Merge branch 'topic/asoc' into to-push
diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt
index f370e7d..bab7711 100644
--- a/Documentation/sound/alsa/soc/machine.txt
+++ b/Documentation/sound/alsa/soc/machine.txt
@@ -9,7 +9,7 @@
the following struct:-
/* SoC machine */
-struct snd_soc_machine {
+struct snd_soc_card {
char *name;
int (*probe)(struct platform_device *pdev);
@@ -67,10 +67,10 @@
.ops = &corgi_ops,
};
-struct snd_soc_machine then sets up the machine with it's DAIs. e.g.
+struct snd_soc_card then sets up the machine with it's DAIs. e.g.
/* corgi audio machine driver */
-static struct snd_soc_machine snd_soc_machine_corgi = {
+static struct snd_soc_card snd_soc_corgi = {
.name = "Corgi",
.dai_link = &corgi_dai,
.num_links = 1,
@@ -90,7 +90,7 @@
/* corgi audio subsystem */
static struct snd_soc_device corgi_snd_devdata = {
- .machine = &snd_soc_machine_corgi,
+ .machine = &snd_soc_corgi,
.platform = &pxa2xx_soc_platform,
.codec_dev = &soc_codec_dev_wm8731,
.codec_data = &corgi_wm8731_setup,
diff --git a/MAINTAINERS b/MAINTAINERS
index fbc8fa5..7010257 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -3977,7 +3977,7 @@
L: alsa-devel@alsa-project.org (subscribers-only)
S: Maintained
-SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT
+SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC)
P: Liam Girdwood
M: lrg@slimlogic.co.uk
P: Mark Brown
diff --git a/arch/arm/mach-pxa/include/mach/palmasoc.h b/arch/arm/mach-pxa/include/mach/palmasoc.h
new file mode 100644
index 0000000..6c4b1f7
--- /dev/null
+++ b/arch/arm/mach-pxa/include/mach/palmasoc.h
@@ -0,0 +1,13 @@
+#ifndef _INCLUDE_PALMASOC_H_
+#define _INCLUDE_PALMASOC_H_
+struct palm27x_asoc_info {
+ int jack_gpio;
+};
+
+#ifdef CONFIG_SND_PXA2XX_SOC_PALM27X
+void __init palm27x_asoc_set_pdata(struct palm27x_asoc_info *data);
+#else
+static inline void palm27x_asoc_set_pdata(struct palm27x_asoc_info *data) {}
+#endif
+
+#endif
diff --git a/include/linux/mfd/wm8350/audio.h b/include/linux/mfd/wm8350/audio.h
index 217bb22..af95a1d 100644
--- a/include/linux/mfd/wm8350/audio.h
+++ b/include/linux/mfd/wm8350/audio.h
@@ -1,7 +1,7 @@
/*
* audio.h -- Audio Driver for Wolfson WM8350 PMIC
*
- * Copyright 2007 Wolfson Microelectronics PLC
+ * Copyright 2007, 2008 Wolfson Microelectronics PLC
*
* 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
@@ -70,9 +70,9 @@
#define WM8350_CODEC_ISEL_0_5 3 /* x0.5 */
#define WM8350_VMID_OFF 0
-#define WM8350_VMID_500K 1
-#define WM8350_VMID_100K 2
-#define WM8350_VMID_10K 3
+#define WM8350_VMID_300K 1
+#define WM8350_VMID_50K 2
+#define WM8350_VMID_5K 3
/*
* R40 (0x28) - Clock Control 1
@@ -591,8 +591,38 @@
#define WM8350_IRQ_CODEC_MICSCD 41
#define WM8350_IRQ_CODEC_MICD 42
+/*
+ * WM8350 Platform data.
+ *
+ * This must be initialised per platform for best audio performance.
+ * Please see WM8350 datasheet for information.
+ */
+struct wm8350_audio_platform_data {
+ int vmid_discharge_msecs; /* VMID --> OFF discharge time */
+ int drain_msecs; /* OFF drain time */
+ int cap_discharge_msecs; /* Cap ON (from OFF) discharge time */
+ int vmid_charge_msecs; /* vmid power up time */
+ u32 vmid_s_curve:2; /* vmid enable s curve speed */
+ u32 dis_out4:2; /* out4 discharge speed */
+ u32 dis_out3:2; /* out3 discharge speed */
+ u32 dis_out2:2; /* out2 discharge speed */
+ u32 dis_out1:2; /* out1 discharge speed */
+ u32 vroi_out4:1; /* out4 tie off */
+ u32 vroi_out3:1; /* out3 tie off */
+ u32 vroi_out2:1; /* out2 tie off */
+ u32 vroi_out1:1; /* out1 tie off */
+ u32 vroi_enable:1; /* enable tie off */
+ u32 codec_current_on:2; /* current level ON */
+ u32 codec_current_standby:2; /* current level STANDBY */
+ u32 codec_current_charge:2; /* codec current @ vmid charge */
+};
+
+struct snd_soc_codec;
+
struct wm8350_codec {
struct platform_device *pdev;
+ struct snd_soc_codec *codec;
+ struct wm8350_audio_platform_data *platform_data;
};
#endif
diff --git a/include/sound/l3.h b/include/sound/l3.h
new file mode 100644
index 0000000..423a08f
--- /dev/null
+++ b/include/sound/l3.h
@@ -0,0 +1,18 @@
+#ifndef _L3_H_
+#define _L3_H_ 1
+
+struct l3_pins {
+ void (*setdat)(int);
+ void (*setclk)(int);
+ void (*setmode)(int);
+ int data_hold;
+ int data_setup;
+ int clock_high;
+ int mode_hold;
+ int mode;
+ int mode_setup;
+};
+
+int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len);
+
+#endif
diff --git a/include/sound/s3c24xx_uda134x.h b/include/sound/s3c24xx_uda134x.h
new file mode 100644
index 0000000..33df4cb
--- /dev/null
+++ b/include/sound/s3c24xx_uda134x.h
@@ -0,0 +1,14 @@
+#ifndef _S3C24XX_UDA134X_H_
+#define _S3C24XX_UDA134X_H_ 1
+
+#include <sound/uda134x.h>
+
+struct s3c24xx_uda134x_platform_data {
+ int l3_clk;
+ int l3_mode;
+ int l3_data;
+ void (*power) (int);
+ int model;
+};
+
+#endif
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
new file mode 100644
index 0000000..24247f7
--- /dev/null
+++ b/include/sound/soc-dai.h
@@ -0,0 +1,231 @@
+/*
+ * linux/sound/soc-dai.h -- ALSA SoC Layer
+ *
+ * Copyright: 2005-2008 Wolfson Microelectronics. PLC.
+ *
+ * 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.
+ *
+ * Digital Audio Interface (DAI) API.
+ */
+
+#ifndef __LINUX_SND_SOC_DAI_H
+#define __LINUX_SND_SOC_DAI_H
+
+
+#include <linux/list.h>
+
+struct snd_pcm_substream;
+
+/*
+ * DAI hardware audio formats.
+ *
+ * Describes the physical PCM data formating and clocking. Add new formats
+ * to the end.
+ */
+#define SND_SOC_DAIFMT_I2S 0 /* I2S mode */
+#define SND_SOC_DAIFMT_RIGHT_J 1 /* Right Justified mode */
+#define SND_SOC_DAIFMT_LEFT_J 2 /* Left Justified mode */
+#define SND_SOC_DAIFMT_DSP_A 3 /* L data msb after FRM LRC */
+#define SND_SOC_DAIFMT_DSP_B 4 /* L data msb during FRM LRC */
+#define SND_SOC_DAIFMT_AC97 5 /* AC97 */
+
+/* left and right justified also known as MSB and LSB respectively */
+#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J
+#define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J
+
+/*
+ * DAI Clock gating.
+ *
+ * DAI bit clocks can be be gated (disabled) when not the DAI is not
+ * sending or receiving PCM data in a frame. This can be used to save power.
+ */
+#define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */
+#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated */
+
+/*
+ * DAI Left/Right Clocks.
+ *
+ * Specifies whether the DAI can support different samples for similtanious
+ * playback and capture. This usually requires a seperate physical frame
+ * clock for playback and capture.
+ */
+#define SND_SOC_DAIFMT_SYNC (0 << 5) /* Tx FRM = Rx FRM */
+#define SND_SOC_DAIFMT_ASYNC (1 << 5) /* Tx FRM ~ Rx FRM */
+
+/*
+ * TDM
+ *
+ * Time Division Multiplexing. Allows PCM data to be multplexed with other
+ * data on the DAI.
+ */
+#define SND_SOC_DAIFMT_TDM (1 << 6)
+
+/*
+ * DAI hardware signal inversions.
+ *
+ * Specifies whether the DAI can also support inverted clocks for the specified
+ * format.
+ */
+#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
+#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal bclk + inv frm */
+#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert bclk + nor frm */
+#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert bclk + frm */
+
+/*
+ * DAI hardware clock masters.
+ *
+ * This is wrt the codec, the inverse is true for the interface
+ * i.e. if the codec is clk and frm master then the interface is
+ * clk and frame slave.
+ */
+#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & frm master */
+#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & frm master */
+#define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */
+#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & frm slave */
+
+#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f
+#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0
+#define SND_SOC_DAIFMT_INV_MASK 0x0f00
+#define SND_SOC_DAIFMT_MASTER_MASK 0xf000
+
+/*
+ * Master Clock Directions
+ */
+#define SND_SOC_CLOCK_IN 0
+#define SND_SOC_CLOCK_OUT 1
+
+struct snd_soc_dai_ops;
+struct snd_soc_dai;
+struct snd_ac97_bus_ops;
+
+/* Digital Audio Interface registration */
+int snd_soc_register_dai(struct snd_soc_dai *dai);
+void snd_soc_unregister_dai(struct snd_soc_dai *dai);
+int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count);
+void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count);
+
+/* Digital Audio Interface clocking API.*/
+int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+ unsigned int freq, int dir);
+
+int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
+ int div_id, int div);
+
+int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
+ int pll_id, unsigned int freq_in, unsigned int freq_out);
+
+/* Digital Audio interface formatting */
+int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
+
+int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
+ unsigned int mask, int slots);
+
+int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
+
+/* Digital Audio Interface mute */
+int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute);
+
+/*
+ * Digital Audio Interface.
+ *
+ * Describes the Digital Audio Interface in terms of it's ALSA, DAI and AC97
+ * operations an capabilities. Codec and platfom drivers will register a this
+ * structure for every DAI they have.
+ *
+ * This structure covers the clocking, formating and ALSA operations for each
+ * interface a
+ */
+struct snd_soc_dai_ops {
+ /*
+ * DAI clocking configuration, all optional.
+ * Called by soc_card drivers, normally in their hw_params.
+ */
+ int (*set_sysclk)(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir);
+ int (*set_pll)(struct snd_soc_dai *dai,
+ int pll_id, unsigned int freq_in, unsigned int freq_out);
+ int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
+
+ /*
+ * DAI format configuration
+ * Called by soc_card drivers, normally in their hw_params.
+ */
+ int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
+ int (*set_tdm_slot)(struct snd_soc_dai *dai,
+ unsigned int mask, int slots);
+ int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
+
+ /*
+ * DAI digital mute - optional.
+ * Called by soc-core to minimise any pops.
+ */
+ int (*digital_mute)(struct snd_soc_dai *dai, int mute);
+
+ /*
+ * ALSA PCM audio operations - all optional.
+ * Called by soc-core during audio PCM operations.
+ */
+ int (*startup)(struct snd_pcm_substream *,
+ struct snd_soc_dai *);
+ void (*shutdown)(struct snd_pcm_substream *,
+ struct snd_soc_dai *);
+ int (*hw_params)(struct snd_pcm_substream *,
+ struct snd_pcm_hw_params *, struct snd_soc_dai *);
+ int (*hw_free)(struct snd_pcm_substream *,
+ struct snd_soc_dai *);
+ int (*prepare)(struct snd_pcm_substream *,
+ struct snd_soc_dai *);
+ int (*trigger)(struct snd_pcm_substream *, int,
+ struct snd_soc_dai *);
+};
+
+/*
+ * Digital Audio Interface runtime data.
+ *
+ * Holds runtime data for a DAI.
+ */
+struct snd_soc_dai {
+ /* DAI description */
+ char *name;
+ unsigned int id;
+ int ac97_control;
+
+ struct device *dev;
+
+ /* DAI callbacks */
+ int (*probe)(struct platform_device *pdev,
+ struct snd_soc_dai *dai);
+ void (*remove)(struct platform_device *pdev,
+ struct snd_soc_dai *dai);
+ int (*suspend)(struct snd_soc_dai *dai);
+ int (*resume)(struct snd_soc_dai *dai);
+
+ /* ops */
+ struct snd_soc_dai_ops ops;
+
+ /* DAI capabilities */
+ struct snd_soc_pcm_stream capture;
+ struct snd_soc_pcm_stream playback;
+
+ /* DAI runtime info */
+ struct snd_pcm_runtime *runtime;
+ struct snd_soc_codec *codec;
+ unsigned int active;
+ unsigned char pop_wait:1;
+ void *dma_data;
+
+ /* DAI private data */
+ void *private_data;
+
+ /* parent codec/platform */
+ union {
+ struct snd_soc_codec *codec;
+ struct snd_soc_platform *platform;
+ };
+
+ struct list_head list;
+};
+
+#endif
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index ca699a3..7ee2f70 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -221,8 +221,6 @@
int num);
/* dapm path setup */
-int __deprecated snd_soc_dapm_connect_input(struct snd_soc_codec *codec,
- const char *sink_name, const char *control_name, const char *src_name);
int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec);
void snd_soc_dapm_free(struct snd_soc_device *socdev);
int snd_soc_dapm_add_routes(struct snd_soc_codec *codec,
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 5e01898..f86e455 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -21,8 +21,6 @@
#include <sound/control.h>
#include <sound/ac97_codec.h>
-#define SND_SOC_VERSION "0.13.2"
-
/*
* Convenience kcontrol builders
*/
@@ -145,105 +143,31 @@
SND_SOC_BIAS_OFF,
};
-/*
- * Digital Audio Interface (DAI) types
- */
-#define SND_SOC_DAI_AC97 0x1
-#define SND_SOC_DAI_I2S 0x2
-#define SND_SOC_DAI_PCM 0x4
-#define SND_SOC_DAI_AC97_BUS 0x8 /* for custom i.e. non ac97_codec.c */
-
-/*
- * DAI hardware audio formats
- */
-#define SND_SOC_DAIFMT_I2S 0 /* I2S mode */
-#define SND_SOC_DAIFMT_RIGHT_J 1 /* Right justified mode */
-#define SND_SOC_DAIFMT_LEFT_J 2 /* Left Justified mode */
-#define SND_SOC_DAIFMT_DSP_A 3 /* L data msb after FRM or LRC */
-#define SND_SOC_DAIFMT_DSP_B 4 /* L data msb during FRM or LRC */
-#define SND_SOC_DAIFMT_AC97 5 /* AC97 */
-
-#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J
-#define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J
-
-/*
- * DAI Gating
- */
-#define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */
-#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated when not Tx/Rx */
-
-/*
- * DAI Sync
- * Synchronous LR (Left Right) clocks and Frame signals.
- */
-#define SND_SOC_DAIFMT_SYNC (0 << 5) /* Tx FRM = Rx FRM */
-#define SND_SOC_DAIFMT_ASYNC (1 << 5) /* Tx FRM ~ Rx FRM */
-
-/*
- * TDM
- */
-#define SND_SOC_DAIFMT_TDM (1 << 6)
-
-/*
- * DAI hardware signal inversions
- */
-#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bclk + frm */
-#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal bclk + inv frm */
-#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert bclk + nor frm */
-#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert bclk + frm */
-
-/*
- * DAI hardware clock masters
- * This is wrt the codec, the inverse is true for the interface
- * i.e. if the codec is clk and frm master then the interface is
- * clk and frame slave.
- */
-#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & frm master */
-#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & frm master */
-#define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */
-#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & frm slave */
-
-#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f
-#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0
-#define SND_SOC_DAIFMT_INV_MASK 0x0f00
-#define SND_SOC_DAIFMT_MASTER_MASK 0xf000
-
-
-/*
- * Master Clock Directions
- */
-#define SND_SOC_CLOCK_IN 0
-#define SND_SOC_CLOCK_OUT 1
-
-/*
- * AC97 codec ID's bitmask
- */
-#define SND_SOC_DAI_AC97_ID0 (1 << 0)
-#define SND_SOC_DAI_AC97_ID1 (1 << 1)
-#define SND_SOC_DAI_AC97_ID2 (1 << 2)
-#define SND_SOC_DAI_AC97_ID3 (1 << 3)
-
struct snd_soc_device;
struct snd_soc_pcm_stream;
struct snd_soc_ops;
struct snd_soc_dai_mode;
struct snd_soc_pcm_runtime;
struct snd_soc_dai;
+struct snd_soc_platform;
struct snd_soc_codec;
-struct snd_soc_machine_config;
struct soc_enum;
struct snd_soc_ac97_ops;
-struct snd_soc_clock_info;
typedef int (*hw_write_t)(void *,const char* ,int);
typedef int (*hw_read_t)(void *,char* ,int);
extern struct snd_ac97_bus_ops soc_ac97_ops;
+int snd_soc_register_platform(struct snd_soc_platform *platform);
+void snd_soc_unregister_platform(struct snd_soc_platform *platform);
+int snd_soc_register_codec(struct snd_soc_codec *codec);
+void snd_soc_unregister_codec(struct snd_soc_codec *codec);
+
/* pcm <-> DAI connect */
void snd_soc_free_pcms(struct snd_soc_device *socdev);
int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid);
-int snd_soc_register_card(struct snd_soc_device *socdev);
+int snd_soc_init_card(struct snd_soc_device *socdev);
/* set runtime hw params */
int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
@@ -263,27 +187,6 @@
struct snd_ac97_bus_ops *ops, int num);
void snd_soc_free_ac97_codec(struct snd_soc_codec *codec);
-/* Digital Audio Interface clocking API.*/
-int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
- unsigned int freq, int dir);
-
-int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
- int div_id, int div);
-
-int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out);
-
-/* Digital Audio interface formatting */
-int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
-
-int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
- unsigned int mask, int slots);
-
-int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
-
-/* Digital Audio Interface mute */
-int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute);
-
/*
*Controls
*/
@@ -341,66 +244,14 @@
int (*trigger)(struct snd_pcm_substream *, int);
};
-/* ASoC DAI ops */
-struct snd_soc_dai_ops {
- /* DAI clocking configuration */
- int (*set_sysclk)(struct snd_soc_dai *dai,
- int clk_id, unsigned int freq, int dir);
- int (*set_pll)(struct snd_soc_dai *dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out);
- int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
-
- /* DAI format configuration */
- int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
- int (*set_tdm_slot)(struct snd_soc_dai *dai,
- unsigned int mask, int slots);
- int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
-
- /* digital mute */
- int (*digital_mute)(struct snd_soc_dai *dai, int mute);
-};
-
-/* SoC DAI (Digital Audio Interface) */
-struct snd_soc_dai {
- /* DAI description */
- char *name;
- unsigned int id;
- unsigned char type;
-
- /* DAI callbacks */
- int (*probe)(struct platform_device *pdev,
- struct snd_soc_dai *dai);
- void (*remove)(struct platform_device *pdev,
- struct snd_soc_dai *dai);
- int (*suspend)(struct platform_device *pdev,
- struct snd_soc_dai *dai);
- int (*resume)(struct platform_device *pdev,
- struct snd_soc_dai *dai);
-
- /* ops */
- struct snd_soc_ops ops;
- struct snd_soc_dai_ops dai_ops;
-
- /* DAI capabilities */
- struct snd_soc_pcm_stream capture;
- struct snd_soc_pcm_stream playback;
-
- /* DAI runtime info */
- struct snd_pcm_runtime *runtime;
- struct snd_soc_codec *codec;
- unsigned int active;
- unsigned char pop_wait:1;
- void *dma_data;
-
- /* DAI private data */
- void *private_data;
-};
-
/* SoC Audio Codec */
struct snd_soc_codec {
char *name;
struct module *owner;
struct mutex mutex;
+ struct device *dev;
+
+ struct list_head list;
/* callbacks */
int (*set_bias_level)(struct snd_soc_codec *,
@@ -426,6 +277,7 @@
short reg_cache_step;
/* dapm */
+ u32 pop_time;
struct list_head dapm_widgets;
struct list_head dapm_paths;
enum snd_soc_bias_level bias_level;
@@ -435,6 +287,11 @@
/* codec DAI's */
struct snd_soc_dai *dai;
unsigned int num_dai;
+
+#ifdef CONFIG_DEBUG_FS
+ struct dentry *debugfs_reg;
+ struct dentry *debugfs_pop_time;
+#endif
};
/* codec device */
@@ -448,13 +305,12 @@
/* SoC platform interface */
struct snd_soc_platform {
char *name;
+ struct list_head list;
int (*probe)(struct platform_device *pdev);
int (*remove)(struct platform_device *pdev);
- int (*suspend)(struct platform_device *pdev,
- struct snd_soc_dai *dai);
- int (*resume)(struct platform_device *pdev,
- struct snd_soc_dai *dai);
+ int (*suspend)(struct snd_soc_dai *dai);
+ int (*resume)(struct snd_soc_dai *dai);
/* pcm creation and destruction */
int (*pcm_new)(struct snd_card *, struct snd_soc_dai *,
@@ -484,9 +340,14 @@
struct snd_pcm *pcm;
};
-/* SoC machine */
-struct snd_soc_machine {
+/* SoC card */
+struct snd_soc_card {
char *name;
+ struct device *dev;
+
+ struct list_head list;
+
+ int instantiated;
int (*probe)(struct platform_device *pdev);
int (*remove)(struct platform_device *pdev);
@@ -499,23 +360,26 @@
int (*resume_post)(struct platform_device *pdev);
/* callbacks */
- int (*set_bias_level)(struct snd_soc_machine *,
+ int (*set_bias_level)(struct snd_soc_card *,
enum snd_soc_bias_level level);
/* CPU <--> Codec DAI links */
struct snd_soc_dai_link *dai_link;
int num_links;
+
+ struct snd_soc_device *socdev;
+
+ struct snd_soc_platform *platform;
+ struct delayed_work delayed_work;
+ struct work_struct deferred_resume_work;
};
/* SoC Device - the audio subsystem */
struct snd_soc_device {
struct device *dev;
- struct snd_soc_machine *machine;
- struct snd_soc_platform *platform;
+ struct snd_soc_card *card;
struct snd_soc_codec *codec;
struct snd_soc_codec_device *codec_dev;
- struct delayed_work delayed_work;
- struct work_struct deferred_resume_work;
void *codec_data;
};
@@ -542,4 +406,6 @@
void *dapm;
};
+#include <sound/soc-dai.h>
+
#endif
diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h
new file mode 100644
index 0000000..475ef8b
--- /dev/null
+++ b/include/sound/uda134x.h
@@ -0,0 +1,26 @@
+/*
+ * uda134x.h -- UDA134x ALSA SoC Codec driver
+ *
+ * Copyright 2007 Dension Audio Systems Ltd.
+ * Author: Zoltan Devai
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _UDA134X_H
+#define _UDA134X_H
+
+#include <sound/l3.h>
+
+struct uda134x_platform_data {
+ struct l3_pins l3;
+ void (*power) (int);
+ int model;
+#define UDA134X_UDA1340 1
+#define UDA134X_UDA1341 2
+#define UDA134X_UDA1344 3
+};
+
+#endif /* _UDA134X_H */
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 4dfda66..ef025c6 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -22,17 +22,16 @@
config SND_SOC_AC97_BUS
bool
-# All the supported Soc's
-source "sound/soc/at32/Kconfig"
-source "sound/soc/at91/Kconfig"
+# All the supported SoCs
+source "sound/soc/atmel/Kconfig"
source "sound/soc/au1x/Kconfig"
+source "sound/soc/blackfin/Kconfig"
+source "sound/soc/davinci/Kconfig"
+source "sound/soc/fsl/Kconfig"
+source "sound/soc/omap/Kconfig"
source "sound/soc/pxa/Kconfig"
source "sound/soc/s3c24xx/Kconfig"
source "sound/soc/sh/Kconfig"
-source "sound/soc/fsl/Kconfig"
-source "sound/soc/davinci/Kconfig"
-source "sound/soc/omap/Kconfig"
-source "sound/soc/blackfin/Kconfig"
# Supported codecs
source "sound/soc/codecs/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index d849349..86a9b1f 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,5 +1,13 @@
snd-soc-core-objs := soc-core.o soc-dapm.o
obj-$(CONFIG_SND_SOC) += snd-soc-core.o
-obj-$(CONFIG_SND_SOC) += codecs/ at32/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/
-obj-$(CONFIG_SND_SOC) += omap/ au1x/ blackfin/
+obj-$(CONFIG_SND_SOC) += codecs/
+obj-$(CONFIG_SND_SOC) += atmel/
+obj-$(CONFIG_SND_SOC) += au1x/
+obj-$(CONFIG_SND_SOC) += blackfin/
+obj-$(CONFIG_SND_SOC) += davinci/
+obj-$(CONFIG_SND_SOC) += fsl/
+obj-$(CONFIG_SND_SOC) += omap/
+obj-$(CONFIG_SND_SOC) += pxa/
+obj-$(CONFIG_SND_SOC) += s3c24xx/
+obj-$(CONFIG_SND_SOC) += sh/
diff --git a/sound/soc/at32/Kconfig b/sound/soc/at32/Kconfig
deleted file mode 100644
index b0765e8..0000000
--- a/sound/soc/at32/Kconfig
+++ /dev/null
@@ -1,34 +0,0 @@
-config SND_AT32_SOC
- tristate "SoC Audio for the Atmel AT32 System-on-a-Chip"
- depends on AVR32 && SND_SOC
- help
- Say Y or M if you want to add support for codecs attached to
- the AT32 SSC interface. You will also need to
- to select the audio interfaces to support below.
-
-
-config SND_AT32_SOC_SSC
- tristate
-
-
-
-config SND_AT32_SOC_PLAYPAQ
- tristate "SoC Audio support for PlayPaq with WM8510"
- depends on SND_AT32_SOC && BOARD_PLAYPAQ
- select SND_AT32_SOC_SSC
- select SND_SOC_WM8510
- help
- Say Y or M here if you want to add support for SoC audio
- on the LRS PlayPaq.
-
-
-
-config SND_AT32_SOC_PLAYPAQ_SLAVE
- bool "Run CODEC on PlayPaq in slave mode"
- depends on SND_AT32_SOC_PLAYPAQ
- default n
- help
- Say Y if you want to run with the AT32 SSC generating the BCLK
- and FRAME signals on the PlayPaq. Unless you want to play
- with the AT32 as the SSC master, you probably want to say N here,
- as this will give you better sound quality.
diff --git a/sound/soc/at32/Makefile b/sound/soc/at32/Makefile
deleted file mode 100644
index c03e55e..0000000
--- a/sound/soc/at32/Makefile
+++ /dev/null
@@ -1,11 +0,0 @@
-# AT32 Platform Support
-snd-soc-at32-objs := at32-pcm.o
-snd-soc-at32-ssc-objs := at32-ssc.o
-
-obj-$(CONFIG_SND_AT32_SOC) += snd-soc-at32.o
-obj-$(CONFIG_SND_AT32_SOC_SSC) += snd-soc-at32-ssc.o
-
-# AT32 Machine Support
-snd-soc-playpaq-objs := playpaq_wm8510.o
-
-obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o
diff --git a/sound/soc/at32/at32-pcm.c b/sound/soc/at32/at32-pcm.c
deleted file mode 100644
index c83584f..0000000
--- a/sound/soc/at32/at32-pcm.c
+++ /dev/null
@@ -1,492 +0,0 @@
-/* sound/soc/at32/at32-pcm.c
- * ASoC PCM interface for Atmel AT32 SoC
- *
- * Copyright (C) 2008 Long Range Systems
- * Geoffrey Wossum <gwossum@acm.org>
- *
- * 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.
- *
- * Note that this is basically a port of the sound/soc/at91-pcm.c to
- * the AVR32 kernel. Thanks to Frank Mandarino for that code.
- */
-
-#include <linux/module.h>
-#include <linux/init.h>
-#include <linux/platform_device.h>
-#include <linux/slab.h>
-#include <linux/dma-mapping.h>
-#include <linux/atmel_pdc.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-
-#include "at32-pcm.h"
-
-
-
-/*--------------------------------------------------------------------------*\
- * Hardware definition
-\*--------------------------------------------------------------------------*/
-/* TODO: These values were taken from the AT91 platform driver, check
- * them against real values for AT32
- */
-static const struct snd_pcm_hardware at32_pcm_hardware = {
- .info = (SNDRV_PCM_INFO_MMAP |
- SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_PAUSE),
-
- .formats = SNDRV_PCM_FMTBIT_S16,
- .period_bytes_min = 32,
- .period_bytes_max = 8192, /* 512 frames * 16 bytes / frame */
- .periods_min = 2,
- .periods_max = 1024,
- .buffer_bytes_max = 32 * 1024,
-};
-
-
-
-/*--------------------------------------------------------------------------*\
- * Data types
-\*--------------------------------------------------------------------------*/
-struct at32_runtime_data {
- struct at32_pcm_dma_params *params;
- dma_addr_t dma_buffer; /* physical address of DMA buffer */
- dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */
- size_t period_size;
-
- dma_addr_t period_ptr; /* physical address of next period */
- int periods; /* period index of period_ptr */
-
- /* Save PDC registers (for power management) */
- u32 pdc_xpr_save;
- u32 pdc_xcr_save;
- u32 pdc_xnpr_save;
- u32 pdc_xncr_save;
-};
-
-
-
-/*--------------------------------------------------------------------------*\
- * Helper functions
-\*--------------------------------------------------------------------------*/
-static int at32_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
-{
- struct snd_pcm_substream *substream = pcm->streams[stream].substream;
- struct snd_dma_buffer *dmabuf = &substream->dma_buffer;
- size_t size = at32_pcm_hardware.buffer_bytes_max;
-
- dmabuf->dev.type = SNDRV_DMA_TYPE_DEV;
- dmabuf->dev.dev = pcm->card->dev;
- dmabuf->private_data = NULL;
- dmabuf->area = dma_alloc_coherent(pcm->card->dev, size,
- &dmabuf->addr, GFP_KERNEL);
- pr_debug("at32_pcm: preallocate_dma_buffer: "
- "area=%p, addr=%p, size=%ld\n",
- (void *)dmabuf->area, (void *)dmabuf->addr, size);
-
- if (!dmabuf->area)
- return -ENOMEM;
-
- dmabuf->bytes = size;
- return 0;
-}
-
-
-
-/*--------------------------------------------------------------------------*\
- * ISR
-\*--------------------------------------------------------------------------*/
-static void at32_pcm_dma_irq(u32 ssc_sr, struct snd_pcm_substream *substream)
-{
- struct snd_pcm_runtime *rtd = substream->runtime;
- struct at32_runtime_data *prtd = rtd->private_data;
- struct at32_pcm_dma_params *params = prtd->params;
- static int count;
-
- count++;
- if (ssc_sr & params->mask->ssc_endbuf) {
- pr_warning("at32-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n",
- substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
- "underrun" : "overrun", params->name, ssc_sr, count);
-
- /* re-start the PDC */
- ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
- params->mask->pdc_disable);
- prtd->period_ptr += prtd->period_size;
- if (prtd->period_ptr >= prtd->dma_buffer_end)
- prtd->period_ptr = prtd->dma_buffer;
-
-
- ssc_writex(params->ssc->regs, params->pdc->xpr,
- prtd->period_ptr);
- ssc_writex(params->ssc->regs, params->pdc->xcr,
- prtd->period_size / params->pdc_xfer_size);
- ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
- params->mask->pdc_enable);
- }
-
-
- if (ssc_sr & params->mask->ssc_endx) {
- /* Load the PDC next pointer and counter registers */
- prtd->period_ptr += prtd->period_size;
- if (prtd->period_ptr >= prtd->dma_buffer_end)
- prtd->period_ptr = prtd->dma_buffer;
- ssc_writex(params->ssc->regs, params->pdc->xnpr,
- prtd->period_ptr);
- ssc_writex(params->ssc->regs, params->pdc->xncr,
- prtd->period_size / params->pdc_xfer_size);
- }
-
-
- snd_pcm_period_elapsed(substream);
-}
-
-
-
-/*--------------------------------------------------------------------------*\
- * PCM operations
-\*--------------------------------------------------------------------------*/
-static int at32_pcm_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct at32_runtime_data *prtd = runtime->private_data;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
-
- /* this may get called several times by oss emulation
- * with different params
- */
- snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
- runtime->dma_bytes = params_buffer_bytes(params);
-
- prtd->params = rtd->dai->cpu_dai->dma_data;
- prtd->params->dma_intr_handler = at32_pcm_dma_irq;
-
- prtd->dma_buffer = runtime->dma_addr;
- prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes;
- prtd->period_size = params_period_bytes(params);
-
- pr_debug("hw_params: DMA for %s initialized "
- "(dma_bytes=%ld, period_size=%ld)\n",
- prtd->params->name, runtime->dma_bytes, prtd->period_size);
-
- return 0;
-}
-
-
-
-static int at32_pcm_hw_free(struct snd_pcm_substream *substream)
-{
- struct at32_runtime_data *prtd = substream->runtime->private_data;
- struct at32_pcm_dma_params *params = prtd->params;
-
- if (params != NULL) {
- ssc_writex(params->ssc->regs, SSC_PDC_PTCR,
- params->mask->pdc_disable);
- prtd->params->dma_intr_handler = NULL;
- }
-
- return 0;
-}
-
-
-
-static int at32_pcm_prepare(struct snd_pcm_substream *substream)
-{
- struct at32_runtime_data *prtd = substream->runtime->private_data;
- struct at32_pcm_dma_params *params = prtd->params;
-
- ssc_writex(params->ssc->regs, SSC_IDR,
- params->mask->ssc_endx | params->mask->ssc_endbuf);
- ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
- params->mask->pdc_disable);
-
- return 0;
-}
-
-
-static int at32_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
-{
- struct snd_pcm_runtime *rtd = substream->runtime;
- struct at32_runtime_data *prtd = rtd->private_data;
- struct at32_pcm_dma_params *params = prtd->params;
- int ret = 0;
-
- pr_debug("at32_pcm_trigger: buffer_size = %ld, "
- "dma_area = %p, dma_bytes = %ld\n",
- rtd->buffer_size, rtd->dma_area, rtd->dma_bytes);
-
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- prtd->period_ptr = prtd->dma_buffer;
-
- ssc_writex(params->ssc->regs, params->pdc->xpr,
- prtd->period_ptr);
- ssc_writex(params->ssc->regs, params->pdc->xcr,
- prtd->period_size / params->pdc_xfer_size);
-
- prtd->period_ptr += prtd->period_size;
- ssc_writex(params->ssc->regs, params->pdc->xnpr,
- prtd->period_ptr);
- ssc_writex(params->ssc->regs, params->pdc->xncr,
- prtd->period_size / params->pdc_xfer_size);
-
- pr_debug("trigger: period_ptr=%lx, xpr=%x, "
- "xcr=%d, xnpr=%x, xncr=%d\n",
- (unsigned long)prtd->period_ptr,
- ssc_readx(params->ssc->regs, params->pdc->xpr),
- ssc_readx(params->ssc->regs, params->pdc->xcr),
- ssc_readx(params->ssc->regs, params->pdc->xnpr),
- ssc_readx(params->ssc->regs, params->pdc->xncr));
-
- ssc_writex(params->ssc->regs, SSC_IER,
- params->mask->ssc_endx | params->mask->ssc_endbuf);
- ssc_writex(params->ssc->regs, SSC_PDC_PTCR,
- params->mask->pdc_enable);
-
- pr_debug("sr=%x, imr=%x\n",
- ssc_readx(params->ssc->regs, SSC_SR),
- ssc_readx(params->ssc->regs, SSC_IER));
- break; /* SNDRV_PCM_TRIGGER_START */
-
-
-
- case SNDRV_PCM_TRIGGER_STOP:
- case SNDRV_PCM_TRIGGER_SUSPEND:
- case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
- params->mask->pdc_disable);
- break;
-
-
- case SNDRV_PCM_TRIGGER_RESUME:
- case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
- params->mask->pdc_enable);
- break;
-
- default:
- ret = -EINVAL;
- }
-
- return ret;
-}
-
-
-
-static snd_pcm_uframes_t at32_pcm_pointer(struct snd_pcm_substream *substream)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct at32_runtime_data *prtd = runtime->private_data;
- struct at32_pcm_dma_params *params = prtd->params;
- dma_addr_t ptr;
- snd_pcm_uframes_t x;
-
- ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr);
- x = bytes_to_frames(runtime, ptr - prtd->dma_buffer);
-
- if (x == runtime->buffer_size)
- x = 0;
-
- return x;
-}
-
-
-
-static int at32_pcm_open(struct snd_pcm_substream *substream)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct at32_runtime_data *prtd;
- int ret = 0;
-
- snd_soc_set_runtime_hwparams(substream, &at32_pcm_hardware);
-
- /* ensure that buffer size is a multiple of period size */
- ret = snd_pcm_hw_constraint_integer(runtime,
- SNDRV_PCM_HW_PARAM_PERIODS);
- if (ret < 0)
- goto out;
-
- prtd = kzalloc(sizeof(*prtd), GFP_KERNEL);
- if (prtd == NULL) {
- ret = -ENOMEM;
- goto out;
- }
- runtime->private_data = prtd;
-
-
-out:
- return ret;
-}
-
-
-
-static int at32_pcm_close(struct snd_pcm_substream *substream)
-{
- struct at32_runtime_data *prtd = substream->runtime->private_data;
-
- kfree(prtd);
- return 0;
-}
-
-
-static int at32_pcm_mmap(struct snd_pcm_substream *substream,
- struct vm_area_struct *vma)
-{
- return remap_pfn_range(vma, vma->vm_start,
- substream->dma_buffer.addr >> PAGE_SHIFT,
- vma->vm_end - vma->vm_start, vma->vm_page_prot);
-}
-
-
-
-static struct snd_pcm_ops at32_pcm_ops = {
- .open = at32_pcm_open,
- .close = at32_pcm_close,
- .ioctl = snd_pcm_lib_ioctl,
- .hw_params = at32_pcm_hw_params,
- .hw_free = at32_pcm_hw_free,
- .prepare = at32_pcm_prepare,
- .trigger = at32_pcm_trigger,
- .pointer = at32_pcm_pointer,
- .mmap = at32_pcm_mmap,
-};
-
-
-
-/*--------------------------------------------------------------------------*\
- * ASoC platform driver
-\*--------------------------------------------------------------------------*/
-static u64 at32_pcm_dmamask = 0xffffffff;
-
-static int at32_pcm_new(struct snd_card *card,
- struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
-{
- int ret = 0;
-
- if (!card->dev->dma_mask)
- card->dev->dma_mask = &at32_pcm_dmamask;
- if (!card->dev->coherent_dma_mask)
- card->dev->coherent_dma_mask = 0xffffffff;
-
- if (dai->playback.channels_min) {
- ret = at32_pcm_preallocate_dma_buffer(
- pcm, SNDRV_PCM_STREAM_PLAYBACK);
- if (ret)
- goto out;
- }
-
- if (dai->capture.channels_min) {
- pr_debug("at32-pcm: Allocating PCM capture DMA buffer\n");
- ret = at32_pcm_preallocate_dma_buffer(
- pcm, SNDRV_PCM_STREAM_CAPTURE);
- if (ret)
- goto out;
- }
-
-
-out:
- return ret;
-}
-
-
-
-static void at32_pcm_free_dma_buffers(struct snd_pcm *pcm)
-{
- struct snd_pcm_substream *substream;
- struct snd_dma_buffer *buf;
- int stream;
-
- for (stream = 0; stream < 2; stream++) {
- substream = pcm->streams[stream].substream;
- if (substream == NULL)
- continue;
-
- buf = &substream->dma_buffer;
- if (!buf->area)
- continue;
- dma_free_coherent(pcm->card->dev, buf->bytes,
- buf->area, buf->addr);
- buf->area = NULL;
- }
-}
-
-
-
-#ifdef CONFIG_PM
-static int at32_pcm_suspend(struct platform_device *pdev,
- struct snd_soc_dai *dai)
-{
- struct snd_pcm_runtime *runtime = dai->runtime;
- struct at32_runtime_data *prtd;
- struct at32_pcm_dma_params *params;
-
- if (runtime == NULL)
- return 0;
- prtd = runtime->private_data;
- params = prtd->params;
-
- /* Disable the PDC and save the PDC registers */
- ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
- params->mask->pdc_disable);
-
- prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr);
- prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr);
- prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr);
- prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr);
-
- return 0;
-}
-
-
-
-static int at32_pcm_resume(struct platform_device *pdev,
- struct snd_soc_dai *dai)
-{
- struct snd_pcm_runtime *runtime = dai->runtime;
- struct at32_runtime_data *prtd;
- struct at32_pcm_dma_params *params;
-
- if (runtime == NULL)
- return 0;
- prtd = runtime->private_data;
- params = prtd->params;
-
- /* Restore the PDC registers and enable the PDC */
- ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save);
- ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save);
- ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save);
- ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save);
-
- ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, params->mask->pdc_enable);
- return 0;
-}
-#else /* CONFIG_PM */
-# define at32_pcm_suspend NULL
-# define at32_pcm_resume NULL
-#endif /* CONFIG_PM */
-
-
-
-struct snd_soc_platform at32_soc_platform = {
- .name = "at32-audio",
- .pcm_ops = &at32_pcm_ops,
- .pcm_new = at32_pcm_new,
- .pcm_free = at32_pcm_free_dma_buffers,
- .suspend = at32_pcm_suspend,
- .resume = at32_pcm_resume,
-};
-EXPORT_SYMBOL_GPL(at32_soc_platform);
-
-
-
-MODULE_AUTHOR("Geoffrey Wossum <gwossum@acm.org>");
-MODULE_DESCRIPTION("Atmel AT32 PCM module");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/at32/at32-pcm.h b/sound/soc/at32/at32-pcm.h
deleted file mode 100644
index 2a52430..0000000
--- a/sound/soc/at32/at32-pcm.h
+++ /dev/null
@@ -1,79 +0,0 @@
-/* sound/soc/at32/at32-pcm.h
- * ASoC PCM interface for Atmel AT32 SoC
- *
- * Copyright (C) 2008 Long Range Systems
- * Geoffrey Wossum <gwossum@acm.org>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
-
-#ifndef __SOUND_SOC_AT32_AT32_PCM_H
-#define __SOUND_SOC_AT32_AT32_PCM_H __FILE__
-
-#include <linux/atmel-ssc.h>
-
-
-/*
- * Registers and status bits that are required by the PCM driver
- * TODO: Is ptcr really used?
- */
-struct at32_pdc_regs {
- u32 xpr; /* PDC RX/TX pointer */
- u32 xcr; /* PDC RX/TX counter */
- u32 xnpr; /* PDC next RX/TX pointer */
- u32 xncr; /* PDC next RX/TX counter */
- u32 ptcr; /* PDC transfer control */
-};
-
-
-
-/*
- * SSC mask info
- */
-struct at32_ssc_mask {
- u32 ssc_enable; /* SSC RX/TX enable */
- u32 ssc_disable; /* SSC RX/TX disable */
- u32 ssc_endx; /* SSC ENDTX or ENDRX */
- u32 ssc_endbuf; /* SSC TXBUFF or RXBUFF */
- u32 pdc_enable; /* PDC RX/TX enable */
- u32 pdc_disable; /* PDC RX/TX disable */
-};
-
-
-
-/*
- * This structure, shared between the PCM driver and the interface,
- * contains all information required by the PCM driver to perform the
- * PDC DMA operation. All fields except dma_intr_handler() are initialized
- * by the interface. The dms_intr_handler() pointer is set by the PCM
- * driver and called by the interface SSC interrupt handler if it is
- * non-NULL.
- */
-struct at32_pcm_dma_params {
- char *name; /* stream identifier */
- int pdc_xfer_size; /* PDC counter increment in bytes */
- struct ssc_device *ssc; /* SSC device for stream */
- struct at32_pdc_regs *pdc; /* PDC register info */
- struct at32_ssc_mask *mask; /* SSC mask info */
- struct snd_pcm_substream *substream;
- void (*dma_intr_handler) (u32, struct snd_pcm_substream *);
-};
-
-
-
-/*
- * The AT32 ASoC platform driver
- */
-extern struct snd_soc_platform at32_soc_platform;
-
-
-
-/*
- * SSC register access (since ssc_writel() / ssc_readl() require literal name)
- */
-#define ssc_readx(base, reg) (__raw_readl((base) + (reg)))
-#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg))
-
-#endif /* __SOUND_SOC_AT32_AT32_PCM_H */
diff --git a/sound/soc/at32/at32-ssc.c b/sound/soc/at32/at32-ssc.c
deleted file mode 100644
index 4ef6492..0000000
--- a/sound/soc/at32/at32-ssc.c
+++ /dev/null
@@ -1,849 +0,0 @@
-/* sound/soc/at32/at32-ssc.c
- * ASoC platform driver for AT32 using SSC as DAI
- *
- * Copyright (C) 2008 Long Range Systems
- * Geoffrey Wossum <gwossum@acm.org>
- *
- * 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.
- *
- * Note that this is basically a port of the sound/soc/at91-ssc.c to
- * the AVR32 kernel. Thanks to Frank Mandarino for that code.
