>
#include <linux/io.h>
-#include <linux/clk.h>
+#include <linux/pm_runtime.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <linux/of_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -782,20 +785,17 @@
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- if (!dev->clk_active) {
- clk_enable(dev->clk);
- dev->clk_active = 1;
- }
+ ret = pm_runtime_get_sync(dev->dev);
+ if (IS_ERR_VALUE(ret))
+ dev_err(dev->dev, "pm_runtime_get_sync() failed\n");
davinci_mcasp_start(dev, substream->stream);
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
davinci_mcasp_stop(dev, substream->stream);
- if (dev->clk_active) {
- clk_disable(dev->clk);
- dev->clk_active = 0;
- }
-
+ ret = pm_runtime_put_sync(dev->dev);
+ if (IS_ERR_VALUE(ret))
+ dev_err(dev->dev, "pm_runtime_put_sync() failed\n");
break;
case SNDRV_PCM_TRIGGER_STOP:
@@ -865,6 +865,114 @@
};
+static const struct of_device_id mcasp_dt_ids[] = {
+ {
+ .compatible = "ti,dm646x-mcasp-audio",
+ .data = (void *)MCASP_VERSION_1,
+ },
+ {
+ .compatible = "ti,da830-mcasp-audio",
+ .data = (void *)MCASP_VERSION_2,
+ },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, mcasp_dt_ids);
+
+static struct snd_platform_data *davinci_mcasp_set_pdata_from_of(
+ struct platform_device *pdev)
+{
+ struct device_node *np = pdev->dev.of_node;
+ struct snd_platform_data *pdata = NULL;
+ const struct of_device_id *match =
+ of_match_device(of_match_ptr(mcasp_dt_ids), &pdev->dev);
+
+ const u32 *of_serial_dir32;
+ u8 *of_serial_dir;
+ u32 val;
+ int i, ret = 0;
+
+ if (pdev->dev.platform_data) {
+ pdata = pdev->dev.platform_data;
+ return pdata;
+ } else if (match) {
+ pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL);
+ if (!pdata) {
+ ret = -ENOMEM;
+ goto nodata;
+ }
+ } else {
+ /* control shouldn't reach here. something is wrong */
+ ret = -EINVAL;
+ goto nodata;
+ }
+
+ if (match->data)
+ pdata->version = (u8)((int)match->data);
+
+ ret = of_property_read_u32(np, "op-mode", &val);
+ if (ret >= 0)
+ pdata->op_mode = val;
+
+ ret = of_property_read_u32(np, "tdm-slots", &val);
+ if (ret >= 0)
+ pdata->tdm_slots = val;
+
+ ret = of_property_read_u32(np, "num-serializer", &val);
+ if (ret >= 0)
+ pdata->num_serializer = val;
+
+ of_serial_dir32 = of_get_property(np, "serial-dir", &val);
+ val /= sizeof(u32);
+ if (val != pdata->num_serializer) {
+ dev_err(&pdev->dev,
+ "num-serializer(%d) != serial-dir size(%d)\n",
+ pdata->num_serializer, val);
+ ret = -EINVAL;
+ goto nodata;
+ }
+
+ if (of_serial_dir32) {
+ of_serial_dir = devm_kzalloc(&pdev->dev,
+ (sizeof(*of_serial_dir) * val),
+ GFP_KERNEL);
+ if (!of_serial_dir) {
+ ret = -ENOMEM;
+ goto nodata;
+ }
+
+ for (i = 0; i < pdata->num_serializer; i++)
+ of_serial_dir[i] = be32_to_cpup(&of_serial_dir32[i]);
+
+ pdata->serial_dir = of_serial_dir;
+ }
+
+ ret = of_property_read_u32(np, "tx-num-evt", &val);
+ if (ret >= 0)
+ pdata->txnumevt = val;
+
+ ret = of_property_read_u32(np, "rx-num-evt", &val);
+ if (ret >= 0)
+ pdata->rxnumevt = val;
+
+ ret = of_property_read_u32(np, "sram-size-playback", &val);
+ if (ret >= 0)
+ pdata->sram_size_playback = val;
+
+ ret = of_property_read_u32(np, "sram-size-capture", &val);
+ if (ret >= 0)
+ pdata->sram_size_capture = val;
+
+ return pdata;
+
+nodata:
+ if (ret < 0) {
+ dev_err(&pdev->dev, "Error populating platform data, err %d\n",
+ ret);
+ pdata = NULL;
+ }
+ return pdata;
+}
+
static int davinci_mcasp_probe(struct platform_device *pdev)
{
struct davinci_pcm_dma_params *dma_data;
@@ -873,11 +981,22 @@
struct davinci_audio_dev *dev;
int ret;
+ if (!pdev->dev.platform_data && !pdev->dev.of_node) {
+ dev_err(&pdev->dev, "No platform data supplied\n");
+ return -EINVAL;
+ }
+
dev = devm_kzalloc(&pdev->dev, sizeof(struct davinci_audio_dev),
GFP_KERNEL);
if (!dev)
return -ENOMEM;
+ pdata = davinci_mcasp_set_pdata_from_of(pdev);
+ if (!pdata) {
+ dev_err(&pdev->dev, "no platform data\n");
+ return -EINVAL;
+ }
+
mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (!mem) {
dev_err(&pdev->dev, "no mem resource?\n");
@@ -891,13 +1010,13 @@
return -EBUSY;
}
- pdata = pdev->dev.platform_data;
- dev->clk = clk_get(&pdev->dev, NULL);
- if (IS_ERR(dev->clk))
- return -ENODEV;
+ pm_runtime_enable(&pdev->dev);
- clk_enable(dev->clk);
- dev->clk_active = 1;
+ ret = pm_runtime_get_sync(&pdev->dev);
+ if (IS_ERR_VALUE(ret)) {
+ dev_err(&pdev->dev, "pm_runtime_get_sync() failed\n");
+ return ret;
+ }
dev->base = devm_ioremap(&pdev->dev, mem->start, resource_size(mem));
if (!dev->base) {
@@ -914,6 +1033,7 @@
dev->version = pdata->version;
dev->txnumevt = pdata->txnumevt;
dev->rxnumevt = pdata->rxnumevt;
+ dev->dev = &pdev->dev;
dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
dma_data->asp_chan_q = pdata->asp_chan_q;
@@ -952,22 +1072,31 @@
if (ret != 0)
goto err_release_clk;
+
+ ret = davinci_soc_platform_register(&pdev->dev);
+ if (ret) {
+ dev_err(&pdev->dev, "register PCM failed: %d\n", ret);
+ goto err_unregister_dai;
+ }
+
return 0;
+err_unregister_dai:
+ snd_soc_unregister_dai(&pdev->dev);
err_release_clk:
- clk_disable(dev->clk);
- clk_put(dev->clk);
+ pm_runtime_put_sync(&pdev->dev);
+ pm_runtime_disable(&pdev->dev);
return ret;
}
static int davinci_mcasp_remove(struct platform_device *pdev)
{
- struct davinci_audio_dev *dev = dev_get_drvdata(&pdev->dev);
snd_soc_unregister_dai(&pdev->dev);
- clk_disable(dev->clk);
- clk_put(dev->clk);
- dev->clk = NULL;
+ davinci_soc_platform_unregister(&pdev->dev);
+
+ pm_runtime_put_sync(&pdev->dev);
+ pm_runtime_disable(&pdev->dev);
return 0;
}
@@ -978,6 +1107,7 @@
.driver = {
.name = "davinci-mcasp",
.owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(mcasp_dt_ids),
},
};
diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h
index 4681acc..0de9ed6 100644
--- a/sound/soc/davinci/davinci-mcasp.h
+++ b/sound/soc/davinci/davinci-mcasp.h
@@ -19,7 +19,8 @@
#define DAVINCI_MCASP_H
#include <linux/io.h>
-#include <mach/asp.h>