- */
-
-/* #define DEBUG */
-
-#include <linux/init.h>
-#include <linux/module.h>
-#include <linux/interrupt.h>
-#include <linux/device.h>
-#include <linux/delay.h>
-#include <linux/clk.h>
-#include <linux/io.h>
-#include <linux/atmel_pdc.h>
-#include <linux/atmel-ssc.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/initval.h>
-#include <sound/soc.h>
-
-#include "at32-pcm.h"
-#include "at32-ssc.h"
-
-
-
-/*-------------------------------------------------------------------------*\
- * Constants
-\*-------------------------------------------------------------------------*/
-#define NUM_SSC_DEVICES 3
-
-/*
- * SSC direction masks
- */
-#define SSC_DIR_MASK_UNUSED 0
-#define SSC_DIR_MASK_PLAYBACK 1
-#define SSC_DIR_MASK_CAPTURE 2
-
-/*
- * SSC register values that Atmel left out of <linux/atmel-ssc.h>. These
- * are expected to be used with SSC_BF
- */
-/* START bit field values */
-#define SSC_START_CONTINUOUS 0
-#define SSC_START_TX_RX 1
-#define SSC_START_LOW_RF 2
-#define SSC_START_HIGH_RF 3
-#define SSC_START_FALLING_RF 4
-#define SSC_START_RISING_RF 5
-#define SSC_START_LEVEL_RF 6
-#define SSC_START_EDGE_RF 7
-#define SSS_START_COMPARE_0 8
-
-/* CKI bit field values */
-#define SSC_CKI_FALLING 0
-#define SSC_CKI_RISING 1
-
-/* CKO bit field values */
-#define SSC_CKO_NONE 0
-#define SSC_CKO_CONTINUOUS 1
-#define SSC_CKO_TRANSFER 2
-
-/* CKS bit field values */
-#define SSC_CKS_DIV 0
-#define SSC_CKS_CLOCK 1
-#define SSC_CKS_PIN 2
-
-/* FSEDGE bit field values */
-#define SSC_FSEDGE_POSITIVE 0
-#define SSC_FSEDGE_NEGATIVE 1
-
-/* FSOS bit field values */
-#define SSC_FSOS_NONE 0
-#define SSC_FSOS_NEGATIVE 1
-#define SSC_FSOS_POSITIVE 2
-#define SSC_FSOS_LOW 3
-#define SSC_FSOS_HIGH 4
-#define SSC_FSOS_TOGGLE 5
-
-#define START_DELAY 1
-
-
-
-/*-------------------------------------------------------------------------*\
- * Module data
-\*-------------------------------------------------------------------------*/
-/*
- * SSC PDC registered required by the PCM DMA engine
- */
-static struct at32_pdc_regs pdc_tx_reg = {
- .xpr = SSC_PDC_TPR,
- .xcr = SSC_PDC_TCR,
- .xnpr = SSC_PDC_TNPR,
- .xncr = SSC_PDC_TNCR,
-};
-
-
-
-static struct at32_pdc_regs pdc_rx_reg = {
- .xpr = SSC_PDC_RPR,
- .xcr = SSC_PDC_RCR,
- .xnpr = SSC_PDC_RNPR,
- .xncr = SSC_PDC_RNCR,
-};
-
-
-
-/*
- * SSC and PDC status bits for transmit and receive
- */
-static struct at32_ssc_mask ssc_tx_mask = {
- .ssc_enable = SSC_BIT(CR_TXEN),
- .ssc_disable = SSC_BIT(CR_TXDIS),
- .ssc_endx = SSC_BIT(SR_ENDTX),
- .ssc_endbuf = SSC_BIT(SR_TXBUFE),
- .pdc_enable = SSC_BIT(PDC_PTCR_TXTEN),
- .pdc_disable = SSC_BIT(PDC_PTCR_TXTDIS),
-};
-
-
-
-static struct at32_ssc_mask ssc_rx_mask = {
- .ssc_enable = SSC_BIT(CR_RXEN),
- .ssc_disable = SSC_BIT(CR_RXDIS),
- .ssc_endx = SSC_BIT(SR_ENDRX),
- .ssc_endbuf = SSC_BIT(SR_RXBUFF),
- .pdc_enable = SSC_BIT(PDC_PTCR_RXTEN),
- .pdc_disable = SSC_BIT(PDC_PTCR_RXTDIS),
-};
-
-
-
-/*
- * DMA parameters for each SSC
- */
-static struct at32_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = {
- {
- {
- .name = "SSC0 PCM out",
- .pdc = &pdc_tx_reg,
- .mask = &ssc_tx_mask,
- },
- {
- .name = "SSC0 PCM in",
- .pdc = &pdc_rx_reg,
- .mask = &ssc_rx_mask,
- },
- },
- {
- {
- .name = "SSC1 PCM out",
- .pdc = &pdc_tx_reg,
- .mask = &ssc_tx_mask,
- },
- {
- .name = "SSC1 PCM in",
- .pdc = &pdc_rx_reg,
- .mask = &ssc_rx_mask,
- },
- },
- {
- {
- .name = "SSC2 PCM out",
- .pdc = &pdc_tx_reg,
- .mask = &ssc_tx_mask,
- },
- {
- .name = "SSC2 PCM in",
- .pdc = &pdc_rx_reg,
- .mask = &ssc_rx_mask,
- },
- },
-};
-
-
-
-static struct at32_ssc_info ssc_info[NUM_SSC_DEVICES] = {
- {
- .name = "ssc0",
- .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock),
- .dir_mask = SSC_DIR_MASK_UNUSED,
- .initialized = 0,
- },
- {
- .name = "ssc1",
- .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock),
- .dir_mask = SSC_DIR_MASK_UNUSED,
- .initialized = 0,
- },
- {
- .name = "ssc2",
- .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock),
- .dir_mask = SSC_DIR_MASK_UNUSED,
- .initialized = 0,
- },
-};
-
-
-
-
-/*-------------------------------------------------------------------------*\
- * ISR
-\*-------------------------------------------------------------------------*/
-/*
- * SSC interrupt handler. Passes PDC interrupts to the DMA interrupt
- * handler in the PCM driver.
- */
-static irqreturn_t at32_ssc_interrupt(int irq, void *dev_id)
-{
- struct at32_ssc_info *ssc_p = dev_id;
- struct at32_pcm_dma_params *dma_params;
- u32 ssc_sr;
- u32 ssc_substream_mask;
- int i;
-
- ssc_sr = (ssc_readl(ssc_p->ssc->regs, SR) &
- ssc_readl(ssc_p->ssc->regs, IMR));
-
- /*
- * Loop through substreams attached to this SSC. If a DMA-related
- * interrupt occured on that substream, call the DMA interrupt
- * handler function, if one has been registered in the dma_param
- * structure by the PCM driver.
- */
- for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) {
- dma_params = ssc_p->dma_params[i];
-
- if ((dma_params != NULL) &&
- (dma_params->dma_intr_handler != NULL)) {
- ssc_substream_mask = (dma_params->mask->ssc_endx |
- dma_params->mask->ssc_endbuf);
- if (ssc_sr & ssc_substream_mask) {
- dma_params->dma_intr_handler(ssc_sr,
- dma_params->
- substream);
- }
- }
- }
-
-
- return IRQ_HANDLED;
-}
-
-/*-------------------------------------------------------------------------*\
- * DAI functions
-\*-------------------------------------------------------------------------*/
-/*
- * Startup. Only that one substream allowed in each direction.
- */
-static int at32_ssc_startup(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
- int dir_mask;
-
- dir_mask = ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
- SSC_DIR_MASK_PLAYBACK : SSC_DIR_MASK_CAPTURE);
-
- spin_lock_irq(&ssc_p->lock);
- if (ssc_p->dir_mask & dir_mask) {
- spin_unlock_irq(&ssc_p->lock);
- return -EBUSY;
- }
- ssc_p->dir_mask |= dir_mask;
- spin_unlock_irq(&ssc_p->lock);
-
- return 0;
-}
-
-
-
-/*
- * Shutdown. Clear DMA parameters and shutdown the SSC if there
- * are no other substreams open.
- */
-static void at32_ssc_shutdown(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
- struct at32_pcm_dma_params *dma_params;
- int dir_mask;
-
- dma_params = ssc_p->dma_params[substream->stream];
-
- if (dma_params != NULL) {
- ssc_writel(dma_params->ssc->regs, CR,
- dma_params->mask->ssc_disable);
- pr_debug("%s disabled SSC_SR=0x%08x\n",
- (substream->stream ? "receiver" : "transmit"),
- ssc_readl(ssc_p->ssc->regs, SR));
-
- dma_params->ssc = NULL;
- dma_params->substream = NULL;
- ssc_p->dma_params[substream->stream] = NULL;
- }
-
-
- dir_mask = 1 << substream->stream;
- spin_lock_irq(&ssc_p->lock);
- ssc_p->dir_mask &= ~dir_mask;
- if (!ssc_p->dir_mask) {
- /* Shutdown the SSC clock */
- pr_debug("at32-ssc: Stopping user %d clock\n",
- ssc_p->ssc->user);
- clk_disable(ssc_p->ssc->clk);
-
- if (ssc_p->initialized) {
- free_irq(ssc_p->ssc->irq, ssc_p);
- ssc_p->initialized = 0;
- }
-
- /* Reset the SSC */
- ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST));
-
- /* clear the SSC dividers */
- ssc_p->cmr_div = 0;
- ssc_p->tcmr_period = 0;
- ssc_p->rcmr_period = 0;
- }
- spin_unlock_irq(&ssc_p->lock);
-}
-
-
-
-/*
- * Set the SSC system clock rate
- */
-static int at32_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
- int clk_id, unsigned int freq, int dir)
-{
- /* TODO: What the heck do I do here? */
- return 0;
-}
-
-
-
-/*
- * Record DAI format for use by hw_params()
- */
-static int at32_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai,
- unsigned int fmt)
-{
- struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
-
- ssc_p->daifmt = fmt;
- return 0;
-}
-
-
-
-/*
- * Record SSC clock dividers for use in hw_params()
- */
-static int at32_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
- int div_id, int div)
-{
- struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
-
- switch (div_id) {
- case AT32_SSC_CMR_DIV:
- /*
- * The same master clock divider is used for both
- * transmit and receive, so if a value has already
- * been set, it must match this value
- */
- if (ssc_p->cmr_div == 0)
- ssc_p->cmr_div = div;
- else if (div != ssc_p->cmr_div)
- return -EBUSY;
- break;
-
- case AT32_SSC_TCMR_PERIOD:
- ssc_p->tcmr_period = div;
- break;
-
- case AT32_SSC_RCMR_PERIOD:
- ssc_p->rcmr_period = div;
- break;
-
- default:
- return -EINVAL;
- }
-
- return 0;
-}
-
-
-
-/*
- * Configure the SSC
- */
-static int at32_ssc_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- int id = rtd->dai->cpu_dai->id;
- struct at32_ssc_info *ssc_p = &ssc_info[id];
- struct at32_pcm_dma_params *dma_params;
- int channels, bits;
- u32 tfmr, rfmr, tcmr, rcmr;
- int start_event;
- int ret;
-
-
- /*
- * Currently, there is only one set of dma_params for each direction.
- * If more are added, this code will have to be changed to select
- * the proper set
- */
- dma_params = &ssc_dma_params[id][substream->stream];
- dma_params->ssc = ssc_p->ssc;
- dma_params->substream = substream;
-
- ssc_p->dma_params[substream->stream] = dma_params;
-
-
- /*
- * The cpu_dai->dma_data field is only used to communicate the
- * appropriate DMA parameters to the PCM driver's hw_params()
- * function. It should not be used for other purposes as it
- * is common to all substreams.
- */
- rtd->dai->cpu_dai->dma_data = dma_params;
-
- channels = params_channels(params);
-
-
- /*
- * Determine sample size in bits and the PDC increment
- */
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S8:
- bits = 8;
- dma_params->pdc_xfer_size = 1;
- break;
-
- case SNDRV_PCM_FORMAT_S16:
- bits = 16;
- dma_params->pdc_xfer_size = 2;
- break;
-
- case SNDRV_PCM_FORMAT_S24:
- bits = 24;
- dma_params->pdc_xfer_size = 4;
- break;
-
- case SNDRV_PCM_FORMAT_S32:
- bits = 32;
- dma_params->pdc_xfer_size = 4;
- break;
-
- default:
- pr_warning("at32-ssc: Unsupported PCM format %d",
- params_format(params));
- return -EINVAL;
- }
- pr_debug("at32-ssc: bits = %d, pdc_xfer_size = %d, channels = %d\n",
- bits, dma_params->pdc_xfer_size, channels);
-
-
- /*
- * The SSC only supports up to 16-bit samples in I2S format, due
- * to the size of the Frame Mode Register FSLEN field.
- */
- if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S)
- if (bits > 16) {
- pr_warning("at32-ssc: "
- "sample size %d is too large for I2S\n",
- bits);
- return -EINVAL;
- }
-
-
- /*
- * Compute the SSC register settings
- */
- switch (ssc_p->daifmt & (SND_SOC_DAIFMT_FORMAT_MASK |
- SND_SOC_DAIFMT_MASTER_MASK)) {
- case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS:
- /*
- * I2S format, SSC provides BCLK and LRS clocks.
- *
- * The SSC transmit and receive clocks are generated from the
- * MCK divider, and the BCLK signal is output on the SSC TK line
- */
- pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME master\n");
- rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) |
- SSC_BF(RCMR_STTDLY, START_DELAY) |
- SSC_BF(RCMR_START, SSC_START_FALLING_RF) |
- SSC_BF(RCMR_CKI, SSC_CKI_RISING) |
- SSC_BF(RCMR_CKO, SSC_CKO_NONE) |
- SSC_BF(RCMR_CKS, SSC_CKS_DIV));
-
- rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) |
- SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) |
- SSC_BF(RFMR_FSLEN, bits - 1) |
- SSC_BF(RFMR_DATNB, channels - 1) |
- SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1));
-
- tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) |
- SSC_BF(TCMR_STTDLY, START_DELAY) |
- SSC_BF(TCMR_START, SSC_START_FALLING_RF) |
- SSC_BF(TCMR_CKI, SSC_CKI_FALLING) |
- SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) |
- SSC_BF(TCMR_CKS, SSC_CKS_DIV));
-
- tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) |
- SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) |
- SSC_BF(TFMR_FSLEN, bits - 1) |
- SSC_BF(TFMR_DATNB, channels - 1) | SSC_BIT(TFMR_MSBF) |
- SSC_BF(TFMR_DATLEN, bits - 1));
- break;
-
-
- case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM:
- /*
- * I2S format, CODEC supplies BCLK and LRC clock.
- *
- * The SSC transmit clock is obtained from the BCLK signal
- * on the TK line, and the SSC receive clock is generated from
- * the transmit clock.
- *
- * For single channel data, one sample is transferred on the
- * falling edge of the LRC clock. For two channel data, one
- * sample is transferred on both edges of the LRC clock.
- */
- pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME slave\n");
- start_event = ((channels == 1) ?
- SSC_START_FALLING_RF : SSC_START_EDGE_RF);
-
- rcmr = (SSC_BF(RCMR_STTDLY, START_DELAY) |
- SSC_BF(RCMR_START, start_event) |
- SSC_BF(RCMR_CKI, SSC_CKI_RISING) |
- SSC_BF(RCMR_CKO, SSC_CKO_NONE) |
- SSC_BF(RCMR_CKS, SSC_CKS_CLOCK));
-
- rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) |
- SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) |
- SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1));
-
- tcmr = (SSC_BF(TCMR_STTDLY, START_DELAY) |
- SSC_BF(TCMR_START, start_event) |
- SSC_BF(TCMR_CKI, SSC_CKI_FALLING) |
- SSC_BF(TCMR_CKO, SSC_CKO_NONE) |
- SSC_BF(TCMR_CKS, SSC_CKS_PIN));
-
- tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) |
- SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) |
- SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1));
- break;
-
-
- case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS:
- /*
- * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks.
- *
- * The SSC transmit and receive clocks are generated from the
- * MCK divider, and the BCLK signal is output on the SSC TK line
- */
- pr_debug("at32-ssc: SSC mode is DSP A BCLK / FRAME master\n");
- rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) |
- SSC_BF(RCMR_STTDLY, 1) |
- SSC_BF(RCMR_START, SSC_START_RISING_RF) |
- SSC_BF(RCMR_CKI, SSC_CKI_RISING) |
- SSC_BF(RCMR_CKO, SSC_CKO_NONE) |
- SSC_BF(RCMR_CKS, SSC_CKS_DIV));
-
- rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) |
- SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) |
- SSC_BF(RFMR_DATNB, channels - 1) |
- SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1));
-
- tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) |
- SSC_BF(TCMR_STTDLY, 1) |
- SSC_BF(TCMR_START, SSC_START_RISING_RF) |
- SSC_BF(TCMR_CKI, SSC_CKI_RISING) |
- SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) |
- SSC_BF(TCMR_CKS, SSC_CKS_DIV));
-
- tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) |
- SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) |
- SSC_BF(TFMR_DATNB, channels - 1) |
- SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1));
- break;
-
-
- case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM:
- default:
- pr_warning("at32-ssc: unsupported DAI format 0x%x\n",
- ssc_p->daifmt);
- return -EINVAL;
- break;
- }
- pr_debug("at32-ssc: RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n",
- rcmr, rfmr, tcmr, tfmr);
-
-
- if (!ssc_p->initialized) {
- /* enable peripheral clock */
- pr_debug("at32-ssc: Starting clock\n");
- clk_enable(ssc_p->ssc->clk);
-
- /* Reset the SSC and its PDC registers */
- ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST));
-
- ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0);
- ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0);
- ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0);
- ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0);
-
- ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0);
- ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0);
- ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0);
- ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0);
-
- ret = request_irq(ssc_p->ssc->irq, at32_ssc_interrupt, 0,
- ssc_p->name, ssc_p);
- if (ret < 0) {
- pr_warning("at32-ssc: request irq failed (%d)\n", ret);
- pr_debug("at32-ssc: Stopping clock\n");
- clk_disable(ssc_p->ssc->clk);
- return ret;
- }
-
- ssc_p->initialized = 1;
- }
-
- /* Set SSC clock mode register */
- ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div);
-
- /* set receive clock mode and format */
- ssc_writel(ssc_p->ssc->regs, RCMR, rcmr);
- ssc_writel(ssc_p->ssc->regs, RFMR, rfmr);
-
- /* set transmit clock mode and format */
- ssc_writel(ssc_p->ssc->regs, TCMR, tcmr);
- ssc_writel(ssc_p->ssc->regs, TFMR, tfmr);
-
- pr_debug("at32-ssc: SSC initialized\n");
- return 0;
-}
-
-
-
-static int at32_ssc_prepare(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
- struct at32_pcm_dma_params *dma_params;
-
- dma_params = ssc_p->dma_params[substream->stream];
-
- ssc_writel(dma_params->ssc->regs, CR, dma_params->mask->ssc_enable);
-
- return 0;
-}
-
-
-
-#ifdef CONFIG_PM
-static int at32_ssc_suspend(struct platform_device *pdev,
- struct snd_soc_dai *cpu_dai)
-{
- struct at32_ssc_info *ssc_p;
-
- if (!cpu_dai->active)
- return 0;
-
- ssc_p = &ssc_info[cpu_dai->id];
-
- /* Save the status register before disabling transmit and receive */
- ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR);
- ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS));
-
- /* Save the current interrupt mask, then disable unmasked interrupts */
- ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR);
- ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr);
-
- ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR);
- ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR);
- ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR);
- ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR);
- ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR);
-
- return 0;
-}
-
-
-
-static int at32_ssc_resume(struct platform_device *pdev,
- struct snd_soc_dai *cpu_dai)
-{
- struct at32_ssc_info *ssc_p;
- u32 cr;
-
- if (!cpu_dai->active)
- return 0;
-
- ssc_p = &ssc_info[cpu_dai->id];
-
- /* restore SSC register settings */
- ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr);
- ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr);
- ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr);
- ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr);
- ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr);
-
- /* re-enable interrupts */
- ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr);
-
- /* Re-enable recieve and transmit as appropriate */
- cr = 0;
- cr |=
- (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0;
- cr |=
- (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0;
- ssc_writel(ssc_p->ssc->regs, CR, cr);
-
- return 0;
-}
-#else /* CONFIG_PM */
-# define at32_ssc_suspend NULL
-# define at32_ssc_resume NULL
-#endif /* CONFIG_PM */
-
-
-#define AT32_SSC_RATES \
- (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
- SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
- SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
-
-
-#define AT32_SSC_FORMATS \
- (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16 | \
- SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32)
-
-
-struct snd_soc_dai at32_ssc_dai[NUM_SSC_DEVICES] = {
- {
- .name = "at32-ssc0",
- .id = 0,
- .type = SND_SOC_DAI_PCM,
- .suspend = at32_ssc_suspend,
- .resume = at32_ssc_resume,
- .playback = {
- .channels_min = 1,
- .channels_max = 2,
- .rates = AT32_SSC_RATES,
- .formats = AT32_SSC_FORMATS,
- },
- .capture = {
- .channels_min = 1,
- .channels_max = 2,
- .rates = AT32_SSC_RATES,
- .formats = AT32_SSC_FORMATS,
- },
- .ops = {
- .startup = at32_ssc_startup,
- .shutdown = at32_ssc_shutdown,
- .prepare = at32_ssc_prepare,
- .hw_params = at32_ssc_hw_params,
- },
- .dai_ops = {
- .set_sysclk = at32_ssc_set_dai_sysclk,
- .set_fmt = at32_ssc_set_dai_fmt,
- .set_clkdiv = at32_ssc_set_dai_clkdiv,
- },
- .private_data = &ssc_info[0],
- },
- {
- .name = "at32-ssc1",
- .id = 1,
- .type = SND_SOC_DAI_PCM,
- .suspend = at32_ssc_suspend,
- .resume = at32_ssc_resume,
- .playback = {
- .channels_min = 1,
- .channels_max = 2,
- .rates = AT32_SSC_RATES,
- .formats = AT32_SSC_FORMATS,
- },
- .capture = {
- .channels_min = 1,
- .channels_max = 2,
- .rates = AT32_SSC_RATES,
- .formats = AT32_SSC_FORMATS,
- },
- .ops = {
- .startup = at32_ssc_startup,
- .shutdown = at32_ssc_shutdown,
- .prepare = at32_ssc_prepare,
- .hw_params = at32_ssc_hw_params,
- },
- .dai_ops = {
- .set_sysclk = at32_ssc_set_dai_sysclk,
- .set_fmt = at32_ssc_set_dai_fmt,
- .set_clkdiv = at32_ssc_set_dai_clkdiv,
- },
- .private_data = &ssc_info[1],
- },
- {
- .name = "at32-ssc2",
- .id = 2,
- .type = SND_SOC_DAI_PCM,
- .suspend = at32_ssc_suspend,
- .resume = at32_ssc_resume,
- .playback = {
- .channels_min = 1,
- .channels_max = 2,
- .rates = AT32_SSC_RATES,
- .formats = AT32_SSC_FORMATS,
- },
- .capture = {
- .channels_min = 1,
- .channels_max = 2,
- .rates = AT32_SSC_RATES,
- .formats = AT32_SSC_FORMATS,
- },
- .ops = {
- .startup = at32_ssc_startup,
- .shutdown = at32_ssc_shutdown,
- .prepare = at32_ssc_prepare,
- .hw_params = at32_ssc_hw_params,
- },
- .dai_ops = {
- .set_sysclk = at32_ssc_set_dai_sysclk,
- .set_fmt = at32_ssc_set_dai_fmt,
- .set_clkdiv = at32_ssc_set_dai_clkdiv,
- },
- .private_data = &ssc_info[2],
- },
-};
-EXPORT_SYMBOL_GPL(at32_ssc_dai);
-
-
-MODULE_AUTHOR("Geoffrey Wossum <gwossum@acm.org>");
-MODULE_DESCRIPTION("AT32 SSC ASoC Interface");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/at32/at32-ssc.h b/sound/soc/at32/at32-ssc.h
deleted file mode 100644
index 3c052db..0000000
--- a/sound/soc/at32/at32-ssc.h
+++ /dev/null
@@ -1,59 +0,0 @@
-/* sound/soc/at32/at32-ssc.h
- * ASoC SSC interface for Atmel AT32 SoC
- *
- * Copyright (C) 2008 Long Range Systems
- * Geoffrey Wossum <gwossum@acm.org>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
-
-#ifndef __SOUND_SOC_AT32_AT32_SSC_H
-#define __SOUND_SOC_AT32_AT32_SSC_H __FILE__
-
-#include <linux/types.h>
-#include <linux/atmel-ssc.h>
-
-#include "at32-pcm.h"
-
-
-
-struct at32_ssc_state {
- u32 ssc_cmr;
- u32 ssc_rcmr;
- u32 ssc_rfmr;
- u32 ssc_tcmr;
- u32 ssc_tfmr;
- u32 ssc_sr;
- u32 ssc_imr;
-};
-
-
-
-struct at32_ssc_info {
- char *name;
- struct ssc_device *ssc;
- spinlock_t lock; /* lock for dir_mask */
- unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */
- unsigned short initialized; /* true if SSC has been initialized */
- unsigned short daifmt;
- unsigned short cmr_div;
- unsigned short tcmr_period;
- unsigned short rcmr_period;
- struct at32_pcm_dma_params *dma_params[2];
- struct at32_ssc_state ssc_state;
-};
-
-
-/* SSC divider ids */
-#define AT32_SSC_CMR_DIV 0 /* MCK divider for BCLK */
-#define AT32_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */
-#define AT32_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */
-
-
-extern struct snd_soc_dai at32_ssc_dai[];
-
-
-
-#endif /* __SOUND_SOC_AT32_AT32_SSC_H */
diff --git a/sound/soc/at91/Kconfig b/sound/soc/at91/Kconfig
deleted file mode 100644
index 85a8832..0000000
--- a/sound/soc/at91/Kconfig
+++ /dev/null
@@ -1,10 +0,0 @@
-config SND_AT91_SOC
- tristate "SoC Audio for the Atmel AT91 System-on-Chip"
- depends on ARCH_AT91
- help
- Say Y or M if you want to add support for codecs attached to
- the AT91 SSC interface. You will also need
- to select the audio interfaces to support below.
-
-config SND_AT91_SOC_SSC
- tristate
diff --git a/sound/soc/at91/Makefile b/sound/soc/at91/Makefile
deleted file mode 100644
index b817f11..0000000
--- a/sound/soc/at91/Makefile
+++ /dev/null
@@ -1,6 +0,0 @@
-# AT91 Platform Support
-snd-soc-at91-objs := at91-pcm.o
-snd-soc-at91-ssc-objs := at91-ssc.o
-
-obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o
-obj-$(CONFIG_SND_AT91_SOC_SSC) += snd-soc-at91-ssc.o
diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c
deleted file mode 100644
index 7ab48bd..0000000
--- a/sound/soc/at91/at91-pcm.c
+++ /dev/null
@@ -1,434 +0,0 @@
-/*
- * at91-pcm.c -- ALSA PCM interface for the Atmel AT91 SoC
- *
- * Author: Frank Mandarino <fmandarino@endrelia.com>
- * Endrelia Technologies Inc.
- * Created: Mar 3, 2006
- *
- * Based on pxa2xx-pcm.c by:
- *
- * Author: Nicolas Pitre
- * Created: Nov 30, 2004
- * Copyright: (C) 2004 MontaVista Software, Inc.
- *
- * 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/init.h>
-#include <linux/platform_device.h>
-#include <linux/slab.h>
-#include <linux/dma-mapping.h>
-#include <linux/atmel_pdc.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-
-#include <mach/hardware.h>
-#include <mach/at91_ssc.h>
-
-#include "at91-pcm.h"
-
-#if 0
-#define DBG(x...) printk(KERN_INFO "at91-pcm: " x)
-#else
-#define DBG(x...)
-#endif
-
-static const struct snd_pcm_hardware at91_pcm_hardware = {
- .info = SNDRV_PCM_INFO_MMAP |
- SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_PAUSE,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
- .period_bytes_min = 32,
- .period_bytes_max = 8192,
- .periods_min = 2,
- .periods_max = 1024,
- .buffer_bytes_max = 32 * 1024,
-};
-
-struct at91_runtime_data {
- struct at91_pcm_dma_params *params;
- dma_addr_t dma_buffer; /* physical address of dma buffer */
- dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */
- size_t period_size;
- dma_addr_t period_ptr; /* physical address of next period */
- u32 pdc_xpr_save; /* PDC register save */
- u32 pdc_xcr_save;
- u32 pdc_xnpr_save;
- u32 pdc_xncr_save;
-};
-
-static void at91_pcm_dma_irq(u32 ssc_sr,
- struct snd_pcm_substream *substream)
-{
- struct at91_runtime_data *prtd = substream->runtime->private_data;
- struct at91_pcm_dma_params *params = prtd->params;
- static int count = 0;
-
- count++;
-
- if (ssc_sr & params->mask->ssc_endbuf) {
-
- printk(KERN_WARNING
- "at91-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n",
- substream->stream == SNDRV_PCM_STREAM_PLAYBACK
- ? "underrun" : "overrun",
- params->name, ssc_sr, count);
-
- /* re-start the PDC */
- at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable);
-
- prtd->period_ptr += prtd->period_size;
- if (prtd->period_ptr >= prtd->dma_buffer_end) {
- prtd->period_ptr = prtd->dma_buffer;
- }
-
- at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->period_ptr);
- at91_ssc_write(params->ssc_base + params->pdc->xcr,
- prtd->period_size / params->pdc_xfer_size);
-
- at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable);
- }
-
- if (ssc_sr & params->mask->ssc_endx) {
-
- /* Load the PDC next pointer and counter registers */
- prtd->period_ptr += prtd->period_size;
- if (prtd->period_ptr >= prtd->dma_buffer_end) {
- prtd->period_ptr = prtd->dma_buffer;
- }
- at91_ssc_write(params->ssc_base + params->pdc->xnpr,
- prtd->period_ptr);
- at91_ssc_write(params->ssc_base + params->pdc->xncr,
- prtd->period_size / params->pdc_xfer_size);
- }
-
- snd_pcm_period_elapsed(substream);
-}
-
-static int at91_pcm_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct at91_runtime_data *prtd = runtime->private_data;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
-
- /* this may get called several times by oss emulation
- * with different params */
-
- snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
- runtime->dma_bytes = params_buffer_bytes(params);
-
- prtd->params = rtd->dai->cpu_dai->dma_data;
- prtd->params->dma_intr_handler = at91_pcm_dma_irq;
-
- prtd->dma_buffer = runtime->dma_addr;
- prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes;
- prtd->period_size = params_period_bytes(params);
-
- DBG("hw_params: DMA for %s initialized (dma_bytes=%d, period_size=%d)\n",
- prtd->params->name, runtime->dma_bytes, prtd->period_size);
- return 0;
-}
-
-static int at91_pcm_hw_free(struct snd_pcm_substream *substream)
-{
- struct at91_runtime_data *prtd = substream->runtime->private_data;
- struct at91_pcm_dma_params *params = prtd->params;
-
- if (params != NULL) {
- at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable);
- prtd->params->dma_intr_handler = NULL;
- }
-
- return 0;
-}
-
-static int at91_pcm_prepare(struct snd_pcm_substream *substream)
-{
- struct at91_runtime_data *prtd = substream->runtime->private_data;
- struct at91_pcm_dma_params *params = prtd->params;
-
- at91_ssc_write(params->ssc_base + AT91_SSC_IDR,
- params->mask->ssc_endx | params->mask->ssc_endbuf);
-
- at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable);
- return 0;
-}
-
-static int at91_pcm_trigger(struct snd_pcm_substream *substream,
- int cmd)
-{
- struct at91_runtime_data *prtd = substream->runtime->private_data;
- struct at91_pcm_dma_params *params = prtd->params;
- int ret = 0;
-
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- prtd->period_ptr = prtd->dma_buffer;
-
- at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->period_ptr);
- at91_ssc_write(params->ssc_base + params->pdc->xcr,
- prtd->period_size / params->pdc_xfer_size);
-
- prtd->period_ptr += prtd->period_size;
- at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->period_ptr);
- at91_ssc_write(params->ssc_base + params->pdc->xncr,
- prtd->period_size / params->pdc_xfer_size);
-
- DBG("trigger: period_ptr=%lx, xpr=%lx, xcr=%ld, xnpr=%lx, xncr=%ld\n",
- (unsigned long) prtd->period_ptr,
- at91_ssc_read(params->ssc_base + params->pdc->xpr),
- at91_ssc_read(params->ssc_base + params->pdc->xcr),
- at91_ssc_read(params->ssc_base + params->pdc->xnpr),
- at91_ssc_read(params->ssc_base + params->pdc->xncr));
-
- at91_ssc_write(params->ssc_base + AT91_SSC_IER,
- params->mask->ssc_endx | params->mask->ssc_endbuf);
-
- at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR,
- params->mask->pdc_enable);
-
- DBG("sr=%lx imr=%lx\n",
- at91_ssc_read(params->ssc_base + AT91_SSC_SR),
- at91_ssc_read(params->ssc_base + AT91_SSC_IMR));
- break;
-
- case SNDRV_PCM_TRIGGER_STOP:
- case SNDRV_PCM_TRIGGER_SUSPEND:
- case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable);
- break;
-
- case SNDRV_PCM_TRIGGER_RESUME:
- case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable);
- break;
-
- default:
- ret = -EINVAL;
- }
-
- return ret;
-}
-
-static snd_pcm_uframes_t at91_pcm_pointer(
- struct snd_pcm_substream *substream)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct at91_runtime_data *prtd = runtime->private_data;
- struct at91_pcm_dma_params *params = prtd->params;
- dma_addr_t ptr;
- snd_pcm_uframes_t x;
-
- ptr = (dma_addr_t) at91_ssc_read(params->ssc_base + params->pdc->xpr);
- x = bytes_to_frames(runtime, ptr - prtd->dma_buffer);
-
- if (x == runtime->buffer_size)
- x = 0;
- return x;
-}
-
-static int at91_pcm_open(struct snd_pcm_substream *substream)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct at91_runtime_data *prtd;
- int ret = 0;
-
- snd_soc_set_runtime_hwparams(substream, &at91_pcm_hardware);
-
- /* ensure that buffer size is a multiple of period size */
- ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
- if (ret < 0)
- goto out;
-
- prtd = kzalloc(sizeof(struct at91_runtime_data), GFP_KERNEL);
- if (prtd == NULL) {
- ret = -ENOMEM;
- goto out;
- }
- runtime->private_data = prtd;
-
- out:
- return ret;
-}
-
-static int at91_pcm_close(struct snd_pcm_substream *substream)
-{
- struct at91_runtime_data *prtd = substream->runtime->private_data;
-
- kfree(prtd);
- return 0;
-}
-
-static int at91_pcm_mmap(struct snd_pcm_substream *substream,
- struct vm_area_struct *vma)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
-
- return dma_mmap_writecombine(substream->pcm->card->dev, vma,
- runtime->dma_area,
- runtime->dma_addr,
- runtime->dma_bytes);
-}
-
-struct snd_pcm_ops at91_pcm_ops = {
- .open = at91_pcm_open,
- .close = at91_pcm_close,
- .ioctl = snd_pcm_lib_ioctl,
- .hw_params = at91_pcm_hw_params,
- .hw_free = at91_pcm_hw_free,
- .prepare = at91_pcm_prepare,
- .trigger = at91_pcm_trigger,
- .pointer = at91_pcm_pointer,
- .mmap = at91_pcm_mmap,
-};
-
-static int at91_pcm_preallocate_dma_buffer(struct snd_pcm *pcm,
- int stream)
-{
- struct snd_pcm_substream *substream = pcm->streams[stream].substream;
- struct snd_dma_buffer *buf = &substream->dma_buffer;
- size_t size = at91_pcm_hardware.buffer_bytes_max;
-
- buf->dev.type = SNDRV_DMA_TYPE_DEV;
- buf->dev.dev = pcm->card->dev;
- buf->private_data = NULL;
- buf->area = dma_alloc_writecombine(pcm->card->dev, size,
- &buf->addr, GFP_KERNEL);
-
- DBG("preallocate_dma_buffer: area=%p, addr=%p, size=%d\n",
- (void *) buf->area,
- (void *) buf->addr,
- size);
-
- if (!buf->area)
- return -ENOMEM;
-
- buf->bytes = size;
- return 0;
-}
-
-static u64 at91_pcm_dmamask = 0xffffffff;
-
-static int at91_pcm_new(struct snd_card *card,
- struct snd_soc_dai *dai, struct snd_pcm *pcm)
-{
- int ret = 0;
-
- if (!card->dev->dma_mask)
- card->dev->dma_mask = &at91_pcm_dmamask;
- if (!card->dev->coherent_dma_mask)
- card->dev->coherent_dma_mask = 0xffffffff;
-
- if (dai->playback.channels_min) {
- ret = at91_pcm_preallocate_dma_buffer(pcm,
- SNDRV_PCM_STREAM_PLAYBACK);
- if (ret)
- goto out;
- }
-
- if (dai->capture.channels_min) {
- ret = at91_pcm_preallocate_dma_buffer(pcm,
- SNDRV_PCM_STREAM_CAPTURE);
- if (ret)
- goto out;
- }
- out:
- return ret;
-}
-
-static void at91_pcm_free_dma_buffers(struct snd_pcm *pcm)
-{
- struct snd_pcm_substream *substream;
- struct snd_dma_buffer *buf;
- int stream;
-
- for (stream = 0; stream < 2; stream++) {
- substream = pcm->streams[stream].substream;
- if (!substream)
- continue;
-
- buf = &substream->dma_buffer;
- if (!buf->area)
- continue;
-
- dma_free_writecombine(pcm->card->dev, buf->bytes,
- buf->area, buf->addr);
- buf->area = NULL;
- }
-}
-
-#ifdef CONFIG_PM
-static int at91_pcm_suspend(struct platform_device *pdev,
- struct snd_soc_dai *dai)
-{
- struct snd_pcm_runtime *runtime = dai->runtime;
- struct at91_runtime_data *prtd;
- struct at91_pcm_dma_params *params;
-
- if (!runtime)
- return 0;
-
- prtd = runtime->private_data;
- params = prtd->params;
-
- /* disable the PDC and save the PDC registers */
-
- at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable);
-
- prtd->pdc_xpr_save = at91_ssc_read(params->ssc_base + params->pdc->xpr);
- prtd->pdc_xcr_save = at91_ssc_read(params->ssc_base + params->pdc->xcr);
- prtd->pdc_xnpr_save = at91_ssc_read(params->ssc_base + params->pdc->xnpr);
- prtd->pdc_xncr_save = at91_ssc_read(params->ssc_base + params->pdc->xncr);
-
- return 0;
-}
-
-static int at91_pcm_resume(struct platform_device *pdev,
- struct snd_soc_dai *dai)
-{
- struct snd_pcm_runtime *runtime = dai->runtime;
- struct at91_runtime_data *prtd;
- struct at91_pcm_dma_params *params;
-
- if (!runtime)
- return 0;
-
- prtd = runtime->private_data;
- params = prtd->params;
-
- /* restore the PDC registers and enable the PDC */
- at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->pdc_xpr_save);
- at91_ssc_write(params->ssc_base + params->pdc->xcr, prtd->pdc_xcr_save);
- at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->pdc_xnpr_save);
- at91_ssc_write(params->ssc_base + params->pdc->xncr, prtd->pdc_xncr_save);
-
- at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable);
- return 0;
-}
-#else
-#define at91_pcm_suspend NULL
-#define at91_pcm_resume NULL
-#endif
-
-struct snd_soc_platform at91_soc_platform = {
- .name = "at91-audio",
- .pcm_ops = &at91_pcm_ops,
- .pcm_new = at91_pcm_new,
- .pcm_free = at91_pcm_free_dma_buffers,
- .suspend = at91_pcm_suspend,
- .resume = at91_pcm_resume,
-};
-
-EXPORT_SYMBOL_GPL(at91_soc_platform);
-
-MODULE_AUTHOR("Frank Mandarino <fmandarino@endrelia.com>");
-MODULE_DESCRIPTION("Atmel AT91 PCM module");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/at91/at91-pcm.h b/sound/soc/at91/at91-pcm.h
deleted file mode 100644
index e5aada2..0000000
--- a/sound/soc/at91/at91-pcm.h
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
- * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC
- *
- * Author: Frank Mandarino <fmandarino@endrelia.com>
- * Endrelia Technologies Inc.
- * Created: Mar 3, 2006
- *
- * Based on pxa2xx-pcm.h by:
- *
- * Author: Nicolas Pitre
- * Created: Nov 30, 2004
- * Copyright: MontaVista Software, Inc.
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
-
-#ifndef _AT91_PCM_H
-#define _AT91_PCM_H
-
-#include <mach/hardware.h>
-
-struct at91_ssc_periph {
- void __iomem *base;
- u32 pid;
-};
-
-/*
- * Registers and status bits that are required by the PCM driver.
- */
-struct at91_pdc_regs {
- unsigned int xpr; /* PDC recv/trans pointer */
- unsigned int xcr; /* PDC recv/trans counter */
- unsigned int xnpr; /* PDC next recv/trans pointer */
- unsigned int xncr; /* PDC next recv/trans counter */
- unsigned int ptcr; /* PDC transfer control */
-};
-
-struct at91_ssc_mask {
- u32 ssc_enable; /* SSC recv/trans enable */
- u32 ssc_disable; /* SSC recv/trans disable */
- u32 ssc_endx; /* SSC ENDTX or ENDRX */
- u32 ssc_endbuf; /* SSC TXBUFE or RXBUFF */
- u32 pdc_enable; /* PDC recv/trans enable */
- u32 pdc_disable; /* PDC recv/trans disable */
-};
-
-/*
- * This structure, shared between the PCM driver and the interface,
- * contains all information required by the PCM driver to perform the
- * PDC DMA operation. All fields except dma_intr_handler() are initialized
- * by the interface. The dms_intr_handler() pointer is set by the PCM
- * driver and called by the interface SSC interrupt handler if it is
- * non-NULL.
- */
-struct at91_pcm_dma_params {
- char *name; /* stream identifier */
- int pdc_xfer_size; /* PDC counter increment in bytes */
- void __iomem *ssc_base; /* SSC base address */
- struct at91_pdc_regs *pdc; /* PDC receive or transmit registers */
- struct at91_ssc_mask *mask;/* SSC & PDC status bits */
- struct snd_pcm_substream *substream;
- void (*dma_intr_handler)(u32, struct snd_pcm_substream *);
-};
-
-extern struct snd_soc_platform at91_soc_platform;
-
-#define at91_ssc_read(a) ((unsigned long) __raw_readl(a))
-#define at91_ssc_write(a,v) __raw_writel((v),(a))
-
-#endif /* _AT91_PCM_H */
diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c
deleted file mode 100644
index 1b61cc4..0000000
--- a/sound/soc/at91/at91-ssc.c
+++ /dev/null
@@ -1,791 +0,0 @@
-/*
- * at91-ssc.c -- ALSA SoC AT91 SSC Audio Layer Platform driver
- *
- * Author: Frank Mandarino <fmandarino@endrelia.com>
- * Endrelia Technologies Inc.
- *
- * Based on pxa2xx Platform drivers by
- * Liam Girdwood <lrg@slimlogic.co.uk>
- *
- * 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/init.h>
-#include <linux/module.h>
-#include <linux/interrupt.h>
-#include <linux/device.h>
-#include <linux/delay.h>
-#include <linux/clk.h>
-#include <linux/atmel_pdc.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/initval.h>
-#include <sound/soc.h>
-
-#include <mach/hardware.h>
-#include <mach/at91_pmc.h>
-#include <mach/at91_ssc.h>
-
-#include "at91-pcm.h"
-#include "at91-ssc.h"
-
-#if 0
-#define DBG(x...) printk(KERN_DEBUG "at91-ssc:" x)
-#else
-#define DBG(x...)
-#endif
-
-#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20)
-#define NUM_SSC_DEVICES 1
-#else
-#define NUM_SSC_DEVICES 3
-#endif
-
-
-/*
- * SSC PDC registers required by the PCM DMA engine.
- */
-static struct at91_pdc_regs pdc_tx_reg = {
- .xpr = ATMEL_PDC_TPR,
- .xcr = ATMEL_PDC_TCR,
- .xnpr = ATMEL_PDC_TNPR,
- .xncr = ATMEL_PDC_TNCR,
-};
-
-static struct at91_pdc_regs pdc_rx_reg = {
- .xpr = ATMEL_PDC_RPR,
- .xcr = ATMEL_PDC_RCR,
- .xnpr = ATMEL_PDC_RNPR,
- .xncr = ATMEL_PDC_RNCR,
-};
-
-/*
- * SSC & PDC status bits for transmit and receive.
- */
-static struct at91_ssc_mask ssc_tx_mask = {
- .ssc_enable = AT91_SSC_TXEN,
- .ssc_disable = AT91_SSC_TXDIS,
- .ssc_endx = AT91_SSC_ENDTX,
- .ssc_endbuf = AT91_SSC_TXBUFE,
- .pdc_enable = ATMEL_PDC_TXTEN,
- .pdc_disable = ATMEL_PDC_TXTDIS,
-};
-
-static struct at91_ssc_mask ssc_rx_mask = {
- .ssc_enable = AT91_SSC_RXEN,
- .ssc_disable = AT91_SSC_RXDIS,
- .ssc_endx = AT91_SSC_ENDRX,
- .ssc_endbuf = AT91_SSC_RXBUFF,
- .pdc_enable = ATMEL_PDC_RXTEN,
- .pdc_disable = ATMEL_PDC_RXTDIS,
-};
-
-
-/*
- * DMA parameters.