+#include <linux/platform_data/davinci_asp.h>
+
#include "davinci-pcm.h"
#define DAVINCI_MCASP_RATES SNDRV_PCM_RATE_8000_96000
@@ -40,9 +41,8 @@
struct davinci_pcm_dma_params dma_params[2];
void __iomem *base;
int sample_rate;
- struct clk *clk;
+ struct device *dev;
unsigned int codec_fmt;
- u8 clk_active;
/* McASP specific data */
int tdm_slots;
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 97d77b2..93ea3bf 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -23,7 +23,6 @@
#include <sound/soc.h>
#include <asm/dma.h>
-#include <mach/edma.h>
#include <mach/sram.h>
#include "davinci-pcm.h"
@@ -864,28 +863,17 @@
.pcm_free = davinci_pcm_free,
};
-static int __devinit davinci_soc_platform_probe(struct platform_device *pdev)
+int davinci_soc_platform_register(struct device *dev)
{
- return snd_soc_register_platform(&pdev->dev, &davinci_soc_platform);
+ return snd_soc_register_platform(dev, &davinci_soc_platform);
}
+EXPORT_SYMBOL_GPL(davinci_soc_platform_register);
-static int __devexit davinci_soc_platform_remove(struct platform_device *pdev)
+void davinci_soc_platform_unregister(struct device *dev)
{
- snd_soc_unregister_platform(&pdev->dev);
- return 0;
+ snd_soc_unregister_platform(dev);
}
-
-static struct platform_driver davinci_pcm_driver = {
- .driver = {
- .name = "davinci-pcm-audio",
- .owner = THIS_MODULE,
- },
-
- .probe = davinci_soc_platform_probe,
- .remove = __devexit_p(davinci_soc_platform_remove),
-};
-
-module_platform_driver(davinci_pcm_driver);
+EXPORT_SYMBOL_GPL(davinci_soc_platform_unregister);
MODULE_AUTHOR("Vladimir Barinov");
MODULE_DESCRIPTION("TI DAVINCI PCM DMA module");
diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h
index c0d6c9b..fc4d01c 100644
--- a/sound/soc/davinci/davinci-pcm.h
+++ b/sound/soc/davinci/davinci-pcm.h
@@ -12,9 +12,8 @@
#ifndef _DAVINCI_PCM_H
#define _DAVINCI_PCM_H
+#include <linux/platform_data/davinci_asp.h>
#include <mach/edma.h>
-#include <mach/asp.h>
-
struct davinci_pcm_dma_params {
int channel; /* sync dma channel ID */
@@ -28,4 +27,7 @@
unsigned int fifo_level;
};
+int davinci_soc_platform_register(struct device *dev);
+void davinci_soc_platform_unregister(struct device *dev);
+
#endif
diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c
index f71175b..5be65aa 100644
--- a/sound/soc/davinci/davinci-sffsdr.c
+++ b/sound/soc/davinci/davinci-sffsdr.c
@@ -86,7 +86,7 @@
.cpu_dai_name = "davinci-mcbsp",
.codec_dai_name = "pcm3008-hifi",
.codec_name = "pcm3008-codec",
- .platform_name = "davinci-pcm-audio",
+ .platform_name = "davinci-mcbsp",
.ops = &sffsdr_ops,
};
diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c
index da030ff..07bde2e 100644
--- a/sound/soc/davinci/davinci-vcif.c
+++ b/sound/soc/davinci/davinci-vcif.c
@@ -240,12 +240,20 @@
return ret;
}
+ ret = davinci_soc_platform_register(&pdev->dev);
+ if (ret) {
+ dev_err(&pdev->dev, "register PCM failed: %d\n", ret);
+ snd_soc_unregister_dai(&pdev->dev);
+ return ret;
+ }
+
return 0;
}
static int davinci_vcif_remove(struct platform_device *pdev)
{
snd_soc_unregister_dai(&pdev->dev);
+ davinci_soc_platform_unregister(&pdev->dev);
return 0;
}
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index 81d7728..7074ae6 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -524,7 +524,7 @@
int ret = 0;
struct snd_soc_dai_driver *dai;
- ssi = kzalloc(sizeof(*ssi), GFP_KERNEL);
+ ssi = devm_kzalloc(&pdev->dev, sizeof(*ssi), GFP_KERNEL);
if (!ssi)
return -ENOMEM;
dev_set_drvdata(&pdev->dev, ssi);
@@ -537,7 +537,7 @@
ssi->irq = platform_get_irq(pdev, 0);
- ssi->clk = clk_get(&pdev->dev, NULL);
+ ssi->clk = devm_clk_get(&pdev->dev, NULL);
if (IS_ERR(ssi->clk)) {
ret = PTR_ERR(ssi->clk);
dev_err(&pdev->dev, "Cannot get the clock: %d\n",
@@ -552,23 +552,18 @@
goto failed_get_resource;
}
- if (!request_mem_region(res->start, resource_size(res), DRV_NAME)) {
- dev_err(&pdev->dev, "request_mem_region failed\n");
- ret = -EBUSY;
- goto failed_get_resource;
- }
-
- ssi->base = ioremap(res->start, resource_size(res));
+ ssi->base = devm_request_and_ioremap(&pdev->dev, res);
if (!ssi->base) {
dev_err(&pdev->dev, "ioremap failed\n");
ret = -ENODEV;
- goto failed_ioremap;
+ goto failed_register;
}
if (ssi->flags & IMX_SSI_USE_AC97) {
if (ac97_ssi) {
+ dev_err(&pdev->dev, "AC'97 SSI already registered\n");
ret = -EBUSY;
- goto failed_ac97;
+ goto failed_register;
}
ac97_ssi = ssi;
setup_channel_to_ac97(ssi);
@@ -637,15 +632,10 @@
failed_pdev_fiq_alloc:
snd_soc_unregister_dai(&pdev->dev);
failed_register:
-failed_ac97:
- iounmap(ssi->base);
-failed_ioremap:
release_mem_region(res->start, resource_size(res));
failed_get_resource:
clk_disable_unprepare(ssi->clk);
- clk_put(ssi->clk);
failed_clk:
- kfree(ssi);
return ret;
}
@@ -663,11 +653,8 @@
if (ssi->flags & IMX_SSI_USE_AC97)
ac97_ssi = NULL;
- iounmap(ssi->base);
release_mem_region(res->start, resource_size(res));
clk_disable_unprepare(ssi->clk);
- clk_put(ssi->clk);
- kfree(ssi);
return 0;
}
diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c
index 2937e54..2cc7782 100644
--- a/sound/soc/mid-x86/mfld_machine.c
+++ b/sound/soc/mid-x86/mfld_machine.c
@@ -318,6 +318,15 @@
.platform_name = "sst-platform",
.init = NULL,
},
+ {
+ .name = "Medfield Compress",
+ .stream_name = "Speaker",
+ .cpu_dai_name = "Compress-cpu-dai",
+ .codec_dai_name = "SN95031 Speaker",
+ .codec_name = "sn95031",
+ .platform_name = "sst-platform",
+ .init = NULL,
+ },
};
/* SoC card */
diff --git a/sound/soc/mid-x86/sst_dsp.h b/sound/soc/mid-x86/sst_dsp.h
new file mode 100644
index 0000000..0fce1de
--- /dev/null
+++ b/sound/soc/mid-x86/sst_dsp.h
@@ -0,0 +1,134 @@
+#ifndef __SST_DSP_H__
+#define __SST_DSP_H__
+/*
+ * sst_dsp.h - Intel SST Driver for audio engine
+ *
+ * Copyright (C) 2008-12 Intel Corporation
+ * Authors: Vinod Koul <vinod.koul@linux.intel.com>
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ * 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; version 2 of the License.