- */
-static struct at91_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = {
- {{
- .name = "SSC0 PCM out",
- .pdc = &pdc_tx_reg,
- .mask = &ssc_tx_mask,
- },
- {
- .name = "SSC0 PCM in",
- .pdc = &pdc_rx_reg,
- .mask = &ssc_rx_mask,
- }},
-#if NUM_SSC_DEVICES == 3
- {{
- .name = "SSC1 PCM out",
- .pdc = &pdc_tx_reg,
- .mask = &ssc_tx_mask,
- },
- {
- .name = "SSC1 PCM in",
- .pdc = &pdc_rx_reg,
- .mask = &ssc_rx_mask,
- }},
- {{
- .name = "SSC2 PCM out",
- .pdc = &pdc_tx_reg,
- .mask = &ssc_tx_mask,
- },
- {
- .name = "SSC2 PCM in",
- .pdc = &pdc_rx_reg,
- .mask = &ssc_rx_mask,
- }},
-#endif
-};
-
-struct at91_ssc_state {
- u32 ssc_cmr;
- u32 ssc_rcmr;
- u32 ssc_rfmr;
- u32 ssc_tcmr;
- u32 ssc_tfmr;
- u32 ssc_sr;
- u32 ssc_imr;
-};
-
-static struct at91_ssc_info {
- char *name;
- struct at91_ssc_periph ssc;
- spinlock_t lock; /* lock for dir_mask */
- unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */
- unsigned short initialized; /* 1=SSC has been initialized */
- unsigned short daifmt;
- unsigned short cmr_div;
- unsigned short tcmr_period;
- unsigned short rcmr_period;
- struct at91_pcm_dma_params *dma_params[2];
- struct at91_ssc_state ssc_state;
-
-} ssc_info[NUM_SSC_DEVICES] = {
- {
- .name = "ssc0",
- .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock),
- .dir_mask = 0,
- .initialized = 0,
- },
-#if NUM_SSC_DEVICES == 3
- {
- .name = "ssc1",
- .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock),
- .dir_mask = 0,
- .initialized = 0,
- },
- {
- .name = "ssc2",
- .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock),
- .dir_mask = 0,
- .initialized = 0,
- },
-#endif
-};
-
-static unsigned int at91_ssc_sysclk;
-
-/*
- * SSC interrupt handler. Passes PDC interrupts to the DMA
- * interrupt handler in the PCM driver.
- */
-static irqreturn_t at91_ssc_interrupt(int irq, void *dev_id)
-{
- struct at91_ssc_info *ssc_p = dev_id;
- struct at91_pcm_dma_params *dma_params;
- u32 ssc_sr;
- int i;
-
- ssc_sr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR)
- & at91_ssc_read(ssc_p->ssc.base + AT91_SSC_IMR);
-
- /*
- * Loop through the substreams attached to this SSC. If
- * a DMA-related interrupt occurred on that substream, call
- * the DMA interrupt handler function, if one has been
- * registered in the dma_params structure by the PCM driver.
- */
- for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) {
- dma_params = ssc_p->dma_params[i];
-
- if (dma_params != NULL && dma_params->dma_intr_handler != NULL &&
- (ssc_sr &
- (dma_params->mask->ssc_endx | dma_params->mask->ssc_endbuf)))
-
- dma_params->dma_intr_handler(ssc_sr, dma_params->substream);
- }
-
- return IRQ_HANDLED;
-}
-
-/*
- * Startup. Only that one substream allowed in each direction.
- */
-static int at91_ssc_startup(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
- int dir_mask;
-
- DBG("ssc_startup: SSC_SR=0x%08lx\n",
- at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR));
- dir_mask = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0x1 : 0x2;
-
- spin_lock_irq(&ssc_p->lock);
- if (ssc_p->dir_mask & dir_mask) {
- spin_unlock_irq(&ssc_p->lock);
- return -EBUSY;
- }
- ssc_p->dir_mask |= dir_mask;
- spin_unlock_irq(&ssc_p->lock);
-
- return 0;
-}
-
-/*
- * Shutdown. Clear DMA parameters and shutdown the SSC if there
- * are no other substreams open.
- */
-static void at91_ssc_shutdown(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
- struct at91_pcm_dma_params *dma_params;
- int dir, dir_mask;
-
- dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
- dma_params = ssc_p->dma_params[dir];
-
- if (dma_params != NULL) {
- at91_ssc_write(dma_params->ssc_base + AT91_SSC_CR,
- dma_params->mask->ssc_disable);
- DBG("%s disabled SSC_SR=0x%08lx\n", (dir ? "receive" : "transmit"),
- at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR));
-
- dma_params->ssc_base = NULL;
- dma_params->substream = NULL;
- ssc_p->dma_params[dir] = NULL;
- }
-
- dir_mask = 1 << dir;
-
- spin_lock_irq(&ssc_p->lock);
- ssc_p->dir_mask &= ~dir_mask;
- if (!ssc_p->dir_mask) {
- /* Shutdown the SSC clock. */
- DBG("Stopping pid %d clock\n", ssc_p->ssc.pid);
- at91_sys_write(AT91_PMC_PCDR, 1<<ssc_p->ssc.pid);
-
- if (ssc_p->initialized) {
- free_irq(ssc_p->ssc.pid, ssc_p);
- ssc_p->initialized = 0;
- }
-
- /* Reset the SSC */
- at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, AT91_SSC_SWRST);
-
- /* Clear the SSC dividers */
- ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0;
- }
- spin_unlock_irq(&ssc_p->lock);
-}
-
-/*
- * Record the SSC system clock rate.
- */
-static int at91_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
- int clk_id, unsigned int freq, int dir)
-{
- /*
- * The only clock supplied to the SSC is the AT91 master clock,
- * which is only used if the SSC is generating BCLK and/or
- * LRC clocks.
- */
- switch (clk_id) {
- case AT91_SYSCLK_MCK:
- at91_ssc_sysclk = freq;
- break;
- default:
- return -EINVAL;
- }
-
- return 0;
-}
-
-/*
- * Record the DAI format for use in hw_params().
- */
-static int at91_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai,
- unsigned int fmt)
-{
- struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
-
- ssc_p->daifmt = fmt;
- return 0;
-}
-
-/*
- * Record SSC clock dividers for use in hw_params().
- */
-static int at91_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
- int div_id, int div)
-{
- struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
-
- switch (div_id) {
- case AT91SSC_CMR_DIV:
- /*
- * The same master clock divider is used for both
- * transmit and receive, so if a value has already
- * been set, it must match this value.
- */
- if (ssc_p->cmr_div == 0)
- ssc_p->cmr_div = div;
- else
- if (div != ssc_p->cmr_div)
- return -EBUSY;
- break;
-
- case AT91SSC_TCMR_PERIOD:
- ssc_p->tcmr_period = div;
- break;
-
- case AT91SSC_RCMR_PERIOD:
- ssc_p->rcmr_period = div;
- break;
-
- default:
- return -EINVAL;
- }
-
- return 0;
-}
-
-/*
- * Configure the SSC.
- */
-static int at91_ssc_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- int id = rtd->dai->cpu_dai->id;
- struct at91_ssc_info *ssc_p = &ssc_info[id];
- struct at91_pcm_dma_params *dma_params;
- int dir, channels, bits;
- u32 tfmr, rfmr, tcmr, rcmr;
- int start_event;
- int ret;
-
- /*
- * Currently, there is only one set of dma params for
- * each direction. If more are added, this code will
- * have to be changed to select the proper set.
- */
- dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
-
- dma_params = &ssc_dma_params[id][dir];
- dma_params->ssc_base = ssc_p->ssc.base;
- dma_params->substream = substream;
-
- ssc_p->dma_params[dir] = dma_params;
-
- /*
- * The cpu_dai->dma_data field is only used to communicate the
- * appropriate DMA parameters to the pcm driver hw_params()
- * function. It should not be used for other purposes
- * as it is common to all substreams.
- */
- rtd->dai->cpu_dai->dma_data = dma_params;
-
- channels = params_channels(params);
-
- /*
- * Determine sample size in bits and the PDC increment.
- */
- switch(params_format(params)) {
- case SNDRV_PCM_FORMAT_S8:
- bits = 8;
- dma_params->pdc_xfer_size = 1;
- break;
- case SNDRV_PCM_FORMAT_S16_LE:
- bits = 16;
- dma_params->pdc_xfer_size = 2;
- break;
- case SNDRV_PCM_FORMAT_S24_LE:
- bits = 24;
- dma_params->pdc_xfer_size = 4;
- break;
- case SNDRV_PCM_FORMAT_S32_LE:
- bits = 32;
- dma_params->pdc_xfer_size = 4;
- break;
- default:
- printk(KERN_WARNING "at91-ssc: unsupported PCM format\n");
- return -EINVAL;
- }
-
- /*
- * The SSC only supports up to 16-bit samples in I2S format, due
- * to the size of the Frame Mode Register FSLEN field.
- */
- if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S
- && bits > 16) {
- printk(KERN_WARNING
- "at91-ssc: sample size %d is too large for I2S\n", bits);
- return -EINVAL;
- }
-
- /*
- * Compute SSC register settings.
- */
- switch (ssc_p->daifmt
- & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) {
-
- case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS:
- /*
- * I2S format, SSC provides BCLK and LRC clocks.
- *
- * The SSC transmit and receive clocks are generated from the
- * MCK divider, and the BCLK signal is output on the SSC TK line.
- */
- rcmr = (( ssc_p->rcmr_period << 24) & AT91_SSC_PERIOD)
- | (( 1 << 16) & AT91_SSC_STTDLY)
- | (( AT91_SSC_START_FALLING_RF ) & AT91_SSC_START)
- | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI)
- | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO)
- | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS);
-
- rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE)
- | (( AT91_SSC_FSOS_NEGATIVE ) & AT91_SSC_FSOS)
- | (((bits - 1) << 16) & AT91_SSC_FSLEN)
- | (((channels - 1) << 8) & AT91_SSC_DATNB)
- | (( 1 << 7) & AT91_SSC_MSBF)
- | (( 0 << 5) & AT91_SSC_LOOP)
- | (((bits - 1) << 0) & AT91_SSC_DATALEN);
-
- tcmr = (( ssc_p->tcmr_period << 24) & AT91_SSC_PERIOD)
- | (( 1 << 16) & AT91_SSC_STTDLY)
- | (( AT91_SSC_START_FALLING_RF ) & AT91_SSC_START)
- | (( AT91_SSC_CKI_FALLING ) & AT91_SSC_CKI)
- | (( AT91_SSC_CKO_CONTINUOUS ) & AT91_SSC_CKO)
- | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS);
-
- tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE)
- | (( 0 << 23) & AT91_SSC_FSDEN)
- | (( AT91_SSC_FSOS_NEGATIVE ) & AT91_SSC_FSOS)
- | (((bits - 1) << 16) & AT91_SSC_FSLEN)
- | (((channels - 1) << 8) & AT91_SSC_DATNB)
- | (( 1 << 7) & AT91_SSC_MSBF)
- | (( 0 << 5) & AT91_SSC_DATDEF)
- | (((bits - 1) << 0) & AT91_SSC_DATALEN);
- break;
-
- case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM:
- /*
- * I2S format, CODEC supplies BCLK and LRC clocks.
- *
- * The SSC transmit clock is obtained from the BCLK signal on
- * on the TK line, and the SSC receive clock is generated from the
- * transmit clock.
- *
- * For single channel data, one sample is transferred on the falling
- * edge of the LRC clock. For two channel data, one sample is
- * transferred on both edges of the LRC clock.
- */
- start_event = channels == 1
- ? AT91_SSC_START_FALLING_RF
- : AT91_SSC_START_EDGE_RF;
-
- rcmr = (( 0 << 24) & AT91_SSC_PERIOD)
- | (( 1 << 16) & AT91_SSC_STTDLY)
- | (( start_event ) & AT91_SSC_START)
- | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI)
- | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO)
- | (( AT91_SSC_CKS_CLOCK ) & AT91_SSC_CKS);
-
- rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE)
- | (( AT91_SSC_FSOS_NONE ) & AT91_SSC_FSOS)
- | (( 0 << 16) & AT91_SSC_FSLEN)
- | (( 0 << 8) & AT91_SSC_DATNB)
- | (( 1 << 7) & AT91_SSC_MSBF)
- | (( 0 << 5) & AT91_SSC_LOOP)
- | (((bits - 1) << 0) & AT91_SSC_DATALEN);
-
- tcmr = (( 0 << 24) & AT91_SSC_PERIOD)
- | (( 1 << 16) & AT91_SSC_STTDLY)
- | (( start_event ) & AT91_SSC_START)
- | (( AT91_SSC_CKI_FALLING ) & AT91_SSC_CKI)
- | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO)
- | (( AT91_SSC_CKS_PIN ) & AT91_SSC_CKS);
-
- tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE)
- | (( 0 << 23) & AT91_SSC_FSDEN)
- | (( AT91_SSC_FSOS_NONE ) & AT91_SSC_FSOS)
- | (( 0 << 16) & AT91_SSC_FSLEN)
- | (( 0 << 8) & AT91_SSC_DATNB)
- | (( 1 << 7) & AT91_SSC_MSBF)
- | (( 0 << 5) & AT91_SSC_DATDEF)
- | (((bits - 1) << 0) & AT91_SSC_DATALEN);
- break;
-
- case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS:
- /*
- * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks.
- *
- * The SSC transmit and receive clocks are generated from the
- * MCK divider, and the BCLK signal is output on the SSC TK line.
- */
- rcmr = (( ssc_p->rcmr_period << 24) & AT91_SSC_PERIOD)
- | (( 1 << 16) & AT91_SSC_STTDLY)
- | (( AT91_SSC_START_RISING_RF ) & AT91_SSC_START)
- | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI)
- | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO)
- | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS);
-
- rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE)
- | (( AT91_SSC_FSOS_POSITIVE ) & AT91_SSC_FSOS)
- | (( 0 << 16) & AT91_SSC_FSLEN)
- | (((channels - 1) << 8) & AT91_SSC_DATNB)
- | (( 1 << 7) & AT91_SSC_MSBF)
- | (( 0 << 5) & AT91_SSC_LOOP)
- | (((bits - 1) << 0) & AT91_SSC_DATALEN);
-
- tcmr = (( ssc_p->tcmr_period << 24) & AT91_SSC_PERIOD)
- | (( 1 << 16) & AT91_SSC_STTDLY)
- | (( AT91_SSC_START_RISING_RF ) & AT91_SSC_START)
- | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI)
- | (( AT91_SSC_CKO_CONTINUOUS ) & AT91_SSC_CKO)
- | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS);
-
- tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE)
- | (( 0 << 23) & AT91_SSC_FSDEN)
- | (( AT91_SSC_FSOS_POSITIVE ) & AT91_SSC_FSOS)
- | (( 0 << 16) & AT91_SSC_FSLEN)
- | (((channels - 1) << 8) & AT91_SSC_DATNB)
- | (( 1 << 7) & AT91_SSC_MSBF)
- | (( 0 << 5) & AT91_SSC_DATDEF)
- | (((bits - 1) << 0) & AT91_SSC_DATALEN);
-
-
-
- break;
-
- case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM:
- default:
- printk(KERN_WARNING "at91-ssc: unsupported DAI format 0x%x.\n",
- ssc_p->daifmt);
- return -EINVAL;
- break;
- }
- DBG("RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", rcmr, rfmr, tcmr, tfmr);
-
- if (!ssc_p->initialized) {
-
- /* Enable PMC peripheral clock for this SSC */
- DBG("Starting pid %d clock\n", ssc_p->ssc.pid);
- at91_sys_write(AT91_PMC_PCER, 1<<ssc_p->ssc.pid);
-
- /* Reset the SSC and its PDC registers */
- at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, AT91_SSC_SWRST);
-
- at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RPR, 0);
- at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RCR, 0);
- at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RNPR, 0);
- at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RNCR, 0);
- at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TPR, 0);
- at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TCR, 0);
- at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TNPR, 0);
- at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TNCR, 0);
-
- if ((ret = request_irq(ssc_p->ssc.pid, at91_ssc_interrupt,
- 0, ssc_p->name, ssc_p)) < 0) {
- printk(KERN_WARNING "at91-ssc: request_irq failure\n");
-
- DBG("Stopping pid %d clock\n", ssc_p->ssc.pid);
- at91_sys_write(AT91_PMC_PCDR, 1<<ssc_p->ssc.pid);
- return ret;
- }
-
- ssc_p->initialized = 1;
- }
-
- /* set SSC clock mode register */
- at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CMR, ssc_p->cmr_div);
-
- /* set receive clock mode and format */
- at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RCMR, rcmr);
- at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RFMR, rfmr);
-
- /* set transmit clock mode and format */
- at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TCMR, tcmr);
- at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TFMR, tfmr);
-
- DBG("hw_params: SSC initialized\n");
- return 0;
-}
-
-
-static int at91_ssc_prepare(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
- struct at91_pcm_dma_params *dma_params;
- int dir;
-
- dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
- dma_params = ssc_p->dma_params[dir];
-
- at91_ssc_write(dma_params->ssc_base + AT91_SSC_CR,
- dma_params->mask->ssc_enable);
-
- DBG("%s enabled SSC_SR=0x%08lx\n", dir ? "receive" : "transmit",
- at91_ssc_read(dma_params->ssc_base + AT91_SSC_SR));
- return 0;
-}
-
-
-#ifdef CONFIG_PM
-static int at91_ssc_suspend(struct platform_device *pdev,
- struct snd_soc_dai *cpu_dai)
-{
- struct at91_ssc_info *ssc_p;
-
- if(!cpu_dai->active)
- return 0;
-
- ssc_p = &ssc_info[cpu_dai->id];
-
- /* Save the status register before disabling transmit and receive. */
- ssc_p->ssc_state.ssc_sr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR);
- at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR,
- AT91_SSC_TXDIS | AT91_SSC_RXDIS);
-
- /* Save the current interrupt mask, then disable unmasked interrupts. */
- ssc_p->ssc_state.ssc_imr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_IMR);
- at91_ssc_write(ssc_p->ssc.base + AT91_SSC_IDR, ssc_p->ssc_state.ssc_imr);
-
- ssc_p->ssc_state.ssc_cmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_CMR);
- ssc_p->ssc_state.ssc_rcmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_RCMR);
- ssc_p->ssc_state.ssc_rfmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_RFMR);
- ssc_p->ssc_state.ssc_tcmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_TCMR);
- ssc_p->ssc_state.ssc_tfmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_TFMR);
-
- return 0;
-}
-
-static int at91_ssc_resume(struct platform_device *pdev,
- struct snd_soc_dai *cpu_dai)
-{
- struct at91_ssc_info *ssc_p;
-
- if(!cpu_dai->active)
- return 0;
-
- ssc_p = &ssc_info[cpu_dai->id];
-
- at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TFMR, ssc_p->ssc_state.ssc_tfmr);
- at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TCMR, ssc_p->ssc_state.ssc_tcmr);
- at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RFMR, ssc_p->ssc_state.ssc_rfmr);
- at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RCMR, ssc_p->ssc_state.ssc_rcmr);
- at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CMR, ssc_p->ssc_state.ssc_cmr);
-
- at91_ssc_write(ssc_p->ssc.base + AT91_SSC_IER, ssc_p->ssc_state.ssc_imr);
-
- at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR,
- ((ssc_p->ssc_state.ssc_sr & AT91_SSC_RXENA) ? AT91_SSC_RXEN : 0) |
- ((ssc_p->ssc_state.ssc_sr & AT91_SSC_TXENA) ? AT91_SSC_TXEN : 0));
-
- return 0;
-}
-
-#else
-#define at91_ssc_suspend NULL
-#define at91_ssc_resume NULL
-#endif
-
-#define AT91_SSC_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
- SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
- SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
- SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
- SNDRV_PCM_RATE_96000)
-
-#define AT91_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\
- SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
-
-struct snd_soc_dai at91_ssc_dai[NUM_SSC_DEVICES] = {
- { .name = "at91-ssc0",
- .id = 0,
- .type = SND_SOC_DAI_PCM,
- .suspend = at91_ssc_suspend,
- .resume = at91_ssc_resume,
- .playback = {
- .channels_min = 1,
- .channels_max = 2,
- .rates = AT91_SSC_RATES,
- .formats = AT91_SSC_FORMATS,},
- .capture = {
- .channels_min = 1,
- .channels_max = 2,
- .rates = AT91_SSC_RATES,
- .formats = AT91_SSC_FORMATS,},
- .ops = {
- .startup = at91_ssc_startup,
- .shutdown = at91_ssc_shutdown,
- .prepare = at91_ssc_prepare,
- .hw_params = at91_ssc_hw_params,},
- .dai_ops = {
- .set_sysclk = at91_ssc_set_dai_sysclk,
- .set_fmt = at91_ssc_set_dai_fmt,
- .set_clkdiv = at91_ssc_set_dai_clkdiv,},
- .private_data = &ssc_info[0].ssc,
- },
-#if NUM_SSC_DEVICES == 3
- { .name = "at91-ssc1",
- .id = 1,
- .type = SND_SOC_DAI_PCM,
- .suspend = at91_ssc_suspend,
- .resume = at91_ssc_resume,
- .playback = {
- .channels_min = 1,
- .channels_max = 2,
- .rates = AT91_SSC_RATES,
- .formats = AT91_SSC_FORMATS,},
- .capture = {
- .channels_min = 1,
- .channels_max = 2,
- .rates = AT91_SSC_RATES,
- .formats = AT91_SSC_FORMATS,},
- .ops = {
- .startup = at91_ssc_startup,
- .shutdown = at91_ssc_shutdown,
- .prepare = at91_ssc_prepare,
- .hw_params = at91_ssc_hw_params,},
- .dai_ops = {
- .set_sysclk = at91_ssc_set_dai_sysclk,
- .set_fmt = at91_ssc_set_dai_fmt,
- .set_clkdiv = at91_ssc_set_dai_clkdiv,},
- .private_data = &ssc_info[1].ssc,
- },
- { .name = "at91-ssc2",
- .id = 2,
- .type = SND_SOC_DAI_PCM,
- .suspend = at91_ssc_suspend,
- .resume = at91_ssc_resume,
- .playback = {
- .channels_min = 1,
- .channels_max = 2,
- .rates = AT91_SSC_RATES,
- .formats = AT91_SSC_FORMATS,},
- .capture = {
- .channels_min = 1,
- .channels_max = 2,
- .rates = AT91_SSC_RATES,
- .formats = AT91_SSC_FORMATS,},
- .ops = {
- .startup = at91_ssc_startup,
- .shutdown = at91_ssc_shutdown,
- .prepare = at91_ssc_prepare,
- .hw_params = at91_ssc_hw_params,},
- .dai_ops = {
- .set_sysclk = at91_ssc_set_dai_sysclk,
- .set_fmt = at91_ssc_set_dai_fmt,
- .set_clkdiv = at91_ssc_set_dai_clkdiv,},
- .private_data = &ssc_info[2].ssc,
- },
-#endif
-};
-
-EXPORT_SYMBOL_GPL(at91_ssc_dai);
-
-/* Module information */
-MODULE_AUTHOR("Frank Mandarino, fmandarino@endrelia.com, www.endrelia.com");
-MODULE_DESCRIPTION("AT91 SSC ASoC Interface");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/at91/at91-ssc.h b/sound/soc/at91/at91-ssc.h
deleted file mode 100644
index 6b7bf38..0000000
--- a/sound/soc/at91/at91-ssc.h
+++ /dev/null
@@ -1,27 +0,0 @@
-/*
- * at91-ssc.h - ALSA SSC interface for the Atmel AT91 SoC
- *
- * Author: Frank Mandarino <fmandarino@endrelia.com>
- * Endrelia Technologies Inc.
- * Created: Jan 9, 2007
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
-
-#ifndef _AT91_SSC_H
-#define _AT91_SSC_H
-
-/* SSC system clock ids */
-#define AT91_SYSCLK_MCK 0 /* SSC uses AT91 MCK as system clock */
-
-/* SSC divider ids */
-#define AT91SSC_CMR_DIV 0 /* MCK divider for BCLK */
-#define AT91SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */
-#define AT91SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */
-
-extern struct snd_soc_dai at91_ssc_dai[];
-
-#endif /* _AT91_SSC_H */
-
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
new file mode 100644
index 0000000..a608d70
--- /dev/null
+++ b/sound/soc/atmel/Kconfig
@@ -0,0 +1,43 @@
+config SND_ATMEL_SOC
+ tristate "SoC Audio for the Atmel System-on-Chip"
+ depends on ARCH_AT91 || AVR32
+ help
+ Say Y or M if you want to add support for codecs attached to
+ the ATMEL SSC interface. You will also need
+ to select the audio interfaces to support below.
+
+config SND_ATMEL_SOC_SSC
+ tristate
+ depends on SND_ATMEL_SOC
+ help
+ Say Y or M if you want to add support for codecs the
+ ATMEL SSC interface. You will also needs to select the individual
+ machine drivers to support below.
+
+config SND_AT91_SOC_SAM9G20_WM8731
+ tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board"
+ depends on ATMEL_SSC && ARCH_AT91SAM9G20 && SND_ATMEL_SOC
+ select SND_ATMEL_SOC_SSC
+ select SND_SOC_WM8731
+ help
+ Say Y if you want to add support for SoC audio on WM8731-based
+ AT91sam9g20 evaluation board.
+
+config SND_AT32_SOC_PLAYPAQ
+ tristate "SoC Audio support for PlayPaq with WM8510"
+ depends on SND_ATMEL_SOC && BOARD_PLAYPAQ
+ select SND_ATMEL_SOC_SSC
+ select SND_SOC_WM8510
+ help
+ Say Y or M here if you want to add support for SoC audio
+ on the LRS PlayPaq.
+
+config SND_AT32_SOC_PLAYPAQ_SLAVE
+ bool "Run CODEC on PlayPaq in slave mode"
+ depends on SND_AT32_SOC_PLAYPAQ
+ default n
+ help
+ Say Y if you want to run with the AT32 SSC generating the BCLK
+ and FRAME signals on the PlayPaq. Unless you want to play
+ with the AT32 as the SSC master, you probably want to say N here,
+ as this will give you better sound quality.
diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile
new file mode 100644
index 0000000..f54a7cc
--- /dev/null
+++ b/sound/soc/atmel/Makefile
@@ -0,0 +1,15 @@
+# AT91 Platform Support
+snd-soc-atmel-pcm-objs := atmel-pcm.o
+snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o
+
+obj-$(CONFIG_SND_ATMEL_SOC) += snd-soc-atmel-pcm.o
+obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o
+
+# AT91 Machine Support
+snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o
+
+# AT32 Machine Support
+snd-soc-playpaq-objs := playpaq_wm8510.o
+
+obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
+obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o
diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
new file mode 100644
index 0000000..1fac5ef
--- /dev/null
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -0,0 +1,494 @@
+/*
+ * atmel-pcm.c -- ALSA PCM interface for the Atmel atmel SoC.
+ *
+ * Copyright (C) 2005 SAN People
+ * Copyright (C) 2008 Atmel
+ *
+ * Authors: Sedji Gaouaou <sedji.gaouaou@atmel.com>
+ *
+ * Based on at91-pcm. by:
+ * Frank Mandarino <fmandarino@endrelia.com>
+ * Copyright 2006 Endrelia Technologies Inc.
+ *
+ * Based on pxa2xx-pcm.c by:
+ *
+ * Author: Nicolas Pitre
+ * Created: Nov 30, 2004
+ * Copyright: (C) 2004 MontaVista Software, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <linux/atmel_pdc.h>
+#include <linux/atmel-ssc.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <mach/hardware.h>
+
+#include "atmel-pcm.h"
+
+
+/*--------------------------------------------------------------------------*\
+ * Hardware definition
+\*--------------------------------------------------------------------------*/
+/* TODO: These values were taken from the AT91 platform driver, check
+ * them against real values for AT32
+ */
+static const struct snd_pcm_hardware atmel_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .period_bytes_min = 32,
+ .period_bytes_max = 8192,
+ .periods_min = 2,
+ .periods_max = 1024,
+ .buffer_bytes_max = 32 * 1024,
+};
+
+
+/*--------------------------------------------------------------------------*\
+ * Data types
+\*--------------------------------------------------------------------------*/
+struct atmel_runtime_data {
+ struct atmel_pcm_dma_params *params;
+ dma_addr_t dma_buffer; /* physical address of dma buffer */
+ dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */
+ size_t period_size;
+
+ dma_addr_t period_ptr; /* physical address of next period */
+ int periods; /* period index of period_ptr */
+
+ /* PDC register save */
+ u32 pdc_xpr_save;
+ u32 pdc_xcr_save;
+ u32 pdc_xnpr_save;
+ u32 pdc_xncr_save;
+};
+
+
+/*--------------------------------------------------------------------------*\
+ * Helper functions
+\*--------------------------------------------------------------------------*/
+static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm,
+ int stream)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size = atmel_pcm_hardware.buffer_bytes_max;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = pcm->card->dev;
+ buf->private_data = NULL;
+ buf->area = dma_alloc_coherent(pcm->card->dev, size,
+ &buf->addr, GFP_KERNEL);
+ pr_debug("atmel-pcm:"
+ "preallocate_dma_buffer: area=%p, addr=%p, size=%d\n",
+ (void *) buf->area,
+ (void *) buf->addr,
+ size);
+
+ if (!buf->area)
+ return -ENOMEM;
+
+ buf->bytes = size;
+ return 0;
+}
+/*--------------------------------------------------------------------------*\
+ * ISR
+\*--------------------------------------------------------------------------*/
+static void atmel_pcm_dma_irq(u32 ssc_sr,
+ struct snd_pcm_substream *substream)
+{
+ struct atmel_runtime_data *prtd = substream->runtime->private_data;
+ struct atmel_pcm_dma_params *params = prtd->params;
+ static int count;
+
+ count++;
+
+ if (ssc_sr & params->mask->ssc_endbuf) {
+ pr_warning("atmel-pcm: buffer %s on %s"
+ " (SSC_SR=%#x, count=%d)\n",
+ substream->stream == SNDRV_PCM_STREAM_PLAYBACK
+ ? "underrun" : "overrun",
+ params->name, ssc_sr, count);
+
+ /* re-start the PDC */
+ ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
+ params->mask->pdc_disable);
+ prtd->period_ptr += prtd->period_size;
+ if (prtd->period_ptr >= prtd->dma_buffer_end)
+ prtd->period_ptr = prtd->dma_buffer;
+
+ ssc_writex(params->ssc->regs, params->pdc->xpr,
+ prtd->period_ptr);
+ ssc_writex(params->ssc->regs, params->pdc->xcr,
+ prtd->period_size / params->pdc_xfer_size);
+ ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
+ params->mask->pdc_enable);
+ }
+
+ if (ssc_sr & params->mask->ssc_endx) {
+ /* Load the PDC next pointer and counter registers */
+ prtd->period_ptr += prtd->period_size;
+ if (prtd->period_ptr >= prtd->dma_buffer_end)
+ prtd->period_ptr = prtd->dma_buffer;
+
+ ssc_writex(params->ssc->regs, params->pdc->xnpr,
+ prtd->period_ptr);
+ ssc_writex(params->ssc->regs, params->pdc->xncr,
+ prtd->period_size / params->pdc_xfer_size);
+ }
+
+ snd_pcm_period_elapsed(substream);
+}
+
+
+/*--------------------------------------------------------------------------*\
+ * PCM operations
+\*--------------------------------------------------------------------------*/
+static int atmel_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct atmel_runtime_data *prtd = runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+ /* this may get called several times by oss emulation
+ * with different params */
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ runtime->dma_bytes = params_buffer_bytes(params);
+
+ prtd->params = rtd->dai->cpu_dai->dma_data;
+ prtd->params->dma_intr_handler = atmel_pcm_dma_irq;
+
+ prtd->dma_buffer = runtime->dma_addr;
+ prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes;
+ prtd->period_size = params_period_bytes(params);
+
+ pr_debug("atmel-pcm: "
+ "hw_params: DMA for %s initialized "
+ "(dma_bytes=%u, period_size=%u)\n",
+ prtd->params->name,
+ runtime->dma_bytes,
+ prtd->period_size);
+ return 0;
+}
+
+static int atmel_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct atmel_runtime_data *prtd = substream->runtime->private_data;
+ struct atmel_pcm_dma_params *params = prtd->params;
+
+ if (params != NULL) {
+ ssc_writex(params->ssc->regs, SSC_PDC_PTCR,
+ params->mask->pdc_disable);
+ prtd->params->dma_intr_handler = NULL;
+ }
+
+ return 0;
+}
+
+static int atmel_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct atmel_runtime_data *prtd = substream->runtime->private_data;
+ struct atmel_pcm_dma_params *params = prtd->params;
+
+ ssc_writex(params->ssc->regs, SSC_IDR,
+ params->mask->ssc_endx | params->mask->ssc_endbuf);
+ ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
+ params->mask->pdc_disable);
+ return 0;
+}
+
+static int atmel_pcm_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_pcm_runtime *rtd = substream->runtime;
+ struct atmel_runtime_data *prtd = rtd->private_data;
+ struct atmel_pcm_dma_params *params = prtd->params;
+ int ret = 0;
+
+ pr_debug("atmel-pcm:buffer_size = %ld,"
+ "dma_area = %p, dma_bytes = %u\n",
+ rtd->buffer_size, rtd->dma_area, rtd->dma_bytes);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ prtd->period_ptr = prtd->dma_buffer;
+
+ ssc_writex(params->ssc->regs, params->pdc->xpr,
+ prtd->period_ptr);
+ ssc_writex(params->ssc->regs, params->pdc->xcr,
+ prtd->period_size / params->pdc_xfer_size);
+
+ prtd->period_ptr += prtd->period_size;
+ ssc_writex(params->ssc->regs, params->pdc->xnpr,
+ prtd->period_ptr);
+ ssc_writex(params->ssc->regs, params->pdc->xncr,
+ prtd->period_size / params->pdc_xfer_size);
+
+ pr_debug("atmel-pcm: trigger: "
+ "period_ptr=%lx, xpr=%u, "
+ "xcr=%u, xnpr=%u, xncr=%u\n",
+ (unsigned long)prtd->period_ptr,
+ ssc_readx(params->ssc->regs, params->pdc->xpr),
+ ssc_readx(params->ssc->regs, params->pdc->xcr),
+ ssc_readx(params->ssc->regs, params->pdc->xnpr),
+ ssc_readx(params->ssc->regs, params->pdc->xncr));
+
+ ssc_writex(params->ssc->regs, SSC_IER,
+ params->mask->ssc_endx | params->mask->ssc_endbuf);
+ ssc_writex(params->ssc->regs, SSC_PDC_PTCR,
+ params->mask->pdc_enable);
+
+ pr_debug("sr=%u imr=%u\n",
+ ssc_readx(params->ssc->regs, SSC_SR),
+ ssc_readx(params->ssc->regs, SSC_IER));
+ break; /* SNDRV_PCM_TRIGGER_START */
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
+ params->mask->pdc_disable);
+ break;
+
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
+ params->mask->pdc_enable);
+ break;
+
+ default:
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+static snd_pcm_uframes_t atmel_pcm_pointer(
+ struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct atmel_runtime_data *prtd = runtime->private_data;
+ struct atmel_pcm_dma_params *params = prtd->params;
+ dma_addr_t ptr;
+ snd_pcm_uframes_t x;
+
+ ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr);
+ x = bytes_to_frames(runtime, ptr - prtd->dma_buffer);
+
+ if (x == runtime->buffer_size)
+ x = 0;
+
+ return x;
+}
+
+static int atmel_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct atmel_runtime_data *prtd;
+ int ret = 0;
+
+ snd_soc_set_runtime_hwparams(substream, &atmel_pcm_hardware);
+
+ /* ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ goto out;
+
+ prtd = kzalloc(sizeof(struct atmel_runtime_data), GFP_KERNEL);
+ if (prtd == NULL) {
+ ret = -ENOMEM;
+ goto out;
+ }
+ runtime->private_data = prtd;
+
+ out:
+ return ret;
+}
+
+static int atmel_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct atmel_runtime_data *prtd = substream->runtime->private_data;
+
+ kfree(prtd);
+ return 0;
+}
+
+static int atmel_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ return remap_pfn_range(vma, vma->vm_start,
+ substream->dma_buffer.addr >> PAGE_SHIFT,
+ vma->vm_end - vma->vm_start, vma->vm_page_prot);
+}
+
+struct snd_pcm_ops atmel_pcm_ops = {
+ .open = atmel_pcm_open,
+ .close = atmel_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = atmel_pcm_hw_params,
+ .hw_free = atmel_pcm_hw_free,
+ .prepare = atmel_pcm_prepare,
+ .trigger = atmel_pcm_trigger,
+ .pointer = atmel_pcm_pointer,
+ .mmap = atmel_pcm_mmap,
+};
+
+
+/*--------------------------------------------------------------------------*\
+ * ASoC platform driver
+\*--------------------------------------------------------------------------*/
+static u64 atmel_pcm_dmamask = 0xffffffff;
+
+static int atmel_pcm_new(struct snd_card *card,
+ struct snd_soc_dai *dai, struct snd_pcm *pcm)
+{
+ int ret = 0;
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &atmel_pcm_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = 0xffffffff;
+
+ if (dai->playback.channels_min) {
+ ret = atmel_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret)
+ goto out;
+ }
+
+ if (dai->capture.channels_min) {
+ pr_debug("at32-pcm:"
+ "Allocating PCM capture DMA buffer\n");
+ ret = atmel_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_CAPTURE);
+ if (ret)
+ goto out;
+ }
+ out:
+ return ret;
+}
+
+static void atmel_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ continue;
+
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+ dma_free_coherent(pcm->card->dev, buf->bytes,
+ buf->area, buf->addr);
+ buf->area = NULL;
+ }
+}
+
+#ifdef CONFIG_PM
+static int atmel_pcm_suspend(struct snd_soc_dai *dai)
+{
+ struct snd_pcm_runtime *runtime = dai->runtime;
+ struct atmel_runtime_data *prtd;
+ struct atmel_pcm_dma_params *params;
+
+ if (!runtime)
+ return 0;
+
+ prtd = runtime->private_data;
+ params = prtd->params;
+
+ /* disable the PDC and save the PDC registers */
+
+ ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable);
+
+ prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr);
+ prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr);
+ prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr);
+ prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr);
+
+ return 0;
+}
+
+static int atmel_pcm_resume(struct snd_soc_dai *dai)
+{
+ struct snd_pcm_runtime *runtime = dai->runtime;
+ struct atmel_runtime_data *prtd;
+ struct atmel_pcm_dma_params *params;
+
+ if (!runtime)
+ return 0;
+
+ prtd = runtime->private_data;
+ params = prtd->params;
+
+ /* restore the PDC registers and enable the PDC */
+ ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save);
+ ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save);
+ ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save);
+ ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save);
+
+ ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable);
+ return 0;
+}
+#else
+#define atmel_pcm_suspend NULL
+#define atmel_pcm_resume NULL
+#endif
+
+struct snd_soc_platform atmel_soc_platform = {
+ .name = "atmel-audio",
+ .pcm_ops = &atmel_pcm_ops,
+ .pcm_new = atmel_pcm_new,
+ .pcm_free = atmel_pcm_free_dma_buffers,
+ .suspend = atmel_pcm_suspend,
+ .resume = atmel_pcm_resume,
+};
+EXPORT_SYMBOL_GPL(atmel_soc_platform);
+
+static int __init atmel_pcm_modinit(void)
+{
+ return snd_soc_register_platform(&atmel_soc_platform);
+}
+module_init(atmel_pcm_modinit);
+
+static void __exit atmel_pcm_modexit(void)
+{
+ snd_soc_unregister_platform(&atmel_soc_platform);
+}
+module_exit(atmel_pcm_modexit);
+
+MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou@atmel.com>");
+MODULE_DESCRIPTION("Atmel PCM module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h
new file mode 100644
index 0000000..ec9b282
--- /dev/null
+++ b/sound/soc/atmel/atmel-pcm.h
@@ -0,0 +1,86 @@
+/*
+ * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC.
+ *
+ * Copyright (C) 2005 SAN People
+ * Copyright (C) 2008 Atmel
+ *
+ * Authors: Sedji Gaouaou <sedji.gaouaou@atmel.com>
+ *
+ * Based on at91-pcm. by:
+ * Frank Mandarino <fmandarino@endrelia.com>
+ * Copyright 2006 Endrelia Technologies Inc.
+ *
+ * Based on pxa2xx-pcm.c by:
+ *
+ * Author: Nicolas Pitre
+ * Created: Nov 30, 2004
+ * Copyright: (C) 2004 MontaVista Software, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef _ATMEL_PCM_H
+#define _ATMEL_PCM_H
+
+#include <linux/atmel-ssc.h>
+
+/*
+ * Registers and status bits that are required by the PCM driver.
+ */
+struct atmel_pdc_regs {
+ unsigned int xpr; /* PDC recv/trans pointer */
+ unsigned int xcr; /* PDC recv/trans counter */
+ unsigned int xnpr; /* PDC next recv/trans pointer */
+ unsigned int xncr; /* PDC next recv/trans counter */
+ unsigned int ptcr; /* PDC transfer control */
+};
+
+struct atmel_ssc_mask {
+ u32 ssc_enable; /* SSC recv/trans enable */
+ u32 ssc_disable; /* SSC recv/trans disable */
+ u32 ssc_endx; /* SSC ENDTX or ENDRX */
+ u32 ssc_endbuf; /* SSC TXBUFE or RXBUFF */
+ u32 pdc_enable; /* PDC recv/trans enable */
+ u32 pdc_disable; /* PDC recv/trans disable */
+};
+
+/*
+ * This structure, shared between the PCM driver and the interface,
+ * contains all information required by the PCM driver to perform the
+ * PDC DMA operation. All fields except dma_intr_handler() are initialized
+ * by the interface. The dms_intr_handler() pointer is set by the PCM
+ * driver and called by the interface SSC interrupt handler if it is
+ * non-NULL.
+ */
+struct atmel_pcm_dma_params {
+ char *name; /* stream identifier */
+ int pdc_xfer_size; /* PDC counter increment in bytes */
+ struct ssc_device *ssc; /* SSC device for stream */
+ struct atmel_pdc_regs *pdc; /* PDC receive or transmit registers */
+ struct atmel_ssc_mask *mask; /* SSC & PDC status bits */
+ struct snd_pcm_substream *substream;
+ void (*dma_intr_handler)(u32, struct snd_pcm_substream *);
+};
+
+extern struct snd_soc_platform atmel_soc_platform;
+
+
+/*
+ * SSC register access (since ssc_writel() / ssc_readl() require literal name)
+ */
+#define ssc_readx(base, reg) (__raw_readl((base) + (reg)))
+#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg))
+
+#endif /* _ATMEL_PCM_H */
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
new file mode 100644
index 0000000..c5d6790
--- /dev/null
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -0,0 +1,790 @@
+/*
+ * atmel_ssc_dai.c -- ALSA SoC ATMEL SSC Audio Layer Platform driver
+ *
+ * Copyright (C) 2005 SAN People
+ * Copyright (C) 2008 Atmel
+ *
+ * Author: Sedji Gaouaou <sedji.gaouaou@atmel.com>
+ * ATMEL CORP.
+ *
+ * Based on at91-ssc.c by
+ * Frank Mandarino <fmandarino@endrelia.com>
+ * Based on pxa2xx Platform drivers by
+ * Liam Girdwood <liam.girdwood@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.
+ *
+ * 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
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/atmel_pdc.h>
+
+#include <linux/atmel-ssc.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <mach/hardware.h>
+
+#include "atmel-pcm.h"
+#include "atmel_ssc_dai.h"
+
+
+#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20)
+#define NUM_SSC_DEVICES 1
+#else
+#define NUM_SSC_DEVICES 3
+#endif
+
+/*
+ * SSC PDC registers required by the PCM DMA engine.
+ */
+static struct atmel_pdc_regs pdc_tx_reg = {
+ .xpr = ATMEL_PDC_TPR,
+ .xcr = ATMEL_PDC_TCR,
+ .xnpr = ATMEL_PDC_TNPR,
+ .xncr = ATMEL_PDC_TNCR,
+};
+
+static struct atmel_pdc_regs pdc_rx_reg = {
+ .xpr = ATMEL_PDC_RPR,
+ .xcr = ATMEL_PDC_RCR,
+ .xnpr = ATMEL_PDC_RNPR,
+ .xncr = ATMEL_PDC_RNCR,
+};
+
+/*
+ * SSC & PDC status bits for transmit and receive.
+ */
+static struct atmel_ssc_mask ssc_tx_mask = {
+ .ssc_enable = SSC_BIT(CR_TXEN),
+ .ssc_disable = SSC_BIT(CR_TXDIS),
+ .ssc_endx = SSC_BIT(SR_ENDTX),
+ .ssc_endbuf = SSC_BIT(SR_TXBUFE),
+ .pdc_enable = ATMEL_PDC_TXTEN,
+ .pdc_disable = ATMEL_PDC_TXTDIS,
+};
+
+static struct atmel_ssc_mask ssc_rx_mask = {
+ .ssc_enable = SSC_BIT(CR_RXEN),
+ .ssc_disable = SSC_BIT(CR_RXDIS),
+ .ssc_endx = SSC_BIT(SR_ENDRX),
+ .ssc_endbuf = SSC_BIT(SR_RXBUFF),
+ .pdc_enable = ATMEL_PDC_RXTEN,
+ .pdc_disable = ATMEL_PDC_RXTDIS,
+};
+
+
+/*
+ * DMA parameters.
+ */
+static struct atmel_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = {
+ {{
+ .name = "SSC0 PCM out",
+ .pdc = &pdc_tx_reg,
+ .mask = &ssc_tx_mask,
+ },
+ {
+ .name = "SSC0 PCM in",
+ .pdc = &pdc_rx_reg,
+ .mask = &ssc_rx_mask,
+ } },
+#if NUM_SSC_DEVICES == 3
+ {{
+ .name = "SSC1 PCM out",
+ .pdc = &pdc_tx_reg,
+ .mask = &ssc_tx_mask,
+ },
+ {
+ .name = "SSC1 PCM in",
+ .pdc = &pdc_rx_reg,
+ .mask = &ssc_rx_mask,
+ } },
+ {{
+ .name = "SSC2 PCM out",
+ .pdc = &pdc_tx_reg,
+ .mask = &ssc_tx_mask,
+ },
+ {
+ .name = "SSC2 PCM in",
+ .pdc = &pdc_rx_reg,
+ .mask = &ssc_rx_mask,
+ } },
+#endif
+};
+
+
+static struct atmel_ssc_info ssc_info[NUM_SSC_DEVICES] = {
+ {
+ .name = "ssc0",
+ .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock),
+ .dir_mask = SSC_DIR_MASK_UNUSED,
+ .initialized = 0,
+ },
+#if NUM_SSC_DEVICES == 3
+ {
+ .name = "ssc1",
+ .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock),
+ .dir_mask = SSC_DIR_MASK_UNUSED,
+ .initialized = 0,
+ },
+ {
+ .name = "ssc2",
+ .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock),
+ .dir_mask = SSC_DIR_MASK_UNUSED,
+ .initialized = 0,
+ },
+#endif
+};
+
+
+/*
+ * SSC interrupt handler. Passes PDC interrupts to the DMA
+ * interrupt handler in the PCM driver.