+ *
+ * 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.,
+ * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ */
+
+enum sst_codec_types {
+ /* AUDIO/MUSIC CODEC Type Definitions */
+ SST_CODEC_TYPE_UNKNOWN = 0,
+ SST_CODEC_TYPE_PCM, /* Pass through Audio codec */
+ SST_CODEC_TYPE_MP3,
+ SST_CODEC_TYPE_MP24,
+ SST_CODEC_TYPE_AAC,
+ SST_CODEC_TYPE_AACP,
+ SST_CODEC_TYPE_eAACP,
+};
+
+enum stream_type {
+ SST_STREAM_TYPE_NONE = 0,
+ SST_STREAM_TYPE_MUSIC = 1,
+};
+
+struct snd_pcm_params {
+ u16 codec; /* codec type */
+ u8 num_chan; /* 1=Mono, 2=Stereo */
+ u8 pcm_wd_sz; /* 16/24 - bit*/
+ u32 reserved; /* Bitrate in bits per second */
+ u32 sfreq; /* Sampling rate in Hz */
+ u8 use_offload_path;
+ u8 reserved2;
+ u16 reserved3;
+ u8 channel_map[8];
+} __packed;
+
+/* MP3 Music Parameters Message */
+struct snd_mp3_params {
+ u16 codec;
+ u8 num_chan; /* 1=Mono, 2=Stereo */
+ u8 pcm_wd_sz; /* 16/24 - bit*/
+ u8 crc_check; /* crc_check - disable (0) or enable (1) */
+ u8 reserved1; /* unused*/
+ u16 reserved2; /* Unused */
+} __packed;
+
+#define AAC_BIT_STREAM_ADTS 0
+#define AAC_BIT_STREAM_ADIF 1
+#define AAC_BIT_STREAM_RAW 2
+
+/* AAC Music Parameters Message */
+struct snd_aac_params {
+ u16 codec;
+ u8 num_chan; /* 1=Mono, 2=Stereo*/
+ u8 pcm_wd_sz; /* 16/24 - bit*/
+ u8 bdownsample; /*SBR downsampling 0 - disable 1 -enabled AAC+ only */
+ u8 bs_format; /* input bit stream format adts=0, adif=1, raw=2 */
+ u16 reser2;
+ u32 externalsr; /*sampling rate of basic AAC raw bit stream*/
+ u8 sbr_signalling;/*disable/enable/set automode the SBR tool.AAC+*/
+ u8 reser1;
+ u16 reser3;
+} __packed;
+
+/* WMA Music Parameters Message */
+struct snd_wma_params {
+ u16 codec;
+ u8 num_chan; /* 1=Mono, 2=Stereo */
+ u8 pcm_wd_sz; /* 16/24 - bit*/
+ u32 brate; /* Use the hard coded value. */
+ u32 sfreq; /* Sampling freq eg. 8000, 441000, 48000 */
+ u32 channel_mask; /* Channel Mask */
+ u16 format_tag; /* Format Tag */
+ u16 block_align; /* packet size */
+ u16 wma_encode_opt;/* Encoder option */
+ u8 op_align; /* op align 0- 16 bit, 1- MSB, 2 LSB */
+ u8 reserved; /* reserved */
+} __packed;
+
+/* Codec params struture */
+union snd_sst_codec_params {
+ struct snd_pcm_params pcm_params;
+ struct snd_mp3_params mp3_params;
+ struct snd_aac_params aac_params;
+ struct snd_wma_params wma_params;
+} __packed;
+
+/* Address and size info of a frame buffer */
+struct sst_address_info {
+ u32 addr; /* Address at IA */
+ u32 size; /* Size of the buffer */
+};
+
+struct snd_sst_alloc_params_ext {
+ struct sst_address_info ring_buf_info[8];
+ u8 sg_count;
+ u8 reserved;
+ u16 reserved2;
+ u32 frag_size; /*Number of samples after which period elapsed
+ message is sent valid only if path = 0*/
+} __packed;
+
+struct snd_sst_stream_params {
+ union snd_sst_codec_params uc;
+} __packed;
+
+struct snd_sst_params {
+ u32 stream_id;
+ u8 codec;
+ u8 ops;
+ u8 stream_type;
+ u8 device_type;
+ struct snd_sst_stream_params sparams;
+ struct snd_sst_alloc_params_ext aparams;
+};
+
+#endif /* __SST_DSP_H__ */
diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c
index d34563b..a263cbe 100644
--- a/sound/soc/mid-x86/sst_platform.c
+++ b/sound/soc/mid-x86/sst_platform.c
@@ -1,7 +1,7 @@
/*
* sst_platform.c - Intel MID Platform driver
*
- * Copyright (C) 2010 Intel Corp
+ * Copyright (C) 2010-2012 Intel Corp
* Author: Vinod Koul <vinod.koul@intel.com>
* Author: Harsha Priya <priya.harsha@intel.com>
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
@@ -32,6 +32,7 @@
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
+#include <sound/compress_driver.h>
#include "sst_platform.h"
static struct sst_device *sst;
@@ -152,6 +153,16 @@
.formats = SNDRV_PCM_FMTBIT_S24_LE,
},
},
+{
+ .name = "Compress-cpu-dai",
+ .compress_dai = 1,
+ .playback = {
+ .channels_min = SST_STEREO,
+ .channels_max = SST_STEREO,
+ .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+},
};
/* helper functions */
@@ -463,8 +474,199 @@
}
return retval;
}
+
+/* compress stream operations */
+static void sst_compr_fragment_elapsed(void *arg)
+{
+ struct snd_compr_stream *cstream = (struct snd_compr_stream *)arg;
+
+ pr_debug("fragment elapsed by driver\n");
+ if (cstream)
+ snd_compr_fragment_elapsed(cstream);
+}
+
+static int sst_platform_compr_open(struct snd_compr_stream *cstream)
+{
+
+ int ret_val = 0;
+ struct snd_compr_runtime *runtime = cstream->runtime;
+ struct sst_runtime_stream *stream;
+
+ stream = kzalloc(sizeof(*stream), GFP_KERNEL);
+ if (!stream)
+ return -ENOMEM;
+
+ spin_lock_init(&stream->status_lock);
+
+ /* get the sst ops */
+ if (!sst || !try_module_get(sst->dev->driver->owner)) {
+ pr_err("no device available to run\n");
+ ret_val = -ENODEV;
+ goto out_ops;
+ }
+ stream->compr_ops = sst->compr_ops;
+
+ stream->id = 0;
+ sst_set_stream_status(stream, SST_PLATFORM_INIT);
+ runtime->private_data = stream;
+ return 0;
+out_ops:
+ kfree(stream);
+ return ret_val;
+}
+
+static int sst_platform_compr_free(struct snd_compr_stream *cstream)
+{
+ struct sst_runtime_stream *stream;
+ int ret_val = 0, str_id;
+
+ stream = cstream->runtime->private_data;
+ /*need to check*/
+ str_id = stream->id;
+ if (str_id)
+ ret_val = stream->compr_ops->close(str_id);
+ module_put(sst->dev->driver->owner);
+ kfree(stream);
+ pr_debug("%s: %d\n", __func__, ret_val);
+ return 0;
+}
+
+static int sst_platform_compr_set_params(struct snd_compr_stream *cstream,
+ struct snd_compr_params *params)
+{
+ struct sst_runtime_stream *stream;
+ int retval;
+ struct snd_sst_params str_params;
+ struct sst_compress_cb cb;
+
+ stream = cstream->runtime->private_data;
+ /* construct fw structure for this*/
+ memset(&str_params, 0, sizeof(str_params));
+
+ str_params.ops = STREAM_OPS_PLAYBACK;
+ str_params.stream_type = SST_STREAM_TYPE_MUSIC;
+ str_params.device_type = SND_SST_DEVICE_COMPRESS;
+
+ switch (params->codec.id) {
+ case SND_AUDIOCODEC_MP3: {
+ str_params.codec = SST_CODEC_TYPE_MP3;
+ str_params.sparams.uc.mp3_params.codec = SST_CODEC_TYPE_MP3;
+ str_params.sparams.uc.mp3_params.num_chan = params->codec.ch_in;
+ str_params.sparams.uc.mp3_params.pcm_wd_sz = 16;
+ break;
+ }
+
+ case SND_AUDIOCODEC_AAC: {
+ str_params.codec = SST_CODEC_TYPE_AAC;
+ str_params.sparams.uc.aac_params.codec = SST_CODEC_TYPE_AAC;
+ str_params.sparams.uc.aac_params.num_chan = params->codec.ch_in;
+ str_params.sparams.uc.aac_params.pcm_wd_sz = 16;
+ if (params->codec.format == SND_AUDIOSTREAMFORMAT_MP4ADTS)
+ str_params.sparams.uc.aac_params.bs_format =
+ AAC_BIT_STREAM_ADTS;
+ else if (params->codec.format == SND_AUDIOSTREAMFORMAT_RAW)
+ str_params.sparams.uc.aac_params.bs_format =
+ AAC_BIT_STREAM_RAW;
+ else {
+ pr_err("Undefined format%d\n", params->codec.format);
+ return -EINVAL;
+ }
+ str_params.sparams.uc.aac_params.externalsr =
+ params->codec.sample_rate;
+ break;
+ }
+
+ default:
+ pr_err("codec not supported, id =%d\n", params->codec.id);
+ return -EINVAL;
+ }
+
+ str_params.aparams.ring_buf_info[0].addr =
+ virt_to_phys(cstream->runtime->buffer);
+ str_params.aparams.ring_buf_info[0].size =
+ cstream->runtime->buffer_size;
+ str_params.aparams.sg_count = 1;
+ str_params.aparams.frag_size = cstream->runtime->fragment_size;
+
+ cb.param = cstream;
+ cb.compr_cb = sst_compr_fragment_elapsed;
+
+ retval = stream->compr_ops->open(&str_params, &cb);
+ if (retval < 0) {