+ */
+static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id)
+{
+ struct atmel_ssc_info *ssc_p = dev_id;
+ struct atmel_pcm_dma_params *dma_params;
+ u32 ssc_sr;
+ u32 ssc_substream_mask;
+ int i;
+
+ ssc_sr = (unsigned long)ssc_readl(ssc_p->ssc->regs, SR)
+ & (unsigned long)ssc_readl(ssc_p->ssc->regs, IMR);
+
+ /*
+ * Loop through the substreams attached to this SSC. If
+ * a DMA-related interrupt occurred on that substream, call
+ * the DMA interrupt handler function, if one has been
+ * registered in the dma_params structure by the PCM driver.
+ */
+ for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) {
+ dma_params = ssc_p->dma_params[i];
+
+ if ((dma_params != NULL) &&
+ (dma_params->dma_intr_handler != NULL)) {
+ ssc_substream_mask = (dma_params->mask->ssc_endx |
+ dma_params->mask->ssc_endbuf);
+ if (ssc_sr & ssc_substream_mask) {
+ dma_params->dma_intr_handler(ssc_sr,
+ dma_params->
+ substream);
+ }
+ }
+ }
+
+ return IRQ_HANDLED;
+}
+
+
+/*-------------------------------------------------------------------------*\
+ * DAI functions
+\*-------------------------------------------------------------------------*/
+/*
+ * Startup. Only that one substream allowed in each direction.
+ */
+static int atmel_ssc_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+ struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
+ int dir_mask;
+
+ pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n",
+ ssc_readl(ssc_p->ssc->regs, SR));
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dir_mask = SSC_DIR_MASK_PLAYBACK;
+ else
+ dir_mask = SSC_DIR_MASK_CAPTURE;
+
+ spin_lock_irq(&ssc_p->lock);
+ if (ssc_p->dir_mask & dir_mask) {
+ spin_unlock_irq(&ssc_p->lock);
+ return -EBUSY;
+ }
+ ssc_p->dir_mask |= dir_mask;
+ spin_unlock_irq(&ssc_p->lock);
+
+ return 0;
+}
+
+/*
+ * Shutdown. Clear DMA parameters and shutdown the SSC if there
+ * are no other substreams open.
+ */
+static void atmel_ssc_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+ struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
+ struct atmel_pcm_dma_params *dma_params;
+ int dir, dir_mask;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dir = 0;
+ else
+ dir = 1;
+
+ dma_params = ssc_p->dma_params[dir];
+
+ if (dma_params != NULL) {
+ ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable);
+ pr_debug("atmel_ssc_shutdown: %s disabled SSC_SR=0x%08x\n",
+ (dir ? "receive" : "transmit"),
+ ssc_readl(ssc_p->ssc->regs, SR));
+
+ dma_params->ssc = NULL;
+ dma_params->substream = NULL;
+ ssc_p->dma_params[dir] = NULL;
+ }
+
+ dir_mask = 1 << dir;
+
+ spin_lock_irq(&ssc_p->lock);
+ ssc_p->dir_mask &= ~dir_mask;
+ if (!ssc_p->dir_mask) {
+ if (ssc_p->initialized) {
+ /* Shutdown the SSC clock. */
+ pr_debug("atmel_ssc_dau: Stopping clock\n");
+ clk_disable(ssc_p->ssc->clk);
+
+ free_irq(ssc_p->ssc->irq, ssc_p);
+ ssc_p->initialized = 0;
+ }
+
+ /* Reset the SSC */
+ ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST));
+ /* Clear the SSC dividers */
+ ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0;
+ }
+ spin_unlock_irq(&ssc_p->lock);
+}
+
+
+/*
+ * Record the DAI format for use in hw_params().
+ */
+static int atmel_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+ unsigned int fmt)
+{
+ struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
+
+ ssc_p->daifmt = fmt;
+ return 0;
+}
+
+/*
+ * Record SSC clock dividers for use in hw_params().
+ */
+static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
+ int div_id, int div)
+{
+ struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
+
+ switch (div_id) {
+ case ATMEL_SSC_CMR_DIV:
+ /*
+ * The same master clock divider is used for both
+ * transmit and receive, so if a value has already
+ * been set, it must match this value.
+ */
+ if (ssc_p->cmr_div == 0)
+ ssc_p->cmr_div = div;
+ else
+ if (div != ssc_p->cmr_div)
+ return -EBUSY;
+ break;
+
+ case ATMEL_SSC_TCMR_PERIOD:
+ ssc_p->tcmr_period = div;
+ break;
+
+ case ATMEL_SSC_RCMR_PERIOD:
+ ssc_p->rcmr_period = div;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+/*
+ * Configure the SSC.
+ */
+static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+ int id = rtd->dai->cpu_dai->id;
+ struct atmel_ssc_info *ssc_p = &ssc_info[id];
+ struct atmel_pcm_dma_params *dma_params;
+ int dir, channels, bits;
+ u32 tfmr, rfmr, tcmr, rcmr;
+ int start_event;
+ int ret;
+
+ /*
+ * Currently, there is only one set of dma params for
+ * each direction. If more are added, this code will
+ * have to be changed to select the proper set.
+ */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dir = 0;
+ else
+ dir = 1;
+
+ dma_params = &ssc_dma_params[id][dir];
+ dma_params->ssc = ssc_p->ssc;
+ dma_params->substream = substream;
+
+ ssc_p->dma_params[dir] = dma_params;
+
+ /*
+ * The cpu_dai->dma_data field is only used to communicate the
+ * appropriate DMA parameters to the pcm driver hw_params()
+ * function. It should not be used for other purposes
+ * as it is common to all substreams.
+ */
+ rtd->dai->cpu_dai->dma_data = dma_params;
+
+ channels = params_channels(params);
+
+ /*
+ * Determine sample size in bits and the PDC increment.
+ */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8:
+ bits = 8;
+ dma_params->pdc_xfer_size = 1;
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ bits = 16;
+ dma_params->pdc_xfer_size = 2;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ bits = 24;
+ dma_params->pdc_xfer_size = 4;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ bits = 32;
+ dma_params->pdc_xfer_size = 4;
+ break;
+ default:
+ printk(KERN_WARNING "atmel_ssc_dai: unsupported PCM format");
+ return -EINVAL;
+ }
+
+ /*
+ * The SSC only supports up to 16-bit samples in I2S format, due
+ * to the size of the Frame Mode Register FSLEN field.
+ */
+ if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S
+ && bits > 16) {
+ printk(KERN_WARNING
+ "atmel_ssc_dai: sample size %d"
+ "is too large for I2S\n", bits);
+ return -EINVAL;
+ }
+
+ /*
+ * Compute SSC register settings.
+ */
+ switch (ssc_p->daifmt
+ & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) {
+
+ case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS:
+ /*
+ * I2S format, SSC provides BCLK and LRC clocks.
+ *
+ * The SSC transmit and receive clocks are generated
+ * from the MCK divider, and the BCLK signal
+ * is output on the SSC TK line.
+ */
+ rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period)
+ | SSC_BF(RCMR_STTDLY, START_DELAY)
+ | SSC_BF(RCMR_START, SSC_START_FALLING_RF)
+ | SSC_BF(RCMR_CKI, SSC_CKI_RISING)
+ | SSC_BF(RCMR_CKO, SSC_CKO_NONE)
+ | SSC_BF(RCMR_CKS, SSC_CKS_DIV);
+
+ rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+ | SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE)
+ | SSC_BF(RFMR_FSLEN, (bits - 1))
+ | SSC_BF(RFMR_DATNB, (channels - 1))
+ | SSC_BIT(RFMR_MSBF)
+ | SSC_BF(RFMR_LOOP, 0)
+ | SSC_BF(RFMR_DATLEN, (bits - 1));
+
+ tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period)
+ | SSC_BF(TCMR_STTDLY, START_DELAY)
+ | SSC_BF(TCMR_START, SSC_START_FALLING_RF)
+ | SSC_BF(TCMR_CKI, SSC_CKI_FALLING)
+ | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS)
+ | SSC_BF(TCMR_CKS, SSC_CKS_DIV);
+
+ tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+ | SSC_BF(TFMR_FSDEN, 0)
+ | SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE)
+ | SSC_BF(TFMR_FSLEN, (bits - 1))
+ | SSC_BF(TFMR_DATNB, (channels - 1))
+ | SSC_BIT(TFMR_MSBF)
+ | SSC_BF(TFMR_DATDEF, 0)
+ | SSC_BF(TFMR_DATLEN, (bits - 1));
+ break;
+
+ case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM:
+ /*
+ * I2S format, CODEC supplies BCLK and LRC clocks.
+ *
+ * The SSC transmit clock is obtained from the BCLK signal on
+ * on the TK line, and the SSC receive clock is
+ * generated from the transmit clock.
+ *
+ * For single channel data, one sample is transferred
+ * on the falling edge of the LRC clock.
+ * For two channel data, one sample is
+ * transferred on both edges of the LRC clock.
+ */
+ start_event = ((channels == 1)
+ ? SSC_START_FALLING_RF
+ : SSC_START_EDGE_RF);
+
+ rcmr = SSC_BF(RCMR_PERIOD, 0)
+ | SSC_BF(RCMR_STTDLY, START_DELAY)
+ | SSC_BF(RCMR_START, start_event)
+ | SSC_BF(RCMR_CKI, SSC_CKI_RISING)
+ | SSC_BF(RCMR_CKO, SSC_CKO_NONE)
+ | SSC_BF(RCMR_CKS, SSC_CKS_CLOCK);
+
+ rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+ | SSC_BF(RFMR_FSOS, SSC_FSOS_NONE)
+ | SSC_BF(RFMR_FSLEN, 0)
+ | SSC_BF(RFMR_DATNB, 0)
+ | SSC_BIT(RFMR_MSBF)
+ | SSC_BF(RFMR_LOOP, 0)
+ | SSC_BF(RFMR_DATLEN, (bits - 1));
+
+ tcmr = SSC_BF(TCMR_PERIOD, 0)
+ | SSC_BF(TCMR_STTDLY, START_DELAY)
+ | SSC_BF(TCMR_START, start_event)
+ | SSC_BF(TCMR_CKI, SSC_CKI_FALLING)
+ | SSC_BF(TCMR_CKO, SSC_CKO_NONE)
+ | SSC_BF(TCMR_CKS, SSC_CKS_PIN);
+
+ tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+ | SSC_BF(TFMR_FSDEN, 0)
+ | SSC_BF(TFMR_FSOS, SSC_FSOS_NONE)
+ | SSC_BF(TFMR_FSLEN, 0)
+ | SSC_BF(TFMR_DATNB, 0)
+ | SSC_BIT(TFMR_MSBF)
+ | SSC_BF(TFMR_DATDEF, 0)
+ | SSC_BF(TFMR_DATLEN, (bits - 1));
+ break;
+
+ case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS:
+ /*
+ * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks.
+ *
+ * The SSC transmit and receive clocks are generated from the
+ * MCK divider, and the BCLK signal is output
+ * on the SSC TK line.
+ */
+ rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period)
+ | SSC_BF(RCMR_STTDLY, 1)
+ | SSC_BF(RCMR_START, SSC_START_RISING_RF)
+ | SSC_BF(RCMR_CKI, SSC_CKI_RISING)
+ | SSC_BF(RCMR_CKO, SSC_CKO_NONE)
+ | SSC_BF(RCMR_CKS, SSC_CKS_DIV);
+
+ rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+ | SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE)
+ | SSC_BF(RFMR_FSLEN, 0)
+ | SSC_BF(RFMR_DATNB, (channels - 1))
+ | SSC_BIT(RFMR_MSBF)
+ | SSC_BF(RFMR_LOOP, 0)
+ | SSC_BF(RFMR_DATLEN, (bits - 1));
+
+ tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period)
+ | SSC_BF(TCMR_STTDLY, 1)
+ | SSC_BF(TCMR_START, SSC_START_RISING_RF)
+ | SSC_BF(TCMR_CKI, SSC_CKI_RISING)
+ | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS)
+ | SSC_BF(TCMR_CKS, SSC_CKS_DIV);
+
+ tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+ | SSC_BF(TFMR_FSDEN, 0)
+ | SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE)
+ | SSC_BF(TFMR_FSLEN, 0)
+ | SSC_BF(TFMR_DATNB, (channels - 1))
+ | SSC_BIT(TFMR_MSBF)
+ | SSC_BF(TFMR_DATDEF, 0)
+ | SSC_BF(TFMR_DATLEN, (bits - 1));
+ break;
+
+ case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM:
+ default:
+ printk(KERN_WARNING "atmel_ssc_dai: unsupported DAI format 0x%x\n",
+ ssc_p->daifmt);
+ return -EINVAL;
+ break;
+ }
+ pr_debug("atmel_ssc_hw_params: "
+ "RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n",
+ rcmr, rfmr, tcmr, tfmr);
+
+ if (!ssc_p->initialized) {
+
+ /* Enable PMC peripheral clock for this SSC */
+ pr_debug("atmel_ssc_dai: Starting clock\n");
+ clk_enable(ssc_p->ssc->clk);
+
+ /* Reset the SSC and its PDC registers */
+ ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST));
+
+ ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0);
+ ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0);
+ ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0);
+ ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0);
+
+ ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0);
+ ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0);
+ ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0);
+ ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0);
+
+ ret = request_irq(ssc_p->ssc->irq, atmel_ssc_interrupt, 0,
+ ssc_p->name, ssc_p);
+ if (ret < 0) {
+ printk(KERN_WARNING
+ "atmel_ssc_dai: request_irq failure\n");
+ pr_debug("Atmel_ssc_dai: Stoping clock\n");
+ clk_disable(ssc_p->ssc->clk);
+ return ret;
+ }
+
+ ssc_p->initialized = 1;
+ }
+
+ /* set SSC clock mode register */
+ ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div);
+
+ /* set receive clock mode and format */
+ ssc_writel(ssc_p->ssc->regs, RCMR, rcmr);
+ ssc_writel(ssc_p->ssc->regs, RFMR, rfmr);
+
+ /* set transmit clock mode and format */
+ ssc_writel(ssc_p->ssc->regs, TCMR, tcmr);
+ ssc_writel(ssc_p->ssc->regs, TFMR, tfmr);
+
+ pr_debug("atmel_ssc_dai,hw_params: SSC initialized\n");
+ return 0;
+}
+
+
+static int atmel_ssc_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+ struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
+ struct atmel_pcm_dma_params *dma_params;
+ int dir;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dir = 0;
+ else
+ dir = 1;
+
+ dma_params = ssc_p->dma_params[dir];
+
+ ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable);
+
+ pr_debug("%s enabled SSC_SR=0x%08x\n",
+ dir ? "receive" : "transmit",
+ ssc_readl(ssc_p->ssc->regs, SR));
+ return 0;
+}
+
+
+#ifdef CONFIG_PM
+static int atmel_ssc_suspend(struct snd_soc_dai *cpu_dai)
+{
+ struct atmel_ssc_info *ssc_p;
+
+ if (!cpu_dai->active)
+ return 0;
+
+ ssc_p = &ssc_info[cpu_dai->id];
+
+ /* Save the status register before disabling transmit and receive */
+ ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR);
+ ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS));
+
+ /* Save the current interrupt mask, then disable unmasked interrupts */
+ ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR);
+ ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr);
+
+ ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR);
+ ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR);
+ ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR);
+ ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR);
+ ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR);
+
+ return 0;
+}
+
+
+
+static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai)
+{
+ struct atmel_ssc_info *ssc_p;
+ u32 cr;
+
+ if (!cpu_dai->active)
+ return 0;
+
+ ssc_p = &ssc_info[cpu_dai->id];
+
+ /* restore SSC register settings */
+ ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr);
+ ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr);
+ ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr);
+ ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr);
+ ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr);
+
+ /* re-enable interrupts */
+ ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr);
+
+ /* Re-enable recieve and transmit as appropriate */
+ cr = 0;
+ cr |=
+ (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0;
+ cr |=
+ (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0;
+ ssc_writel(ssc_p->ssc->regs, CR, cr);
+
+ return 0;
+}
+#else /* CONFIG_PM */
+# define atmel_ssc_suspend NULL
+# define atmel_ssc_resume NULL
+#endif /* CONFIG_PM */
+
+
+#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000_96000)
+
+#define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
+ { .name = "atmel-ssc0",
+ .id = 0,
+ .suspend = atmel_ssc_suspend,
+ .resume = atmel_ssc_resume,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = ATMEL_SSC_RATES,
+ .formats = ATMEL_SSC_FORMATS,},
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = ATMEL_SSC_RATES,
+ .formats = ATMEL_SSC_FORMATS,},
+ .ops = {
+ .startup = atmel_ssc_startup,
+ .shutdown = atmel_ssc_shutdown,
+ .prepare = atmel_ssc_prepare,
+ .hw_params = atmel_ssc_hw_params,
+ .set_fmt = atmel_ssc_set_dai_fmt,
+ .set_clkdiv = atmel_ssc_set_dai_clkdiv,},
+ .private_data = &ssc_info[0],
+ },
+#if NUM_SSC_DEVICES == 3
+ { .name = "atmel-ssc1",
+ .id = 1,
+ .suspend = atmel_ssc_suspend,
+ .resume = atmel_ssc_resume,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = ATMEL_SSC_RATES,
+ .formats = ATMEL_SSC_FORMATS,},
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = ATMEL_SSC_RATES,
+ .formats = ATMEL_SSC_FORMATS,},
+ .ops = {
+ .startup = atmel_ssc_startup,
+ .shutdown = atmel_ssc_shutdown,
+ .prepare = atmel_ssc_prepare,
+ .hw_params = atmel_ssc_hw_params,
+ .set_fmt = atmel_ssc_set_dai_fmt,
+ .set_clkdiv = atmel_ssc_set_dai_clkdiv,},
+ .private_data = &ssc_info[1],
+ },
+ { .name = "atmel-ssc2",
+ .id = 2,
+ .suspend = atmel_ssc_suspend,
+ .resume = atmel_ssc_resume,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = ATMEL_SSC_RATES,
+ .formats = ATMEL_SSC_FORMATS,},
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = ATMEL_SSC_RATES,
+ .formats = ATMEL_SSC_FORMATS,},
+ .ops = {
+ .startup = atmel_ssc_startup,
+ .shutdown = atmel_ssc_shutdown,
+ .prepare = atmel_ssc_prepare,
+ .hw_params = atmel_ssc_hw_params,
+ .set_fmt = atmel_ssc_set_dai_fmt,
+ .set_clkdiv = atmel_ssc_set_dai_clkdiv,},
+ .private_data = &ssc_info[2],
+ },
+#endif
+};
+EXPORT_SYMBOL_GPL(atmel_ssc_dai);
+
+static int __init atmel_ssc_modinit(void)
+{
+ return snd_soc_register_dais(atmel_ssc_dai, ARRAY_SIZE(atmel_ssc_dai));
+}
+module_init(atmel_ssc_modinit);
+
+static void __exit atmel_ssc_modexit(void)
+{
+ snd_soc_unregister_dais(atmel_ssc_dai, ARRAY_SIZE(atmel_ssc_dai));
+}
+module_exit(atmel_ssc_modexit);
+
+/* Module information */
+MODULE_AUTHOR("Sedji Gaouaou, sedji.gaouaou@atmel.com, www.atmel.com");
+MODULE_DESCRIPTION("ATMEL SSC ASoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h
new file mode 100644
index 0000000..a828746
--- /dev/null
+++ b/sound/soc/atmel/atmel_ssc_dai.h
@@ -0,0 +1,121 @@
+/*
+ * atmel_ssc_dai.h - ALSA SSC interface for the Atmel SoC
+ *
+ * Copyright (C) 2005 SAN People
+ * Copyright (C) 2008 Atmel
+ *
+ * Author: Sedji Gaouaou <sedji.gaouaou@atmel.com>
+ * ATMEL CORP.
+ *
+ * Based on at91-ssc.c by
+ * Frank Mandarino <fmandarino@endrelia.com>
+ * Based on pxa2xx Platform drivers by
+ * Liam Girdwood <liam.girdwood@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.
+ *
+ * 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
+ */
+
+#ifndef _ATMEL_SSC_DAI_H
+#define _ATMEL_SSC_DAI_H
+
+#include <linux/types.h>
+#include <linux/atmel-ssc.h>
+
+#include "atmel-pcm.h"
+
+/* SSC system clock ids */
+#define ATMEL_SYSCLK_MCK 0 /* SSC uses AT91 MCK as system clock */
+
+/* SSC divider ids */
+#define ATMEL_SSC_CMR_DIV 0 /* MCK divider for BCLK */
+#define ATMEL_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */
+#define ATMEL_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */
+/*
+ * SSC direction masks
+ */
+#define SSC_DIR_MASK_UNUSED 0
+#define SSC_DIR_MASK_PLAYBACK 1
+#define SSC_DIR_MASK_CAPTURE 2
+
+/*
+ * SSC register values that Atmel left out of <linux/atmel-ssc.h>. These
+ * are expected to be used with SSC_BF
+ */
+/* START bit field values */
+#define SSC_START_CONTINUOUS 0
+#define SSC_START_TX_RX 1
+#define SSC_START_LOW_RF 2
+#define SSC_START_HIGH_RF 3
+#define SSC_START_FALLING_RF 4
+#define SSC_START_RISING_RF 5
+#define SSC_START_LEVEL_RF 6
+#define SSC_START_EDGE_RF 7
+#define SSS_START_COMPARE_0 8
+
+/* CKI bit field values */
+#define SSC_CKI_FALLING 0
+#define SSC_CKI_RISING 1
+
+/* CKO bit field values */
+#define SSC_CKO_NONE 0
+#define SSC_CKO_CONTINUOUS 1
+#define SSC_CKO_TRANSFER 2
+
+/* CKS bit field values */
+#define SSC_CKS_DIV 0
+#define SSC_CKS_CLOCK 1
+#define SSC_CKS_PIN 2
+
+/* FSEDGE bit field values */
+#define SSC_FSEDGE_POSITIVE 0
+#define SSC_FSEDGE_NEGATIVE 1
+
+/* FSOS bit field values */
+#define SSC_FSOS_NONE 0
+#define SSC_FSOS_NEGATIVE 1
+#define SSC_FSOS_POSITIVE 2
+#define SSC_FSOS_LOW 3
+#define SSC_FSOS_HIGH 4
+#define SSC_FSOS_TOGGLE 5
+
+#define START_DELAY 1
+
+struct atmel_ssc_state {
+ u32 ssc_cmr;
+ u32 ssc_rcmr;
+ u32 ssc_rfmr;
+ u32 ssc_tcmr;
+ u32 ssc_tfmr;
+ u32 ssc_sr;
+ u32 ssc_imr;
+};
+
+
+struct atmel_ssc_info {
+ char *name;
+ struct ssc_device *ssc;
+ spinlock_t lock; /* lock for dir_mask */
+ unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */
+ unsigned short initialized; /* true if SSC has been initialized */
+ unsigned short daifmt;
+ unsigned short cmr_div;
+ unsigned short tcmr_period;
+ unsigned short rcmr_period;
+ struct atmel_pcm_dma_params *dma_params[2];
+ struct atmel_ssc_state ssc_state;
+};
+extern struct snd_soc_dai atmel_ssc_dai[];
+
+#endif /* _AT91_SSC_DAI_H */
diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c
similarity index 98%
rename from sound/soc/at32/playpaq_wm8510.c
rename to sound/soc/atmel/playpaq_wm8510.c
index b1966e4..43dd8ce 100644
--- a/sound/soc/at32/playpaq_wm8510.c
+++ b/sound/soc/atmel/playpaq_wm8510.c
@@ -22,7 +22,6 @@
#include <linux/module.h>
#include <linux/moduleparam.h>
-#include <linux/version.h>
#include <linux/kernel.h>
#include <linux/errno.h>
#include <linux/clk.h>
@@ -40,8 +39,8 @@
#include <mach/portmux.h>
#include "../codecs/wm8510.h"
-#include "at32-pcm.h"
-#include "at32-ssc.h"
+#include "atmel-pcm.h"
+#include "atmel_ssc_dai.h"
/*-------------------------------------------------------------------------*\
@@ -362,8 +361,9 @@
-static struct snd_soc_machine snd_soc_machine_playpaq = {
+static struct snd_soc_card snd_soc_playpaq = {
.name = "LRS_PlayPaq_WM8510",
+ .platform = &at32_soc_platform,
.dai_link = &playpaq_wm8510_dai,
.num_links = 1,
};
@@ -378,8 +378,7 @@
static struct snd_soc_device playpaq_wm8510_snd_devdata = {
- .machine = &snd_soc_machine_playpaq,
- .platform = &at32_soc_platform,
+ .card = &snd_soc_playpaq,
.codec_dev = &soc_codec_dev_wm8510,
.codec_data = &playpaq_wm8510_setup,
};
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
new file mode 100644
index 0000000..1fb59a9
--- /dev/null
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -0,0 +1,328 @@
+/*
+ * sam9g20_wm8731 -- SoC audio for AT91SAM9G20-based
+ * ATMEL AT91SAM9G20ek board.
+ *
+ * Copyright (C) 2005 SAN People
+ * Copyright (C) 2008 Atmel
+ *
+ * Authors: Sedji Gaouaou <sedji.gaouaou@atmel.com>
+ *
+ * Based on ati_b1_wm8731.c by:
+ * Frank Mandarino <fmandarino@endrelia.com>
+ * Copyright 2006 Endrelia Technologies Inc.
+ * Based on corgi.c by:
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/clk.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+
+#include <linux/atmel-ssc.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+
+#include "../codecs/wm8731.h"
+#include "atmel-pcm.h"
+#include "atmel_ssc_dai.h"
+
+
+static int at91sam9g20ek_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ int ret;
+
+ /* codec system clock is supplied by PCK0, set to 12MHz */
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK,
+ 12000000, SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static void at91sam9g20ek_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+
+ dev_dbg(rtd->socdev->dev, "shutdown");
+}
+
+static int at91sam9g20ek_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct atmel_ssc_info *ssc_p = cpu_dai->private_data;
+ struct ssc_device *ssc = ssc_p->ssc;
+ int ret;
+
+ unsigned int rate;
+ int cmr_div, period;
+
+ if (ssc == NULL) {
+ printk(KERN_INFO "at91sam9g20ek_hw_params: ssc is NULL!\n");
+ return -EINVAL;
+ }
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /*
+ * The SSC clock dividers depend on the sample rate. The CMR.DIV
+ * field divides the system master clock MCK to drive the SSC TK
+ * signal which provides the codec BCLK. The TCMR.PERIOD and
+ * RCMR.PERIOD fields further divide the BCLK signal to drive
+ * the SSC TF and RF signals which provide the codec DACLRC and
+ * ADCLRC clocks.
+ *
+ * The dividers were determined through trial and error, where a
+ * CMR.DIV value is chosen such that the resulting BCLK value is
+ * divisible, or almost divisible, by (2 * sample rate), and then
+ * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1.
+ */
+ rate = params_rate(params);
+
+ switch (rate) {
+ case 8000:
+ cmr_div = 55; /* BCLK = 133MHz/(2*55) = 1.209MHz */
+ period = 74; /* LRC = BCLK/(2*(74+1)) ~= 8060,6Hz */
+ break;
+ case 11025:
+ cmr_div = 67; /* BCLK = 133MHz/(2*60) = 1.108MHz */
+ period = 45; /* LRC = BCLK/(2*(49+1)) = 11083,3Hz */
+ break;
+ case 16000:
+ cmr_div = 63; /* BCLK = 133MHz/(2*63) = 1.055MHz */
+ period = 32; /* LRC = BCLK/(2*(32+1)) = 15993,2Hz */
+ break;
+ case 22050:
+ cmr_div = 52; /* BCLK = 133MHz/(2*52) = 1.278MHz */
+ period = 28; /* LRC = BCLK/(2*(28+1)) = 22049Hz */
+ break;
+ case 32000:
+ cmr_div = 66; /* BCLK = 133MHz/(2*66) = 1.007MHz */
+ period = 15; /* LRC = BCLK/(2*(15+1)) = 31486,742Hz */
+ break;
+ case 44100:
+ cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */
+ period = 25; /* LRC = BCLK/(2*(25+1)) = 44098Hz */
+ break;
+ case 48000:
+ cmr_div = 33; /* BCLK = 133MHz/(2*33) = 2.015MHz */
+ period = 20; /* LRC = BCLK/(2*(20+1)) = 47979,79Hz */
+ break;
+ case 88200:
+ cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */
+ period = 12; /* LRC = BCLK/(2*(12+1)) = 88196Hz */
+ break;
+ case 96000:
+ cmr_div = 23; /* BCLK = 133MHz/(2*23) = 2.891MHz */
+ period = 14; /* LRC = BCLK/(2*(14+1)) = 96376Hz */
+ break;
+ default:
+ printk(KERN_WARNING "unsupported rate %d"
+ " on at91sam9g20ek board\n", rate);
+ return -EINVAL;
+ }
+
+ /* set the MCK divider for BCLK */
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, ATMEL_SSC_CMR_DIV, cmr_div);
+ if (ret < 0)
+ return ret;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ /* set the BCLK divider for DACLRC */
+ ret = snd_soc_dai_set_clkdiv(cpu_dai,
+ ATMEL_SSC_TCMR_PERIOD, period);
+ } else {
+ /* set the BCLK divider for ADCLRC */
+ ret = snd_soc_dai_set_clkdiv(cpu_dai,
+ ATMEL_SSC_RCMR_PERIOD, period);
+ }
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops at91sam9g20ek_ops = {
+ .startup = at91sam9g20ek_startup,
+ .hw_params = at91sam9g20ek_hw_params,
+ .shutdown = at91sam9g20ek_shutdown,
+};
+
+
+static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = {
+ SND_SOC_DAPM_MIC("Int Mic", NULL),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+
+ /* speaker connected to LHPOUT */
+ {"Ext Spk", NULL, "LHPOUT"},
+
+ /* mic is connected to Mic Jack, with WM8731 Mic Bias */
+ {"MICIN", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Int Mic"},
+};
+
+/*
+ * Logic for a wm8731 as connected on a at91sam9g20ek board.
+ */
+static int at91sam9g20ek_wm8731_init(struct snd_soc_codec *codec)
+{
+ printk(KERN_DEBUG
+ "at91sam9g20ek_wm8731 "
+ ": at91sam9g20ek_wm8731_init() called\n");
+
+ /* Add specific widgets */
+ snd_soc_dapm_new_controls(codec, at91sam9g20ek_dapm_widgets,
+ ARRAY_SIZE(at91sam9g20ek_dapm_widgets));
+ /* Set up specific audio path interconnects */
+ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+ /* not connected */
+ snd_soc_dapm_disable_pin(codec, "RLINEIN");
+ snd_soc_dapm_disable_pin(codec, "LLINEIN");
+
+ /* always connected */
+ snd_soc_dapm_enable_pin(codec, "Int Mic");
+ snd_soc_dapm_enable_pin(codec, "Ext Spk");
+
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
+
+static struct snd_soc_dai_link at91sam9g20ek_dai = {
+ .name = "WM8731",
+ .stream_name = "WM8731 PCM",
+ .cpu_dai = &atmel_ssc_dai[0],
+ .codec_dai = &wm8731_dai,
+ .init = at91sam9g20ek_wm8731_init,
+ .ops = &at91sam9g20ek_ops,
+};
+
+static struct snd_soc_card snd_soc_at91sam9g20ek = {
+ .name = "WM8731",
+ .platform = &atmel_soc_platform,
+ .dai_link = &at91sam9g20ek_dai,
+ .num_links = 1,
+};
+
+static struct wm8731_setup_data at91sam9g20ek_wm8731_setup = {
+ .i2c_bus = 0,
+ .i2c_address = 0x1b,
+};
+
+static struct snd_soc_device at91sam9g20ek_snd_devdata = {
+ .card = &snd_soc_at91sam9g20ek,
+ .codec_dev = &soc_codec_dev_wm8731,
+ .codec_data = &at91sam9g20ek_wm8731_setup,
+};
+
+static struct platform_device *at91sam9g20ek_snd_device;
+
+static int __init at91sam9g20ek_init(void)
+{
+ struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data;
+ struct ssc_device *ssc = NULL;
+ int ret;
+
+ /*
+ * Request SSC device
+ */
+ ssc = ssc_request(0);
+ if (IS_ERR(ssc)) {
+ ret = PTR_ERR(ssc);
+ ssc = NULL;
+ goto err_ssc;
+ }
+ ssc_p->ssc = ssc;
+
+ at91sam9g20ek_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!at91sam9g20ek_snd_device) {
+ printk(KERN_DEBUG
+ "platform device allocation failed\n");
+ ret = -ENOMEM;
+ }
+
+ platform_set_drvdata(at91sam9g20ek_snd_device,
+ &at91sam9g20ek_snd_devdata);
+ at91sam9g20ek_snd_devdata.dev = &at91sam9g20ek_snd_device->dev;
+
+ ret = platform_device_add(at91sam9g20ek_snd_device);
+ if (ret) {
+ printk(KERN_DEBUG
+ "platform device allocation failed\n");
+ platform_device_put(at91sam9g20ek_snd_device);
+ }
+
+ return ret;
+
+err_ssc:
+ return ret;
+}
+
+static void __exit at91sam9g20ek_exit(void)
+{
+ struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data;
+ struct ssc_device *ssc;
+
+ if (ssc_p != NULL) {
+ ssc = ssc_p->ssc;
+ if (ssc != NULL)
+ ssc_free(ssc);
+ ssc_p->ssc = NULL;
+ }
+
+ platform_device_unregister(at91sam9g20ek_snd_device);
+ at91sam9g20ek_snd_device = NULL;
+}
+
+module_init(at91sam9g20ek_init);
+module_exit(at91sam9g20ek_exit);
+
+/* Module information */
+MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou@atmel.com>");
+MODULE_DESCRIPTION("ALSA SoC AT91SAM9G20EK_WM8731");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 1466d93..74c823d 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -406,11 +406,12 @@
{
au1xpsc_audio_pcmdma[PCM_TX] = NULL;
au1xpsc_audio_pcmdma[PCM_RX] = NULL;
- return 0;
+ return snd_soc_register_platform(&au1xpsc_soc_platform);
}
static void __exit au1xpsc_audio_dbdma_exit(void)
{
+ snd_soc_unregister_platform(&au1xpsc_soc_platform);
}
module_init(au1xpsc_audio_dbdma_init);
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index 57facba..f0e30ae 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -160,7 +160,8 @@
EXPORT_SYMBOL_GPL(soc_ac97_ops);
static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
/* FIXME */
struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
@@ -210,7 +211,7 @@
}
static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream,
- int cmd)
+ int cmd, struct snd_soc_dai *dai)
{
/* FIXME */
struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
@@ -313,8 +314,7 @@
au1xpsc_ac97_workdata = NULL;
}
-static int au1xpsc_ac97_suspend(struct platform_device *pdev,
- struct snd_soc_dai *dai)
+static int au1xpsc_ac97_suspend(struct snd_soc_dai *dai)
{
/* save interesting registers and disable PSC */
au1xpsc_ac97_workdata->pm[0] =
@@ -328,8 +328,7 @@
return 0;
}
-static int au1xpsc_ac97_resume(struct platform_device *pdev,
- struct snd_soc_dai *dai)
+static int au1xpsc_ac97_resume(struct snd_soc_dai *dai)
{
/* restore PSC clock config */
au_writel(au1xpsc_ac97_workdata->pm[0] | PSC_SEL_PS_AC97MODE,
@@ -345,7 +344,7 @@
struct snd_soc_dai au1xpsc_ac97_dai = {
.name = "au1xpsc_ac97",
- .type = SND_SOC_DAI_AC97,
+ .ac97_control = 1,
.probe = au1xpsc_ac97_probe,
.remove = au1xpsc_ac97_remove,
.suspend = au1xpsc_ac97_suspend,
@@ -372,11 +371,12 @@
static int __init au1xpsc_ac97_init(void)
{
au1xpsc_ac97_workdata = NULL;
- return 0;
+ return snd_soc_register_dai(&au1xpsc_ac97_dai);
}
static void __exit au1xpsc_ac97_exit(void)
{
+ snd_soc_unregister_dai(&au1xpsc_ac97_dai);
}
module_init(au1xpsc_ac97_init);
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index 9384702..f916de4 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -116,7 +116,8 @@
}
static int au1xpsc_i2s_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata;
@@ -240,7 +241,8 @@
return 0;
}
-static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
{
struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata;
int ret, stype = SUBSTREAM_TYPE(substream);
@@ -337,8 +339,7 @@
au1xpsc_i2s_workdata = NULL;
}
-static int au1xpsc_i2s_suspend(struct platform_device *pdev,
- struct snd_soc_dai *cpu_dai)
+static int au1xpsc_i2s_suspend(struct snd_soc_dai *cpu_dai)
{
/* save interesting register and disable PSC */
au1xpsc_i2s_workdata->pm[0] =
@@ -352,8 +353,7 @@
return 0;
}
-static int au1xpsc_i2s_resume(struct platform_device *pdev,
- struct snd_soc_dai *cpu_dai)
+static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai)
{
/* select I2S mode and PSC clock */
au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata));
@@ -369,7 +369,6 @@
struct snd_soc_dai au1xpsc_i2s_dai = {
.name = "au1xpsc_i2s",
- .type = SND_SOC_DAI_I2S,
.probe = au1xpsc_i2s_probe,
.remove = au1xpsc_i2s_remove,
.suspend = au1xpsc_i2s_suspend,
@@ -389,8 +388,6 @@
.ops = {
.trigger = au1xpsc_i2s_trigger,
.hw_params = au1xpsc_i2s_hw_params,
- },
- .dai_ops = {
.set_fmt = au1xpsc_i2s_set_fmt,
},
};
@@ -399,11 +396,12 @@
static int __init au1xpsc_i2s_init(void)
{
au1xpsc_i2s_workdata = NULL;
- return 0;
+ return snd_soc_register_dai(&au1xpsc_i2s_dai);
}
static void __exit au1xpsc_i2s_exit(void)
{
+ snd_soc_unregister_dai(&au1xpsc_i2s_dai);
}
module_init(au1xpsc_i2s_init);
diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c
index f75ae7f..27683eb 100644
--- a/sound/soc/au1x/sample-ac97.c
+++ b/sound/soc/au1x/sample-ac97.c
@@ -42,14 +42,14 @@
.ops = NULL,
};
-static struct snd_soc_machine au1xpsc_sample_ac97_machine = {
+static struct snd_soc_card au1xpsc_sample_ac97_machine = {
.name = "Au1xxx PSC AC97 Audio",
.dai_link = &au1xpsc_sample_ac97_dai,
.num_links = 1,
};
static struct snd_soc_device au1xpsc_sample_ac97_devdata = {
- .machine = &au1xpsc_sample_ac97_machine,
+ .card = &au1xpsc_sample_ac97_machine,
.platform = &au1xpsc_soc_platform, /* see dbdma2.c */
.codec_dev = &soc_codec_dev_ac97,
};
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index dc00620..0a2f8f9 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -1,6 +1,6 @@
config SND_BF5XX_I2S
tristate "SoC I2S Audio for the ADI BF5xx chip"
- depends on BLACKFIN && SND_SOC
+ depends on BLACKFIN
help
Say Y or M if you want to add support for codecs attached to
the Blackfin SPORT (synchronous serial ports) interface in I2S
@@ -13,7 +13,6 @@
select SND_BF5XX_SOC_I2S
select SND_SOC_SSM2602
select I2C
- select I2C_BLACKFIN_TWI
help
Say Y if you want to add support for SoC audio on BF527-EZKIT.
@@ -35,7 +34,7 @@
config SND_BF5XX_AC97
tristate "SoC AC97 Audio for the ADI BF5xx chip"
- depends on BLACKFIN && SND_SOC
+ depends on BLACKFIN
help
Say Y or M if you want to add support for codecs attached to
the Blackfin SPORT (synchronous serial ports) interface in slot 16
@@ -47,7 +46,7 @@
properly with this driver. This driver is known to work with the
Analog Devices line of AC97 codecs.
-config SND_MMAP_SUPPORT
+config SND_BF5XX_MMAP_SUPPORT
bool "Enable MMAP Support"
depends on SND_BF5XX_AC97
default y
@@ -55,9 +54,17 @@
Say y if you want AC97 driver to support mmap mode.
We introduce an intermediate buffer to simulate mmap.
+config SND_BF5XX_MULTICHAN_SUPPORT
+ bool "Enable Multichannel Support"
+ depends on SND_BF5XX_AC97
+ default n
+ help
+ Say y if you want AC97 driver to support up to 5.1 channel audio.
+ this mode will consume much more memory for DMA.
+
config SND_BF5XX_SOC_SPORT
tristate
-
+
config SND_BF5XX_SOC_I2S
tristate
select SND_BF5XX_SOC_SPORT
@@ -80,7 +87,7 @@
int "Set a SPORT for Sound chip"
depends on (SND_BF5XX_I2S || SND_BF5XX_AC97)
range 0 3 if BF54x
- range 0 1 if (BF53x || BF561)
+ range 0 1 if !BF54x
default 0
help
Set the correct SPORT for sound chip.
@@ -90,12 +97,13 @@
depends on SND_BF5XX_AC97
default y if BFIN548_EZKIT
default n if !BFIN548_EZKIT
-
+
config SND_BF5XX_RESET_GPIO_NUM
int "Set a GPIO for cold reset"
depends on SND_BF5XX_HAVE_COLD_RESET
range 0 159
default 19 if BFIN548_EZKIT
default 5 if BFIN537_STAMP
+ default 0
help
Set the correct GPIO for RESET the sound chip.
diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c
index 25e50d2..8067cfa 100644
--- a/sound/soc/blackfin/bf5xx-ac97-pcm.c
+++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c
@@ -43,24 +43,34 @@
#include "bf5xx-ac97.h"
#include "bf5xx-sport.h"
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+static unsigned int ac97_chan_mask[] = {
+ SP_FL, /* Mono */
+ SP_STEREO, /* Stereo */
+ SP_2DOT1, /* 2.1*/
+ SP_QUAD,/*Quadraquic*/
+ SP_FL | SP_FR | SP_FC | SP_SL | SP_SR,/*5 channels */
+ SP_5DOT1, /* 5.1 */
+};
+
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
static void bf5xx_mmap_copy(struct snd_pcm_substream *substream,
snd_pcm_uframes_t count)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct sport_device *sport = runtime->private_data;
+ unsigned int chan_mask = ac97_chan_mask[runtime->channels - 1];
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- bf5xx_pcm_to_ac97(
- (struct ac97_frame *)sport->tx_dma_buf + sport->tx_pos,
- (__u32 *)runtime->dma_area + sport->tx_pos, count);
+ bf5xx_pcm_to_ac97((struct ac97_frame *)sport->tx_dma_buf +
+ sport->tx_pos, (__u16 *)runtime->dma_area + sport->tx_pos *
+ runtime->channels, count, chan_mask);
sport->tx_pos += runtime->period_size;
if (sport->tx_pos >= runtime->buffer_size)
sport->tx_pos %= runtime->buffer_size;
sport->tx_delay_pos = sport->tx_pos;
} else {
- bf5xx_ac97_to_pcm(
- (struct ac97_frame *)sport->rx_dma_buf + sport->rx_pos,
- (__u32 *)runtime->dma_area + sport->rx_pos, count);
+ bf5xx_ac97_to_pcm((struct ac97_frame *)sport->rx_dma_buf +
+ sport->rx_pos, (__u16 *)runtime->dma_area + sport->rx_pos *
+ runtime->channels, count);
sport->rx_pos += runtime->period_size;
if (sport->rx_pos >= runtime->buffer_size)
sport->rx_pos %= runtime->buffer_size;
@@ -71,7 +81,7 @@
static void bf5xx_dma_irq(void *data)
{
struct snd_pcm_substream *pcm = data;
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
struct snd_pcm_runtime *runtime = pcm->runtime;
struct sport_device *sport = runtime->private_data;
bf5xx_mmap_copy(pcm, runtime->period_size);
@@ -90,17 +100,14 @@
* The total rx/tx buffer is for ac97 frame to hold all pcm data
* is 0x20000 * sizeof(struct ac97_frame) / 4.
*/
-#ifdef CONFIG_SND_MMAP_SUPPORT
static const struct snd_pcm_hardware bf5xx_pcm_hardware = {
.info = SNDRV_PCM_INFO_INTERLEAVED |
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_BLOCK_TRANSFER,
-#else
-static const struct snd_pcm_hardware bf5xx_pcm_hardware = {
- .info = SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_BLOCK_TRANSFER,
#endif
+ SNDRV_PCM_INFO_BLOCK_TRANSFER,
+
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.period_bytes_min = 32,
.period_bytes_max = 0x10000,
@@ -123,10 +130,20 @@
static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream)
{
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
struct snd_pcm_runtime *runtime = substream->runtime;
+ struct sport_device *sport = runtime->private_data;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- memset(runtime->dma_area, 0, runtime->buffer_size);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ sport->once = 0;
+ if (runtime->dma_area)
+ memset(runtime->dma_area, 0, runtime->buffer_size);
+ memset(sport->tx_dma_buf, 0, runtime->buffer_size *
+ sizeof(struct ac97_frame));
+ } else
+ memset(sport->rx_dma_buf, 0, runtime->buffer_size *
+ sizeof(struct ac97_frame));
+#endif
snd_pcm_lib_free_pages(substream);
return 0;
}
@@ -139,7 +156,7 @@
/* An intermediate buffer is introduced for implementing mmap for
* SPORT working in TMD mode(include AC97).