+ pr_err("stream allocation failed %d\n", retval);
+ return retval;
+ }
+
+ stream->id = retval;
+ return 0;
+}
+
+static int sst_platform_compr_trigger(struct snd_compr_stream *cstream, int cmd)
+{
+ struct sst_runtime_stream *stream =
+ cstream->runtime->private_data;
+
+ return stream->compr_ops->control(cmd, stream->id);
+}
+
+static int sst_platform_compr_pointer(struct snd_compr_stream *cstream,
+ struct snd_compr_tstamp *tstamp)
+{
+ struct sst_runtime_stream *stream;
+
+ stream = cstream->runtime->private_data;
+ stream->compr_ops->tstamp(stream->id, tstamp);
+ tstamp->byte_offset = tstamp->copied_total %
+ (u32)cstream->runtime->buffer_size;
+ pr_debug("calc bytes offset/copied bytes as %d\n", tstamp->byte_offset);
+ return 0;
+}
+
+static int sst_platform_compr_ack(struct snd_compr_stream *cstream,
+ size_t bytes)
+{
+ struct sst_runtime_stream *stream;
+
+ stream = cstream->runtime->private_data;
+ stream->compr_ops->ack(stream->id, (unsigned long)bytes);
+ stream->bytes_written += bytes;
+
+ return 0;
+}
+
+static int sst_platform_compr_get_caps(struct snd_compr_stream *cstream,
+ struct snd_compr_caps *caps)
+{
+ struct sst_runtime_stream *stream =
+ cstream->runtime->private_data;
+
+ return stream->compr_ops->get_caps(caps);
+}
+
+static int sst_platform_compr_get_codec_caps(struct snd_compr_stream *cstream,
+ struct snd_compr_codec_caps *codec)
+{
+ struct sst_runtime_stream *stream =
+ cstream->runtime->private_data;
+
+ return stream->compr_ops->get_codec_caps(codec);
+}
+
+static struct snd_compr_ops sst_platform_compr_ops = {
+
+ .open = sst_platform_compr_open,
+ .free = sst_platform_compr_free,
+ .set_params = sst_platform_compr_set_params,
+ .trigger = sst_platform_compr_trigger,
+ .pointer = sst_platform_compr_pointer,
+ .ack = sst_platform_compr_ack,
+ .get_caps = sst_platform_compr_get_caps,
+ .get_codec_caps = sst_platform_compr_get_codec_caps,
+};
+
static struct snd_soc_platform_driver sst_soc_platform_drv = {
.ops = &sst_platform_ops,
+ .compr_ops = &sst_platform_compr_ops,
.pcm_new = sst_pcm_new,
.pcm_free = sst_pcm_free,
};
diff --git a/sound/soc/mid-x86/sst_platform.h b/sound/soc/mid-x86/sst_platform.h
index f04f4f7..d61c5d5 100644
--- a/sound/soc/mid-x86/sst_platform.h
+++ b/sound/soc/mid-x86/sst_platform.h
@@ -27,6 +27,8 @@
#ifndef __SST_PLATFORMDRV_H__
#define __SST_PLATFORMDRV_H__
+#include "sst_dsp.h"
+
#define SST_MONO 1
#define SST_STEREO 2
#define SST_MAX_CAP 5
@@ -42,7 +44,6 @@
#define SST_MIN_PERIODS 2
#define SST_MAX_PERIODS (1024*2)
#define SST_FIFO_SIZE 0
-#define SST_CODEC_TYPE_PCM 1
struct pcm_stream_info {
int str_id;
@@ -83,6 +84,7 @@
SND_SST_DEVICE_VIBRA,
SND_SST_DEVICE_HAPTIC,
SND_SST_DEVICE_CAPTURE,
+ SND_SST_DEVICE_COMPRESS,
};
/* PCM Parameters */
@@ -107,6 +109,24 @@
struct sst_pcm_params sparams;
};
+struct sst_compress_cb {
+ void *param;
+ void (*compr_cb)(void *param);
+};
+
+struct compress_sst_ops {
+ const char *name;
+ int (*open) (struct snd_sst_params *str_params,
+ struct sst_compress_cb *cb);
+ int (*control) (unsigned int cmd, unsigned int str_id);
+ int (*tstamp) (unsigned int str_id, struct snd_compr_tstamp *tstamp);
+ int (*ack) (unsigned int str_id, unsigned long bytes);
+ int (*close) (unsigned int str_id);
+ int (*get_caps) (struct snd_compr_caps *caps);
+ int (*get_codec_caps) (struct snd_compr_codec_caps *codec);
+
+};
+
struct sst_ops {
int (*open) (struct sst_stream_params *str_param);
int (*device_control) (int cmd, void *arg);
@@ -115,8 +135,11 @@
struct sst_runtime_stream {
int stream_status;
+ unsigned int id;
+ size_t bytes_written;
struct pcm_stream_info stream_info;
struct sst_ops *ops;
+ struct compress_sst_ops *compr_ops;
spinlock_t status_lock;
};
@@ -124,6 +147,7 @@
char *name;
struct device *dev;
struct sst_ops *ops;
+ struct compress_sst_ops *compr_ops;
};
int sst_register_dsp(struct sst_device *sst);
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index b303071..aa037b2 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -704,7 +704,7 @@
return ret;
}
- saif->clk = clk_get(&pdev->dev, NULL);
+ saif->clk = devm_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",
@@ -717,8 +717,7 @@
saif->base = devm_request_and_ioremap(&pdev->dev, iores);
if (!saif->base) {
dev_err(&pdev->dev, "ioremap failed\n");
- ret = -ENODEV;
- goto failed_get_resource;
+ return -ENODEV;
}
dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0);
@@ -731,7 +730,7 @@
&saif->dma_param.chan_num);
if (ret) {
dev_err(&pdev->dev, "failed to get dma channel\n");
- goto failed_get_resource;
+ return ret;
}
} else {
saif->dma_param.chan_num = dmares->start;
@@ -742,7 +741,7 @@
ret = saif->irq;
dev_err(&pdev->dev, "failed to get irq resource: %d\n",
ret);
- goto failed_get_resource;
+ return ret;
}
saif->dev = &pdev->dev;
@@ -750,7 +749,7 @@
"mxs-saif", saif);
if (ret) {
dev_err(&pdev->dev, "failed to request irq\n");
- goto failed_get_resource;
+ return ret;
}
saif->dma_param.chan_irq = platform_get_irq(pdev, 1);
@@ -758,7 +757,7 @@
ret = saif->dma_param.chan_irq;
dev_err(&pdev->dev, "failed to get dma irq resource: %d\n",
ret);
- goto failed_get_resource;
+ return ret;
}
platform_set_drvdata(pdev, saif);
@@ -766,7 +765,7 @@
ret = snd_soc_register_dai(&pdev->dev, &mxs_saif_dai);
if (ret) {
dev_err(&pdev->dev, "register DAI failed\n");
- goto failed_get_resource;
+ return ret;
}
ret = mxs_pcm_platform_register(&pdev->dev);
@@ -779,19 +778,14 @@
failed_pdev_alloc:
snd_soc_unregister_dai(&pdev->dev);
-failed_get_resource:
- clk_put(saif->clk);
return ret;
}
static int __devexit mxs_saif_remove(struct platform_device *pdev)
{
- struct mxs_saif *saif = platform_get_drvdata(pdev);
-
mxs_pcm_platform_unregister(&pdev->dev);
snd_soc_unregister_dai(&pdev->dev);
- clk_put(saif->clk);
return 0;
}
diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
index 9d93793..be525df 100644
--- a/sound/soc/omap/omap-abe-twl6040.c
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -25,6 +25,7 @@
#include <linux/mfd/twl6040.h>
#include <linux/platform_data/omap-abe-twl6040.h>
#include <linux/module.h>
+#include <linux/of.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -43,6 +44,8 @@
struct abe_twl6040 {
int jack_detection; /* board can detect jack events */
int mclk_freq; /* MCLK frequency speed for twl6040 */
+
+ struct platform_device *dmic_codec_dev;
};
static int omap_abe_hw_params(struct snd_pcm_substream *substream,
@@ -185,17 +188,6 @@
int hs_trim;
int ret = 0;
- /* Disable not connected paths if not used */
- twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone");
- twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk");
- twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk");
- twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out");
- twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator");
- twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic");
- twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic");
- twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic");
- twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In");
-
/*
* Configure McPDM offset cancellation based on the HSOTRIM value from
* twl6040.
@@ -216,6 +208,24 @@
twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET);
}
+ /*
+ * NULL pdata means we booted with DT. In this case the routing is
+ * provided and the card is fully routed, no need to mark pins.