*/
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
sport_set_tx_callback(sport, bf5xx_dma_irq, substream);
sport_config_tx_dma(sport, sport->tx_dma_buf, runtime->periods,
@@ -173,24 +190,24 @@
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
bf5xx_mmap_copy(substream, runtime->period_size);
- snd_pcm_period_elapsed(substream);
sport->tx_delay_pos = 0;
+#endif
sport_tx_start(sport);
- }
- else
+ } else
sport_rx_start(sport);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
sport->tx_pos = 0;
#endif
sport_tx_stop(sport);
} else {
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
sport->rx_pos = 0;
#endif
sport_rx_stop(sport);
@@ -208,7 +225,7 @@
struct sport_device *sport = runtime->private_data;
unsigned int curr;
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
curr = sport->tx_delay_pos;
else
@@ -249,22 +266,7 @@
return ret;
}
-static int bf5xx_pcm_close(struct snd_pcm_substream *substream)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct sport_device *sport = runtime->private_data;
-
- pr_debug("%s enter\n", __func__);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- sport->once = 0;
- memset(sport->tx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame));
- } else
- memset(sport->rx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame));
-
- return 0;
-}
-
-#ifdef CONFIG_SND_MMAP_SUPPORT
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream,
struct vm_area_struct *vma)
{
@@ -281,32 +283,29 @@
void __user *buf, snd_pcm_uframes_t count)
{
struct snd_pcm_runtime *runtime = substream->runtime;
-
+ unsigned int chan_mask = ac97_chan_mask[runtime->channels - 1];
pr_debug("%s copy pos:0x%lx count:0x%lx\n",
substream->stream ? "Capture" : "Playback", pos, count);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- bf5xx_pcm_to_ac97(
- (struct ac97_frame *)runtime->dma_area + pos,
- buf, count);
+ bf5xx_pcm_to_ac97((struct ac97_frame *)runtime->dma_area + pos,
+ (__u16 *)buf, count, chan_mask);
else
- bf5xx_ac97_to_pcm(
- (struct ac97_frame *)runtime->dma_area + pos,
- buf, count);
+ bf5xx_ac97_to_pcm((struct ac97_frame *)runtime->dma_area + pos,
+ (__u16 *)buf, count);
return 0;
}
#endif
struct snd_pcm_ops bf5xx_pcm_ac97_ops = {
.open = bf5xx_pcm_open,
- .close = bf5xx_pcm_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = bf5xx_pcm_hw_params,
.hw_free = bf5xx_pcm_hw_free,
.prepare = bf5xx_pcm_prepare,
.trigger = bf5xx_pcm_trigger,
.pointer = bf5xx_pcm_pointer,
-#ifdef CONFIG_SND_MMAP_SUPPORT
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
.mmap = bf5xx_pcm_mmap,
#else
.copy = bf5xx_pcm_copy,
@@ -344,7 +343,7 @@
* Need to allocate local buffer when enable
* MMAP for SPORT working in TMD mode (include AC97).
*/
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
if (!sport_handle->tx_dma_buf) {
sport_handle->tx_dma_buf = dma_alloc_coherent(NULL, \
@@ -381,7 +380,7 @@
struct snd_pcm_substream *substream;
struct snd_dma_buffer *buf;
int stream;
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
size_t size = bf5xx_pcm_hardware.buffer_bytes_max *
sizeof(struct ac97_frame) / 4;
#endif
@@ -395,7 +394,7 @@
continue;
dma_free_coherent(NULL, buf->bytes, buf->area, 0);
buf->area = NULL;
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
if (sport_handle->tx_dma_buf)
dma_free_coherent(NULL, size, \
@@ -452,6 +451,18 @@
};
EXPORT_SYMBOL_GPL(bf5xx_ac97_soc_platform);
+static int __init bfin_ac97_init(void)
+{
+ return snd_soc_register_platform(&bf5xx_ac97_soc_platform);
+}
+module_init(bfin_ac97_init);
+
+static void __exit bfin_ac97_exit(void)
+{
+ snd_soc_unregister_platform(&bf5xx_ac97_soc_platform);
+}
+module_exit(bfin_ac97_exit);
+
MODULE_AUTHOR("Cliff Cai");
MODULE_DESCRIPTION("ADI Blackfin AC97 PCM DMA module");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index 5e5aafb..3be2be6 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -54,71 +54,103 @@
static int *cmd_count;
static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM;
-#if defined(CONFIG_BF54x)
+static u16 sport_req[][7] = {
+ PIN_REQ_SPORT_0,
+#ifdef PIN_REQ_SPORT_1
+ PIN_REQ_SPORT_1,
+#endif
+#ifdef PIN_REQ_SPORT_2
+ PIN_REQ_SPORT_2,
+#endif
+#ifdef PIN_REQ_SPORT_3
+ PIN_REQ_SPORT_3,
+#endif
+ };
+
static struct sport_param sport_params[4] = {
{
.dma_rx_chan = CH_SPORT0_RX,
.dma_tx_chan = CH_SPORT0_TX,
- .err_irq = IRQ_SPORT0_ERR,
- .regs = (struct sport_register *)SPORT0_TCR1,
- },
- {
- .dma_rx_chan = CH_SPORT1_RX,
- .dma_tx_chan = CH_SPORT1_TX,
- .err_irq = IRQ_SPORT1_ERR,
- .regs = (struct sport_register *)SPORT1_TCR1,
- },
- {
- .dma_rx_chan = CH_SPORT2_RX,
- .dma_tx_chan = CH_SPORT2_TX,
- .err_irq = IRQ_SPORT2_ERR,
- .regs = (struct sport_register *)SPORT2_TCR1,
- },
- {
- .dma_rx_chan = CH_SPORT3_RX,
- .dma_tx_chan = CH_SPORT3_TX,
- .err_irq = IRQ_SPORT3_ERR,
- .regs = (struct sport_register *)SPORT3_TCR1,
- }
-};
-#else
-static struct sport_param sport_params[2] = {
- {
- .dma_rx_chan = CH_SPORT0_RX,
- .dma_tx_chan = CH_SPORT0_TX,
.err_irq = IRQ_SPORT0_ERROR,
.regs = (struct sport_register *)SPORT0_TCR1,
},
+#ifdef PIN_REQ_SPORT_1
{
.dma_rx_chan = CH_SPORT1_RX,
.dma_tx_chan = CH_SPORT1_TX,
.err_irq = IRQ_SPORT1_ERROR,
.regs = (struct sport_register *)SPORT1_TCR1,
- }
-};
+ },
#endif
+#ifdef PIN_REQ_SPORT_2
+ {
+ .dma_rx_chan = CH_SPORT2_RX,
+ .dma_tx_chan = CH_SPORT2_TX,
+ .err_irq = IRQ_SPORT2_ERROR,
+ .regs = (struct sport_register *)SPORT2_TCR1,
+ },
+#endif
+#ifdef PIN_REQ_SPORT_3
+ {
+ .dma_rx_chan = CH_SPORT3_RX,
+ .dma_tx_chan = CH_SPORT3_TX,
+ .err_irq = IRQ_SPORT3_ERROR,
+ .regs = (struct sport_register *)SPORT3_TCR1,
+ }
+#endif
+};
-void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u32 *src, \
- size_t count)
+void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u16 *src,
+ size_t count, unsigned int chan_mask)
{
while (count--) {
- dst->ac97_tag = TAG_VALID | TAG_PCM;
- (dst++)->ac97_pcm = *src++;
+ dst->ac97_tag = TAG_VALID;
+ if (chan_mask & SP_FL) {
+ dst->ac97_pcm_r = *src++;
+ dst->ac97_tag |= TAG_PCM_RIGHT;
+ }
+ if (chan_mask & SP_FR) {
+ dst->ac97_pcm_l = *src++;
+ dst->ac97_tag |= TAG_PCM_LEFT;
+
+ }
+#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT)
+ if (chan_mask & SP_SR) {
+ dst->ac97_sl = *src++;
+ dst->ac97_tag |= TAG_PCM_SL;
+ }
+ if (chan_mask & SP_SL) {
+ dst->ac97_sr = *src++;
+ dst->ac97_tag |= TAG_PCM_SR;
+ }
+ if (chan_mask & SP_LFE) {
+ dst->ac97_lfe = *src++;
+ dst->ac97_tag |= TAG_PCM_LFE;
+ }
+ if (chan_mask & SP_FC) {
+ dst->ac97_center = *src++;
+ dst->ac97_tag |= TAG_PCM_CENTER;
+ }
+#endif
+ dst++;
}
}
EXPORT_SYMBOL(bf5xx_pcm_to_ac97);
-void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u32 *dst, \
+void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u16 *dst,
size_t count)
{
- while (count--)
- *(dst++) = (src++)->ac97_pcm;
+ while (count--) {
+ *(dst++) = src->ac97_pcm_l;
+ *(dst++) = src->ac97_pcm_r;
+ src++;
+ }
}
EXPORT_SYMBOL(bf5xx_ac97_to_pcm);
static unsigned int sport_tx_curr_frag(struct sport_device *sport)
{
- return sport->tx_curr_frag = sport_curr_offset_tx(sport) / \
+ return sport->tx_curr_frag = sport_curr_offset_tx(sport) /
sport->tx_fragsize;
}
@@ -130,7 +162,7 @@
sport_incfrag(sport, &nextfrag, 1);
- nextwrite = (struct ac97_frame *)(sport->tx_buf + \
+ nextwrite = (struct ac97_frame *)(sport->tx_buf +
nextfrag * sport->tx_fragsize);
pr_debug("sport->tx_buf:%p, nextfrag:0x%x nextwrite:%p, cmd_count:%d\n",
sport->tx_buf, nextfrag, nextwrite, cmd_count[nextfrag]);
@@ -237,8 +269,7 @@
EXPORT_SYMBOL_GPL(soc_ac97_ops);
#ifdef CONFIG_PM
-static int bf5xx_ac97_suspend(struct platform_device *pdev,
- struct snd_soc_dai *dai)
+static int bf5xx_ac97_suspend(struct snd_soc_dai *dai)
{
struct sport_device *sport =
(struct sport_device *)dai->private_data;
@@ -253,8 +284,7 @@
return 0;
}
-static int bf5xx_ac97_resume(struct platform_device *pdev,
- struct snd_soc_dai *dai)
+static int bf5xx_ac97_resume(struct snd_soc_dai *dai)
{
int ret;
struct sport_device *sport =
@@ -297,20 +327,15 @@
static int bf5xx_ac97_probe(struct platform_device *pdev,
struct snd_soc_dai *dai)
{
- int ret;
-#if defined(CONFIG_BF54x)
- u16 sport_req[][7] = {PIN_REQ_SPORT_0, PIN_REQ_SPORT_1,
- PIN_REQ_SPORT_2, PIN_REQ_SPORT_3};
-#else
- u16 sport_req[][7] = {PIN_REQ_SPORT_0, PIN_REQ_SPORT_1};
-#endif
+ int ret = 0;
cmd_count = (int *)get_zeroed_page(GFP_KERNEL);
if (cmd_count == NULL)
return -ENOMEM;
if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) {
pr_err("Requesting Peripherals failed\n");
- return -EFAULT;
+ ret = -EFAULT;
+ goto peripheral_err;
}
#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
@@ -318,54 +343,54 @@
if (gpio_request(CONFIG_SND_BF5XX_RESET_GPIO_NUM, "SND_AD198x RESET")) {
pr_err("Failed to request GPIO_%d for reset\n",
CONFIG_SND_BF5XX_RESET_GPIO_NUM);
- peripheral_free_list(&sport_req[sport_num][0]);
- return -1;
+ ret = -1;
+ goto gpio_err;
}
gpio_direction_output(CONFIG_SND_BF5XX_RESET_GPIO_NUM, 1);
#endif
sport_handle = sport_init(&sport_params[sport_num], 2, \
sizeof(struct ac97_frame), NULL);
if (!sport_handle) {
- peripheral_free_list(&sport_req[sport_num][0]);
-#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
- gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
-#endif
- return -ENODEV;
+ ret = -ENODEV;
+ goto sport_err;
}
/*SPORT works in TDM mode to simulate AC97 transfers*/
ret = sport_set_multichannel(sport_handle, 16, 0x1F, 1);
if (ret) {
pr_err("SPORT is busy!\n");
- kfree(sport_handle);
- peripheral_free_list(&sport_req[sport_num][0]);
-#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
- gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
-#endif
- return -EBUSY;
+ ret = -EBUSY;
+ goto sport_config_err;
}
ret = sport_config_rx(sport_handle, IRFS, 0xF, 0, (16*16-1));
if (ret) {
pr_err("SPORT is busy!\n");
- kfree(sport_handle);
- peripheral_free_list(&sport_req[sport_num][0]);
-#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
- gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
-#endif
- return -EBUSY;
+ ret = -EBUSY;
+ goto sport_config_err;
}
ret = sport_config_tx(sport_handle, ITFS, 0xF, 0, (16*16-1));
if (ret) {
pr_err("SPORT is busy!\n");
- kfree(sport_handle);
- peripheral_free_list(&sport_req[sport_num][0]);
-#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
- gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
-#endif
- return -EBUSY;
+ ret = -EBUSY;
+ goto sport_config_err;
}
+
return 0;
+
+sport_config_err:
+ kfree(sport_handle);
+sport_err:
+#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
+ gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
+#endif
+gpio_err:
+ peripheral_free_list(&sport_req[sport_num][0]);
+peripheral_err:
+ free_page((unsigned long)cmd_count);
+ cmd_count = NULL;
+
+ return ret;
}
static void bf5xx_ac97_remove(struct platform_device *pdev,
@@ -373,6 +398,7 @@
{
free_page((unsigned long)cmd_count);
cmd_count = NULL;
+ peripheral_free_list(&sport_req[sport_num][0]);
#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
#endif
@@ -381,7 +407,7 @@
struct snd_soc_dai bfin_ac97_dai = {
.name = "bf5xx-ac97",
.id = 0,
- .type = SND_SOC_DAI_AC97,
+ .ac97_control = 1,
.probe = bf5xx_ac97_probe,
.remove = bf5xx_ac97_remove,
.suspend = bf5xx_ac97_suspend,
@@ -389,7 +415,11 @@
.playback = {
.stream_name = "AC97 Playback",
.channels_min = 2,
+#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT)
+ .channels_max = 6,
+#else
.channels_max = 2,
+#endif
.rates = SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE, },
.capture = {
@@ -401,6 +431,18 @@
};
EXPORT_SYMBOL_GPL(bfin_ac97_dai);
+static int __init bfin_ac97_init(void)
+{
+ return snd_soc_register_dai(&bfin_ac97_dai);
+}
+module_init(bfin_ac97_init);
+
+static void __exit bfin_ac97_exit(void)
+{
+ snd_soc_unregister_dai(&bfin_ac97_dai);
+}
+module_exit(bfin_ac97_exit);
+
MODULE_AUTHOR("Roy Huang");
MODULE_DESCRIPTION("AC97 driver for ADI Blackfin");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h
index 3f77cc5..3f2a911 100644
--- a/sound/soc/blackfin/bf5xx-ac97.h
+++ b/sound/soc/blackfin/bf5xx-ac97.h
@@ -16,21 +16,46 @@
u16 ac97_tag; /* slot 0 */
u16 ac97_addr; /* slot 1 */
u16 ac97_data; /* slot 2 */
- u32 ac97_pcm; /* slot 3 and 4: left and right pcm data */
+ u16 ac97_pcm_l; /*slot 3:front left*/
+ u16 ac97_pcm_r; /*slot 4:front left*/
+#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT)
+ u16 ac97_mdm_l1;
+ u16 ac97_center; /*slot 6:center*/
+ u16 ac97_sl; /*slot 7:surround left*/
+ u16 ac97_sr; /*slot 8:surround right*/
+ u16 ac97_lfe; /*slot 9:lfe*/
+#endif
} __attribute__ ((packed));
+/* Speaker location */
+#define SP_FL 0x0001
+#define SP_FR 0x0010
+#define SP_FC 0x0002
+#define SP_LFE 0x0020
+#define SP_SL 0x0004
+#define SP_SR 0x0040
+
+#define SP_STEREO (SP_FL | SP_FR)
+#define SP_2DOT1 (SP_FL | SP_FR | SP_LFE)
+#define SP_QUAD (SP_FL | SP_FR | SP_SL | SP_SR)
+#define SP_5DOT1 (SP_FL | SP_FR | SP_FC | SP_LFE | SP_SL | SP_SR)
+
#define TAG_VALID 0x8000
#define TAG_CMD 0x6000
#define TAG_PCM_LEFT 0x1000
#define TAG_PCM_RIGHT 0x0800
-#define TAG_PCM (TAG_PCM_LEFT | TAG_PCM_RIGHT)
+#define TAG_PCM_MDM_L1 0x0400
+#define TAG_PCM_CENTER 0x0200
+#define TAG_PCM_SL 0x0100
+#define TAG_PCM_SR 0x0080
+#define TAG_PCM_LFE 0x0040
extern struct snd_soc_dai bfin_ac97_dai;
-void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u32 *src, \
- size_t count);
+void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u16 *src, \
+ size_t count, unsigned int chan_mask);
-void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u32 *dst, \
+void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u16 *dst, \
size_t count);
#endif
diff --git a/sound/soc/blackfin/bf5xx-ad1980.c b/sound/soc/blackfin/bf5xx-ad1980.c
index 124425d..d8f5912 100644
--- a/sound/soc/blackfin/bf5xx-ad1980.c
+++ b/sound/soc/blackfin/bf5xx-ad1980.c
@@ -43,7 +43,7 @@
#include "bf5xx-ac97-pcm.h"
#include "bf5xx-ac97.h"
-static struct snd_soc_machine bf5xx_board;
+static struct snd_soc_card bf5xx_board;
static int bf5xx_board_startup(struct snd_pcm_substream *substream)
{
@@ -67,15 +67,15 @@
.ops = &bf5xx_board_ops,
};
-static struct snd_soc_machine bf5xx_board = {
+static struct snd_soc_card bf5xx_board = {
.name = "bf5xx-board",
+ .platform = &bf5xx_ac97_soc_platform,
.dai_link = &bf5xx_board_dai,
.num_links = 1,
};
static struct snd_soc_device bf5xx_board_snd_devdata = {
- .machine = &bf5xx_board,
- .platform = &bf5xx_ac97_soc_platform,
+ .card = &bf5xx_board,
.codec_dev = &soc_codec_dev_ad1980,
};
diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c
index 622c9b9..7f2a5e1 100644
--- a/sound/soc/blackfin/bf5xx-ad73311.c
+++ b/sound/soc/blackfin/bf5xx-ad73311.c
@@ -65,7 +65,7 @@
#define GPIO_SE CONFIG_SND_BFIN_AD73311_SE
-static struct snd_soc_machine bf5xx_ad73311;
+static struct snd_soc_card bf5xx_ad73311;
static int snd_ad73311_startup(void)
{
@@ -168,7 +168,7 @@
params_format(params));
/* set cpu DAI configuration */
- ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
if (ret < 0)
return ret;
@@ -190,16 +190,16 @@
.ops = &bf5xx_ad73311_ops,
};
-static struct snd_soc_machine bf5xx_ad73311 = {
+static struct snd_soc_card bf5xx_ad73311 = {
.name = "bf5xx_ad73311",
+ .platform = &bf5xx_i2s_soc_platform,
.probe = bf5xx_probe,
.dai_link = &bf5xx_ad73311_dai,
.num_links = 1,
};
static struct snd_soc_device bf5xx_ad73311_snd_devdata = {
- .machine = &bf5xx_ad73311,
- .platform = &bf5xx_i2s_soc_platform,
+ .card = &bf5xx_ad73311,
.codec_dev = &soc_codec_dev_ad73311,
};
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index 61fccf9..53d290b 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -283,6 +283,18 @@
};
EXPORT_SYMBOL_GPL(bf5xx_i2s_soc_platform);
+static int __init bfin_i2s_init(void)
+{
+ return snd_soc_register_platform(&bf5xx_i2s_soc_platform);
+}
+module_init(bfin_i2s_init);
+
+static void __exit bfin_i2s_exit(void)
+{
+ snd_soc_unregister_platform(&bf5xx_i2s_soc_platform);
+}
+module_exit(bfin_i2s_exit);
+
MODULE_AUTHOR("Cliff Cai");
MODULE_DESCRIPTION("ADI Blackfin I2S PCM DMA module");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index e020c16..d1d95d2 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -132,7 +132,8 @@
return ret;
}
-static int bf5xx_i2s_startup(struct snd_pcm_substream *substream)
+static int bf5xx_i2s_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
pr_debug("%s enter\n", __func__);
@@ -142,7 +143,8 @@
}
static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
int ret = 0;
@@ -193,7 +195,8 @@
return 0;
}
-static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream)
+static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
pr_debug("%s enter\n", __func__);
bf5xx_i2s.counter--;
@@ -219,16 +222,14 @@
return 0;
}
-static void bf5xx_i2s_remove(struct platform_device *pdev,
- struct snd_soc_dai *dai)
+static void bf5xx_i2s_remove(struct snd_soc_dai *dai)
{
pr_debug("%s enter\n", __func__);
peripheral_free_list(&sport_req[sport_num][0]);
}
#ifdef CONFIG_PM
-static int bf5xx_i2s_suspend(struct platform_device *dev,
- struct snd_soc_dai *dai)
+static int bf5xx_i2s_suspend(struct snd_soc_dai *dai)
{
struct sport_device *sport =
(struct sport_device *)dai->private_data;
@@ -289,7 +290,6 @@
struct snd_soc_dai bf5xx_i2s_dai = {
.name = "bf5xx-i2s",
.id = 0,
- .type = SND_SOC_DAI_I2S,
.probe = bf5xx_i2s_probe,
.remove = bf5xx_i2s_remove,
.suspend = bf5xx_i2s_suspend,
@@ -307,13 +307,24 @@
.ops = {
.startup = bf5xx_i2s_startup,
.shutdown = bf5xx_i2s_shutdown,
- .hw_params = bf5xx_i2s_hw_params,},
- .dai_ops = {
+ .hw_params = bf5xx_i2s_hw_params,
.set_fmt = bf5xx_i2s_set_dai_fmt,
},
};
EXPORT_SYMBOL_GPL(bf5xx_i2s_dai);
+static int __init bfin_i2s_init(void)
+{
+ return snd_soc_register_dai(&bf5xx_i2s_dai);
+}
+module_init(bfin_i2s_init);
+
+static void __exit bfin_i2s_exit(void)
+{
+ snd_soc_unregister_dai(&bf5xx_i2s_dai);
+}
+module_exit(bfin_i2s_exit);
+
/* Module information */
MODULE_AUTHOR("Cliff Cai");
MODULE_DESCRIPTION("I2S driver for ADI Blackfin");
diff --git a/sound/soc/blackfin/bf5xx-sport.h b/sound/soc/blackfin/bf5xx-sport.h
index fcadcc0..2e63dea 100644
--- a/sound/soc/blackfin/bf5xx-sport.h
+++ b/sound/soc/blackfin/bf5xx-sport.h
@@ -116,7 +116,7 @@
void *err_data;
unsigned char *tx_dma_buf;
unsigned char *rx_dma_buf;
-#ifdef CONFIG_SND_MMAP_SUPPORT
+#ifdef CONFIG_SND_BF5XX_MMAP_SUPPORT
dma_addr_t tx_dma_phy;
dma_addr_t rx_dma_phy;
int tx_pos;/*pcm sample count*/
diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c
index e15f67f..bc0cdde 100644
--- a/sound/soc/blackfin/bf5xx-ssm2602.c
+++ b/sound/soc/blackfin/bf5xx-ssm2602.c
@@ -44,7 +44,7 @@
#include "bf5xx-i2s-pcm.h"
#include "bf5xx-i2s.h"
-static struct snd_soc_machine bf5xx_ssm2602;
+static struct snd_soc_card bf5xx_ssm2602;
static int bf5xx_ssm2602_startup(struct snd_pcm_substream *substream)
{
@@ -92,17 +92,17 @@
*/
/* set codec DAI configuration */
- ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
if (ret < 0)
return ret;
/* set cpu DAI configuration */
- ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
if (ret < 0)
return ret;
- ret = codec_dai->dai_ops.set_sysclk(codec_dai, SSM2602_SYSCLK, clk,
+ ret = snd_soc_dai_set_sysclk(codec_dai, SSM2602_SYSCLK, clk,
SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
@@ -135,15 +135,15 @@
.i2c_address = 0x1b,
};
-static struct snd_soc_machine bf5xx_ssm2602 = {
+static struct snd_soc_card bf5xx_ssm2602 = {
.name = "bf5xx_ssm2602",
+ .platform = &bf5xx_i2s_soc_platform,
.dai_link = &bf5xx_ssm2602_dai,
.num_links = 1,
};
static struct snd_soc_device bf5xx_ssm2602_snd_devdata = {
- .machine = &bf5xx_ssm2602,
- .platform = &bf5xx_i2s_soc_platform,
+ .card = &bf5xx_ssm2602,
.codec_dev = &soc_codec_dev_ssm2602,
.codec_data = &bf5xx_ssm2602_setup,
};
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 38a0e3b..c41289b 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -1,31 +1,40 @@
config SND_SOC_ALL_CODECS
tristate "Build all ASoC CODEC drivers"
- depends on I2C
- select SPI
- select SPI_MASTER
- select SND_SOC_AD73311
- select SND_SOC_AK4535
- select SND_SOC_CS4270
- select SND_SOC_SSM2602
- select SND_SOC_TLV320AIC23
- select SND_SOC_TLV320AIC26
- select SND_SOC_TLV320AIC3X
- select SND_SOC_UDA1380
- select SND_SOC_WM8510
- select SND_SOC_WM8580
- select SND_SOC_WM8731
- select SND_SOC_WM8750
- select SND_SOC_WM8753
- select SND_SOC_WM8900
- select SND_SOC_WM8903
- select SND_SOC_WM8971
- select SND_SOC_WM8990
+ select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS
+ select SND_SOC_AD1980 if SND_SOC_AC97_BUS
+ select SND_SOC_AD73311 if I2C
+ select SND_SOC_AK4535 if I2C
+ select SND_SOC_CS4270 if I2C
+ select SND_SOC_PCM3008
+ select SND_SOC_SSM2602 if I2C
+ select SND_SOC_TLV320AIC23 if I2C
+ select SND_SOC_TLV320AIC26 if SPI_MASTER
+ select SND_SOC_TLV320AIC3X if I2C
+ select SND_SOC_TWL4030 if TWL4030_CORE
+ select SND_SOC_UDA134X
+ select SND_SOC_UDA1380 if I2C
+ select SND_SOC_WM8350 if MFD_WM8350
+ select SND_SOC_WM8510 if (I2C || SPI_MASTER)
+ select SND_SOC_WM8580 if I2C
+ select SND_SOC_WM8728 if (I2C || SPI_MASTER)
+ select SND_SOC_WM8731 if (I2C || SPI_MASTER)
+ select SND_SOC_WM8750 if (I2C || SPI_MASTER)
+ select SND_SOC_WM8753 if (I2C || SPI_MASTER)
+ select SND_SOC_WM8900 if I2C
+ select SND_SOC_WM8903 if I2C
+ select SND_SOC_WM8971 if I2C
+ select SND_SOC_WM8990 if I2C
+ select SND_SOC_WM9712 if SND_SOC_AC97_BUS
+ select SND_SOC_WM9713 if SND_SOC_AC97_BUS
help
Normally ASoC codec drivers are only built if a machine driver which
uses them is also built since they are only usable with a machine
driver. Selecting this option will allow these drivers to be built
without an explicit machine driver for test and development purposes.
+ Support for the bus types used to access the codecs to be built must
+ be selected separately.
+
If unsure select "N".
@@ -60,6 +69,12 @@
bool
depends on SND_SOC_CS4270
+config SND_SOC_L3
+ tristate
+
+config SND_SOC_PCM3008
+ tristate
+
config SND_SOC_SSM2602
tristate
@@ -75,15 +90,29 @@
tristate
depends on I2C
+config SND_SOC_TWL4030
+ tristate
+ depends on TWL4030_CORE
+
+config SND_SOC_UDA134X
+ tristate
+ select SND_SOC_L3
+
config SND_SOC_UDA1380
tristate
+config SND_SOC_WM8350
+ tristate
+
config SND_SOC_WM8510
tristate
config SND_SOC_WM8580
tristate
+config SND_SOC_WM8728
+ tristate
+
config SND_SOC_WM8731
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 90f0a58..c4ddc9a 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -3,13 +3,19 @@
snd-soc-ad73311-objs := ad73311.o
snd-soc-ak4535-objs := ak4535.o
snd-soc-cs4270-objs := cs4270.o
+snd-soc-l3-objs := l3.o
+snd-soc-pcm3008-objs := pcm3008.o
snd-soc-ssm2602-objs := ssm2602.o
snd-soc-tlv320aic23-objs := tlv320aic23.o
snd-soc-tlv320aic26-objs := tlv320aic26.o
snd-soc-tlv320aic3x-objs := tlv320aic3x.o
+snd-soc-twl4030-objs := twl4030.o
+snd-soc-uda134x-objs := uda134x.o
snd-soc-uda1380-objs := uda1380.o
+snd-soc-wm8350-objs := wm8350.o
snd-soc-wm8510-objs := wm8510.o
snd-soc-wm8580-objs := wm8580.o
+snd-soc-wm8728-objs := wm8728.o
snd-soc-wm8731-objs := wm8731.o
snd-soc-wm8750-objs := wm8750.o
snd-soc-wm8753-objs := wm8753.o
@@ -25,13 +31,19 @@
obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
+obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o
+obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o
obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o
obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o
obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o
obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o
+obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o
+obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o
obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o
+obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o
obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o
obj-$(CONFIG_SND_SOC_WM8580) += snd-soc-wm8580.o
+obj-$(CONFIG_SND_SOC_WM8728) += snd-soc-wm8728.o
obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o
obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o
obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index bd1ebdc..fb53e65 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -24,7 +24,8 @@
#define AC97_VERSION "0.6"
-static int ac97_prepare(struct snd_pcm_substream *substream)
+static int ac97_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -42,7 +43,7 @@
struct snd_soc_dai ac97_dai = {
.name = "AC97 HiFi",
- .type = SND_SOC_DAI_AC97,
+ .ac97_control = 1,
.playback = {
.stream_name = "AC97 Playback",
.channels_min = 1,
@@ -113,7 +114,7 @@
if (ret < 0)
goto bus_err;
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0)
goto bus_err;
return 0;
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index 1397b8e..73fdbb4 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -85,6 +85,9 @@
SOC_DOUBLE("Surround Playback Volume", AC97_SURROUND_MASTER, 8, 0, 31, 1),
SOC_DOUBLE("Surround Playback Switch", AC97_SURROUND_MASTER, 15, 7, 1, 1),
+SOC_DOUBLE("Center/LFE Playback Volume", AC97_CENTER_LFE_MASTER, 8, 0, 31, 1),
+SOC_DOUBLE("Center/LFE Playback Switch", AC97_CENTER_LFE_MASTER, 15, 7, 1, 1),
+
SOC_ENUM("Capture Source", ad1980_cap_src),
SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0),
@@ -142,10 +145,11 @@
struct snd_soc_dai ad1980_dai = {
.name = "AC97",
+ .ac97_control = 1,
.playback = {
.stream_name = "Playback",
.channels_min = 2,
- .channels_max = 2,
+ .channels_max = 6,
.rates = SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE, },
.capture = {
@@ -192,6 +196,7 @@
struct snd_soc_codec *codec;
int ret = 0;
u16 vendor_id2;
+ u16 ext_status;
printk(KERN_INFO "AD1980 SoC Audio Codec\n");
@@ -234,7 +239,7 @@
ret = ad1980_reset(codec, 0);
if (ret < 0) {
- printk(KERN_ERR "AC97 link error\n");
+ printk(KERN_ERR "Failed to reset AD1980: AC97 link error\n");
goto reset_err;
}
@@ -253,12 +258,19 @@
"supported\n");
}
- ac97_write(codec, AC97_MASTER, 0x0000); /* unmute line out volume */
- ac97_write(codec, AC97_PCM, 0x0000); /* unmute PCM out volume */
- ac97_write(codec, AC97_REC_GAIN, 0x0000);/* unmute record volume */
+ /* unmute captures and playbacks volume */
+ ac97_write(codec, AC97_MASTER, 0x0000);
+ ac97_write(codec, AC97_PCM, 0x0000);
+ ac97_write(codec, AC97_REC_GAIN, 0x0000);
+ ac97_write(codec, AC97_CENTER_LFE_MASTER, 0x0000);
+ ac97_write(codec, AC97_SURROUND_MASTER, 0x0000);
+
+ /*power on LFE/CENTER/Surround DACs*/
+ ext_status = ac97_read(codec, AC97_EXTENDED_STATUS);
+ ac97_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800);
ad1980_add_controls(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "ad1980: failed to register card\n");
goto reset_err;
diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c
index 37af860..b09289a 100644
--- a/sound/soc/codecs/ad73311.c
+++ b/sound/soc/codecs/ad73311.c
@@ -8,14 +8,10 @@
* 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.
- *
- * Revision history
- * 25th Sep 2008 Initial version.
*/
#include <linux/init.h>
#include <linux/module.h>
-#include <linux/version.h>
#include <linux/kernel.h>
#include <linux/device.h>
#include <sound/core.h>
@@ -68,7 +64,7 @@
goto pcm_err;
}
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "ad73311: failed to register card\n");
goto register_err;
@@ -102,6 +98,18 @@
};
EXPORT_SYMBOL_GPL(soc_codec_dev_ad73311);
+static int __init ad73311_init(void)
+{
+ return snd_soc_register_dai(&ad73311_dai);
+}
+module_init(ad73311_init);
+
+static void __exit ad73311_exit(void)
+{
+ snd_soc_unregister_dai(&ad73311_dai);
+}
+module_exit(ad73311_exit);
+
MODULE_DESCRIPTION("ASoC ad73311 driver");
MODULE_AUTHOR("Cliff Cai ");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 2a89b58..81300d8 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -339,7 +339,8 @@
}
static int ak4535_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -451,8 +452,6 @@
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
.ops = {
.hw_params = ak4535_hw_params,
- },
- .dai_ops = {
.set_fmt = ak4535_set_dai_fmt,
.digital_mute = ak4535_mute,
.set_sysclk = ak4535_set_dai_sysclk,
@@ -513,7 +512,7 @@
ak4535_add_controls(codec);
ak4535_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "ak4535: failed to register card\n");
goto card_err;
@@ -689,6 +688,18 @@
};
EXPORT_SYMBOL_GPL(soc_codec_dev_ak4535);
+static int __init ak4535_modinit(void)
+{
+ return snd_soc_register_dai(&ak4535_dai);
+}
+module_init(ak4535_modinit);
+
+static void __exit ak4535_exit(void)
+{
+ snd_soc_unregister_dai(&ak4535_dai);
+}
+module_exit(ak4535_exit);
+
MODULE_DESCRIPTION("Soc AK4535 driver");
MODULE_AUTHOR("Richard Purdie");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 0bbd945..f1aa0c3 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -360,13 +360,14 @@
/*
* Program the CS4270 with the given hardware parameters.
*
- * The .dai_ops functions are used to provide board-specific data, like
+ * The .ops functions are used to provide board-specific data, like
* input frequencies, to this driver. This function takes that information,
* combines it with the hardware parameters provided, and programs the
* hardware accordingly.
*/
static int cs4270_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -450,6 +451,19 @@
return ret;
}
+ /* Disable automatic volume control. It's enabled by default, and
+ * it causes volume change commands to be delayed, sometimes until
+ * after playback has started.
+ */
+
+ reg = cs4270_read_reg_cache(codec, CS4270_TRANS);
+ reg &= ~(CS4270_TRANS_SOFT | CS4270_TRANS_ZERO);
+ ret = cs4270_i2c_write(codec, CS4270_TRANS, reg);
+ if (ret < 0) {
+ printk(KERN_ERR "I2C write failed\n");
+ return ret;
+ }
+
/* Thaw and power-up the codec */
ret = snd_soc_write(codec, CS4270_PWRCTL, 0);
@@ -697,10 +711,10 @@
if (codec->control_data) {
/* Initialize codec ops */
cs4270_dai.ops.hw_params = cs4270_hw_params;
- cs4270_dai.dai_ops.set_sysclk = cs4270_set_dai_sysclk;
- cs4270_dai.dai_ops.set_fmt = cs4270_set_dai_fmt;
+ cs4270_dai.ops.set_sysclk = cs4270_set_dai_sysclk;
+ cs4270_dai.ops.set_fmt = cs4270_set_dai_fmt;
#ifdef CONFIG_SND_SOC_CS4270_HWMUTE
- cs4270_dai.dai_ops.digital_mute = cs4270_mute;
+ cs4270_dai.ops.digital_mute = cs4270_mute;
#endif
} else
printk(KERN_INFO "cs4270: no I2C device found, "
@@ -709,7 +723,7 @@
printk(KERN_INFO "cs4270: I2C disabled, using stand-alone mode\n");
#endif
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "cs4270: failed to register card\n");
goto error_del_driver;
@@ -760,6 +774,18 @@
};
EXPORT_SYMBOL_GPL(soc_codec_device_cs4270);
+static int __init cs4270_init(void)
+{
+ return snd_soc_register_dai(&cs4270_dai);
+}
+module_init(cs4270_init);
+
+static void __exit cs4270_exit(void)
+{
+ snd_soc_unregister_dai(&cs4270_dai);
+}
+module_exit(cs4270_exit);
+
MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
MODULE_DESCRIPTION("Cirrus Logic CS4270 ALSA SoC Codec Driver");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/l3.c b/sound/soc/codecs/l3.c
new file mode 100644
index 0000000..5353af5
--- /dev/null
+++ b/sound/soc/codecs/l3.c
@@ -0,0 +1,91 @@
+/*
+ * L3 code
+ *
+ * Copyright (C) 2008, Christian Pellegrin <chripell@evolware.org>
+ *
+ * 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.
+ *
+ *
+ * based on:
+ *
+ * L3 bus algorithm module.
+ *
+ * Copyright (C) 2001 Russell King, All Rights Reserved.
+ *
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/delay.h>
+
+#include <sound/l3.h>
+
+/*
+ * Send one byte of data to the chip. Data is latched into the chip on
+ * the rising edge of the clock.
+ */
+static void sendbyte(struct l3_pins *adap, unsigned int byte)
+{
+ int i;
+
+ for (i = 0; i < 8; i++) {
+ adap->setclk(0);
+ udelay(adap->data_hold);
+ adap->setdat(byte & 1);
+ udelay(adap->data_setup);
+ adap->setclk(1);
+ udelay(adap->clock_high);
+ byte >>= 1;
+ }
+}
+
+/*
+ * Send a set of bytes to the chip. We need to pulse the MODE line
+ * between each byte, but never at the start nor at the end of the
+ * transfer.
+ */
+static void sendbytes(struct l3_pins *adap, const u8 *buf,
+ int len)
+{
+ int i;
+
+ for (i = 0; i < len; i++) {
+ if (i) {
+ udelay(adap->mode_hold);
+ adap->setmode(0);
+ udelay(adap->mode);
+ }
+ adap->setmode(1);
+ udelay(adap->mode_setup);
+ sendbyte(adap, buf[i]);
+ }
+}
+
+int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len)
+{
+ adap->setclk(1);
+ adap->setdat(1);
+ adap->setmode(1);
+ udelay(adap->mode);
+
+ adap->setmode(0);
+ udelay(adap->mode_setup);
+ sendbyte(adap, addr);
+ udelay(adap->mode_hold);
+
+ sendbytes(adap, data, len);
+
+ adap->setclk(1);
+ adap->setdat(1);
+ adap->setmode(0);
+
+ return len;
+}
+EXPORT_SYMBOL_GPL(l3_write);
+
+MODULE_DESCRIPTION("L3 bit-banging driver");
+MODULE_AUTHOR("Christian Pellegrin <chripell@evolware.org>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c
new file mode 100644
index 0000000..9a3e67e
--- /dev/null
+++ b/sound/soc/codecs/pcm3008.c
@@ -0,0 +1,212 @@
+/*
+ * ALSA Soc PCM3008 codec support
+ *
+ * Author: Hugo Villeneuve
+ * Copyright (C) 2008 Lyrtech inc
+ *
+ * Based on AC97 Soc codec, original copyright follow:
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ *
+ * 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.
+ *
+ * Generic PCM3008 support.
+ */
+
+#include <linux/init.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <linux/gpio.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include "pcm3008.h"
+
+#define PCM3008_VERSION "0.2"
+
+#define PCM3008_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000)
+
+struct snd_soc_dai pcm3008_dai = {
+ .name = "PCM3008 HiFi",
+ .playback = {
+ .stream_name = "PCM3008 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = PCM3008_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .stream_name = "PCM3008 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = PCM3008_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+};
+EXPORT_SYMBOL_GPL(pcm3008_dai);
+
+static void pcm3008_gpio_free(struct pcm3008_setup_data *setup)
+{
+ gpio_free(setup->dem0_pin);
+ gpio_free(setup->dem1_pin);
+ gpio_free(setup->pdad_pin);
+ gpio_free(setup->pdda_pin);
+}
+
+static int pcm3008_soc_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ struct pcm3008_setup_data *setup = socdev->codec_data;
+ int ret = 0;
+
+ printk(KERN_INFO "PCM3008 SoC Audio Codec %s\n", PCM3008_VERSION);
+
+ socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (!socdev->codec)
+ return -ENOMEM;
+
+ codec = socdev->codec;
+ mutex_init(&codec->mutex);
+
+ codec->name = "PCM3008";
+ codec->owner = THIS_MODULE;
+ codec->dai = &pcm3008_dai;
+ codec->num_dai = 1;
+ codec->write = NULL;
+ codec->read = NULL;
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ /* Register PCMs. */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "pcm3008: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ /* Register Card. */
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "pcm3008: failed to register card\n");
+ goto card_err;
+ }
+
+ /* DEM1 DEM0 DE-EMPHASIS_MODE
+ * Low Low De-emphasis 44.1 kHz ON
+ * Low High De-emphasis OFF
+ * High Low De-emphasis 48 kHz ON
+ * High High De-emphasis 32 kHz ON
+ */
+
+ /* Configure DEM0 GPIO (turning OFF DAC De-emphasis). */
+ ret = gpio_request(setup->dem0_pin, "codec_dem0");
+ if (ret == 0)
+ ret = gpio_direction_output(setup->dem0_pin, 1);
+ if (ret != 0)
+ goto gpio_err;
+
+ /* Configure DEM1 GPIO (turning OFF DAC De-emphasis). */
+ ret = gpio_request(setup->dem1_pin, "codec_dem1");
+ if (ret == 0)
+ ret = gpio_direction_output(setup->dem1_pin, 0);
+ if (ret != 0)
+ goto gpio_err;
+
+ /* Configure PDAD GPIO. */
+ ret = gpio_request(setup->pdad_pin, "codec_pdad");
+ if (ret == 0)
+ ret = gpio_direction_output(setup->pdad_pin, 1);
+ if (ret != 0)
+ goto gpio_err;
+
+ /* Configure PDDA GPIO. */
+ ret = gpio_request(setup->pdda_pin, "codec_pdda");
+ if (ret == 0)
+ ret = gpio_direction_output(setup->pdda_pin, 1);
+ if (ret != 0)
+ goto gpio_err;
+
+ return ret;
+
+gpio_err:
+ pcm3008_gpio_free(setup);
+card_err:
+ snd_soc_free_pcms(socdev);
+pcm_err:
+ kfree(socdev->codec);
+
+ return ret;
+}
+
+static int pcm3008_soc_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ struct pcm3008_setup_data *setup = socdev->codec_data;
+
+ if (!codec)
+ return 0;
+
+ pcm3008_gpio_free(setup);
+ snd_soc_free_pcms(socdev);
+ kfree(socdev->codec);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int pcm3008_soc_suspend(struct platform_device *pdev, pm_message_t msg)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct pcm3008_setup_data *setup = socdev->codec_data;
+
+ gpio_set_value(setup->pdad_pin, 0);
+ gpio_set_value(setup->pdda_pin, 0);
+
+ return 0;
+}
+
+static int pcm3008_soc_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct pcm3008_setup_data *setup = socdev->codec_data;
+
+ gpio_set_value(setup->pdad_pin, 1);
+ gpio_set_value(setup->pdda_pin, 1);
+
+ return 0;
+}
+#else
+#define pcm3008_soc_suspend NULL
+#define pcm3008_soc_resume NULL
+#endif
+
+struct snd_soc_codec_device soc_codec_dev_pcm3008 = {
+ .probe = pcm3008_soc_probe,
+ .remove = pcm3008_soc_remove,
+ .suspend = pcm3008_soc_suspend,
+ .resume = pcm3008_soc_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_pcm3008);
+
+static int __init pcm3008_init(void)
+{
+ return snd_soc_register_dai(&pcm3008_dai);
+}
+module_init(pcm3008_init);
+
+static void __exit pcm3008_exit(void)
+{
+ snd_soc_unregister_dai(&pcm3008_dai);
+}
+module_exit(pcm3008_exit);
+
+MODULE_DESCRIPTION("Soc PCM3008 driver");
+MODULE_AUTHOR("Hugo Villeneuve");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/pcm3008.h b/sound/soc/codecs/pcm3008.h
new file mode 100644
index 0000000..d04e87d
--- /dev/null
+++ b/sound/soc/codecs/pcm3008.h
@@ -0,0 +1,25 @@
+/*
+ * PCM3008 ALSA SoC Layer
+ *
+ * Author: Hugo Villeneuve
+ * Copyright (C) 2008 Lyrtech inc
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_SOC_PCM3008_H
+#define __LINUX_SND_SOC_PCM3008_H
+
+struct pcm3008_setup_data {
+ unsigned dem0_pin;
+ unsigned dem1_pin;
+ unsigned pdad_pin;
+ unsigned pdda_pin;
+};
+
+extern struct snd_soc_codec_device soc_codec_dev_pcm3008;
+extern struct snd_soc_dai pcm3008_dai;
+
+#endif
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 44ef0da..cac3736 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -285,16 +285,23 @@
}
static int ssm2602_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
u16 srate;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
struct ssm2602_priv *ssm2602 = codec->private_data;
+ struct i2c_client *i2c = codec->control_data;
u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3;
int i = get_coeff(ssm2602->sysclk, params_rate(params));
+ if (substream == ssm2602->slave_substream) {
+ dev_dbg(&i2c->dev, "Ignoring hw_params for slave substream\n");
+ return 0;
+ }
+
/*no match is found*/
if (i == ARRAY_SIZE(coeff_div))
return -EINVAL;
@@ -324,19 +331,26 @@
return 0;
}
-static int ssm2602_startup(struct snd_pcm_substream *substream)
+static int ssm2602_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
struct ssm2602_priv *ssm2602 = codec->private_data;
+ struct i2c_client *i2c = codec->control_data;
struct snd_pcm_runtime *master_runtime;
/* The DAI has shared clocks so if we already have a playback or
* capture going then constrain this substream to match it.
+ * TODO: the ssm2602 allows pairs of non-matching PB/REC rates
*/
if (ssm2602->master_substream) {
master_runtime = ssm2602->master_substream->runtime;
+ dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n",
+ master_runtime->sample_bits,
+ master_runtime->rate);
+
snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_RATE,
master_runtime->rate,
@@ -354,7 +368,8 @@
return 0;
}
-static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream)
+static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -365,14 +380,21 @@
return 0;
}
-static void ssm2602_shutdown(struct snd_pcm_substream *substream)
+static void ssm2602_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
+ struct ssm2602_priv *ssm2602 = codec->private_data;
/* deactivate */
if (!codec->active)
ssm2602_write(codec, SSM2602_ACTIVE, 0);
+
+ if (ssm2602->master_substream == substream)
+ ssm2602->master_substream = ssm2602->slave_substream;
+
+ ssm2602->slave_substream = NULL;
}
static int ssm2602_mute(struct snd_soc_dai *dai, int mute)
@@ -432,10 +454,10 @@
iface |= 0x0001;
break;
case SND_SOC_DAIFMT_DSP_A:
- iface |= 0x0003;
+ iface |= 0x0013;
break;
case SND_SOC_DAIFMT_DSP_B:
- iface |= 0x0013;
+ iface |= 0x0003;
break;
default:
return -EINVAL;
@@ -496,6 +518,9 @@
SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
SNDRV_PCM_RATE_96000)
+#define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
struct snd_soc_dai ssm2602_dai = {
.name = "SSM2602",
.playback = {
@@ -503,20 +528,18 @@
.channels_min = 2,
.channels_max = 2,
.rates = SSM2602_RATES,
- .formats = SNDRV_PCM_FMTBIT_S32_LE,},
+ .formats = SSM2602_FORMATS,},
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
.rates = SSM2602_RATES,
- .formats = SNDRV_PCM_FMTBIT_S32_LE,},
+ .formats = SSM2602_FORMATS,},
.ops = {
.startup = ssm2602_startup,
.prepare = ssm2602_pcm_prepare,
.hw_params = ssm2602_hw_params,
.shutdown = ssm2602_shutdown,
- },
- .dai_ops = {
.digital_mute = ssm2602_mute,
.set_sysclk = ssm2602_set_dai_sysclk,
.set_fmt = ssm2602_set_dai_fmt,
@@ -601,7 +624,7 @@
ssm2602_add_controls(codec);
ssm2602_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
pr_err("ssm2602: failed to register card\n");
goto card_err;
@@ -770,6 +793,18 @@
};
EXPORT_SYMBOL_GPL(soc_codec_dev_ssm2602);
+static int __init ssm2602_modinit(void)
+{
+ return snd_soc_register_dai(&ssm2602_dai);
+}
+module_init(ssm2602_modinit);
+
+static void __exit ssm2602_exit(void)
+{
+ snd_soc_unregister_dai(&ssm2602_dai);
+}
+module_exit(ssm2602_exit);
+
MODULE_DESCRIPTION("ASoC ssm2602 driver");
MODULE_AUTHOR("Cliff Cai");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index 44308da..cfdea00 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -37,12 +37,6 @@
#define AIC23_VERSION "0.1"
-struct tlv320aic23_srate_reg_info {
- u32 sample_rate;
- u8 control; /* SR3, SR2, SR1, SR0 and BOSR */
- u8 divider; /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */
-};
-
/*
* AIC23 register cache
*/
@@ -261,20 +255,156 @@
};
-/* tlv320aic23 related */
-static const struct tlv320aic23_srate_reg_info srate_reg_info[] = {
- {4000, 0x06, 1}, /* 4000 */
- {8000, 0x06, 0}, /* 8000 */
- {16000, 0x0C, 1}, /* 16000 */
- {22050, 0x11, 1}, /* 22050 */
- {24000, 0x00, 1}, /* 24000 */
- {32000, 0x0C, 0}, /* 32000 */
- {44100, 0x11, 0}, /* 44100 */
- {48000, 0x00, 0}, /* 48000 */
- {88200, 0x1F, 0}, /* 88200 */
- {96000, 0x0E, 0}, /* 96000 */
+/* AIC23 driver data */
+struct aic23 {
+ struct snd_soc_codec codec;
+ int mclk;
+ int requested_adc;
+ int requested_dac;
};
+/*
+ * Common Crystals used
+ * 11.2896 Mhz /128 = *88.2k /192 = 58.8k
+ * 12.0000 Mhz /125 = *96k /136 = 88.235K
+ * 12.2880 Mhz /128 = *96k /192 = 64k
+ * 16.9344 Mhz /128 = 132.3k /192 = *88.2k
+ * 18.4320 Mhz /128 = 144k /192 = *96k
+ */
+
+/*
+ * Normal BOSR 0-256/2 = 128, 1-384/2 = 192
+ * USB BOSR 0-250/2 = 125, 1-272/2 = 136
+ */
+static const int bosr_usb_divisor_table[] = {
+ 128, 125, 192, 136
+};
+#define LOWER_GROUP ((1<<0) | (1<<1) | (1<<2) | (1<<3) | (1<<6) | (1<<7))
+#define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11) | (1<<15))
+static const unsigned short sr_valid_mask[] = {
+ LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 0*/
+ LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/
+ LOWER_GROUP, /* Usb, bosr - 0*/
+ UPPER_GROUP, /* Usb, bosr - 1*/
+};
+/*
+ * Every divisor is a factor of 11*12
+ */
+#define SR_MULT (11*12)
+#define A(x) (x) ? (SR_MULT/x) : 0
+static const unsigned char sr_adc_mult_table[] = {
+ A(2), A(2), A(12), A(12), A(0), A(0), A(3), A(1),
+ A(2), A(2), A(11), A(11), A(0), A(0), A(0), A(1)
+};
+static const unsigned char sr_dac_mult_table[] = {
+ A(2), A(12), A(2), A(12), A(0), A(0), A(3), A(1),
+ A(2), A(11), A(2), A(11), A(0), A(0), A(0), A(1)
+};
+
+static unsigned get_score(int adc, int adc_l, int adc_h, int need_adc,
+ int dac, int dac_l, int dac_h, int need_dac)
+{
+ if ((adc >= adc_l) && (adc <= adc_h) &&
+ (dac >= dac_l) && (dac <= dac_h)) {
+ int diff_adc = need_adc - adc;
+ int diff_dac = need_dac - dac;
+ return abs(diff_adc) + abs(diff_dac);
+ }
+ return UINT_MAX;
+}
+
+static int find_rate(int mclk, u32 need_adc, u32 need_dac)
+{
+ int i, j;
+ int best_i = -1;
+ int best_j = -1;
+ int best_div = 0;
+ unsigned best_score = UINT_MAX;
+ int adc_l, adc_h, dac_l, dac_h;
+
+ need_adc *= SR_MULT;
+ need_dac *= SR_MULT;
+ /*
+ * rates given are +/- 1/32
+ */
+ adc_l = need_adc - (need_adc >> 5);
+ adc_h = need_adc + (need_adc >> 5);
+ dac_l = need_dac - (need_dac >> 5);
+ dac_h = need_dac + (need_dac >> 5);
+ for (i = 0; i < ARRAY_SIZE(bosr_usb_divisor_table); i++) {
+ int base = mclk / bosr_usb_divisor_table[i];
+ int mask = sr_valid_mask[i];
+ for (j = 0; j < ARRAY_SIZE(sr_adc_mult_table);
+ j++, mask >>= 1) {
+ int adc;
+ int dac;
+ int score;
+ if ((mask & 1) == 0)
+ continue;
+ adc = base * sr_adc_mult_table[j];
+ dac = base * sr_dac_mult_table[j];
+ score = get_score(adc, adc_l, adc_h, need_adc,
+ dac, dac_l, dac_h, need_dac);
+ if (best_score > score) {
+ best_score = score;
+ best_i = i;
+ best_j = j;
+ best_div = 0;
+ }
+ score = get_score((adc >> 1), adc_l, adc_h, need_adc,
+ (dac >> 1), dac_l, dac_h, need_dac);
+ /* prefer to have a /2 */
+ if ((score != UINT_MAX) && (best_score >= score)) {
+ best_score = score;
+ best_i = i;
+ best_j = j;
+ best_div = 1;
+ }
+ }
+ }
+ return (best_j << 2) | best_i | (best_div << TLV320AIC23_CLKIN_SHIFT);
+}
+
+#ifdef DEBUG
+static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk,
+ u32 *sample_rate_adc, u32 *sample_rate_dac)
+{
+ int src = tlv320aic23_read_reg_cache(codec, TLV320AIC23_SRATE);
+ int sr = (src >> 2) & 0x0f;
+ int val = (mclk / bosr_usb_divisor_table[src & 3]);
+ int adc = (val * sr_adc_mult_table[sr]) / SR_MULT;
+ int dac = (val * sr_dac_mult_table[sr]) / SR_MULT;
+ if (src & TLV320AIC23_CLKIN_HALF) {
+ adc >>= 1;
+ dac >>= 1;
+ }
+ *sample_rate_adc = adc;
+ *sample_rate_dac = dac;
+}
+#endif
+
+static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk,
+ u32 sample_rate_adc, u32 sample_rate_dac)
+{
+ /* Search for the right sample rate */
+ int data = find_rate(mclk, sample_rate_adc, sample_rate_dac);
+ if (data < 0) {
+ printk(KERN_ERR "%s:Invalid rate %u,%u requested\n",
+ __func__, sample_rate_adc, sample_rate_dac);
+ return -EINVAL;
+ }
+ tlv320aic23_write(codec, TLV320AIC23_SRATE, data);
+#ifdef DEBUG
+ {
+ u32 adc, dac;
+ get_current_sample_rates(codec, mclk, &adc, &dac);
+ printk(KERN_DEBUG "actual samplerate = %u,%u reg=%x\n",
+ adc, dac, data);
+ }
+#endif
+ return 0;
+}
+
static int tlv320aic23_add_widgets(struct snd_soc_codec *codec)
{
snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
@@ -288,32 +418,36 @@
}
static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
- u16 iface_reg, data;
- u8 count = 0;
+ u16 iface_reg;
+ int ret;
+ struct aic23 *aic23 = container_of(codec, struct aic23, codec);
+ u32 sample_rate_adc = aic23->requested_adc;
+ u32 sample_rate_dac = aic23->requested_dac;
+ u32 sample_rate = params_rate(params);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ aic23->requested_dac = sample_rate_dac = sample_rate;
+ if (!sample_rate_adc)
+ sample_rate_adc = sample_rate;
+ } else {
+ aic23->requested_adc = sample_rate_adc = sample_rate;
+ if (!sample_rate_dac)
+ sample_rate_dac = sample_rate;
+ }
+ ret = set_sample_rate_control(codec, aic23->mclk, sample_rate_adc,
+ sample_rate_dac);
+ if (ret < 0)
+ return ret;
iface_reg =
tlv320aic23_read_reg_cache(codec,
TLV320AIC23_DIGT_FMT) & ~(0x03 << 2);
-
- /* Search for the right sample rate */
- /* Verify what happens if the rate is not supported
- * now it goes to 96Khz */
- while ((srate_reg_info[count].sample_rate != params_rate(params)) &&
- (count < ARRAY_SIZE(srate_reg_info))) {
- count++;
- }
-
- data = (srate_reg_info[count].divider << TLV320AIC23_CLKIN_SHIFT) |
- (srate_reg_info[count]. control << TLV320AIC23_BOSR_SHIFT) |
- TLV320AIC23_USB_CLK_ON;
-
- tlv320aic23_write(codec, TLV320AIC23_SRATE, data);
-
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
break;
@@ -332,7 +466,8 @@
return 0;
}
-static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream)
+static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -344,17 +479,23 @@
return 0;
}
-static void tlv320aic23_shutdown(struct snd_pcm_substream *substream)
+static void tlv320aic23_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
+ struct aic23 *aic23 = container_of(codec, struct aic23, codec);
/* deactivate */
if (!codec->active) {
udelay(50);
tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
}
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ aic23->requested_dac = 0;
+ else
+ aic23->requested_adc = 0;
}
static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute)
@@ -400,7 +541,7 @@
case SND_SOC_DAIFMT_I2S:
iface_reg |= TLV320AIC23_FOR_I2S;
break;
- case SND_SOC_DAIFMT_DSP_A:
+ case SND_SOC_DAIFMT_DSP_B:
iface_reg |= TLV320AIC23_FOR_DSP;
break;
case SND_SOC_DAIFMT_RIGHT_J:
@@ -422,12 +563,9 @@
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = codec_dai->codec;
-
- switch (freq) {
- case 12000000:
- return 0;
- }
- return -EINVAL;
+ struct aic23 *aic23 = container_of(codec, struct aic23, codec);
+ aic23->mclk = freq;
+ return 0;
}
static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
@@ -478,12 +616,10 @@
.prepare = tlv320aic23_pcm_prepare,
.hw_params = tlv320aic23_hw_params,
.shutdown = tlv320aic23_shutdown,
- },
- .dai_ops = {
- .digital_mute = tlv320aic23_mute,
- .set_fmt = tlv320aic23_set_dai_fmt,
- .set_sysclk = tlv320aic23_set_dai_sysclk,
- }
+ .digital_mute = tlv320aic23_mute,
+ .set_fmt = tlv320aic23_set_dai_fmt,
+ .set_sysclk = tlv320aic23_set_dai_sysclk,
+ }
};
EXPORT_SYMBOL_GPL(tlv320aic23_dai);
@@ -584,7 +720,7 @@
tlv320aic23_add_controls(codec);
tlv320aic23_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "tlv320aic23: failed to register card\n");
goto card_err;
@@ -659,14 +795,15 @@
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec;
+ struct aic23 *aic23;
int ret = 0;
printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION);
- codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
- if (codec == NULL)
+ aic23 = kzalloc(sizeof(struct aic23), GFP_KERNEL);
+ if (aic23 == NULL)
return -ENOMEM;
-
+ codec = &aic23->codec;
socdev->codec = codec;
mutex_init(&codec->mutex);
INIT_LIST_HEAD(&codec->dapm_widgets);
@@ -687,6 +824,7 @@
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->codec;
+ struct aic23 *aic23 = container_of(codec, struct aic23, codec);
if (codec->control_data)
tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
@@ -697,7 +835,7 @@
i2c_del_driver(&tlv320aic23_i2c_driver);
#endif
kfree(codec->reg_cache);
- kfree(codec);
+ kfree(aic23);
return 0;
}
@@ -709,6 +847,18 @@
};
EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320aic23);
+static int __init tlv320aic23_modinit(void)
+{
+ return snd_soc_register_dai(&tlv320aic23_dai);
+}
+module_init(tlv320aic23_modinit);
+
+static void __exit tlv320aic23_exit(void)
+{
+ snd_soc_unregister_dai(&tlv320aic23_dai);
+}
+module_exit(tlv320aic23_exit);
+
MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver");
MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index bed8a9e..29f2f1a 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -125,7 +125,8 @@
* Digital Audio Interface Operations
*/
static int aic26_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -287,8 +288,6 @@
},
.ops = {
.hw_params = aic26_hw_params,
- },
- .dai_ops = {
.digital_mute = aic26_mute,
.set_sysclk = aic26_set_sysclk,
.set_fmt = aic26_set_fmt,
@@ -360,7 +359,7 @@
/* CODEC is setup, we can register the card now */
dev_dbg(&pdev->dev, "Registering card\n");
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
dev_err(&pdev->dev, "aic26: failed to register card\n");
goto card_err;
@@ -427,7 +426,7 @@
static int aic26_spi_probe(struct spi_device *spi)
{
struct aic26 *aic26;
- int rc, i, reg;
+ int ret, i, reg;
dev_dbg(&spi->dev, "probing tlv320aic26 spi device\n");
@@ -457,6 +456,14 @@
aic26->codec.reg_cache_size = AIC26_NUM_REGS;
aic26->codec.reg_cache = aic26->reg_cache;
+ aic26_dai.dev = &spi->dev;
+ ret = snd_soc_register_dai(&aic26_dai);
+ if (ret != 0) {
+ dev_err(&spi->dev, "Failed to register DAI: %d\n", ret);
+ kfree(aic26);
+ return ret;
+ }
+
/* Reset the codec to power on defaults */
aic26_reg_write(&aic26->codec, AIC26_REG_RESET, 0xBB00);
@@ -475,8 +482,8 @@
/* Register the sysfs files for debugging */
/* Create SysFS files */
- rc = device_create_file(&spi->dev, &dev_attr_keyclick);
- if (rc)
+ ret = device_create_file(&spi->dev, &dev_attr_keyclick);
+ if (ret)
dev_info(&spi->dev, "error creating sysfs files\n");
#if defined(CONFIG_SND_SOC_OF_SIMPLE)
@@ -493,6 +500,7 @@
{
struct aic26 *aic26 = dev_get_drvdata(&spi->dev);
+ snd_soc_unregister_dai(&aic26_dai);
kfree(aic26);
return 0;
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index cff276e..b47a749 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -253,11 +253,17 @@
SOC_DOUBLE_R("Line DAC Playback Volume", DACL1_2_LLOPM_VOL,
DACR1_2_RLOPM_VOL, 0, 0x7f, 1),
- SOC_DOUBLE_R("Line DAC Playback Switch", LLOPM_CTRL, RLOPM_CTRL, 3,
- 0x01, 0),
- SOC_DOUBLE_R("Line PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL,
- PGAR_2_RLOPM_VOL, 0, 0x7f, 1),
- SOC_DOUBLE_R("Line Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL,
+ SOC_SINGLE("LineL Playback Switch", LLOPM_CTRL, 3, 0x01, 0),
+ SOC_SINGLE("LineR Playback Switch", RLOPM_CTRL, 3, 0x01, 0),
+ SOC_DOUBLE_R("LineL DAC Playback Volume", DACL1_2_LLOPM_VOL,
+ DACR1_2_LLOPM_VOL, 0, 0x7f, 1),
+ SOC_SINGLE("LineL Left PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL,
+ 0, 0x7f, 1),
+ SOC_SINGLE("LineR Right PGA Bypass Playback Volume", PGAR_2_RLOPM_VOL,
+ 0, 0x7f, 1),
+ SOC_DOUBLE_R("LineL Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL,
+ LINE2R_2_LLOPM_VOL, 0, 0x7f, 1),
+ SOC_DOUBLE_R("LineR Line2 Bypass Playback Volume", LINE2L_2_RLOPM_VOL,
LINE2R_2_RLOPM_VOL, 0, 0x7f, 1),
SOC_DOUBLE_R("Mono DAC Playback Volume", DACL1_2_MONOLOPM_VOL,
@@ -272,8 +278,12 @@
DACR1_2_HPROUT_VOL, 0, 0x7f, 1),
SOC_DOUBLE_R("HP DAC Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3,
0x01, 0),
- SOC_DOUBLE_R("HP PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL,
+ SOC_DOUBLE_R("HP Right PGA Bypass Playback Volume", PGAR_2_HPLOUT_VOL,
PGAR_2_HPROUT_VOL, 0, 0x7f, 1),
+ SOC_SINGLE("HPL PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL,
+ 0, 0x7f, 1),
+ SOC_SINGLE("HPR PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL,
+ 0, 0x7f, 1),
SOC_DOUBLE_R("HP Line2 Bypass Playback Volume", LINE2L_2_HPLOUT_VOL,
LINE2R_2_HPROUT_VOL, 0, 0x7f, 1),
@@ -281,8 +291,10 @@
DACR1_2_HPRCOM_VOL, 0, 0x7f, 1),
SOC_DOUBLE_R("HPCOM DAC Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3,
0x01, 0),
- SOC_DOUBLE_R("HPCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL,
- PGAR_2_HPRCOM_VOL, 0, 0x7f, 1),
+ SOC_SINGLE("HPLCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL,
+ 0, 0x7f, 1),
+ SOC_SINGLE("HPRCOM PGA Bypass Playback Volume", PGAL_2_HPRCOM_VOL,
+ 0, 0x7f, 1),
SOC_DOUBLE_R("HPCOM Line2 Bypass Playback Volume", LINE2L_2_HPLCOM_VOL,
LINE2R_2_HPRCOM_VOL, 0, 0x7f, 1),
@@ -333,7 +345,8 @@
/* Left DAC_L1 Mixer */
static const struct snd_kcontrol_new aic3x_left_dac_mixer_controls[] = {
- SOC_DAPM_SINGLE("Line Switch", DACL1_2_LLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("LineL Switch", DACL1_2_LLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("LineR Switch", DACL1_2_RLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("Mono Switch", DACL1_2_MONOLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("HP Switch", DACL1_2_HPLOUT_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("HPCOM Switch", DACL1_2_HPLCOM_VOL, 7, 1, 0),
@@ -341,7 +354,8 @@
/* Right DAC_R1 Mixer */
static const struct snd_kcontrol_new aic3x_right_dac_mixer_controls[] = {
- SOC_DAPM_SINGLE("Line Switch", DACR1_2_RLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("LineL Switch", DACR1_2_LLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("LineR Switch", DACR1_2_RLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("Mono Switch", DACR1_2_MONOLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("HP Switch", DACR1_2_HPROUT_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("HPCOM Switch", DACR1_2_HPRCOM_VOL, 7, 1, 0),
@@ -350,14 +364,18 @@
/* Left PGA Mixer */
static const struct snd_kcontrol_new aic3x_left_pga_mixer_controls[] = {
SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_LADC_CTRL, 3, 1, 1),
+ SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_LADC_CTRL, 3, 1, 1),
SOC_DAPM_SINGLE_AIC3X("Line2L Switch", LINE2L_2_LADC_CTRL, 3, 1, 1),
SOC_DAPM_SINGLE_AIC3X("Mic3L Switch", MIC3LR_2_LADC_CTRL, 4, 1, 1),
+ SOC_DAPM_SINGLE_AIC3X("Mic3R Switch", MIC3LR_2_LADC_CTRL, 0, 1, 1),
};
/* Right PGA Mixer */
static const struct snd_kcontrol_new aic3x_right_pga_mixer_controls[] = {
SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_RADC_CTRL, 3, 1, 1),
+ SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_RADC_CTRL, 3, 1, 1),
SOC_DAPM_SINGLE_AIC3X("Line2R Switch", LINE2R_2_RADC_CTRL, 3, 1, 1),
+ SOC_DAPM_SINGLE_AIC3X("Mic3L Switch", MIC3LR_2_RADC_CTRL, 4, 1, 1),
SOC_DAPM_SINGLE_AIC3X("Mic3R Switch", MIC3LR_2_RADC_CTRL, 0, 1, 1),
};
@@ -379,34 +397,42 @@
/* Left PGA Bypass Mixer */
static const struct snd_kcontrol_new aic3x_left_pga_bp_mixer_controls[] = {
- SOC_DAPM_SINGLE("Line Switch", PGAL_2_LLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("LineL Switch", PGAL_2_LLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("LineR Switch", PGAL_2_RLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("Mono Switch", PGAL_2_MONOLOPM_VOL, 7, 1, 0),
- SOC_DAPM_SINGLE("HP Switch", PGAL_2_HPLOUT_VOL, 7, 1, 0),
- SOC_DAPM_SINGLE("HPCOM Switch", PGAL_2_HPLCOM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HPL Switch", PGAL_2_HPLOUT_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HPR Switch", PGAL_2_HPROUT_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HPLCOM Switch", PGAL_2_HPLCOM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HPRCOM Switch", PGAL_2_HPRCOM_VOL, 7, 1, 0),
};
/* Right PGA Bypass Mixer */
static const struct snd_kcontrol_new aic3x_right_pga_bp_mixer_controls[] = {
- SOC_DAPM_SINGLE("Line Switch", PGAR_2_RLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("LineL Switch", PGAR_2_LLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("LineR Switch", PGAR_2_RLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("Mono Switch", PGAR_2_MONOLOPM_VOL, 7, 1, 0),
- SOC_DAPM_SINGLE("HP Switch", PGAR_2_HPROUT_VOL, 7, 1, 0),
- SOC_DAPM_SINGLE("HPCOM Switch", PGAR_2_HPRCOM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HPL Switch", PGAR_2_HPLOUT_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HPR Switch", PGAR_2_HPROUT_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HPLCOM Switch", PGAR_2_HPLCOM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HPRCOM Switch", PGAR_2_HPRCOM_VOL, 7, 1, 0),
};
/* Left Line2 Bypass Mixer */
static const struct snd_kcontrol_new aic3x_left_line2_bp_mixer_controls[] = {
- SOC_DAPM_SINGLE("Line Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("LineL Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("LineR Switch", LINE2L_2_RLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("Mono Switch", LINE2L_2_MONOLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("HP Switch", LINE2L_2_HPLOUT_VOL, 7, 1, 0),
- SOC_DAPM_SINGLE("HPCOM Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HPLCOM Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0),
};
/* Right Line2 Bypass Mixer */
static const struct snd_kcontrol_new aic3x_right_line2_bp_mixer_controls[] = {
- SOC_DAPM_SINGLE("Line Switch", LINE2R_2_RLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("LineL Switch", LINE2R_2_LLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("LineR Switch", LINE2R_2_RLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("Mono Switch", LINE2R_2_MONOLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("HP Switch", LINE2R_2_HPROUT_VOL, 7, 1, 0),
- SOC_DAPM_SINGLE("HPCOM Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HPRCOM Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0),
};
static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
@@ -439,22 +465,26 @@
/* Mono Output */
SND_SOC_DAPM_PGA("Mono Out", MONOLOPM_CTRL, 0, 0, NULL, 0),
- /* Left Inputs to Left ADC */
+ /* Inputs to Left ADC */
SND_SOC_DAPM_ADC("Left ADC", "Left Capture", LINE1L_2_LADC_CTRL, 2, 0),
SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0,
&aic3x_left_pga_mixer_controls[0],
ARRAY_SIZE(aic3x_left_pga_mixer_controls)),
SND_SOC_DAPM_MUX("Left Line1L Mux", SND_SOC_NOPM, 0, 0,
&aic3x_left_line1_mux_controls),
+ SND_SOC_DAPM_MUX("Left Line1R Mux", SND_SOC_NOPM, 0, 0,
+ &aic3x_left_line1_mux_controls),
SND_SOC_DAPM_MUX("Left Line2L Mux", SND_SOC_NOPM, 0, 0,
&aic3x_left_line2_mux_controls),
- /* Right Inputs to Right ADC */
+ /* Inputs to Right ADC */
SND_SOC_DAPM_ADC("Right ADC", "Right Capture",
LINE1R_2_RADC_CTRL, 2, 0),
SND_SOC_DAPM_MIXER("Right PGA Mixer", SND_SOC_NOPM, 0, 0,
&aic3x_right_pga_mixer_controls[0],
ARRAY_SIZE(aic3x_right_pga_mixer_controls)),
+ SND_SOC_DAPM_MUX("Right Line1L Mux", SND_SOC_NOPM, 0, 0,
+ &aic3x_right_line1_mux_controls),
SND_SOC_DAPM_MUX("Right Line1R Mux", SND_SOC_NOPM, 0, 0,
&aic3x_right_line1_mux_controls),
SND_SOC_DAPM_MUX("Right Line2R Mux", SND_SOC_NOPM, 0, 0,
@@ -531,7 +561,8 @@
{"Left DAC Mux", "DAC_L2", "Left DAC"},
{"Left DAC Mux", "DAC_L3", "Left DAC"},
- {"Left DAC_L1 Mixer", "Line Switch", "Left DAC Mux"},
+ {"Left DAC_L1 Mixer", "LineL Switch", "Left DAC Mux"},
+ {"Left DAC_L1 Mixer", "LineR Switch", "Left DAC Mux"},
{"Left DAC_L1 Mixer", "Mono Switch", "Left DAC Mux"},
{"Left DAC_L1 Mixer", "HP Switch", "Left DAC Mux"},
{"Left DAC_L1 Mixer", "HPCOM Switch", "Left DAC Mux"},
@@ -557,7 +588,8 @@
{"Right DAC Mux", "DAC_R2", "Right DAC"},
{"Right DAC Mux", "DAC_R3", "Right DAC"},
- {"Right DAC_R1 Mixer", "Line Switch", "Right DAC Mux"},
+ {"Right DAC_R1 Mixer", "LineL Switch", "Right DAC Mux"},
+ {"Right DAC_R1 Mixer", "LineR Switch", "Right DAC Mux"},
{"Right DAC_R1 Mixer", "Mono Switch", "Right DAC Mux"},
{"Right DAC_R1 Mixer", "HP Switch", "Right DAC Mux"},
{"Right DAC_R1 Mixer", "HPCOM Switch", "Right DAC Mux"},
@@ -592,8 +624,10 @@
{"Left Line2L Mux", "differential", "LINE2L"},
{"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"},
+ {"Left PGA Mixer", "Line1R Switch", "Left Line1R Mux"},
{"Left PGA Mixer", "Line2L Switch", "Left Line2L Mux"},
{"Left PGA Mixer", "Mic3L Switch", "MIC3L"},
+ {"Left PGA Mixer", "Mic3R Switch", "MIC3R"},
{"Left ADC", NULL, "Left PGA Mixer"},
{"Left ADC", NULL, "GPIO1 dmic modclk"},
@@ -605,18 +639,23 @@
{"Right Line2R Mux", "single-ended", "LINE2R"},
{"Right Line2R Mux", "differential", "LINE2R"},
+ {"Right PGA Mixer", "Line1L Switch", "Right Line1L Mux"},
{"Right PGA Mixer", "Line1R Switch", "Right Line1R Mux"},
{"Right PGA Mixer", "Line2R Switch", "Right Line2R Mux"},
+ {"Right PGA Mixer", "Mic3L Switch", "MIC3L"},
{"Right PGA Mixer", "Mic3R Switch", "MIC3R"},
{"Right ADC", NULL, "Right PGA Mixer"},
{"Right ADC", NULL, "GPIO1 dmic modclk"},
/* Left PGA Bypass */
- {"Left PGA Bypass Mixer", "Line Switch", "Left PGA Mixer"},
+ {"Left PGA Bypass Mixer", "LineL Switch", "Left PGA Mixer"},
+ {"Left PGA Bypass Mixer", "LineR Switch", "Left PGA Mixer"},
{"Left PGA Bypass Mixer", "Mono Switch", "Left PGA Mixer"},
- {"Left PGA Bypass Mixer", "HP Switch", "Left PGA Mixer"},
- {"Left PGA Bypass Mixer", "HPCOM Switch", "Left PGA Mixer"},
+ {"Left PGA Bypass Mixer", "HPL Switch", "Left PGA Mixer"},
+ {"Left PGA Bypass Mixer", "HPR Switch", "Left PGA Mixer"},
+ {"Left PGA Bypass Mixer", "HPLCOM Switch", "Left PGA Mixer"},
+ {"Left PGA Bypass Mixer", "HPRCOM Switch", "Left PGA Mixer"},
{"Left HPCOM Mux", "differential of HPLOUT", "Left PGA Bypass Mixer"},
{"Left HPCOM Mux", "constant VCM", "Left PGA Bypass Mixer"},
@@ -627,10 +666,13 @@
{"Left HP Out", NULL, "Left PGA Bypass Mixer"},
/* Right PGA Bypass */
- {"Right PGA Bypass Mixer", "Line Switch", "Right PGA Mixer"},
+ {"Right PGA Bypass Mixer", "LineL Switch", "Right PGA Mixer"},
+ {"Right PGA Bypass Mixer", "LineR Switch", "Right PGA Mixer"},
{"Right PGA Bypass Mixer", "Mono Switch", "Right PGA Mixer"},
- {"Right PGA Bypass Mixer", "HP Switch", "Right PGA Mixer"},
- {"Right PGA Bypass Mixer", "HPCOM Switch", "Right PGA Mixer"},
+ {"Right PGA Bypass Mixer", "HPL Switch", "Right PGA Mixer"},
+ {"Right PGA Bypass Mixer", "HPR Switch", "Right PGA Mixer"},
+ {"Right PGA Bypass Mixer", "HPLCOM Switch", "Right PGA Mixer"},
+ {"Right PGA Bypass Mixer", "HPRCOM Switch", "Right PGA Mixer"},
{"Right HPCOM Mux", "differential of HPROUT", "Right PGA Bypass Mixer"},
{"Right HPCOM Mux", "constant VCM", "Right PGA Bypass Mixer"},
@@ -643,10 +685,11 @@
{"Right HP Out", NULL, "Right PGA Bypass Mixer"},
/* Left Line2 Bypass */
- {"Left Line2 Bypass Mixer", "Line Switch", "Left Line2L Mux"},
+ {"Left Line2 Bypass Mixer", "LineL Switch", "Left Line2L Mux"},
+ {"Left Line2 Bypass Mixer", "LineR Switch", "Left Line2L Mux"},
{"Left Line2 Bypass Mixer", "Mono Switch", "Left Line2L Mux"},
{"Left Line2 Bypass Mixer", "HP Switch", "Left Line2L Mux"},
- {"Left Line2 Bypass Mixer", "HPCOM Switch", "Left Line2L Mux"},
+ {"Left Line2 Bypass Mixer", "HPLCOM Switch", "Left Line2L Mux"},
{"Left HPCOM Mux", "differential of HPLOUT", "Left Line2 Bypass Mixer"},
{"Left HPCOM Mux", "constant VCM", "Left Line2 Bypass Mixer"},
@@ -657,10 +700,11 @@
{"Left HP Out", NULL, "Left Line2 Bypass Mixer"},
/* Right Line2 Bypass */
- {"Right Line2 Bypass Mixer", "Line Switch", "Right Line2R Mux"},
+ {"Right Line2 Bypass Mixer", "LineL Switch", "Right Line2R Mux"},
+ {"Right Line2 Bypass Mixer", "LineR Switch", "Right Line2R Mux"},
{"Right Line2 Bypass Mixer", "Mono Switch", "Right Line2R Mux"},
{"Right Line2 Bypass Mixer", "HP Switch", "Right Line2R Mux"},
- {"Right Line2 Bypass Mixer", "HPCOM Switch", "Right Line2R Mux"},
+ {"Right Line2 Bypass Mixer", "HPRCOM Switch", "Right Line2R Mux"},
{"Right HPCOM Mux", "differential of HPROUT", "Right Line2 Bypass Mixer"},
{"Right HPCOM Mux", "constant VCM", "Right Line2 Bypass Mixer"},
@@ -694,7 +738,8 @@
}
static int aic3x_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -846,6 +891,7 @@
struct snd_soc_codec *codec = codec_dai->codec;
struct aic3x_priv *aic3x = codec->private_data;
u8 iface_areg, iface_breg;
+ int delay = 0;
iface_areg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLA) & 0x3f;
iface_breg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & 0x3f;
@@ -871,6 +917,8 @@
SND_SOC_DAIFMT_INV_MASK)) {
case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF):
break;
+ case (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF):
+ delay = 1;
case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF):
iface_breg |= (0x01 << 6);
break;
@@ -887,6 +935,7 @@
/* set iface */
aic3x_write(codec, AIC3X_ASD_INTF_CTRLA, iface_areg);
aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, iface_breg);
+ aic3x_write(codec, AIC3X_ASD_INTF_CTRLC, delay);
return 0;
}
@@ -981,14 +1030,41 @@
}
EXPORT_SYMBOL_GPL(aic3x_get_gpio);
+void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect,
+ int headset_debounce, int button_debounce)
+{
+ u8 val;
+
+ val = ((detect & AIC3X_HEADSET_DETECT_MASK)
+ << AIC3X_HEADSET_DETECT_SHIFT) |
+ ((headset_debounce & AIC3X_HEADSET_DEBOUNCE_MASK)
+ << AIC3X_HEADSET_DEBOUNCE_SHIFT) |
+ ((button_debounce & AIC3X_BUTTON_DEBOUNCE_MASK)
+ << AIC3X_BUTTON_DEBOUNCE_SHIFT);
+
+ if (detect & AIC3X_HEADSET_DETECT_MASK)
+ val |= AIC3X_HEADSET_DETECT_ENABLED;
+
+ aic3x_write(codec, AIC3X_HEADSET_DETECT_CTRL_A, val);
+}
+EXPORT_SYMBOL_GPL(aic3x_set_headset_detection);
+
int aic3x_headset_detected(struct snd_soc_codec *codec)
{
u8 val;
- aic3x_read(codec, AIC3X_RT_IRQ_FLAGS_REG, &val);
- return (val >> 2) & 1;
+ aic3x_read(codec, AIC3X_HEADSET_DETECT_CTRL_B, &val);
+ return (val >> 4) & 1;
}
EXPORT_SYMBOL_GPL(aic3x_headset_detected);
+int aic3x_button_pressed(struct snd_soc_codec *codec)
+{
+ u8 val;
+ aic3x_read(codec, AIC3X_HEADSET_DETECT_CTRL_B, &val);
+ return (val >> 5) & 1;
+}
+EXPORT_SYMBOL_GPL(aic3x_button_pressed);
+
#define AIC3X_RATES SNDRV_PCM_RATE_8000_96000
#define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
@@ -1009,8 +1085,6 @@
.formats = AIC3X_FORMATS,},
.ops = {
.hw_params = aic3x_hw_params,
- },
- .dai_ops = {
.digital_mute = aic3x_mute,
.set_sysclk = aic3x_set_dai_sysclk,
.set_fmt = aic3x_set_dai_fmt,
@@ -1152,7 +1226,7 @@
aic3x_add_controls(codec);
aic3x_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "aic3x: failed to register card\n");
goto card_err;
@@ -1341,6 +1415,18 @@
};
EXPORT_SYMBOL_GPL(soc_codec_dev_aic3x);
+static int __init aic3x_modinit(void)
+{
+ return snd_soc_register_dai(&aic3x_dai);
+}
+module_init(aic3x_modinit);
+
+static void __exit aic3x_exit(void)
+{
+ snd_soc_unregister_dai(&aic3x_dai);
+}
+module_exit(aic3x_exit);
+
MODULE_DESCRIPTION("ASoC TLV320AIC3X codec driver");
MODULE_AUTHOR("Vladimir Barinov");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h
index 00a195a..ac827e5 100644
--- a/sound/soc/codecs/tlv320aic3x.h
+++ b/sound/soc/codecs/tlv320aic3x.h
@@ -35,11 +35,15 @@
#define AIC3X_ASD_INTF_CTRLA 8
/* Audio serial data interface control register B */
#define AIC3X_ASD_INTF_CTRLB 9
+/* Audio serial data interface control register C */
+#define AIC3X_ASD_INTF_CTRLC 10
/* Audio overflow status and PLL R value programming register */
#define AIC3X_OVRF_STATUS_AND_PLLR_REG 11
/* Audio codec digital filter control register */
#define AIC3X_CODEC_DFILT_CTRL 12
-
+/* Headset/button press detection register */
+#define AIC3X_HEADSET_DETECT_CTRL_A 13
+#define AIC3X_HEADSET_DETECT_CTRL_B 14
/* ADC PGA Gain control registers */
#define LADC_VOL 15
#define RADC_VOL 16
@@ -48,7 +52,9 @@
#define MIC3LR_2_RADC_CTRL 18
/* Line1 Input control registers */
#define LINE1L_2_LADC_CTRL 19
+#define LINE1R_2_LADC_CTRL 21
#define LINE1R_2_RADC_CTRL 22
+#define LINE1L_2_RADC_CTRL 24
/* Line2 Input control registers */
#define LINE2L_2_LADC_CTRL 20
#define LINE2R_2_RADC_CTRL 23
@@ -79,6 +85,8 @@
#define LINE2L_2_HPLOUT_VOL 45
#define LINE2R_2_HPROUT_VOL 62
#define PGAL_2_HPLOUT_VOL 46
+#define PGAL_2_HPROUT_VOL 60
+#define PGAR_2_HPLOUT_VOL 49
#define PGAR_2_HPROUT_VOL 63
#define DACL1_2_HPLOUT_VOL 47
#define DACR1_2_HPROUT_VOL 64
@@ -88,6 +96,8 @@
#define LINE2L_2_HPLCOM_VOL 52
#define LINE2R_2_HPRCOM_VOL 69
#define PGAL_2_HPLCOM_VOL 53
+#define PGAR_2_HPLCOM_VOL 56
+#define PGAL_2_HPRCOM_VOL 67
#define PGAR_2_HPRCOM_VOL 70
#define DACL1_2_HPLCOM_VOL 54
#define DACR1_2_HPRCOM_VOL 71
@@ -103,11 +113,17 @@
#define MONOLOPM_CTRL 79
/* Line Output Plus/Minus control registers */
#define LINE2L_2_LLOPM_VOL 80
+#define LINE2L_2_RLOPM_VOL 87
+#define LINE2R_2_LLOPM_VOL 83
#define LINE2R_2_RLOPM_VOL 90
#define PGAL_2_LLOPM_VOL 81
+#define PGAL_2_RLOPM_VOL 88
+#define PGAR_2_LLOPM_VOL 84
#define PGAR_2_RLOPM_VOL 91
#define DACL1_2_LLOPM_VOL 82
+#define DACL1_2_RLOPM_VOL 89
#define DACR1_2_RLOPM_VOL 92
+#define DACR1_2_LLOPM_VOL 85
#define LLOPM_CTRL 86
#define RLOPM_CTRL 93
/* GPIO/IRQ registers */
@@ -221,7 +237,49 @@
void aic3x_set_gpio(struct snd_soc_codec *codec, int gpio, int state);
int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio);
+
+/* headset detection / button API */
+
+/* The AIC3x supports detection of stereo headsets (GND + left + right signal)
+ * and cellular headsets (GND + speaker output + microphone input).
+ * It is recommended to enable MIC bias for this function to work properly.