+ */
+ if (!pdata)
+ return ret;
+
+ /* Disable not connected paths if not used */
+ twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone");
+ twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk");
+ twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk");
+ twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out");
+ twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator");
+ twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic");
+ twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic");
+ twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic");
+ twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In");
+
return ret;
}
@@ -270,52 +280,116 @@
static __devinit int omap_abe_probe(struct platform_device *pdev)
{
struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev);
+ struct device_node *node = pdev->dev.of_node;
struct snd_soc_card *card = &omap_abe_card;
struct abe_twl6040 *priv;
int num_links = 0;
- int ret;
+ int ret = 0;
card->dev = &pdev->dev;
- if (!pdata) {
- dev_err(&pdev->dev, "Missing pdata\n");
- return -ENODEV;
- }
-
priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL);
if (priv == NULL)
return -ENOMEM;
- if (pdata->card_name) {
- card->name = pdata->card_name;
+ priv->dmic_codec_dev = ERR_PTR(-EINVAL);
+
+ if (node) {
+ struct device_node *dai_node;
+
+ if (snd_soc_of_parse_card_name(card, "ti,model")) {
+ dev_err(&pdev->dev, "Card name is not provided\n");
+ return -ENODEV;
+ }
+
+ ret = snd_soc_of_parse_audio_routing(card,
+ "ti,audio-routing");
+ if (ret) {
+ dev_err(&pdev->dev,
+ "Error while parsing DAPM routing\n");
+ return ret;
+ }
+
+ dai_node = of_parse_phandle(node, "ti,mcpdm", 0);
+ if (!dai_node) {
+ dev_err(&pdev->dev, "McPDM node is not provided\n");
+ return -EINVAL;
+ }
+ abe_twl6040_dai_links[0].cpu_dai_name = NULL;
+ abe_twl6040_dai_links[0].cpu_of_node = dai_node;
+
+ dai_node = of_parse_phandle(node, "ti,dmic", 0);
+ if (dai_node) {
+ num_links = 2;
+ abe_twl6040_dai_links[1].cpu_dai_name = NULL;
+ abe_twl6040_dai_links[1].cpu_of_node = dai_node;
+
+ priv->dmic_codec_dev = platform_device_register_simple(
+ "dmic-codec", -1, NULL, 0);
+ if (IS_ERR(priv->dmic_codec_dev)) {
+ dev_err(&pdev->dev,
+ "Can't instantiate dmic-codec\n");
+ return PTR_ERR(priv->dmic_codec_dev);
+ }
+ } else {
+ num_links = 1;
+ }
+
+ of_property_read_u32(node, "ti,jack-detection",
+ &priv->jack_detection);
+ of_property_read_u32(node, "ti,mclk-freq",
+ &priv->mclk_freq);
+ if (!priv->mclk_freq) {
+ dev_err(&pdev->dev, "MCLK frequency not provided\n");
+ ret = -EINVAL;
+ goto err_unregister;
+ }
+
+ omap_abe_card.fully_routed = 1;
+ } else if (pdata) {
+ if (pdata->card_name) {
+ card->name = pdata->card_name;
+ } else {
+ dev_err(&pdev->dev, "Card name is not provided\n");
+ return -ENODEV;
+ }
+
+ if (pdata->has_dmic)
+ num_links = 2;
+ else
+ num_links = 1;
+
+ priv->jack_detection = pdata->jack_detection;
+ priv->mclk_freq = pdata->mclk_freq;
} else {
- dev_err(&pdev->dev, "Card name is not provided\n");
+ dev_err(&pdev->dev, "Missing pdata\n");
return -ENODEV;
}
- priv->jack_detection = pdata->jack_detection;
- priv->mclk_freq = pdata->mclk_freq;
-
if (!priv->mclk_freq) {
dev_err(&pdev->dev, "MCLK frequency missing\n");
- return -ENODEV;
+ ret = -ENODEV;
+ goto err_unregister;
}
- if (pdata->has_dmic)
- num_links = 2;
- else
- num_links = 1;
-
card->dai_link = abe_twl6040_dai_links;
card->num_links = num_links;
snd_soc_card_set_drvdata(card, priv);
ret = snd_soc_register_card(card);
- if (ret)
+ if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
+ goto err_unregister;
+ }
+
+ return 0;
+
+err_unregister:
+ if (!IS_ERR(priv->dmic_codec_dev))
+ platform_device_unregister(priv->dmic_codec_dev);
return ret;
}
@@ -323,17 +397,28 @@
static int __devexit omap_abe_remove(struct platform_device *pdev)
{
struct snd_soc_card *card = platform_get_drvdata(pdev);
+ struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
snd_soc_unregister_card(card);
+ if (!IS_ERR(priv->dmic_codec_dev))
+ platform_device_unregister(priv->dmic_codec_dev);
+
return 0;
}
+static const struct of_device_id omap_abe_of_match[] = {
+ {.compatible = "ti,abe-twl6040", },
+ { },
+};
+MODULE_DEVICE_TABLE(of, omap_abe_of_match);
+
static struct platform_driver omap_abe_driver = {
.driver = {
.name = "omap-abe-twl6040",
.owner = THIS_MODULE,
.pm = &snd_soc_pm_ops,
+ .of_match_table = omap_abe_of_match,
},
.probe = omap_abe_probe,
.remove = __devexit_p(omap_abe_remove),
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index 2c66e249..f7babb3 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -445,9 +445,8 @@
{
struct omap_mcpdm *mcpdm;
struct resource *res;
- int ret = 0;
- mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL);
+ mcpdm = devm_kzalloc(&pdev->dev, sizeof(struct omap_mcpdm), GFP_KERNEL);
if (!mcpdm)
return -ENOMEM;
@@ -456,55 +455,30 @@
mutex_init(&mcpdm->mutex);
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (res == NULL) {
- dev_err(&pdev->dev, "no resource\n");
- goto err_res;
- }
+ if (res == NULL)
+ return -ENOMEM;
- if (!request_mem_region(res->start, resource_size(res), "McPDM")) {
- ret = -EBUSY;
- goto err_res;
- }
+ if (!devm_request_mem_region(&pdev->dev, res->start,
+ resource_size(res), "McPDM"))
+ return -EBUSY;
- mcpdm->io_base = ioremap(res->start, resource_size(res));
- if (!mcpdm->io_base) {
- ret = -ENOMEM;
- goto err_iomap;
- }
+ mcpdm->io_base = devm_ioremap(&pdev->dev, res->start,
+ resource_size(res));
+ if (!mcpdm->io_base)
+ return -ENOMEM;
mcpdm->irq = platform_get_irq(pdev, 0);
- if (mcpdm->irq < 0) {
- ret = mcpdm->irq;
- goto err_irq;
- }
+ if (mcpdm->irq < 0)
+ return mcpdm->irq;
mcpdm->dev = &pdev->dev;
- ret = snd_soc_register_dai(&pdev->dev, &omap_mcpdm_dai);
- if (!ret)
- return 0;
-
-err_irq:
- iounmap(mcpdm->io_base);
-err_iomap:
- release_mem_region(res->start, resource_size(res));
-err_res:
- kfree(mcpdm);
- return ret;
+ return snd_soc_register_dai(&pdev->dev, &omap_mcpdm_dai);
}
static int __devexit asoc_mcpdm_remove(struct platform_device *pdev)
{
- struct omap_mcpdm *mcpdm = platform_get_drvdata(pdev);
- struct resource *res;
-
snd_soc_unregister_dai(&pdev->dev);
-
- res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- iounmap(mcpdm->io_base);
- release_mem_region(res->start, resource_size(res));
-
- kfree(mcpdm);
return 0;
}
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index fe3995c..e7b8317 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -1,6 +1,6 @@
config SND_SOC_SAMSUNG
tristate "ASoC support for Samsung"
- depends on ARCH_S3C24XX || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 || ARCH_S5P64X0 || ARCH_EXYNOS4
+ depends on PLAT_SAMSUNG
select S3C64XX_DMA if ARCH_S3C64XX
select S3C2410_DMA if ARCH_S3C24XX
help
@@ -191,6 +191,7 @@
select SND_SAMSUNG_I2S
select SND_SOC_WM8996
select SND_SOC_WM9081
+ select SND_SOC_WM0010
select SND_SOC_WM1250_EV1
config SND_SOC_TOBERMORY
@@ -199,6 +200,14 @@
select SND_SAMSUNG_I2S
select SND_SOC_WM8962
+config SND_SOC_BELLS
+ tristate "Audio support for Wolfson Bells"
+ depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410
+ select SND_SAMSUNG_I2S
+ select SND_SOC_WM5102
+ select SND_SOC_WM5110
+ select SND_SOC_WM9081
+
config SND_SOC_LOWLAND
tristate "Audio support for Wolfson Lowland"
depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410
diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile
index 9d03beb..709f605 100644
--- a/sound/soc/samsung/Makefile
+++ b/sound/soc/samsung/Makefile
@@ -42,6 +42,7 @@
snd-soc-tobermory-objs := tobermory.o
snd-soc-lowland-objs := lowland.o
snd-soc-littlemill-objs := littlemill.o
+snd-soc-bells-objs := bells.o
obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o
obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
@@ -65,3 +66,4 @@
obj-$(CONFIG_SND_SOC_TOBERMORY) += snd-soc-tobermory.o
obj-$(CONFIG_SND_SOC_LOWLAND) += snd-soc-lowland.o
obj-$(CONFIG_SND_SOC_LITTLEMILL) += snd-soc-littlemill.o
+obj-$(CONFIG_SND_SOC_BELLS) += snd-soc-bells.o
diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c
new file mode 100644
index 0000000..5dc10df
--- /dev/null
+++ b/sound/soc/samsung/bells.c
@@ -0,0 +1,346 @@
+/*
+ * Bells audio support
+ *
+ * Copyright 2012 Wolfson Microelectronics
+ *
+ * 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.