+ * For more information, please refer to the datasheet. */
+enum {
+ AIC3X_HEADSET_DETECT_OFF = 0,
+ AIC3X_HEADSET_DETECT_STEREO = 1,
+ AIC3X_HEADSET_DETECT_CELLULAR = 2,
+ AIC3X_HEADSET_DETECT_BOTH = 3
+};
+
+enum {
+ AIC3X_HEADSET_DEBOUNCE_16MS = 0,
+ AIC3X_HEADSET_DEBOUNCE_32MS = 1,
+ AIC3X_HEADSET_DEBOUNCE_64MS = 2,
+ AIC3X_HEADSET_DEBOUNCE_128MS = 3,
+ AIC3X_HEADSET_DEBOUNCE_256MS = 4,
+ AIC3X_HEADSET_DEBOUNCE_512MS = 5
+};
+
+enum {
+ AIC3X_BUTTON_DEBOUNCE_0MS = 0,
+ AIC3X_BUTTON_DEBOUNCE_8MS = 1,
+ AIC3X_BUTTON_DEBOUNCE_16MS = 2,
+ AIC3X_BUTTON_DEBOUNCE_32MS = 3
+};
+
+#define AIC3X_HEADSET_DETECT_ENABLED 0x80
+#define AIC3X_HEADSET_DETECT_SHIFT 5
+#define AIC3X_HEADSET_DETECT_MASK 3
+#define AIC3X_HEADSET_DEBOUNCE_SHIFT 2
+#define AIC3X_HEADSET_DEBOUNCE_MASK 7
+#define AIC3X_BUTTON_DEBOUNCE_SHIFT 0
+#define AIC3X_BUTTON_DEBOUNCE_MASK 3
+
+/* see the enums above for valid parameters to this function */
+void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect,
+ int headset_debounce, int button_debounce);
int aic3x_headset_detected(struct snd_soc_codec *codec);
+int aic3x_button_pressed(struct snd_soc_codec *codec);
struct aic3x_setup_data {
int i2c_bus;
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
new file mode 100644
index 0000000..5184888
--- /dev/null
+++ b/sound/soc/codecs/twl4030.c
@@ -0,0 +1,1317 @@
+/*
+ * ALSA SoC TWL4030 codec driver
+ *
+ * Author: Steve Sakoman, <steve@sakoman.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.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/i2c/twl4030.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "twl4030.h"
+
+/*
+ * twl4030 register cache & default register settings
+ */
+static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
+ 0x00, /* this register not used */
+ 0x93, /* REG_CODEC_MODE (0x1) */
+ 0xc3, /* REG_OPTION (0x2) */
+ 0x00, /* REG_UNKNOWN (0x3) */
+ 0x00, /* REG_MICBIAS_CTL (0x4) */
+ 0x20, /* REG_ANAMICL (0x5) */
+ 0x00, /* REG_ANAMICR (0x6) */
+ 0x00, /* REG_AVADC_CTL (0x7) */
+ 0x00, /* REG_ADCMICSEL (0x8) */
+ 0x00, /* REG_DIGMIXING (0x9) */
+ 0x0c, /* REG_ATXL1PGA (0xA) */
+ 0x0c, /* REG_ATXR1PGA (0xB) */
+ 0x00, /* REG_AVTXL2PGA (0xC) */
+ 0x00, /* REG_AVTXR2PGA (0xD) */
+ 0x01, /* REG_AUDIO_IF (0xE) */
+ 0x00, /* REG_VOICE_IF (0xF) */
+ 0x00, /* REG_ARXR1PGA (0x10) */
+ 0x00, /* REG_ARXL1PGA (0x11) */
+ 0x6c, /* REG_ARXR2PGA (0x12) */
+ 0x6c, /* REG_ARXL2PGA (0x13) */
+ 0x00, /* REG_VRXPGA (0x14) */
+ 0x00, /* REG_VSTPGA (0x15) */
+ 0x00, /* REG_VRX2ARXPGA (0x16) */
+ 0x0c, /* REG_AVDAC_CTL (0x17) */
+ 0x00, /* REG_ARX2VTXPGA (0x18) */
+ 0x00, /* REG_ARXL1_APGA_CTL (0x19) */
+ 0x00, /* REG_ARXR1_APGA_CTL (0x1A) */
+ 0x4b, /* REG_ARXL2_APGA_CTL (0x1B) */
+ 0x4b, /* REG_ARXR2_APGA_CTL (0x1C) */
+ 0x00, /* REG_ATX2ARXPGA (0x1D) */
+ 0x00, /* REG_BT_IF (0x1E) */
+ 0x00, /* REG_BTPGA (0x1F) */
+ 0x00, /* REG_BTSTPGA (0x20) */
+ 0x00, /* REG_EAR_CTL (0x21) */
+ 0x24, /* REG_HS_SEL (0x22) */
+ 0x0a, /* REG_HS_GAIN_SET (0x23) */
+ 0x00, /* REG_HS_POPN_SET (0x24) */
+ 0x00, /* REG_PREDL_CTL (0x25) */
+ 0x00, /* REG_PREDR_CTL (0x26) */
+ 0x00, /* REG_PRECKL_CTL (0x27) */
+ 0x00, /* REG_PRECKR_CTL (0x28) */
+ 0x00, /* REG_HFL_CTL (0x29) */
+ 0x00, /* REG_HFR_CTL (0x2A) */
+ 0x00, /* REG_ALC_CTL (0x2B) */
+ 0x00, /* REG_ALC_SET1 (0x2C) */
+ 0x00, /* REG_ALC_SET2 (0x2D) */
+ 0x00, /* REG_BOOST_CTL (0x2E) */
+ 0x00, /* REG_SOFTVOL_CTL (0x2F) */
+ 0x00, /* REG_DTMF_FREQSEL (0x30) */
+ 0x00, /* REG_DTMF_TONEXT1H (0x31) */
+ 0x00, /* REG_DTMF_TONEXT1L (0x32) */
+ 0x00, /* REG_DTMF_TONEXT2H (0x33) */
+ 0x00, /* REG_DTMF_TONEXT2L (0x34) */
+ 0x00, /* REG_DTMF_TONOFF (0x35) */
+ 0x00, /* REG_DTMF_WANONOFF (0x36) */
+ 0x00, /* REG_I2S_RX_SCRAMBLE_H (0x37) */
+ 0x00, /* REG_I2S_RX_SCRAMBLE_M (0x38) */
+ 0x00, /* REG_I2S_RX_SCRAMBLE_L (0x39) */
+ 0x16, /* REG_APLL_CTL (0x3A) */
+ 0x00, /* REG_DTMF_CTL (0x3B) */
+ 0x00, /* REG_DTMF_PGA_CTL2 (0x3C) */
+ 0x00, /* REG_DTMF_PGA_CTL1 (0x3D) */
+ 0x00, /* REG_MISC_SET_1 (0x3E) */
+ 0x00, /* REG_PCMBTMUX (0x3F) */
+ 0x00, /* not used (0x40) */
+ 0x00, /* not used (0x41) */
+ 0x00, /* not used (0x42) */
+ 0x00, /* REG_RX_PATH_SEL (0x43) */
+ 0x00, /* REG_VDL_APGA_CTL (0x44) */
+ 0x00, /* REG_VIBRA_CTL (0x45) */
+ 0x00, /* REG_VIBRA_SET (0x46) */
+ 0x00, /* REG_VIBRA_PWM_SET (0x47) */
+ 0x00, /* REG_ANAMIC_GAIN (0x48) */
+ 0x00, /* REG_MISC_SET_2 (0x49) */
+};
+
+/*
+ * read twl4030 register cache
+ */
+static inline unsigned int twl4030_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u8 *cache = codec->reg_cache;
+
+ return cache[reg];
+}
+
+/*
+ * write twl4030 register cache
+ */
+static inline void twl4030_write_reg_cache(struct snd_soc_codec *codec,
+ u8 reg, u8 value)
+{
+ u8 *cache = codec->reg_cache;
+
+ if (reg >= TWL4030_CACHEREGNUM)
+ return;
+ cache[reg] = value;
+}
+
+/*
+ * write to the twl4030 register space
+ */
+static int twl4030_write(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int value)
+{
+ twl4030_write_reg_cache(codec, reg, value);
+ return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg);
+}
+
+static void twl4030_clear_codecpdz(struct snd_soc_codec *codec)
+{
+ u8 mode;
+
+ mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE);
+ twl4030_write(codec, TWL4030_REG_CODEC_MODE,
+ mode & ~TWL4030_CODECPDZ);
+
+ /* REVISIT: this delay is present in TI sample drivers */
+ /* but there seems to be no TRM requirement for it */
+ udelay(10);
+}
+
+static void twl4030_set_codecpdz(struct snd_soc_codec *codec)
+{
+ u8 mode;
+
+ mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE);
+ twl4030_write(codec, TWL4030_REG_CODEC_MODE,
+ mode | TWL4030_CODECPDZ);
+
+ /* REVISIT: this delay is present in TI sample drivers */
+ /* but there seems to be no TRM requirement for it */
+ udelay(10);
+}
+
+static void twl4030_init_chip(struct snd_soc_codec *codec)
+{
+ int i;
+
+ /* clear CODECPDZ prior to setting register defaults */
+ twl4030_clear_codecpdz(codec);
+
+ /* set all audio section registers to reasonable defaults */
+ for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++)
+ twl4030_write(codec, i, twl4030_reg[i]);
+
+}
+
+/* Earpiece */
+static const char *twl4030_earpiece_texts[] =
+ {"Off", "DACL1", "DACL2", "Invalid", "DACR1"};
+
+static const struct soc_enum twl4030_earpiece_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_EAR_CTL, 1,
+ ARRAY_SIZE(twl4030_earpiece_texts),
+ twl4030_earpiece_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_earpiece_control =
+SOC_DAPM_ENUM("Route", twl4030_earpiece_enum);
+
+/* PreDrive Left */
+static const char *twl4030_predrivel_texts[] =
+ {"Off", "DACL1", "DACL2", "Invalid", "DACR2"};
+
+static const struct soc_enum twl4030_predrivel_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_PREDL_CTL, 1,
+ ARRAY_SIZE(twl4030_predrivel_texts),
+ twl4030_predrivel_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_predrivel_control =
+SOC_DAPM_ENUM("Route", twl4030_predrivel_enum);
+
+/* PreDrive Right */
+static const char *twl4030_predriver_texts[] =
+ {"Off", "DACR1", "DACR2", "Invalid", "DACL2"};
+
+static const struct soc_enum twl4030_predriver_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_PREDR_CTL, 1,
+ ARRAY_SIZE(twl4030_predriver_texts),
+ twl4030_predriver_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_predriver_control =
+SOC_DAPM_ENUM("Route", twl4030_predriver_enum);
+
+/* Headset Left */
+static const char *twl4030_hsol_texts[] =
+ {"Off", "DACL1", "DACL2"};
+
+static const struct soc_enum twl4030_hsol_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 1,
+ ARRAY_SIZE(twl4030_hsol_texts),
+ twl4030_hsol_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_hsol_control =
+SOC_DAPM_ENUM("Route", twl4030_hsol_enum);
+
+/* Headset Right */
+static const char *twl4030_hsor_texts[] =
+ {"Off", "DACR1", "DACR2"};
+
+static const struct soc_enum twl4030_hsor_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 4,
+ ARRAY_SIZE(twl4030_hsor_texts),
+ twl4030_hsor_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_hsor_control =
+SOC_DAPM_ENUM("Route", twl4030_hsor_enum);
+
+/* Carkit Left */
+static const char *twl4030_carkitl_texts[] =
+ {"Off", "DACL1", "DACL2"};
+
+static const struct soc_enum twl4030_carkitl_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_PRECKL_CTL, 1,
+ ARRAY_SIZE(twl4030_carkitl_texts),
+ twl4030_carkitl_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_carkitl_control =
+SOC_DAPM_ENUM("Route", twl4030_carkitl_enum);
+
+/* Carkit Right */
+static const char *twl4030_carkitr_texts[] =
+ {"Off", "DACR1", "DACR2"};
+
+static const struct soc_enum twl4030_carkitr_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_PRECKR_CTL, 1,
+ ARRAY_SIZE(twl4030_carkitr_texts),
+ twl4030_carkitr_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_carkitr_control =
+SOC_DAPM_ENUM("Route", twl4030_carkitr_enum);
+
+/* Handsfree Left */
+static const char *twl4030_handsfreel_texts[] =
+ {"Voice", "DACL1", "DACL2", "DACR2"};
+
+static const struct soc_enum twl4030_handsfreel_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_HFL_CTL, 0,
+ ARRAY_SIZE(twl4030_handsfreel_texts),
+ twl4030_handsfreel_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_handsfreel_control =
+SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum);
+
+/* Handsfree Right */
+static const char *twl4030_handsfreer_texts[] =
+ {"Voice", "DACR1", "DACR2", "DACL2"};
+
+static const struct soc_enum twl4030_handsfreer_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_HFR_CTL, 0,
+ ARRAY_SIZE(twl4030_handsfreer_texts),
+ twl4030_handsfreer_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_handsfreer_control =
+SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum);
+
+static int outmixer_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ int ret = 0;
+ int val;
+
+ switch (e->reg) {
+ case TWL4030_REG_PREDL_CTL:
+ case TWL4030_REG_PREDR_CTL:
+ case TWL4030_REG_EAR_CTL:
+ val = w->value >> e->shift_l;
+ if (val == 3) {
+ printk(KERN_WARNING
+ "Invalid MUX setting for register 0x%02x (%d)\n",
+ e->reg, val);
+ ret = -1;
+ }
+ break;
+ }
+
+ return ret;
+}
+
+static int handsfree_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct soc_enum *e = (struct soc_enum *)w->kcontrols->private_value;
+ unsigned char hs_ctl;
+
+ hs_ctl = twl4030_read_reg_cache(w->codec, e->reg);
+
+ if (hs_ctl & TWL4030_HF_CTL_REF_EN) {
+ hs_ctl |= TWL4030_HF_CTL_RAMP_EN;
+ twl4030_write(w->codec, e->reg, hs_ctl);
+ hs_ctl |= TWL4030_HF_CTL_LOOP_EN;
+ twl4030_write(w->codec, e->reg, hs_ctl);
+ hs_ctl |= TWL4030_HF_CTL_HB_EN;
+ twl4030_write(w->codec, e->reg, hs_ctl);
+ } else {
+ hs_ctl &= ~(TWL4030_HF_CTL_RAMP_EN | TWL4030_HF_CTL_LOOP_EN
+ | TWL4030_HF_CTL_HB_EN);
+ twl4030_write(w->codec, e->reg, hs_ctl);
+ }
+
+ return 0;
+}
+
+/*
+ * Some of the gain controls in TWL (mostly those which are associated with
+ * the outputs) are implemented in an interesting way:
+ * 0x0 : Power down (mute)
+ * 0x1 : 6dB
+ * 0x2 : 0 dB
+ * 0x3 : -6 dB
+ * Inverting not going to help with these.
+ * Custom volsw and volsw_2r get/put functions to handle these gain bits.
+ */
+#define SOC_DOUBLE_TLV_TWL4030(xname, xreg, shift_left, shift_right, xmax,\
+ xinvert, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+ SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw, \
+ .get = snd_soc_get_volsw_twl4030, \
+ .put = snd_soc_put_volsw_twl4030, \
+ .private_value = (unsigned long)&(struct soc_mixer_control) \
+ {.reg = xreg, .shift = shift_left, .rshift = shift_right,\
+ .max = xmax, .invert = xinvert} }
+#define SOC_DOUBLE_R_TLV_TWL4030(xname, reg_left, reg_right, xshift, xmax,\
+ xinvert, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+ SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw_2r, \
+ .get = snd_soc_get_volsw_r2_twl4030,\
+ .put = snd_soc_put_volsw_r2_twl4030, \
+ .private_value = (unsigned long)&(struct soc_mixer_control) \
+ {.reg = reg_left, .rreg = reg_right, .shift = xshift, \
+ .rshift = xshift, .max = xmax, .invert = xinvert} }
+#define SOC_SINGLE_TLV_TWL4030(xname, xreg, xshift, xmax, xinvert, tlv_array) \
+ SOC_DOUBLE_TLV_TWL4030(xname, xreg, xshift, xshift, xmax, \
+ xinvert, tlv_array)
+
+static int snd_soc_get_volsw_twl4030(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = mc->reg;
+ unsigned int shift = mc->shift;
+ unsigned int rshift = mc->rshift;
+ int max = mc->max;
+ int mask = (1 << fls(max)) - 1;
+
+ ucontrol->value.integer.value[0] =
+ (snd_soc_read(codec, reg) >> shift) & mask;
+ if (ucontrol->value.integer.value[0])
+ ucontrol->value.integer.value[0] =
+ max + 1 - ucontrol->value.integer.value[0];
+
+ if (shift != rshift) {
+ ucontrol->value.integer.value[1] =
+ (snd_soc_read(codec, reg) >> rshift) & mask;
+ if (ucontrol->value.integer.value[1])
+ ucontrol->value.integer.value[1] =
+ max + 1 - ucontrol->value.integer.value[1];
+ }
+
+ return 0;
+}
+
+static int snd_soc_put_volsw_twl4030(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = mc->reg;
+ unsigned int shift = mc->shift;
+ unsigned int rshift = mc->rshift;
+ int max = mc->max;
+ int mask = (1 << fls(max)) - 1;
+ unsigned short val, val2, val_mask;
+
+ val = (ucontrol->value.integer.value[0] & mask);
+
+ val_mask = mask << shift;
+ if (val)
+ val = max + 1 - val;
+ val = val << shift;
+ if (shift != rshift) {
+ val2 = (ucontrol->value.integer.value[1] & mask);
+ val_mask |= mask << rshift;
+ if (val2)
+ val2 = max + 1 - val2;
+ val |= val2 << rshift;
+ }
+ return snd_soc_update_bits(codec, reg, val_mask, val);
+}
+
+static int snd_soc_get_volsw_r2_twl4030(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = mc->reg;
+ unsigned int reg2 = mc->rreg;
+ unsigned int shift = mc->shift;
+ int max = mc->max;
+ int mask = (1<<fls(max))-1;
+
+ ucontrol->value.integer.value[0] =
+ (snd_soc_read(codec, reg) >> shift) & mask;
+ ucontrol->value.integer.value[1] =
+ (snd_soc_read(codec, reg2) >> shift) & mask;
+
+ if (ucontrol->value.integer.value[0])
+ ucontrol->value.integer.value[0] =
+ max + 1 - ucontrol->value.integer.value[0];
+ if (ucontrol->value.integer.value[1])
+ ucontrol->value.integer.value[1] =
+ max + 1 - ucontrol->value.integer.value[1];
+
+ return 0;
+}
+
+static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = mc->reg;
+ unsigned int reg2 = mc->rreg;
+ unsigned int shift = mc->shift;
+ int max = mc->max;
+ int mask = (1 << fls(max)) - 1;
+ int err;
+ unsigned short val, val2, val_mask;
+
+ val_mask = mask << shift;
+ val = (ucontrol->value.integer.value[0] & mask);
+ val2 = (ucontrol->value.integer.value[1] & mask);
+
+ if (val)
+ val = max + 1 - val;
+ if (val2)
+ val2 = max + 1 - val2;
+
+ val = val << shift;
+ val2 = val2 << shift;
+
+ err = snd_soc_update_bits(codec, reg, val_mask, val);
+ if (err < 0)
+ return err;
+
+ err = snd_soc_update_bits(codec, reg2, val_mask, val2);
+ return err;
+}
+
+static int twl4030_get_left_input(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = kcontrol->private_data;
+ u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
+ int result = 0;
+
+ /* one bit must be set a time */
+ reg &= TWL4030_CKMIC_EN | TWL4030_AUXL_EN | TWL4030_HSMIC_EN
+ | TWL4030_MAINMIC_EN;
+ if (reg != 0) {
+ result++;
+ while ((reg & 1) == 0) {
+ result++;
+ reg >>= 1;
+ }
+ }
+
+ ucontrol->value.integer.value[0] = result;
+ return 0;
+}
+
+static int twl4030_put_left_input(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = kcontrol->private_data;
+ int value = ucontrol->value.integer.value[0];
+ u8 anamicl, micbias, avadc_ctl;
+
+ anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
+ anamicl &= ~(TWL4030_CKMIC_EN | TWL4030_AUXL_EN | TWL4030_HSMIC_EN
+ | TWL4030_MAINMIC_EN);
+ micbias = twl4030_read_reg_cache(codec, TWL4030_REG_MICBIAS_CTL);
+ micbias &= ~(TWL4030_HSMICBIAS_EN | TWL4030_MICBIAS1_EN);
+ avadc_ctl = twl4030_read_reg_cache(codec, TWL4030_REG_AVADC_CTL);
+
+ switch (value) {
+ case 1:
+ anamicl |= TWL4030_MAINMIC_EN;
+ micbias |= TWL4030_MICBIAS1_EN;
+ break;
+ case 2:
+ anamicl |= TWL4030_HSMIC_EN;
+ micbias |= TWL4030_HSMICBIAS_EN;
+ break;
+ case 3:
+ anamicl |= TWL4030_AUXL_EN;
+ break;
+ case 4:
+ anamicl |= TWL4030_CKMIC_EN;
+ break;
+ default:
+ break;
+ }
+
+ /* If some input is selected, enable amp and ADC */
+ if (value != 0) {
+ anamicl |= TWL4030_MICAMPL_EN;
+ avadc_ctl |= TWL4030_ADCL_EN;
+ } else {
+ anamicl &= ~TWL4030_MICAMPL_EN;
+ avadc_ctl &= ~TWL4030_ADCL_EN;
+ }
+
+ twl4030_write(codec, TWL4030_REG_ANAMICL, anamicl);
+ twl4030_write(codec, TWL4030_REG_MICBIAS_CTL, micbias);
+ twl4030_write(codec, TWL4030_REG_AVADC_CTL, avadc_ctl);
+
+ return 1;
+}
+
+static int twl4030_get_right_input(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = kcontrol->private_data;
+ u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICR);
+ int value = 0;
+
+ reg &= TWL4030_SUBMIC_EN|TWL4030_AUXR_EN;
+ switch (reg) {
+ case TWL4030_SUBMIC_EN:
+ value = 1;
+ break;
+ case TWL4030_AUXR_EN:
+ value = 2;
+ break;
+ default:
+ break;
+ }
+
+ ucontrol->value.integer.value[0] = value;
+ return 0;
+}
+
+static int twl4030_put_right_input(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = kcontrol->private_data;
+ int value = ucontrol->value.integer.value[0];
+ u8 anamicr, micbias, avadc_ctl;
+
+ anamicr = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICR);
+ anamicr &= ~(TWL4030_SUBMIC_EN|TWL4030_AUXR_EN);
+ micbias = twl4030_read_reg_cache(codec, TWL4030_REG_MICBIAS_CTL);
+ micbias &= ~TWL4030_MICBIAS2_EN;
+ avadc_ctl = twl4030_read_reg_cache(codec, TWL4030_REG_AVADC_CTL);
+
+ switch (value) {
+ case 1:
+ anamicr |= TWL4030_SUBMIC_EN;
+ micbias |= TWL4030_MICBIAS2_EN;
+ break;
+ case 2:
+ anamicr |= TWL4030_AUXR_EN;
+ break;
+ default:
+ break;
+ }
+
+ if (value != 0) {
+ anamicr |= TWL4030_MICAMPR_EN;
+ avadc_ctl |= TWL4030_ADCR_EN;
+ } else {
+ anamicr &= ~TWL4030_MICAMPR_EN;
+ avadc_ctl &= ~TWL4030_ADCR_EN;
+ }
+
+ twl4030_write(codec, TWL4030_REG_ANAMICR, anamicr);
+ twl4030_write(codec, TWL4030_REG_MICBIAS_CTL, micbias);
+ twl4030_write(codec, TWL4030_REG_AVADC_CTL, avadc_ctl);
+
+ return 1;
+}
+
+static const char *twl4030_left_in_sel[] = {
+ "None",
+ "Main Mic",
+ "Headset Mic",
+ "Line In",
+ "Carkit Mic",
+};
+
+static const char *twl4030_right_in_sel[] = {
+ "None",
+ "Sub Mic",
+ "Line In",
+};
+
+static const struct soc_enum twl4030_left_input_mux =
+ SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl4030_left_in_sel),
+ twl4030_left_in_sel);
+
+static const struct soc_enum twl4030_right_input_mux =
+ SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl4030_right_in_sel),
+ twl4030_right_in_sel);
+
+/*
+ * FGAIN volume control:
+ * from -62 to 0 dB in 1 dB steps (mute instead of -63 dB)
+ */
+static DECLARE_TLV_DB_SCALE(digital_fine_tlv, -6300, 100, 1);
+
+/*
+ * CGAIN volume control:
+ * 0 dB to 12 dB in 6 dB steps
+ * value 2 and 3 means 12 dB
+ */
+static DECLARE_TLV_DB_SCALE(digital_coarse_tlv, 0, 600, 0);
+
+/*
+ * Analog playback gain
+ * -24 dB to 12 dB in 2 dB steps
+ */
+static DECLARE_TLV_DB_SCALE(analog_tlv, -2400, 200, 0);
+
+/*
+ * Gain controls tied to outputs
+ * -6 dB to 6 dB in 6 dB steps (mute instead of -12)
+ */
+static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1);
+
+/*
+ * Capture gain after the ADCs
+ * from 0 dB to 31 dB in 1 dB steps
+ */
+static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0);
+
+/*
+ * Gain control for input amplifiers
+ * 0 dB to 30 dB in 6 dB steps
+ */
+static DECLARE_TLV_DB_SCALE(input_gain_tlv, 0, 600, 0);
+
+static const struct snd_kcontrol_new twl4030_snd_controls[] = {
+ /* Common playback gain controls */
+ SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume",
+ TWL4030_REG_ARXL1PGA, TWL4030_REG_ARXR1PGA,
+ 0, 0x3f, 0, digital_fine_tlv),
+ SOC_DOUBLE_R_TLV("DAC2 Digital Fine Playback Volume",
+ TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA,
+ 0, 0x3f, 0, digital_fine_tlv),
+
+ SOC_DOUBLE_R_TLV("DAC1 Digital Coarse Playback Volume",
+ TWL4030_REG_ARXL1PGA, TWL4030_REG_ARXR1PGA,
+ 6, 0x2, 0, digital_coarse_tlv),
+ SOC_DOUBLE_R_TLV("DAC2 Digital Coarse Playback Volume",
+ TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA,
+ 6, 0x2, 0, digital_coarse_tlv),
+
+ SOC_DOUBLE_R_TLV("DAC1 Analog Playback Volume",
+ TWL4030_REG_ARXL1_APGA_CTL, TWL4030_REG_ARXR1_APGA_CTL,
+ 3, 0x12, 1, analog_tlv),
+ SOC_DOUBLE_R_TLV("DAC2 Analog Playback Volume",
+ TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL,
+ 3, 0x12, 1, analog_tlv),
+ SOC_DOUBLE_R("DAC1 Analog Playback Switch",
+ TWL4030_REG_ARXL1_APGA_CTL, TWL4030_REG_ARXR1_APGA_CTL,
+ 1, 1, 0),
+ SOC_DOUBLE_R("DAC2 Analog Playback Switch",
+ TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL,
+ 1, 1, 0),
+
+ /* Separate output gain controls */
+ SOC_DOUBLE_R_TLV_TWL4030("PreDriv Playback Volume",
+ TWL4030_REG_PREDL_CTL, TWL4030_REG_PREDR_CTL,
+ 4, 3, 0, output_tvl),
+
+ SOC_DOUBLE_TLV_TWL4030("Headset Playback Volume",
+ TWL4030_REG_HS_GAIN_SET, 0, 2, 3, 0, output_tvl),
+
+ SOC_DOUBLE_R_TLV_TWL4030("Carkit Playback Volume",
+ TWL4030_REG_PRECKL_CTL, TWL4030_REG_PRECKR_CTL,
+ 4, 3, 0, output_tvl),
+
+ SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume",
+ TWL4030_REG_EAR_CTL, 4, 3, 0, output_tvl),
+
+ /* Common capture gain controls */
+ SOC_DOUBLE_R_TLV("Capture Volume",
+ TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA,
+ 0, 0x1f, 0, digital_capture_tlv),
+
+ SOC_DOUBLE_TLV("Input Boost Volume", TWL4030_REG_ANAMIC_GAIN,
+ 0, 3, 5, 0, input_gain_tlv),
+
+ /* Input source controls */
+ SOC_ENUM_EXT("Left Input Source", twl4030_left_input_mux,
+ twl4030_get_left_input, twl4030_put_left_input),
+ SOC_ENUM_EXT("Right Input Source", twl4030_right_input_mux,
+ twl4030_get_right_input, twl4030_put_right_input),
+};
+
+/* add non dapm controls */
+static int twl4030_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(twl4030_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&twl4030_snd_controls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
+ SND_SOC_DAPM_INPUT("INL"),
+ SND_SOC_DAPM_INPUT("INR"),
+
+ SND_SOC_DAPM_OUTPUT("OUTL"),
+ SND_SOC_DAPM_OUTPUT("OUTR"),
+ SND_SOC_DAPM_OUTPUT("EARPIECE"),
+ SND_SOC_DAPM_OUTPUT("PREDRIVEL"),
+ SND_SOC_DAPM_OUTPUT("PREDRIVER"),
+ SND_SOC_DAPM_OUTPUT("HSOL"),
+ SND_SOC_DAPM_OUTPUT("HSOR"),
+ SND_SOC_DAPM_OUTPUT("CARKITL"),
+ SND_SOC_DAPM_OUTPUT("CARKITR"),
+ SND_SOC_DAPM_OUTPUT("HFL"),
+ SND_SOC_DAPM_OUTPUT("HFR"),
+
+ /* DACs */
+ SND_SOC_DAPM_DAC("DAC Right1", "Right Front Playback",
+ TWL4030_REG_AVDAC_CTL, 0, 0),
+ SND_SOC_DAPM_DAC("DAC Left1", "Left Front Playback",
+ TWL4030_REG_AVDAC_CTL, 1, 0),
+ SND_SOC_DAPM_DAC("DAC Right2", "Right Rear Playback",
+ TWL4030_REG_AVDAC_CTL, 2, 0),
+ SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback",
+ TWL4030_REG_AVDAC_CTL, 3, 0),
+
+ /* Analog PGAs */
+ SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL,
+ 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("ARXL1_APGA", TWL4030_REG_ARXL1_APGA_CTL,
+ 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("ARXR2_APGA", TWL4030_REG_ARXR2_APGA_CTL,
+ 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("ARXL2_APGA", TWL4030_REG_ARXL2_APGA_CTL,
+ 0, 0, NULL, 0),
+
+ /* Output MUX controls */
+ /* Earpiece */
+ SND_SOC_DAPM_MUX_E("Earpiece Mux", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_earpiece_control, outmixer_event,
+ SND_SOC_DAPM_PRE_REG),
+ /* PreDrivL/R */
+ SND_SOC_DAPM_MUX_E("PredriveL Mux", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_predrivel_control, outmixer_event,
+ SND_SOC_DAPM_PRE_REG),
+ SND_SOC_DAPM_MUX_E("PredriveR Mux", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_predriver_control, outmixer_event,
+ SND_SOC_DAPM_PRE_REG),
+ /* HeadsetL/R */
+ SND_SOC_DAPM_MUX("HeadsetL Mux", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_hsol_control),
+ SND_SOC_DAPM_MUX("HeadsetR Mux", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_hsor_control),
+ /* CarkitL/R */
+ SND_SOC_DAPM_MUX("CarkitL Mux", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_carkitl_control),
+ SND_SOC_DAPM_MUX("CarkitR Mux", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_carkitr_control),
+ /* HandsfreeL/R */
+ SND_SOC_DAPM_MUX_E("HandsfreeL Mux", TWL4030_REG_HFL_CTL, 5, 0,
+ &twl4030_dapm_handsfreel_control, handsfree_event,
+ SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("HandsfreeR Mux", TWL4030_REG_HFR_CTL, 5, 0,
+ &twl4030_dapm_handsfreer_control, handsfree_event,
+ SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ {"ARXL1_APGA", NULL, "DAC Left1"},
+ {"ARXR1_APGA", NULL, "DAC Right1"},
+ {"ARXL2_APGA", NULL, "DAC Left2"},
+ {"ARXR2_APGA", NULL, "DAC Right2"},
+
+ /* Internal playback routings */
+ /* Earpiece */
+ {"Earpiece Mux", "DACL1", "ARXL1_APGA"},
+ {"Earpiece Mux", "DACL2", "ARXL2_APGA"},
+ {"Earpiece Mux", "DACR1", "ARXR1_APGA"},
+ /* PreDrivL */
+ {"PredriveL Mux", "DACL1", "ARXL1_APGA"},
+ {"PredriveL Mux", "DACL2", "ARXL2_APGA"},
+ {"PredriveL Mux", "DACR2", "ARXR2_APGA"},
+ /* PreDrivR */
+ {"PredriveR Mux", "DACR1", "ARXR1_APGA"},
+ {"PredriveR Mux", "DACR2", "ARXR2_APGA"},
+ {"PredriveR Mux", "DACL2", "ARXL2_APGA"},
+ /* HeadsetL */
+ {"HeadsetL Mux", "DACL1", "ARXL1_APGA"},
+ {"HeadsetL Mux", "DACL2", "ARXL2_APGA"},
+ /* HeadsetR */
+ {"HeadsetR Mux", "DACR1", "ARXR1_APGA"},
+ {"HeadsetR Mux", "DACR2", "ARXR2_APGA"},
+ /* CarkitL */
+ {"CarkitL Mux", "DACL1", "ARXL1_APGA"},
+ {"CarkitL Mux", "DACL2", "ARXL2_APGA"},
+ /* CarkitR */
+ {"CarkitR Mux", "DACR1", "ARXR1_APGA"},
+ {"CarkitR Mux", "DACR2", "ARXR2_APGA"},
+ /* HandsfreeL */
+ {"HandsfreeL Mux", "DACL1", "ARXL1_APGA"},
+ {"HandsfreeL Mux", "DACL2", "ARXL2_APGA"},
+ {"HandsfreeL Mux", "DACR2", "ARXR2_APGA"},
+ /* HandsfreeR */
+ {"HandsfreeR Mux", "DACR1", "ARXR1_APGA"},
+ {"HandsfreeR Mux", "DACR2", "ARXR2_APGA"},
+ {"HandsfreeR Mux", "DACL2", "ARXL2_APGA"},
+
+ /* outputs */
+ {"OUTL", NULL, "ARXL2_APGA"},
+ {"OUTR", NULL, "ARXR2_APGA"},
+ {"EARPIECE", NULL, "Earpiece Mux"},
+ {"PREDRIVEL", NULL, "PredriveL Mux"},
+ {"PREDRIVER", NULL, "PredriveR Mux"},
+ {"HSOL", NULL, "HeadsetL Mux"},
+ {"HSOR", NULL, "HeadsetR Mux"},
+ {"CARKITL", NULL, "CarkitL Mux"},
+ {"CARKITR", NULL, "CarkitR Mux"},
+ {"HFL", NULL, "HandsfreeL Mux"},
+ {"HFR", NULL, "HandsfreeR Mux"},
+
+ /* inputs */
+ {"ADCL", NULL, "INL"},
+ {"ADCR", NULL, "INR"},
+};
+
+static int twl4030_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, twl4030_dapm_widgets,
+ ARRAY_SIZE(twl4030_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+static void twl4030_power_up(struct snd_soc_codec *codec)
+{
+ u8 anamicl, regmisc1, byte, popn;
+ int i = 0;
+
+ /* set CODECPDZ to turn on codec */
+ twl4030_set_codecpdz(codec);
+
+ /* initiate offset cancellation */
+ anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
+ twl4030_write(codec, TWL4030_REG_ANAMICL,
+ anamicl | TWL4030_CNCL_OFFSET_START);
+
+ /* wait for offset cancellation to complete */
+ do {
+ /* this takes a little while, so don't slam i2c */
+ udelay(2000);
+ twl4030_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
+ TWL4030_REG_ANAMICL);
+ } while ((i++ < 100) &&
+ ((byte & TWL4030_CNCL_OFFSET_START) ==
+ TWL4030_CNCL_OFFSET_START));
+
+ /* anti-pop when changing analog gain */
+ regmisc1 = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1);
+ twl4030_write(codec, TWL4030_REG_MISC_SET_1,
+ regmisc1 | TWL4030_SMOOTH_ANAVOL_EN);
+
+ /* toggle CODECPDZ as per TRM */
+ twl4030_clear_codecpdz(codec);
+ twl4030_set_codecpdz(codec);
+
+ /* program anti-pop with bias ramp delay */
+ popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
+ popn &= TWL4030_RAMP_DELAY;
+ popn |= TWL4030_RAMP_DELAY_645MS;
+ twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
+ popn |= TWL4030_VMID_EN;
+ twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
+
+ /* enable anti-pop ramp */
+ popn |= TWL4030_RAMP_EN;
+ twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
+}
+
+static void twl4030_power_down(struct snd_soc_codec *codec)
+{
+ u8 popn;
+
+ /* disable anti-pop ramp */
+ popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
+ popn &= ~TWL4030_RAMP_EN;
+ twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
+
+ /* disable bias out */
+ popn &= ~TWL4030_VMID_EN;
+ twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
+
+ /* power down */
+ twl4030_clear_codecpdz(codec);
+}
+
+static int twl4030_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ twl4030_power_up(codec);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ /* TODO: develop a twl4030_prepare function */
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ /* TODO: develop a twl4030_standby function */
+ twl4030_power_down(codec);
+ break;
+ case SND_SOC_BIAS_OFF:
+ twl4030_power_down(codec);
+ break;
+ }
+ codec->bias_level = level;
+
+ return 0;
+}
+
+static int twl4030_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ u8 mode, old_mode, format, old_format;
+
+
+ /* bit rate */
+ old_mode = twl4030_read_reg_cache(codec,
+ TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ;
+ mode = old_mode & ~TWL4030_APLL_RATE;
+
+ switch (params_rate(params)) {
+ case 8000:
+ mode |= TWL4030_APLL_RATE_8000;
+ break;
+ case 11025:
+ mode |= TWL4030_APLL_RATE_11025;
+ break;
+ case 12000:
+ mode |= TWL4030_APLL_RATE_12000;
+ break;
+ case 16000:
+ mode |= TWL4030_APLL_RATE_16000;
+ break;
+ case 22050:
+ mode |= TWL4030_APLL_RATE_22050;
+ break;
+ case 24000:
+ mode |= TWL4030_APLL_RATE_24000;
+ break;
+ case 32000:
+ mode |= TWL4030_APLL_RATE_32000;
+ break;
+ case 44100:
+ mode |= TWL4030_APLL_RATE_44100;
+ break;
+ case 48000:
+ mode |= TWL4030_APLL_RATE_48000;
+ break;
+ default:
+ printk(KERN_ERR "TWL4030 hw params: unknown rate %d\n",
+ params_rate(params));
+ return -EINVAL;
+ }
+
+ if (mode != old_mode) {
+ /* change rate and set CODECPDZ */
+ twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
+ twl4030_set_codecpdz(codec);
+ }
+
+ /* sample size */
+ old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
+ format = old_format;
+ format &= ~TWL4030_DATA_WIDTH;
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ format |= TWL4030_DATA_WIDTH_16S_16W;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ format |= TWL4030_DATA_WIDTH_32S_24W;
+ break;
+ default:
+ printk(KERN_ERR "TWL4030 hw params: unknown format %d\n",
+ params_format(params));
+ return -EINVAL;
+ }
+
+ if (format != old_format) {
+
+ /* clear CODECPDZ before changing format (codec requirement) */
+ twl4030_clear_codecpdz(codec);
+
+ /* change format */
+ twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
+
+ /* set CODECPDZ afterwards */
+ twl4030_set_codecpdz(codec);
+ }
+ return 0;
+}
+
+static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u8 infreq;
+
+ switch (freq) {
+ case 19200000:
+ infreq = TWL4030_APLL_INFREQ_19200KHZ;
+ break;
+ case 26000000:
+ infreq = TWL4030_APLL_INFREQ_26000KHZ;
+ break;
+ case 38400000:
+ infreq = TWL4030_APLL_INFREQ_38400KHZ;
+ break;
+ default:
+ printk(KERN_ERR "TWL4030 set sysclk: unknown rate %d\n",
+ freq);
+ return -EINVAL;
+ }
+
+ infreq |= TWL4030_APLL_EN;
+ twl4030_write(codec, TWL4030_REG_APLL_CTL, infreq);
+
+ return 0;
+}
+
+static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u8 old_format, format;
+
+ /* get format */
+ old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
+ format = old_format;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ format &= ~(TWL4030_AIF_SLAVE_EN);
+ format &= ~(TWL4030_CLK256FS_EN);
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ format |= TWL4030_AIF_SLAVE_EN;
+ format |= TWL4030_CLK256FS_EN;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* interface format */
+ format &= ~TWL4030_AIF_FORMAT;
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ format |= TWL4030_AIF_FORMAT_CODEC;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (format != old_format) {
+
+ /* clear CODECPDZ before changing format (codec requirement) */
+ twl4030_clear_codecpdz(codec);
+
+ /* change format */
+ twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
+
+ /* set CODECPDZ afterwards */
+ twl4030_set_codecpdz(codec);
+ }
+
+ return 0;
+}
+
+#define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000)
+#define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE)
+
+struct snd_soc_dai twl4030_dai = {
+ .name = "twl4030",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = TWL4030_RATES,
+ .formats = TWL4030_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = TWL4030_RATES,
+ .formats = TWL4030_FORMATS,},
+ .ops = {
+ .hw_params = twl4030_hw_params,
+ .set_sysclk = twl4030_set_dai_sysclk,
+ .set_fmt = twl4030_set_dai_fmt,
+ }
+};
+EXPORT_SYMBOL_GPL(twl4030_dai);
+
+static int twl4030_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ return 0;
+}
+
+static int twl4030_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ twl4030_set_bias_level(codec, codec->suspend_bias_level);
+ return 0;
+}
+
+/*
+ * initialize the driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+
+static int twl4030_init(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret = 0;
+
+ printk(KERN_INFO "TWL4030 Audio Codec init \n");
+
+ codec->name = "twl4030";
+ codec->owner = THIS_MODULE;
+ codec->read = twl4030_read_reg_cache;
+ codec->write = twl4030_write;
+ codec->set_bias_level = twl4030_set_bias_level;
+ codec->dai = &twl4030_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = sizeof(twl4030_reg);
+ codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg),
+ GFP_KERNEL);
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "twl4030: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ twl4030_init_chip(codec);
+
+ /* power on device */
+ twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ twl4030_add_controls(codec);
+ twl4030_add_widgets(codec);
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "twl4030: failed to register card\n");
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ kfree(codec->reg_cache);
+ return ret;
+}
+
+static struct snd_soc_device *twl4030_socdev;
+
+static int twl4030_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ twl4030_socdev = socdev;
+ twl4030_init(socdev);
+
+ return 0;
+}
+
+static int twl4030_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ printk(KERN_INFO "TWL4030 Audio Codec remove\n");
+ kfree(codec);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_twl4030 = {
+ .probe = twl4030_probe,
+ .remove = twl4030_remove,
+ .suspend = twl4030_suspend,
+ .resume = twl4030_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030);
+
+static int __init twl4030_modinit(void)
+{
+ return snd_soc_register_dai(&twl4030_dai);
+}
+module_init(twl4030_modinit);
+
+static void __exit twl4030_exit(void)
+{
+ snd_soc_unregister_dai(&twl4030_dai);
+}
+module_exit(twl4030_exit);
+
+MODULE_DESCRIPTION("ASoC TWL4030 codec driver");
+MODULE_AUTHOR("Steve Sakoman");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h
new file mode 100644
index 0000000..54615c7
--- /dev/null
+++ b/sound/soc/codecs/twl4030.h
@@ -0,0 +1,219 @@
+/*
+ * ALSA SoC TWL4030 codec driver
+ *
+ * Author: Steve Sakoman <steve@sakoman.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.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#ifndef __TWL4030_AUDIO_H__
+#define __TWL4030_AUDIO_H__
+
+#define TWL4030_REG_CODEC_MODE 0x1
+#define TWL4030_REG_OPTION 0x2
+#define TWL4030_REG_UNKNOWN 0x3
+#define TWL4030_REG_MICBIAS_CTL 0x4
+#define TWL4030_REG_ANAMICL 0x5
+#define TWL4030_REG_ANAMICR 0x6
+#define TWL4030_REG_AVADC_CTL 0x7
+#define TWL4030_REG_ADCMICSEL 0x8
+#define TWL4030_REG_DIGMIXING 0x9
+#define TWL4030_REG_ATXL1PGA 0xA
+#define TWL4030_REG_ATXR1PGA 0xB
+#define TWL4030_REG_AVTXL2PGA 0xC
+#define TWL4030_REG_AVTXR2PGA 0xD
+#define TWL4030_REG_AUDIO_IF 0xE
+#define TWL4030_REG_VOICE_IF 0xF
+#define TWL4030_REG_ARXR1PGA 0x10
+#define TWL4030_REG_ARXL1PGA 0x11
+#define TWL4030_REG_ARXR2PGA 0x12
+#define TWL4030_REG_ARXL2PGA 0x13
+#define TWL4030_REG_VRXPGA 0x14
+#define TWL4030_REG_VSTPGA 0x15
+#define TWL4030_REG_VRX2ARXPGA 0x16
+#define TWL4030_REG_AVDAC_CTL 0x17
+#define TWL4030_REG_ARX2VTXPGA 0x18
+#define TWL4030_REG_ARXL1_APGA_CTL 0x19
+#define TWL4030_REG_ARXR1_APGA_CTL 0x1A
+#define TWL4030_REG_ARXL2_APGA_CTL 0x1B
+#define TWL4030_REG_ARXR2_APGA_CTL 0x1C
+#define TWL4030_REG_ATX2ARXPGA 0x1D
+#define TWL4030_REG_BT_IF 0x1E
+#define TWL4030_REG_BTPGA 0x1F
+#define TWL4030_REG_BTSTPGA 0x20
+#define TWL4030_REG_EAR_CTL 0x21
+#define TWL4030_REG_HS_SEL 0x22
+#define TWL4030_REG_HS_GAIN_SET 0x23
+#define TWL4030_REG_HS_POPN_SET 0x24
+#define TWL4030_REG_PREDL_CTL 0x25
+#define TWL4030_REG_PREDR_CTL 0x26
+#define TWL4030_REG_PRECKL_CTL 0x27
+#define TWL4030_REG_PRECKR_CTL 0x28
+#define TWL4030_REG_HFL_CTL 0x29
+#define TWL4030_REG_HFR_CTL 0x2A
+#define TWL4030_REG_ALC_CTL 0x2B
+#define TWL4030_REG_ALC_SET1 0x2C
+#define TWL4030_REG_ALC_SET2 0x2D
+#define TWL4030_REG_BOOST_CTL 0x2E
+#define TWL4030_REG_SOFTVOL_CTL 0x2F
+#define TWL4030_REG_DTMF_FREQSEL 0x30
+#define TWL4030_REG_DTMF_TONEXT1H 0x31
+#define TWL4030_REG_DTMF_TONEXT1L 0x32
+#define TWL4030_REG_DTMF_TONEXT2H 0x33
+#define TWL4030_REG_DTMF_TONEXT2L 0x34
+#define TWL4030_REG_DTMF_TONOFF 0x35
+#define TWL4030_REG_DTMF_WANONOFF 0x36
+#define TWL4030_REG_I2S_RX_SCRAMBLE_H 0x37
+#define TWL4030_REG_I2S_RX_SCRAMBLE_M 0x38
+#define TWL4030_REG_I2S_RX_SCRAMBLE_L 0x39
+#define TWL4030_REG_APLL_CTL 0x3A
+#define TWL4030_REG_DTMF_CTL 0x3B
+#define TWL4030_REG_DTMF_PGA_CTL2 0x3C
+#define TWL4030_REG_DTMF_PGA_CTL1 0x3D
+#define TWL4030_REG_MISC_SET_1 0x3E
+#define TWL4030_REG_PCMBTMUX 0x3F
+#define TWL4030_REG_RX_PATH_SEL 0x43
+#define TWL4030_REG_VDL_APGA_CTL 0x44
+#define TWL4030_REG_VIBRA_CTL 0x45
+#define TWL4030_REG_VIBRA_SET 0x46
+#define TWL4030_REG_VIBRA_PWM_SET 0x47
+#define TWL4030_REG_ANAMIC_GAIN 0x48
+#define TWL4030_REG_MISC_SET_2 0x49
+
+#define TWL4030_CACHEREGNUM (TWL4030_REG_MISC_SET_2 + 1)
+
+/* Bitfield Definitions */
+
+/* TWL4030_CODEC_MODE (0x01) Fields */
+
+#define TWL4030_APLL_RATE 0xF0
+#define TWL4030_APLL_RATE_8000 0x00
+#define TWL4030_APLL_RATE_11025 0x10
+#define TWL4030_APLL_RATE_12000 0x20
+#define TWL4030_APLL_RATE_16000 0x40
+#define TWL4030_APLL_RATE_22050 0x50
+#define TWL4030_APLL_RATE_24000 0x60
+#define TWL4030_APLL_RATE_32000 0x80
+#define TWL4030_APLL_RATE_44100 0x90
+#define TWL4030_APLL_RATE_48000 0xA0
+#define TWL4030_SEL_16K 0x04
+#define TWL4030_CODECPDZ 0x02
+#define TWL4030_OPT_MODE 0x01
+
+/* TWL4030_REG_MICBIAS_CTL (0x04) Fields */
+
+#define TWL4030_MICBIAS2_CTL 0x40
+#define TWL4030_MICBIAS1_CTL 0x20
+#define TWL4030_HSMICBIAS_EN 0x04
+#define TWL4030_MICBIAS2_EN 0x02
+#define TWL4030_MICBIAS1_EN 0x01
+
+/* ANAMICL (0x05) Fields */
+
+#define TWL4030_CNCL_OFFSET_START 0x80
+#define TWL4030_OFFSET_CNCL_SEL 0x60
+#define TWL4030_OFFSET_CNCL_SEL_ARX1 0x00
+#define TWL4030_OFFSET_CNCL_SEL_ARX2 0x20
+#define TWL4030_OFFSET_CNCL_SEL_VRX 0x40
+#define TWL4030_OFFSET_CNCL_SEL_ALL 0x60
+#define TWL4030_MICAMPL_EN 0x10
+#define TWL4030_CKMIC_EN 0x08
+#define TWL4030_AUXL_EN 0x04
+#define TWL4030_HSMIC_EN 0x02
+#define TWL4030_MAINMIC_EN 0x01
+
+/* ANAMICR (0x06) Fields */
+
+#define TWL4030_MICAMPR_EN 0x10
+#define TWL4030_AUXR_EN 0x04
+#define TWL4030_SUBMIC_EN 0x01
+
+/* AVADC_CTL (0x07) Fields */
+
+#define TWL4030_ADCL_EN 0x08
+#define TWL4030_AVADC_CLK_PRIORITY 0x04
+#define TWL4030_ADCR_EN 0x02
+
+/* AUDIO_IF (0x0E) Fields */
+
+#define TWL4030_AIF_SLAVE_EN 0x80
+#define TWL4030_DATA_WIDTH 0x60
+#define TWL4030_DATA_WIDTH_16S_16W 0x00
+#define TWL4030_DATA_WIDTH_32S_16W 0x40
+#define TWL4030_DATA_WIDTH_32S_24W 0x60
+#define TWL4030_AIF_FORMAT 0x18
+#define TWL4030_AIF_FORMAT_CODEC 0x00
+#define TWL4030_AIF_FORMAT_LEFT 0x08
+#define TWL4030_AIF_FORMAT_RIGHT 0x10
+#define TWL4030_AIF_FORMAT_TDM 0x18
+#define TWL4030_AIF_TRI_EN 0x04
+#define TWL4030_CLK256FS_EN 0x02
+#define TWL4030_AIF_EN 0x01
+
+/* HS_GAIN_SET (0x23) Fields */
+
+#define TWL4030_HSR_GAIN 0x0C
+#define TWL4030_HSR_GAIN_PWR_DOWN 0x00
+#define TWL4030_HSR_GAIN_PLUS_6DB 0x04
+#define TWL4030_HSR_GAIN_0DB 0x08
+#define TWL4030_HSR_GAIN_MINUS_6DB 0x0C
+#define TWL4030_HSL_GAIN 0x03
+#define TWL4030_HSL_GAIN_PWR_DOWN 0x00
+#define TWL4030_HSL_GAIN_PLUS_6DB 0x01
+#define TWL4030_HSL_GAIN_0DB 0x02
+#define TWL4030_HSL_GAIN_MINUS_6DB 0x03
+
+/* HS_POPN_SET (0x24) Fields */
+
+#define TWL4030_VMID_EN 0x40
+#define TWL4030_EXTMUTE 0x20
+#define TWL4030_RAMP_DELAY 0x1C
+#define TWL4030_RAMP_DELAY_20MS 0x00
+#define TWL4030_RAMP_DELAY_40MS 0x04
+#define TWL4030_RAMP_DELAY_81MS 0x08
+#define TWL4030_RAMP_DELAY_161MS 0x0C
+#define TWL4030_RAMP_DELAY_323MS 0x10
+#define TWL4030_RAMP_DELAY_645MS 0x14
+#define TWL4030_RAMP_DELAY_1291MS 0x18
+#define TWL4030_RAMP_DELAY_2581MS 0x1C
+#define TWL4030_RAMP_EN 0x02
+
+/* HFL_CTL (0x29, 0x2A) Fields */
+#define TWL4030_HF_CTL_HB_EN 0x04
+#define TWL4030_HF_CTL_LOOP_EN 0x08
+#define TWL4030_HF_CTL_RAMP_EN 0x10
+#define TWL4030_HF_CTL_REF_EN 0x20
+
+/* APLL_CTL (0x3A) Fields */
+
+#define TWL4030_APLL_EN 0x10
+#define TWL4030_APLL_INFREQ 0x0F
+#define TWL4030_APLL_INFREQ_19200KHZ 0x05
+#define TWL4030_APLL_INFREQ_26000KHZ 0x06
+#define TWL4030_APLL_INFREQ_38400KHZ 0x0F
+
+/* REG_MISC_SET_1 (0x3E) Fields */
+
+#define TWL4030_CLK64_EN 0x80
+#define TWL4030_SCRAMBLE_EN 0x40
+#define TWL4030_FMLOOP_EN 0x20
+#define TWL4030_SMOOTH_ANAVOL_EN 0x02
+#define TWL4030_DIGMIC_LR_SWAP_EN 0x01
+
+extern struct snd_soc_dai twl4030_dai;
+extern struct snd_soc_codec_device soc_codec_dev_twl4030;
+
+#endif /* End of __TWL4030_AUDIO_H__ */
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
new file mode 100644
index 0000000..a2c5064
--- /dev/null
+++ b/sound/soc/codecs/uda134x.c
@@ -0,0 +1,668 @@
+/*
+ * uda134x.c -- UDA134X ALSA SoC Codec driver
+ *
+ * Modifications by Christian Pellegrin <chripell@evolware.org>
+ *
+ * Copyright 2007 Dension Audio Systems Ltd.