+ */
+
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/jack.h>
+#include <linux/gpio.h>
+#include <linux/module.h>
+
+#include "../codecs/wm5102.h"
+#include "../codecs/wm9081.h"
+
+/*
+ * 44.1kHz based clocks for the SYSCLK domain, use a very high clock
+ * to allow all the DSP functionality to be enabled if desired.
+ */
+#define SYSCLK_RATE (44100 * 1024)
+
+/* 48kHz based clocks for the ASYNC domain */
+#define ASYNCCLK_RATE (48000 * 512)
+
+/* BCLK2 is fixed at this currently */
+#define BCLK2_RATE (64 * 8000)
+
+/*
+ * Expect a 24.576MHz crystal if one is fitted (the driver will function
+ * if this is not fitted).
+ */
+#define MCLK_RATE 24576000
+
+#define WM9081_AUDIO_RATE 44100
+#define WM9081_MCLK_RATE (WM9081_AUDIO_RATE * 256)
+
+static int bells_set_bias_level(struct snd_soc_card *card,
+ struct snd_soc_dapm_context *dapm,
+ enum snd_soc_bias_level level)
+{
+ struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+ struct snd_soc_codec *codec = codec_dai->codec;
+ int ret;
+
+ if (dapm->dev != codec_dai->dev)
+ return 0;
+
+ switch (level) {
+ case SND_SOC_BIAS_PREPARE:
+ if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
+ ret = snd_soc_codec_set_pll(codec, WM5102_FLL1,
+ ARIZONA_FLL_SRC_MCLK1,
+ MCLK_RATE,
+ SYSCLK_RATE);
+ if (ret < 0)
+ pr_err("Failed to start FLL: %d\n", ret);
+
+ ret = snd_soc_codec_set_pll(codec, WM5102_FLL2,
+ ARIZONA_FLL_SRC_AIF2BCLK,
+ BCLK2_RATE,
+ ASYNCCLK_RATE);
+ if (ret < 0)
+ pr_err("Failed to start FLL: %d\n", ret);
+ }
+ break;
+
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static int bells_set_bias_level_post(struct snd_soc_card *card,
+ struct snd_soc_dapm_context *dapm,
+ enum snd_soc_bias_level level)
+{
+ struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+ struct snd_soc_codec *codec = codec_dai->codec;
+ int ret;
+
+ if (dapm->dev != codec_dai->dev)
+ return 0;
+
+ switch (level) {
+ case SND_SOC_BIAS_STANDBY:
+ ret = snd_soc_codec_set_pll(codec, WM5102_FLL1, 0, 0, 0);
+ if (ret < 0) {
+ pr_err("Failed to stop FLL: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_codec_set_pll(codec, WM5102_FLL2, 0, 0, 0);
+ if (ret < 0) {
+ pr_err("Failed to stop FLL: %d\n", ret);
+ return ret;
+ }
+ break;
+
+ default:
+ break;
+ }
+
+ dapm->bias_level = level;
+
+ return 0;
+}
+
+static int bells_late_probe(struct snd_soc_card *card)
+{
+ struct snd_soc_codec *codec = card->rtd[0].codec;
+ struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai;
+ struct snd_soc_dai *aif2_dai = card->rtd[1].cpu_dai;
+ struct snd_soc_dai *aif3_dai = card->rtd[2].cpu_dai;
+ struct snd_soc_dai *wm9081_dai = card->rtd[2].codec_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_sysclk(aif1_dai, ARIZONA_CLK_SYSCLK, 0, 0);
+ if (ret != 0) {
+ dev_err(aif1_dai->dev, "Failed to set AIF1 clock: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_sysclk(aif2_dai, ARIZONA_CLK_ASYNCCLK, 0, 0);
+ if (ret != 0) {
+ dev_err(aif2_dai->dev, "Failed to set AIF2 clock: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_sysclk(aif3_dai, ARIZONA_CLK_SYSCLK, 0, 0);
+ if (ret != 0) {
+ dev_err(aif1_dai->dev, "Failed to set AIF1 clock: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_SYSCLK,
+ ARIZONA_CLK_SRC_FLL1, SYSCLK_RATE,
+ SND_SOC_CLOCK_IN);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to set SYSCLK: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_OPCLK, 0,
+ WM9081_MCLK_RATE, SND_SOC_CLOCK_OUT);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to set OPCLK: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_ASYNCCLK,
+ ARIZONA_CLK_SRC_FLL2, ASYNCCLK_RATE,
+ SND_SOC_CLOCK_IN);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to set SYSCLK: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_codec_set_sysclk(wm9081_dai->codec, WM9081_SYSCLK_MCLK,
+ 0, WM9081_MCLK_RATE, 0);
+ if (ret != 0) {
+ dev_err(wm9081_dai->dev, "Failed to set MCLK: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_pcm_stream baseband_params = {
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+ .rate_min = 8000,
+ .rate_max = 8000,
+ .channels_min = 2,
+ .channels_max = 2,
+};
+
+static const struct snd_soc_pcm_stream sub_params = {
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+ .rate_min = WM9081_AUDIO_RATE,
+ .rate_max = WM9081_AUDIO_RATE,
+ .channels_min = 2,
+ .channels_max = 2,
+};
+
+static struct snd_soc_dai_link bells_dai_wm5102[] = {
+ {
+ .name = "CPU",
+ .stream_name = "CPU",
+ .cpu_dai_name = "samsung-i2s.0",
+ .codec_dai_name = "wm5102-aif1",
+ .platform_name = "samsung-audio",
+ .codec_name = "wm5102-codec",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBM_CFM,
+ },
+ {
+ .name = "Baseband",
+ .stream_name = "Baseband",
+ .cpu_dai_name = "wm5102-aif2",
+ .codec_dai_name = "wm1250-ev1",
+ .codec_name = "wm1250-ev1.1-0027",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBM_CFM,
+ .ignore_suspend = 1,
+ .params = &baseband_params,
+ },
+ {
+ .name = "Sub",
+ .stream_name = "Sub",
+ .cpu_dai_name = "wm5102-aif3",
+ .codec_dai_name = "wm9081-hifi",
+ .codec_name = "wm9081.1-006c",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBS_CFS,
+ .ignore_suspend = 1,
+ .params = &sub_params,
+ },
+};
+
+static struct snd_soc_dai_link bells_dai_wm5110[] = {
+ {
+ .name = "CPU",
+ .stream_name = "CPU",
+ .cpu_dai_name = "samsung-i2s.0",
+ .codec_dai_name = "wm5110-aif1",
+ .platform_name = "samsung-audio",
+ .codec_name = "wm5110-codec",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBM_CFM,
+ },
+ {
+ .name = "Baseband",
+ .stream_name = "Baseband",
+ .cpu_dai_name = "wm5110-aif2",
+ .codec_dai_name = "wm1250-ev1",
+ .codec_name = "wm1250-ev1.1-0027",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBM_CFM,
+ .ignore_suspend = 1,
+ .params = &baseband_params,
+ },
+ {
+ .name = "Sub",
+ .stream_name = "Sub",
+ .cpu_dai_name = "wm5102-aif3",
+ .codec_dai_name = "wm9081-hifi",
+ .codec_name = "wm9081.1-006c",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBS_CFS,
+ .ignore_suspend = 1,
+ .params = &sub_params,
+ },
+};
+
+static struct snd_soc_codec_conf bells_codec_conf[] = {
+ {
+ .dev_name = "wm9081.1-006c",
+ .name_prefix = "Sub",
+ },
+};
+
+static struct snd_soc_dapm_route bells_routes[] = {
+ { "Sub CLK_SYS", NULL, "OPCLK" },
+};
+
+static struct snd_soc_card bells_cards[] = {
+ {
+ .name = "Bells WM5102",
+ .owner = THIS_MODULE,
+ .dai_link = bells_dai_wm5102,
+ .num_links = ARRAY_SIZE(bells_dai_wm5102),
+ .codec_conf = bells_codec_conf,
+ .num_configs = ARRAY_SIZE(bells_codec_conf),
+
+ .late_probe = bells_late_probe,
+
+ .dapm_routes = bells_routes,
+ .num_dapm_routes = ARRAY_SIZE(bells_routes),
+
+ .set_bias_level = bells_set_bias_level,
+ .set_bias_level_post = bells_set_bias_level_post,
+ },
+ {
+ .name = "Bells WM5110",
+ .owner = THIS_MODULE,
+ .dai_link = bells_dai_wm5110,
+ .num_links = ARRAY_SIZE(bells_dai_wm5110),
+ .codec_conf = bells_codec_conf,
+ .num_configs = ARRAY_SIZE(bells_codec_conf),
+
+ .late_probe = bells_late_probe,
+
+ .dapm_routes = bells_routes,
+ .num_dapm_routes = ARRAY_SIZE(bells_routes),
+
+ .set_bias_level = bells_set_bias_level,
+ .set_bias_level_post = bells_set_bias_level_post,
+ },
+};
+
+
+static __devinit int bells_probe(struct platform_device *pdev)
+{
+ int ret;
+
+ bells_cards[pdev->id].dev = &pdev->dev;
+
+ ret = snd_soc_register_card(&bells_cards[pdev->id]);
+ if (ret) {
+ dev_err(&pdev->dev,
+ "snd_soc_register_card(%s) failed: %d\n",
+ bells_cards[pdev->id].name, ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int __devexit bells_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_card(&bells_cards[pdev->id]);
+
+ return 0;
+}
+
+static struct platform_driver bells_driver = {
+ .driver = {
+ .name = "bells",
+ .owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = bells_probe,
+ .remove = __devexit_p(bells_remove),
+};
+
+module_platform_driver(bells_driver);
+
+MODULE_DESCRIPTION("Bells audio support");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:bells");
diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c
index a4a9fc7..c7e1c28 100644
--- a/sound/soc/samsung/speyside.c
+++ b/sound/soc/samsung/speyside.c
@@ -25,7 +25,7 @@
struct snd_soc_dapm_context *dapm,
enum snd_soc_bias_level level)
{
- struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+ struct snd_soc_dai *codec_dai = card->rtd[1].codec_dai;
int ret;
if (dapm->dev != codec_dai->dev)
@@ -57,7 +57,7 @@
struct snd_soc_dapm_context *dapm,
enum snd_soc_bias_level level)
{
- struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+ struct snd_soc_dai *codec_dai = card->rtd[1].codec_dai;
int ret;
if (dapm->dev != codec_dai->dev)
@@ -126,6 +126,18 @@
snd_soc_dapm_sync(&codec->dapm);
}
+static int speyside_wm0010_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_dai *dai = rtd->codec_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_sysclk(dai, 0, MCLK_AUDIO_RATE, 0);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
static int speyside_wm8996_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_dai *dai = rtd->codec_dai;
@@ -172,17 +184,37 @@
return 0;
}
+static const struct snd_soc_pcm_stream dsp_codec_params = {
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+};
+
static struct snd_soc_dai_link speyside_dai[] = {
{
- .name = "CPU",
- .stream_name = "CPU",
+ .name = "CPU-DSP",
+ .stream_name = "CPU-DSP",
.cpu_dai_name = "samsung-i2s.0",
- .codec_dai_name = "wm8996-aif1",
+ .codec_dai_name = "wm0010-sdi1",
.platform_name = "samsung-audio",
+ .codec_name = "spi0.0",
+ .init = speyside_wm0010_init,
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBM_CFM,
+ },
+ {
+ .name = "DSP-CODEC",
+ .stream_name = "DSP-CODEC",
+ .cpu_dai_name = "wm0010-sdi2",
+ .codec_dai_name = "wm8996-aif1",
.codec_name = "wm8996.1-001a",
.init = speyside_wm8996_init,
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBM_CFM,
+ .params = &dsp_codec_params,
+ .ignore_suspend = 1,
},
{
.name = "Baseband",
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
new file mode 100644
index 0000000..967d0e1
--- /dev/null
+++ b/sound/soc/soc-compress.c
@@ -0,0 +1,294 @@
+/*
+ * soc-compress.c -- ALSA SoC Compress
+ *
+ * Copyright (C) 2012 Intel Corp.
+ *
+ * Authors: Namarta Kohli <namartax.kohli@intel.com>
+ * Ramesh Babu K V <ramesh.babu@linux.intel.com>
+ * Vinod Koul <vinod.koul@linux.intel.com>
+ *
+ * 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.
+ *
+ */
+
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/workqueue.h>
+#include <sound/core.h>
+#include <sound/compress_params.h>
+#include <sound/compress_driver.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+
+static int soc_compr_open(struct snd_compr_stream *cstream)
+{
+ struct snd_soc_pcm_runtime *rtd = cstream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret = 0;
+
+ if (platform->driver->compr_ops && platform->driver->compr_ops->open) {
+ ret = platform->driver->compr_ops->open(cstream);
+ if (ret < 0) {
+ pr_err("compress asoc: can't open platform %s\n", platform->name);
+ goto out;
+ }
+ }
+
+ if (rtd->dai_link->compr_ops && rtd->dai_link->compr_ops->startup) {
+ ret = rtd->dai_link->compr_ops->startup(cstream);
+ if (ret < 0) {
+ pr_err("compress asoc: %s startup failed\n", rtd->dai_link->name);
+ goto machine_err;
+ }
+ }
+
+ if (cstream->direction == SND_COMPRESS_PLAYBACK) {
+ cpu_dai->playback_active++;
+ codec_dai->playback_active++;
+ } else {
+ cpu_dai->capture_active++;
+ codec_dai->capture_active++;
+ }
+
+ cpu_dai->active++;
+ codec_dai->active++;
+ rtd->codec->active++;
+
+ return 0;
+
+machine_err:
+ if (platform->driver->compr_ops && platform->driver->compr_ops->free)
+ platform->driver->compr_ops->free(cstream);
+out:
+ return ret;
+}
+
+static int soc_compr_free(struct snd_compr_stream *cstream)
+{
+ struct snd_soc_pcm_runtime *rtd = cstream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_codec *codec = rtd->codec;
+
+ if (cstream->direction == SND_COMPRESS_PLAYBACK) {
+ cpu_dai->playback_active--;
+ codec_dai->playback_active--;
+ } else {
+ cpu_dai->capture_active--;
+ codec_dai->capture_active--;
+ }
+
+ snd_soc_dai_digital_mute(codec_dai, 1);
+
+ cpu_dai->active--;
+ codec_dai->active--;
+ codec->active--;
+
+ if (!cpu_dai->active)
+ cpu_dai->rate = 0;
+
+ if (!codec_dai->active)
+ codec_dai->rate = 0;
+
+
+ if (rtd->dai_link->compr_ops && rtd->dai_link->compr_ops->shutdown)
+ rtd->dai_link->compr_ops->shutdown(cstream);
+
+ if (platform->driver->compr_ops && platform->driver->compr_ops->free)
+ platform->driver->compr_ops->free(cstream);
+ cpu_dai->runtime = NULL;
+
+ if (cstream->direction == SND_COMPRESS_PLAYBACK) {
+ if (!rtd->pmdown_time || codec->ignore_pmdown_time ||
+ rtd->dai_link->ignore_pmdown_time) {
+ snd_soc_dapm_stream_event(rtd,
+ SNDRV_PCM_STREAM_PLAYBACK,
+ SND_SOC_DAPM_STREAM_STOP);
+ } else
+ codec_dai->pop_wait = 1;
+ schedule_delayed_work(&rtd->delayed_work,
+ msecs_to_jiffies(rtd->pmdown_time));
+ } else {
+ /* capture streams can be powered down now */
+ snd_soc_dapm_stream_event(rtd,
+ SNDRV_PCM_STREAM_CAPTURE,
+ SND_SOC_DAPM_STREAM_STOP);
+ }
+
+ return 0;
+}
+
+static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd)
+{
+
+ struct snd_soc_pcm_runtime *rtd = cstream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret = 0;
+
+ if (platform->driver->compr_ops && platform->driver->compr_ops->trigger) {
+ ret = platform->driver->compr_ops->trigger(cstream, cmd);
+ if (ret < 0)
+ return ret;
+ }
+
+ if (cmd == SNDRV_PCM_TRIGGER_START)
+ snd_soc_dai_digital_mute(codec_dai, 0);
+ else if (cmd == SNDRV_PCM_TRIGGER_STOP)
+ snd_soc_dai_digital_mute(codec_dai, 1);
+
+ return ret;
+}
+
+static int soc_compr_set_params(struct snd_compr_stream *cstream,
+ struct snd_compr_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = cstream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ int ret = 0;
+
+ /* first we call set_params for the platform driver
+ * this should configure the soc side
+ * if the machine has compressed ops then we call that as well
+ * expectation is that platform and machine will configure everything
+ * for this compress path, like configuring pcm port for codec
+ */
+ if (platform->driver->compr_ops && platform->driver->compr_ops->set_params) {
+ ret = platform->driver->compr_ops->set_params(cstream, params);
+ if (ret < 0)
+ return ret;
+ }
+
+ if (rtd->dai_link->compr_ops && rtd->dai_link->compr_ops->set_params) {
+ ret = rtd->dai_link->compr_ops->set_params(cstream);
+ if (ret < 0)
+ return ret;
+ }
+
+ snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
+ SND_SOC_DAPM_STREAM_START);
+
+ return ret;
+}
+
+static int soc_compr_get_params(struct snd_compr_stream *cstream,
+ struct snd_codec *params)
+{
+ struct snd_soc_pcm_runtime *rtd = cstream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ int ret = 0;
+
+ if (platform->driver->compr_ops && platform->driver->compr_ops->get_params)
+ ret = platform->driver->compr_ops->get_params(cstream, params);
+
+ return ret;
+}
+
+static int soc_compr_get_caps(struct snd_compr_stream *cstream,
+ struct snd_compr_caps *caps)
+{
+ struct snd_soc_pcm_runtime *rtd = cstream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ int ret = 0;
+
+ if (platform->driver->compr_ops && platform->driver->compr_ops->get_caps)
+ ret = platform->driver->compr_ops->get_caps(cstream, caps);
+
+ return ret;
+}
+
+static int soc_compr_get_codec_caps(struct snd_compr_stream *cstream,
+ struct snd_compr_codec_caps *codec)
+{
+ struct snd_soc_pcm_runtime *rtd = cstream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ int ret = 0;
+
+ if (platform->driver->compr_ops && platform->driver->compr_ops->get_codec_caps)
+ ret = platform->driver->compr_ops->get_codec_caps(cstream, codec);
+
+ return ret;
+}
+
+static int soc_compr_ack(struct snd_compr_stream *cstream, size_t bytes)
+{
+ struct snd_soc_pcm_runtime *rtd = cstream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ int ret = 0;
+
+ if (platform->driver->compr_ops && platform->driver->compr_ops->ack)
+ ret = platform->driver->compr_ops->ack(cstream, bytes);
+
+ return ret;
+}
+
+static int soc_compr_pointer(struct snd_compr_stream *cstream,
+ struct snd_compr_tstamp *tstamp)
+{
+ struct snd_soc_pcm_runtime *rtd = cstream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+
+ if (platform->driver->compr_ops && platform->driver->compr_ops->pointer)
+ platform->driver->compr_ops->pointer(cstream, tstamp);
+
+ return 0;
+}
+
+/* ASoC Compress operations */
+static struct snd_compr_ops soc_compr_ops = {
+ .open = soc_compr_open,
+ .free = soc_compr_free,
+ .set_params = soc_compr_set_params,
+ .get_params = soc_compr_get_params,
+ .trigger = soc_compr_trigger,
+ .pointer = soc_compr_pointer,
+ .ack = soc_compr_ack,
+ .get_caps = soc_compr_get_caps,
+ .get_codec_caps = soc_compr_get_codec_caps
+};
+
+/* create a new compress */
+int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_compr *compr;
+ char new_name[64];
+ int ret = 0, direction = 0;
+
+ /* check client and interface hw capabilities */
+ snprintf(new_name, sizeof(new_name), "%s %s-%d",
+ rtd->dai_link->stream_name, codec_dai->name, num);
+ direction = SND_COMPRESS_PLAYBACK;
+ compr = kzalloc(sizeof(*compr), GFP_KERNEL);
+ if (compr == NULL) {
+ snd_printk(KERN_ERR "Cannot allocate compr\n");
+ return -ENOMEM;
+ }
+
+ compr->ops = &soc_compr_ops;
+ mutex_init(&compr->lock);
+ ret = snd_compress_new(rtd->card->snd_card, num, direction, compr);
+ if (ret < 0) {
+ pr_err("compress asoc: can't create compress for codec %s\n",
+ codec->name);
+ kfree(compr);
+ return ret;
+ }
+
+ rtd->compr = compr;
+ compr->private_data = rtd;
+
+ printk(KERN_INFO "compress asoc: %s <-> %s mapping ok\n", codec_dai->name,
+ cpu_dai->name);
+ return ret;
+}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index c501af6..b95d1fb 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1388,37 +1388,48 @@
if (ret < 0)
pr_warn("asoc: failed to add pmdown_time sysfs:%d\n", ret);
- if (!dai_link->params) {
- /* create the pcm */
- ret = soc_new_pcm(rtd, num);
+ if (cpu_dai->driver->compress_dai) {
+ /*create compress_device"*/
+ ret = soc_new_compress(rtd, num);
if (ret < 0) {
- pr_err("asoc: can't create pcm %s :%d\n",
- dai_link->stream_name, ret);
+ pr_err("asoc: can't create compress %s\n",
+ dai_link->stream_name);
return ret;
}
} else {
- /* link the DAI widgets */
- play_w = codec_dai->playback_widget;
- capture_w = cpu_dai->capture_widget;
- if (play_w && capture_w) {
- ret = snd_soc_dapm_new_pcm(card, dai_link->params,
- capture_w, play_w);
- if (ret != 0) {
- dev_err(card->dev, "Can't link %s to %s: %d\n",
- play_w->name, capture_w->name, ret);
+
+ if (!dai_link->params) {
+ /* create the pcm */
+ ret = soc_new_pcm(rtd, num);
+ if (ret < 0) {
+ pr_err("asoc: can't create pcm %s :%d\n",
+ dai_link->stream_name, ret);
return ret;
}
- }
-
- play_w = cpu_dai->playback_widget;
- capture_w = codec_dai->capture_widget;
- if (play_w && capture_w) {
- ret = snd_soc_dapm_new_pcm(card, dai_link->params,
+ } else {
+ /* link the DAI widgets */
+ play_w = codec_dai->playback_widget;
+ capture_w = cpu_dai->capture_widget;
+ if (play_w && capture_w) {
+ ret = snd_soc_dapm_new_pcm(card, dai_link->params,
capture_w, play_w);
- if (ret != 0) {
- dev_err(card->dev, "Can't link %s to %s: %d\n",
- play_w->name, capture_w->name, ret);
- return ret;
+ if (ret != 0) {
+ dev_err(card->dev, "Can't link %s to %s: %d\n",
+ play_w->name, capture_w->name, ret);
+ return ret;
+ }
+ }
+
+ play_w = cpu_dai->playback_widget;
+ capture_w = codec_dai->capture_widget;
+ if (play_w && capture_w) {
+ ret = snd_soc_dapm_new_pcm(card, dai_link->params,
+ capture_w, play_w);
+ if (ret != 0) {
+ dev_err(card->dev, "Can't link %s to %s: %d\n",
+ play_w->name, capture_w->name, ret);
+ return ret;
+ }
}
}
}
@@ -1816,7 +1827,6 @@
static int soc_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = platform_get_drvdata(pdev);
- int ret = 0;
/*
* no card, so machine driver should be registering card
@@ -1832,13 +1842,7 @@
/* Bodge while we unpick instantiation */
card->dev = &pdev->dev;
- ret = snd_soc_register_card(card);
- if (ret != 0) {
- dev_err(&pdev->dev, "Failed to register card\n");
- return ret;
- }
-
- return 0;
+ return snd_soc_register_card(card);
}
static int soc_cleanup_card_resources(struct snd_soc_card *card)
@@ -3717,6 +3721,9 @@
}
}
+ if (!dai->codec)
+ dai->dapm.idle_bias_off = 1;
+
list_add(&dai->list, &dai_list);
mutex_unlock(&client_mutex);
@@ -3805,6 +3812,9 @@
}
}
+ if (!dai->codec)
+ dai->dapm.idle_bias_off = 1;
+
list_add(&dai->list, &dai_list);
mutex_unlock(&client_mutex);
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 0c17293..fa0fd8d 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -83,11 +83,6 @@
jack->status &= ~mask;
jack->status |= status & mask;
- /* The DAPM sync is expensive enough to be worth skipping.
- * However, empty mask means pin synchronization is desired. */
- if (mask && (jack->status == oldstatus))
- goto out;
-
trace_snd_soc_jack_notify(jack, status);
list_for_each_entry(pin, &jack->pins, list) {
@@ -109,7 +104,6 @@
snd_jack_report(jack->jack, jack->status);
-out:
mutex_unlock(&jack->mutex);
}
EXPORT_SYMBOL_GPL(snd_soc_jack_report);
diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c
index 057e28e..772cb19 100644
--- a/sound/soc/ux500/ux500_msp_dai.c
+++ b/sound/soc/ux500/ux500_msp_dai.c
@@ -760,6 +760,9 @@
drvdata = devm_kzalloc(&pdev->dev,
sizeof(struct ux500_msp_i2s_drvdata),
GFP_KERNEL);
+ if (!drvdata)
+ return -ENOMEM;
+
drvdata->fmt = 0;
drvdata->slots = 1;
drvdata->tx_mask = 0x01;
diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c
index 5c472f3..36be11e 100644
--- a/sound/soc/ux500/ux500_msp_i2s.c
+++ b/sound/soc/ux500/ux500_msp_i2s.c
@@ -673,6 +673,8 @@
*msp_p = devm_kzalloc(&pdev->dev, sizeof(struct ux500_msp), GFP_KERNEL);
msp = *msp_p;
+ if (!msp)
+ return -ENOMEM;
msp->id = platform_data->id;
msp->dev = &pdev->dev;