+ * Author: Zoltan Devai
+ *
+ * Based on the WM87xx drivers by Liam Girdwood and Richard Purdie
+ *
+ * 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/delay.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include <sound/uda134x.h>
+#include <sound/l3.h>
+
+#include "uda134x.h"
+
+
+#define POWER_OFF_ON_STANDBY 1
+/*
+ ALSA SOC usually puts the device in standby mode when it's not used
+ for sometime. If you define POWER_OFF_ON_STANDBY the driver will
+ turn off the ADC/DAC when this callback is invoked and turn it back
+ on when needed. Unfortunately this will result in a very light bump
+ (it can be audible only with good earphones). If this bothers you
+ just comment this line, you will have slightly higher power
+ consumption . Please note that sending the L3 command for ADC is
+ enough to make the bump, so it doesn't make difference if you
+ completely take off power from the codec.
+ */
+
+#define UDA134X_RATES SNDRV_PCM_RATE_8000_48000
+#define UDA134X_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE)
+
+struct uda134x_priv {
+ int sysclk;
+ int dai_fmt;
+
+ struct snd_pcm_substream *master_substream;
+ struct snd_pcm_substream *slave_substream;
+};
+
+/* In-data addresses are hard-coded into the reg-cache values */
+static const char uda134x_reg[UDA134X_REGS_NUM] = {
+ /* Extended address registers */
+ 0x04, 0x04, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00,
+ /* Status, data regs */
+ 0x00, 0x83, 0x00, 0x40, 0x80, 0x00,
+};
+
+/*
+ * The codec has no support for reading its registers except for peak level...
+ */
+static inline unsigned int uda134x_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u8 *cache = codec->reg_cache;
+
+ if (reg >= UDA134X_REGS_NUM)
+ return -1;
+ return cache[reg];
+}
+
+/*
+ * Write the register cache
+ */
+static inline void uda134x_write_reg_cache(struct snd_soc_codec *codec,
+ u8 reg, unsigned int value)
+{
+ u8 *cache = codec->reg_cache;
+
+ if (reg >= UDA134X_REGS_NUM)
+ return;
+ cache[reg] = value;
+}
+
+/*
+ * Write to the uda134x registers
+ *
+ */
+static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ int ret;
+ u8 addr;
+ u8 data = value;
+ struct uda134x_platform_data *pd = codec->control_data;
+
+ pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value);
+
+ if (reg >= UDA134X_REGS_NUM) {
+ printk(KERN_ERR "%s unkown register: reg: %d",
+ __func__, reg);
+ return -EINVAL;
+ }
+
+ uda134x_write_reg_cache(codec, reg, value);
+
+ switch (reg) {
+ case UDA134X_STATUS0:
+ case UDA134X_STATUS1:
+ addr = UDA134X_STATUS_ADDR;
+ break;
+ case UDA134X_DATA000:
+ case UDA134X_DATA001:
+ case UDA134X_DATA010:
+ addr = UDA134X_DATA0_ADDR;
+ break;
+ case UDA134X_DATA1:
+ addr = UDA134X_DATA1_ADDR;
+ break;
+ default:
+ /* It's an extended address register */
+ addr = (reg | UDA134X_EXTADDR_PREFIX);
+
+ ret = l3_write(&pd->l3,
+ UDA134X_DATA0_ADDR, &addr, 1);
+ if (ret != 1)
+ return -EIO;
+
+ addr = UDA134X_DATA0_ADDR;
+ data = (value | UDA134X_EXTDATA_PREFIX);
+ break;
+ }
+
+ ret = l3_write(&pd->l3,
+ addr, &data, 1);
+ if (ret != 1)
+ return -EIO;
+
+ return 0;
+}
+
+static inline void uda134x_reset(struct snd_soc_codec *codec)
+{
+ u8 reset_reg = uda134x_read_reg_cache(codec, UDA134X_STATUS0);
+ uda134x_write(codec, UDA134X_STATUS0, reset_reg | (1<<6));
+ msleep(1);
+ uda134x_write(codec, UDA134X_STATUS0, reset_reg & ~(1<<6));
+}
+
+static int uda134x_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u8 mute_reg = uda134x_read_reg_cache(codec, UDA134X_DATA010);
+
+ pr_debug("%s mute: %d\n", __func__, mute);
+
+ if (mute)
+ mute_reg |= (1<<2);
+ else
+ mute_reg &= ~(1<<2);
+
+ uda134x_write(codec, UDA134X_DATA010, mute_reg & ~(1<<2));
+
+ return 0;
+}
+
+static int uda134x_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct uda134x_priv *uda134x = codec->private_data;
+ struct snd_pcm_runtime *master_runtime;
+
+ if (uda134x->master_substream) {
+ master_runtime = uda134x->master_substream->runtime;
+
+ pr_debug("%s constraining to %d bits at %d\n", __func__,
+ master_runtime->sample_bits,
+ master_runtime->rate);
+
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE,
+ master_runtime->rate,
+ master_runtime->rate);
+
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+ master_runtime->sample_bits,
+ master_runtime->sample_bits);
+
+ uda134x->slave_substream = substream;
+ } else
+ uda134x->master_substream = substream;
+
+ return 0;
+}
+
+static void uda134x_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct uda134x_priv *uda134x = codec->private_data;
+
+ if (uda134x->master_substream == substream)
+ uda134x->master_substream = uda134x->slave_substream;
+
+ uda134x->slave_substream = NULL;
+}
+
+static int uda134x_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct uda134x_priv *uda134x = codec->private_data;
+ u8 hw_params;
+
+ if (substream == uda134x->slave_substream) {
+ pr_debug("%s ignoring hw_params for slave substream\n",
+ __func__);
+ return 0;
+ }
+
+ hw_params = uda134x_read_reg_cache(codec, UDA134X_STATUS0);
+ hw_params &= STATUS0_SYSCLK_MASK;
+ hw_params &= STATUS0_DAIFMT_MASK;
+
+ pr_debug("%s sysclk: %d, rate:%d\n", __func__,
+ uda134x->sysclk, params_rate(params));
+
+ /* set SYSCLK / fs ratio */
+ switch (uda134x->sysclk / params_rate(params)) {
+ case 512:
+ break;
+ case 384:
+ hw_params |= (1<<4);
+ break;
+ case 256:
+ hw_params |= (1<<5);
+ break;
+ default:
+ printk(KERN_ERR "%s unsupported fs\n", __func__);
+ return -EINVAL;
+ }
+
+ pr_debug("%s dai_fmt: %d, params_format:%d\n", __func__,
+ uda134x->dai_fmt, params_format(params));
+
+ /* set DAI format and word length */
+ switch (uda134x->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ hw_params |= (1<<1);
+ break;
+ case SNDRV_PCM_FORMAT_S18_3LE:
+ hw_params |= (1<<2);
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ hw_params |= ((1<<2) | (1<<1));
+ break;
+ default:
+ printk(KERN_ERR "%s unsupported format (right)\n",
+ __func__);
+ return -EINVAL;
+ }
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ hw_params |= (1<<3);
+ break;
+ default:
+ printk(KERN_ERR "%s unsupported format\n", __func__);
+ return -EINVAL;
+ }
+
+ uda134x_write(codec, UDA134X_STATUS0, hw_params);
+
+ return 0;
+}
+
+static int uda134x_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct uda134x_priv *uda134x = codec->private_data;
+
+ pr_debug("%s clk_id: %d, freq: %d, dir: %d\n", __func__,
+ clk_id, freq, dir);
+
+ /* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable
+ because the codec is slave. Of course limitations of the clock
+ master (the IIS controller) apply.
+ We'll error out on set_hw_params if it's not OK */
+ if ((freq >= (256 * 8000)) && (freq <= (512 * 48000))) {
+ uda134x->sysclk = freq;
+ return 0;
+ }
+
+ printk(KERN_ERR "%s unsupported sysclk\n", __func__);
+ return -EINVAL;
+}
+
+static int uda134x_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct uda134x_priv *uda134x = codec->private_data;
+
+ pr_debug("%s fmt: %08X\n", __func__, fmt);
+
+ /* codec supports only full slave mode */
+ if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) {
+ printk(KERN_ERR "%s unsupported slave mode\n", __func__);
+ return -EINVAL;
+ }
+
+ /* no support for clock inversion */
+ if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) {
+ printk(KERN_ERR "%s unsupported clock inversion\n", __func__);
+ return -EINVAL;
+ }
+
+ /* We can't setup DAI format here as it depends on the word bit num */
+ /* so let's just store the value for later */
+ uda134x->dai_fmt = fmt;
+
+ return 0;
+}
+
+static int uda134x_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u8 reg;
+ struct uda134x_platform_data *pd = codec->control_data;
+ int i;
+ u8 *cache = codec->reg_cache;
+
+ pr_debug("%s bias level %d\n", __func__, level);
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ /* ADC, DAC on */
+ reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
+ uda134x_write(codec, UDA134X_STATUS1, reg | 0x03);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ /* power on */
+ if (pd->power) {
+ pd->power(1);
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < ARRAY_SIZE(uda134x_reg); i++)
+ codec->write(codec, i, *cache++);
+ }
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ /* ADC, DAC power off */
+ reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
+ uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03));
+ break;
+ case SND_SOC_BIAS_OFF:
+ /* power off */
+ if (pd->power)
+ pd->power(0);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+static const char *uda134x_dsp_setting[] = {"Flat", "Minimum1",
+ "Minimum2", "Maximum"};
+static const char *uda134x_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+static const char *uda134x_mixmode[] = {"Differential", "Analog1",
+ "Analog2", "Both"};
+
+static const struct soc_enum uda134x_mixer_enum[] = {
+SOC_ENUM_SINGLE(UDA134X_DATA010, 0, 0x04, uda134x_dsp_setting),
+SOC_ENUM_SINGLE(UDA134X_DATA010, 3, 0x04, uda134x_deemph),
+SOC_ENUM_SINGLE(UDA134X_EA010, 0, 0x04, uda134x_mixmode),
+};
+
+static const struct snd_kcontrol_new uda1341_snd_controls[] = {
+SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1),
+SOC_SINGLE("Capture Volume", UDA134X_EA010, 2, 0x07, 0),
+SOC_SINGLE("Analog1 Volume", UDA134X_EA000, 0, 0x1F, 1),
+SOC_SINGLE("Analog2 Volume", UDA134X_EA001, 0, 0x1F, 1),
+
+SOC_SINGLE("Mic Sensitivity", UDA134X_EA010, 2, 7, 0),
+SOC_SINGLE("Mic Volume", UDA134X_EA101, 0, 0x1F, 0),
+
+SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0),
+SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0),
+
+SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]),
+SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]),
+SOC_ENUM("Input Mux", uda134x_mixer_enum[2]),
+
+SOC_SINGLE("AGC Switch", UDA134X_EA100, 4, 1, 0),
+SOC_SINGLE("AGC Target Volume", UDA134X_EA110, 0, 0x03, 1),
+SOC_SINGLE("AGC Timing", UDA134X_EA110, 2, 0x07, 0),
+
+SOC_SINGLE("DAC +6dB Switch", UDA134X_STATUS1, 6, 1, 0),
+SOC_SINGLE("ADC +6dB Switch", UDA134X_STATUS1, 5, 1, 0),
+SOC_SINGLE("ADC Polarity Switch", UDA134X_STATUS1, 4, 1, 0),
+SOC_SINGLE("DAC Polarity Switch", UDA134X_STATUS1, 3, 1, 0),
+SOC_SINGLE("Double Speed Playback Switch", UDA134X_STATUS1, 2, 1, 0),
+SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new uda1340_snd_controls[] = {
+SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1),
+
+SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0),
+SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0),
+
+SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]),
+SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]),
+
+SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0),
+};
+
+static int uda134x_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i, n;
+ const struct snd_kcontrol_new *ctrls;
+ struct uda134x_platform_data *pd = codec->control_data;
+
+ switch (pd->model) {
+ case UDA134X_UDA1340:
+ case UDA134X_UDA1344:
+ n = ARRAY_SIZE(uda1340_snd_controls);
+ ctrls = uda1340_snd_controls;
+ break;
+ case UDA134X_UDA1341:
+ n = ARRAY_SIZE(uda1341_snd_controls);
+ ctrls = uda1341_snd_controls;
+ break;
+ default:
+ printk(KERN_ERR "%s unkown codec type: %d",
+ __func__, pd->model);
+ return -EINVAL;
+ }
+
+ for (i = 0; i < n; i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&ctrls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+struct snd_soc_dai uda134x_dai = {
+ .name = "UDA134X",
+ /* playback capabilities */
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = UDA134X_RATES,
+ .formats = UDA134X_FORMATS,
+ },
+ /* capture capabilities */
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = UDA134X_RATES,
+ .formats = UDA134X_FORMATS,
+ },
+ /* pcm operations */
+ .ops = {
+ .startup = uda134x_startup,
+ .shutdown = uda134x_shutdown,
+ .hw_params = uda134x_hw_params,
+ .digital_mute = uda134x_mute,
+ .set_sysclk = uda134x_set_dai_sysclk,
+ .set_fmt = uda134x_set_dai_fmt,
+ }
+};
+EXPORT_SYMBOL(uda134x_dai);
+
+
+static int uda134x_soc_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ struct uda134x_priv *uda134x;
+ void *codec_setup_data = socdev->codec_data;
+ int ret = -ENOMEM;
+ struct uda134x_platform_data *pd;
+
+ printk(KERN_INFO "UDA134X SoC Audio Codec\n");
+
+ if (!codec_setup_data) {
+ printk(KERN_ERR "UDA134X SoC codec: "
+ "missing L3 bitbang function\n");
+ return -ENODEV;
+ }
+
+ pd = codec_setup_data;
+ switch (pd->model) {
+ case UDA134X_UDA1340:
+ case UDA134X_UDA1341:
+ case UDA134X_UDA1344:
+ break;
+ default:
+ printk(KERN_ERR "UDA134X SoC codec: "
+ "unsupported model %d\n",
+ pd->model);
+ return -EINVAL;
+ }
+
+ socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (socdev->codec == NULL)
+ return ret;
+
+ codec = socdev->codec;
+
+ uda134x = kzalloc(sizeof(struct uda134x_priv), GFP_KERNEL);
+ if (uda134x == NULL)
+ goto priv_err;
+ codec->private_data = uda134x;
+
+ codec->reg_cache = kmemdup(uda134x_reg, sizeof(uda134x_reg),
+ GFP_KERNEL);
+ if (codec->reg_cache == NULL)
+ goto reg_err;
+
+ mutex_init(&codec->mutex);
+
+ codec->reg_cache_size = sizeof(uda134x_reg);
+ codec->reg_cache_step = 1;
+
+ codec->name = "UDA134X";
+ codec->owner = THIS_MODULE;
+ codec->dai = &uda134x_dai;
+ codec->num_dai = 1;
+ codec->read = uda134x_read_reg_cache;
+ codec->write = uda134x_write;
+#ifdef POWER_OFF_ON_STANDBY
+ codec->set_bias_level = uda134x_set_bias_level;
+#endif
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->control_data = codec_setup_data;
+
+ if (pd->power)
+ pd->power(1);
+
+ uda134x_reset(codec);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "UDA134X: failed to register pcms\n");
+ goto pcm_err;
+ }
+
+ ret = uda134x_add_controls(codec);
+ if (ret < 0) {
+ printk(KERN_ERR "UDA134X: failed to register controls\n");
+ goto pcm_err;
+ }
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "UDA134X: failed to register card\n");
+ goto card_err;
+ }
+
+ return 0;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ kfree(codec->reg_cache);
+reg_err:
+ kfree(codec->private_data);
+priv_err:
+ kfree(codec);
+ return ret;
+}
+
+/* power down chip */
+static int uda134x_soc_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+
+ kfree(codec->private_data);
+ kfree(codec->reg_cache);
+ kfree(codec);
+
+ return 0;
+}
+
+#if defined(CONFIG_PM)
+static int uda134x_soc_suspend(struct platform_device *pdev,
+ pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int uda134x_soc_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ uda134x_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
+ uda134x_set_bias_level(codec, SND_SOC_BIAS_ON);
+ return 0;
+}
+#else
+#define uda134x_soc_suspend NULL
+#define uda134x_soc_resume NULL
+#endif /* CONFIG_PM */
+
+struct snd_soc_codec_device soc_codec_dev_uda134x = {
+ .probe = uda134x_soc_probe,
+ .remove = uda134x_soc_remove,
+ .suspend = uda134x_soc_suspend,
+ .resume = uda134x_soc_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_uda134x);
+
+static int __init uda134x_init(void)
+{
+ return snd_soc_register_dai(&uda134x_dai);
+}
+module_init(uda134x_init);
+
+static void __exit uda134x_exit(void)
+{
+ snd_soc_unregister_dai(&uda134x_dai);
+}
+module_exit(uda134x_exit);
+
+MODULE_DESCRIPTION("UDA134X ALSA soc codec driver");
+MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell@evolware.org>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/uda134x.h b/sound/soc/codecs/uda134x.h
new file mode 100644
index 0000000..94f4404
--- /dev/null
+++ b/sound/soc/codecs/uda134x.h
@@ -0,0 +1,36 @@
+#ifndef _UDA134X_CODEC_H
+#define _UDA134X_CODEC_H
+
+#define UDA134X_L3ADDR 5
+#define UDA134X_DATA0_ADDR ((UDA134X_L3ADDR << 2) | 0)
+#define UDA134X_DATA1_ADDR ((UDA134X_L3ADDR << 2) | 1)
+#define UDA134X_STATUS_ADDR ((UDA134X_L3ADDR << 2) | 2)
+
+#define UDA134X_EXTADDR_PREFIX 0xC0
+#define UDA134X_EXTDATA_PREFIX 0xE0
+
+/* UDA134X registers */
+#define UDA134X_EA000 0
+#define UDA134X_EA001 1
+#define UDA134X_EA010 2
+#define UDA134X_EA011 3
+#define UDA134X_EA100 4
+#define UDA134X_EA101 5
+#define UDA134X_EA110 6
+#define UDA134X_EA111 7
+#define UDA134X_STATUS0 8
+#define UDA134X_STATUS1 9
+#define UDA134X_DATA000 10
+#define UDA134X_DATA001 11
+#define UDA134X_DATA010 12
+#define UDA134X_DATA1 13
+
+#define UDA134X_REGS_NUM 14
+
+#define STATUS0_DAIFMT_MASK (~(7<<1))
+#define STATUS0_SYSCLK_MASK (~(3<<4))
+
+extern struct snd_soc_dai uda134x_dai;
+extern struct snd_soc_codec_device soc_codec_dev_uda134x;
+
+#endif
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index a69ee72..e6bf084 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -407,7 +407,8 @@
* when the DAI is being clocked by the CPU DAI. It's up to the
* machine and cpu DAI driver to do this before we are called.
*/
-static int uda1380_pcm_prepare(struct snd_pcm_substream *substream)
+static int uda1380_pcm_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -439,7 +440,8 @@
}
static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -477,7 +479,8 @@
return 0;
}
-static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream)
+static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -560,8 +563,6 @@
.hw_params = uda1380_pcm_hw_params,
.shutdown = uda1380_pcm_shutdown,
.prepare = uda1380_pcm_prepare,
- },
- .dai_ops = {
.digital_mute = uda1380_mute,
.set_fmt = uda1380_set_dai_fmt,
},
@@ -579,8 +580,6 @@
.hw_params = uda1380_pcm_hw_params,
.shutdown = uda1380_pcm_shutdown,
.prepare = uda1380_pcm_prepare,
- },
- .dai_ops = {
.digital_mute = uda1380_mute,
.set_fmt = uda1380_set_dai_fmt,
},
@@ -598,8 +597,6 @@
.hw_params = uda1380_pcm_hw_params,
.shutdown = uda1380_pcm_shutdown,
.prepare = uda1380_pcm_prepare,
- },
- .dai_ops = {
.set_fmt = uda1380_set_dai_fmt,
},
},
@@ -680,7 +677,7 @@
/* uda1380 init */
uda1380_add_controls(codec);
uda1380_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
pr_err("uda1380: failed to register card\n");
goto card_err;
@@ -844,6 +841,18 @@
};
EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380);
+static int __init uda1380_modinit(void)
+{
+ return snd_soc_register_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai));
+}
+module_init(uda1380_modinit);
+
+static void __exit uda1380_exit(void)
+{
+ snd_soc_unregister_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai));
+}
+module_exit(uda1380_exit);
+
MODULE_AUTHOR("Giorgio Padrin");
MODULE_DESCRIPTION("Audio support for codec Philips UDA1380");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
new file mode 100644
index 0000000..e3989d4
--- /dev/null
+++ b/sound/soc/codecs/wm8350.c
@@ -0,0 +1,1583 @@
+/*
+ * wm8350.c -- WM8350 ALSA SoC audio driver
+ *
+ * Copyright (C) 2007, 2008 Wolfson Microelectronics PLC.
+ *
+ * Author: Liam Girdwood <lg@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/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/platform_device.h>
+#include <linux/mfd/wm8350/audio.h>
+#include <linux/mfd/wm8350/core.h>
+#include <linux/regulator/consumer.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "wm8350.h"
+
+#define WM8350_OUTn_0dB 0x39
+
+#define WM8350_RAMP_NONE 0
+#define WM8350_RAMP_UP 1
+#define WM8350_RAMP_DOWN 2
+
+/* We only include the analogue supplies here; the digital supplies
+ * need to be available well before this driver can be probed.
+ */
+static const char *supply_names[] = {
+ "AVDD",
+ "HPVDD",
+};
+
+struct wm8350_output {
+ u16 active;
+ u16 left_vol;
+ u16 right_vol;
+ u16 ramp;
+ u16 mute;
+};
+
+struct wm8350_data {
+ struct snd_soc_codec codec;
+ struct wm8350_output out1;
+ struct wm8350_output out2;
+ struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
+};
+
+static unsigned int wm8350_codec_cache_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ struct wm8350 *wm8350 = codec->control_data;
+ return wm8350->reg_cache[reg];
+}
+
+static unsigned int wm8350_codec_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ struct wm8350 *wm8350 = codec->control_data;
+ return wm8350_reg_read(wm8350, reg);
+}
+
+static int wm8350_codec_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ struct wm8350 *wm8350 = codec->control_data;
+ return wm8350_reg_write(wm8350, reg, value);
+}
+
+/*
+ * Ramp OUT1 PGA volume to minimise pops at stream startup and shutdown.
+ */
+static inline int wm8350_out1_ramp_step(struct snd_soc_codec *codec)
+{
+ struct wm8350_data *wm8350_data = codec->private_data;
+ struct wm8350_output *out1 = &wm8350_data->out1;
+ struct wm8350 *wm8350 = codec->control_data;
+ int left_complete = 0, right_complete = 0;
+ u16 reg, val;
+
+ /* left channel */
+ reg = wm8350_reg_read(wm8350, WM8350_LOUT1_VOLUME);
+ val = (reg & WM8350_OUT1L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT;
+
+ if (out1->ramp == WM8350_RAMP_UP) {
+ /* ramp step up */
+ if (val < out1->left_vol) {
+ val++;
+ reg &= ~WM8350_OUT1L_VOL_MASK;
+ wm8350_reg_write(wm8350, WM8350_LOUT1_VOLUME,
+ reg | (val << WM8350_OUT1L_VOL_SHIFT));
+ } else
+ left_complete = 1;
+ } else if (out1->ramp == WM8350_RAMP_DOWN) {
+ /* ramp step down */
+ if (val > 0) {
+ val--;
+ reg &= ~WM8350_OUT1L_VOL_MASK;
+ wm8350_reg_write(wm8350, WM8350_LOUT1_VOLUME,
+ reg | (val << WM8350_OUT1L_VOL_SHIFT));
+ } else
+ left_complete = 1;
+ } else
+ return 1;
+
+ /* right channel */
+ reg = wm8350_reg_read(wm8350, WM8350_ROUT1_VOLUME);
+ val = (reg & WM8350_OUT1R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT;
+ if (out1->ramp == WM8350_RAMP_UP) {
+ /* ramp step up */
+ if (val < out1->right_vol) {
+ val++;
+ reg &= ~WM8350_OUT1R_VOL_MASK;
+ wm8350_reg_write(wm8350, WM8350_ROUT1_VOLUME,
+ reg | (val << WM8350_OUT1R_VOL_SHIFT));
+ } else
+ right_complete = 1;
+ } else if (out1->ramp == WM8350_RAMP_DOWN) {
+ /* ramp step down */
+ if (val > 0) {
+ val--;
+ reg &= ~WM8350_OUT1R_VOL_MASK;
+ wm8350_reg_write(wm8350, WM8350_ROUT1_VOLUME,
+ reg | (val << WM8350_OUT1R_VOL_SHIFT));
+ } else
+ right_complete = 1;
+ }
+
+ /* only hit the update bit if either volume has changed this step */
+ if (!left_complete || !right_complete)
+ wm8350_set_bits(wm8350, WM8350_LOUT1_VOLUME, WM8350_OUT1_VU);
+
+ return left_complete & right_complete;
+}
+
+/*
+ * Ramp OUT2 PGA volume to minimise pops at stream startup and shutdown.
+ */
+static inline int wm8350_out2_ramp_step(struct snd_soc_codec *codec)
+{
+ struct wm8350_data *wm8350_data = codec->private_data;
+ struct wm8350_output *out2 = &wm8350_data->out2;
+ struct wm8350 *wm8350 = codec->control_data;
+ int left_complete = 0, right_complete = 0;
+ u16 reg, val;
+
+ /* left channel */
+ reg = wm8350_reg_read(wm8350, WM8350_LOUT2_VOLUME);
+ val = (reg & WM8350_OUT2L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT;
+ if (out2->ramp == WM8350_RAMP_UP) {
+ /* ramp step up */
+ if (val < out2->left_vol) {
+ val++;
+ reg &= ~WM8350_OUT2L_VOL_MASK;
+ wm8350_reg_write(wm8350, WM8350_LOUT2_VOLUME,
+ reg | (val << WM8350_OUT1L_VOL_SHIFT));
+ } else
+ left_complete = 1;
+ } else if (out2->ramp == WM8350_RAMP_DOWN) {
+ /* ramp step down */
+ if (val > 0) {
+ val--;
+ reg &= ~WM8350_OUT2L_VOL_MASK;
+ wm8350_reg_write(wm8350, WM8350_LOUT2_VOLUME,
+ reg | (val << WM8350_OUT1L_VOL_SHIFT));
+ } else
+ left_complete = 1;
+ } else
+ return 1;
+
+ /* right channel */
+ reg = wm8350_reg_read(wm8350, WM8350_ROUT2_VOLUME);
+ val = (reg & WM8350_OUT2R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT;
+ if (out2->ramp == WM8350_RAMP_UP) {
+ /* ramp step up */
+ if (val < out2->right_vol) {
+ val++;
+ reg &= ~WM8350_OUT2R_VOL_MASK;
+ wm8350_reg_write(wm8350, WM8350_ROUT2_VOLUME,
+ reg | (val << WM8350_OUT1R_VOL_SHIFT));
+ } else
+ right_complete = 1;
+ } else if (out2->ramp == WM8350_RAMP_DOWN) {
+ /* ramp step down */
+ if (val > 0) {
+ val--;
+ reg &= ~WM8350_OUT2R_VOL_MASK;
+ wm8350_reg_write(wm8350, WM8350_ROUT2_VOLUME,
+ reg | (val << WM8350_OUT1R_VOL_SHIFT));
+ } else
+ right_complete = 1;
+ }
+
+ /* only hit the update bit if either volume has changed this step */
+ if (!left_complete || !right_complete)
+ wm8350_set_bits(wm8350, WM8350_LOUT2_VOLUME, WM8350_OUT2_VU);
+
+ return left_complete & right_complete;
+}
+
+/*
+ * This work ramps both output PGAs at stream start/stop time to
+ * minimise pop associated with DAPM power switching.
+ * It's best to enable Zero Cross when ramping occurs to minimise any
+ * zipper noises.
+ */
+static void wm8350_pga_work(struct work_struct *work)
+{
+ struct snd_soc_codec *codec =
+ container_of(work, struct snd_soc_codec, delayed_work.work);
+ struct wm8350_data *wm8350_data = codec->private_data;
+ struct wm8350_output *out1 = &wm8350_data->out1,
+ *out2 = &wm8350_data->out2;
+ int i, out1_complete, out2_complete;
+
+ /* do we need to ramp at all ? */
+ if (out1->ramp == WM8350_RAMP_NONE && out2->ramp == WM8350_RAMP_NONE)
+ return;
+
+ /* PGA volumes have 6 bits of resolution to ramp */
+ for (i = 0; i <= 63; i++) {
+ out1_complete = 1, out2_complete = 1;
+ if (out1->ramp != WM8350_RAMP_NONE)
+ out1_complete = wm8350_out1_ramp_step(codec);
+ if (out2->ramp != WM8350_RAMP_NONE)
+ out2_complete = wm8350_out2_ramp_step(codec);
+
+ /* ramp finished ? */
+ if (out1_complete && out2_complete)
+ break;
+
+ /* we need to delay longer on the up ramp */
+ if (out1->ramp == WM8350_RAMP_UP ||
+ out2->ramp == WM8350_RAMP_UP) {
+ /* delay is longer over 0dB as increases are larger */
+ if (i >= WM8350_OUTn_0dB)
+ schedule_timeout_interruptible(msecs_to_jiffies
+ (2));
+ else
+ schedule_timeout_interruptible(msecs_to_jiffies
+ (1));
+ } else
+ udelay(50); /* doesn't matter if we delay longer */
+ }
+
+ out1->ramp = WM8350_RAMP_NONE;
+ out2->ramp = WM8350_RAMP_NONE;
+}
+
+/*
+ * WM8350 Controls
+ */
+
+static int pga_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct wm8350_data *wm8350_data = codec->private_data;
+ struct wm8350_output *out;
+
+ switch (w->shift) {
+ case 0:
+ case 1:
+ out = &wm8350_data->out1;
+ break;
+ case 2:
+ case 3:
+ out = &wm8350_data->out2;
+ break;
+
+ default:
+ BUG();
+ return -1;
+ }
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ out->ramp = WM8350_RAMP_UP;
+ out->active = 1;
+
+ if (!delayed_work_pending(&codec->delayed_work))
+ schedule_delayed_work(&codec->delayed_work,
+ msecs_to_jiffies(1));
+ break;
+
+ case SND_SOC_DAPM_PRE_PMD:
+ out->ramp = WM8350_RAMP_DOWN;
+ out->active = 0;
+
+ if (!delayed_work_pending(&codec->delayed_work))
+ schedule_delayed_work(&codec->delayed_work,
+ msecs_to_jiffies(1));
+ break;
+ }
+
+ return 0;
+}
+
+static int wm8350_put_volsw_2r_vu(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wm8350_data *wm8350_priv = codec->private_data;
+ struct wm8350_output *out = NULL;
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ int ret;
+ unsigned int reg = mc->reg;
+ u16 val;
+
+ /* For OUT1 and OUT2 we shadow the values and only actually write
+ * them out when active in order to ensure the amplifier comes on
+ * as quietly as possible. */
+ switch (reg) {
+ case WM8350_LOUT1_VOLUME:
+ out = &wm8350_priv->out1;
+ break;
+ case WM8350_LOUT2_VOLUME:
+ out = &wm8350_priv->out2;
+ break;
+ default:
+ break;
+ }
+
+ if (out) {
+ out->left_vol = ucontrol->value.integer.value[0];
+ out->right_vol = ucontrol->value.integer.value[1];
+ if (!out->active)
+ return 1;
+ }
+
+ ret = snd_soc_put_volsw_2r(kcontrol, ucontrol);
+ if (ret < 0)
+ return ret;
+
+ /* now hit the volume update bits (always bit 8) */
+ val = wm8350_codec_read(codec, reg);
+ wm8350_codec_write(codec, reg, val | WM8350_OUT1_VU);
+ return 1;
+}
+
+static int wm8350_get_volsw_2r(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wm8350_data *wm8350_priv = codec->private_data;
+ struct wm8350_output *out1 = &wm8350_priv->out1;
+ struct wm8350_output *out2 = &wm8350_priv->out2;
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ unsigned int reg = mc->reg;
+
+ /* If these are cached registers use the cache */
+ switch (reg) {
+ case WM8350_LOUT1_VOLUME:
+ ucontrol->value.integer.value[0] = out1->left_vol;
+ ucontrol->value.integer.value[1] = out1->right_vol;
+ return 0;
+
+ case WM8350_LOUT2_VOLUME:
+ ucontrol->value.integer.value[0] = out2->left_vol;
+ ucontrol->value.integer.value[1] = out2->right_vol;
+ return 0;
+
+ default:
+ break;
+ }
+
+ return snd_soc_get_volsw_2r(kcontrol, ucontrol);
+}
+
+/* double control with volume update */
+#define SOC_WM8350_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, \
+ xinvert, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+ SNDRV_CTL_ELEM_ACCESS_READWRITE | \
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw_2r, \
+ .get = wm8350_get_volsw_2r, .put = wm8350_put_volsw_2r_vu, \
+ .private_value = (unsigned long)&(struct soc_mixer_control) \
+ {.reg = reg_left, .rreg = reg_right, .shift = xshift, \
+ .rshift = xshift, .max = xmax, .invert = xinvert}, }
+
+static const char *wm8350_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" };
+static const char *wm8350_pol[] = { "Normal", "Inv R", "Inv L", "Inv L & R" };
+static const char *wm8350_dacmutem[] = { "Normal", "Soft" };
+static const char *wm8350_dacmutes[] = { "Fast", "Slow" };
+static const char *wm8350_dacfilter[] = { "Normal", "Sloping" };
+static const char *wm8350_adcfilter[] = { "None", "High Pass" };
+static const char *wm8350_adchp[] = { "44.1kHz", "8kHz", "16kHz", "32kHz" };
+static const char *wm8350_lr[] = { "Left", "Right" };
+
+static const struct soc_enum wm8350_enum[] = {
+ SOC_ENUM_SINGLE(WM8350_DAC_CONTROL, 4, 4, wm8350_deemp),
+ SOC_ENUM_SINGLE(WM8350_DAC_CONTROL, 0, 4, wm8350_pol),
+ SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 14, 2, wm8350_dacmutem),
+ SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 13, 2, wm8350_dacmutes),
+ SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 12, 2, wm8350_dacfilter),
+ SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 15, 2, wm8350_adcfilter),
+ SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 8, 4, wm8350_adchp),
+ SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 0, 4, wm8350_pol),
+ SOC_ENUM_SINGLE(WM8350_INPUT_MIXER_VOLUME, 15, 2, wm8350_lr),
+};
+
+static DECLARE_TLV_DB_LINEAR(pre_amp_tlv, -1200, 3525);
+static DECLARE_TLV_DB_LINEAR(out_pga_tlv, -5700, 600);
+static DECLARE_TLV_DB_SCALE(dac_pcm_tlv, -7163, 36, 1);
+static DECLARE_TLV_DB_SCALE(adc_pcm_tlv, -12700, 50, 1);
+static DECLARE_TLV_DB_SCALE(out_mix_tlv, -1500, 300, 1);
+
+static const unsigned int capture_sd_tlv[] = {
+ TLV_DB_RANGE_HEAD(2),
+ 0, 12, TLV_DB_SCALE_ITEM(-3600, 300, 1),
+ 13, 15, TLV_DB_SCALE_ITEM(0, 0, 0),
+};
+
+static const struct snd_kcontrol_new wm8350_snd_controls[] = {
+ SOC_ENUM("Playback Deemphasis", wm8350_enum[0]),
+ SOC_ENUM("Playback DAC Inversion", wm8350_enum[1]),
+ SOC_WM8350_DOUBLE_R_TLV("Playback PCM Volume",
+ WM8350_DAC_DIGITAL_VOLUME_L,
+ WM8350_DAC_DIGITAL_VOLUME_R,
+ 0, 255, 0, dac_pcm_tlv),
+ SOC_ENUM("Playback PCM Mute Function", wm8350_enum[2]),
+ SOC_ENUM("Playback PCM Mute Speed", wm8350_enum[3]),
+ SOC_ENUM("Playback PCM Filter", wm8350_enum[4]),
+ SOC_ENUM("Capture PCM Filter", wm8350_enum[5]),
+ SOC_ENUM("Capture PCM HP Filter", wm8350_enum[6]),
+ SOC_ENUM("Capture ADC Inversion", wm8350_enum[7]),
+ SOC_WM8350_DOUBLE_R_TLV("Capture PCM Volume",
+ WM8350_ADC_DIGITAL_VOLUME_L,
+ WM8350_ADC_DIGITAL_VOLUME_R,
+ 0, 255, 0, adc_pcm_tlv),
+ SOC_DOUBLE_TLV("Capture Sidetone Volume",
+ WM8350_ADC_DIVIDER,
+ 8, 4, 15, 1, capture_sd_tlv),
+ SOC_WM8350_DOUBLE_R_TLV("Capture Volume",
+ WM8350_LEFT_INPUT_VOLUME,
+ WM8350_RIGHT_INPUT_VOLUME,
+ 2, 63, 0, pre_amp_tlv),
+ SOC_DOUBLE_R("Capture ZC Switch",
+ WM8350_LEFT_INPUT_VOLUME,
+ WM8350_RIGHT_INPUT_VOLUME, 13, 1, 0),
+ SOC_SINGLE_TLV("Left Input Left Sidetone Volume",
+ WM8350_OUTPUT_LEFT_MIXER_VOLUME, 1, 7, 0, out_mix_tlv),
+ SOC_SINGLE_TLV("Left Input Right Sidetone Volume",
+ WM8350_OUTPUT_LEFT_MIXER_VOLUME,
+ 5, 7, 0, out_mix_tlv),
+ SOC_SINGLE_TLV("Left Input Bypass Volume",
+ WM8350_OUTPUT_LEFT_MIXER_VOLUME,
+ 9, 7, 0, out_mix_tlv),
+ SOC_SINGLE_TLV("Right Input Left Sidetone Volume",
+ WM8350_OUTPUT_RIGHT_MIXER_VOLUME,
+ 1, 7, 0, out_mix_tlv),
+ SOC_SINGLE_TLV("Right Input Right Sidetone Volume",
+ WM8350_OUTPUT_RIGHT_MIXER_VOLUME,
+ 5, 7, 0, out_mix_tlv),
+ SOC_SINGLE_TLV("Right Input Bypass Volume",
+ WM8350_OUTPUT_RIGHT_MIXER_VOLUME,
+ 13, 7, 0, out_mix_tlv),
+ SOC_SINGLE("Left Input Mixer +20dB Switch",
+ WM8350_INPUT_MIXER_VOLUME_L, 0, 1, 0),
+ SOC_SINGLE("Right Input Mixer +20dB Switch",
+ WM8350_INPUT_MIXER_VOLUME_R, 0, 1, 0),
+ SOC_SINGLE_TLV("Out4 Capture Volume",
+ WM8350_INPUT_MIXER_VOLUME,
+ 1, 7, 0, out_mix_tlv),
+ SOC_WM8350_DOUBLE_R_TLV("Out1 Playback Volume",
+ WM8350_LOUT1_VOLUME,
+ WM8350_ROUT1_VOLUME,
+ 2, 63, 0, out_pga_tlv),
+ SOC_DOUBLE_R("Out1 Playback ZC Switch",
+ WM8350_LOUT1_VOLUME,
+ WM8350_ROUT1_VOLUME, 13, 1, 0),
+ SOC_WM8350_DOUBLE_R_TLV("Out2 Playback Volume",
+ WM8350_LOUT2_VOLUME,
+ WM8350_ROUT2_VOLUME,
+ 2, 63, 0, out_pga_tlv),
+ SOC_DOUBLE_R("Out2 Playback ZC Switch", WM8350_LOUT2_VOLUME,
+ WM8350_ROUT2_VOLUME, 13, 1, 0),
+ SOC_SINGLE("Out2 Right Invert Switch", WM8350_ROUT2_VOLUME, 10, 1, 0),
+ SOC_SINGLE_TLV("Out2 Beep Volume", WM8350_BEEP_VOLUME,
+ 5, 7, 0, out_mix_tlv),
+
+ SOC_DOUBLE_R("Out1 Playback Switch",
+ WM8350_LOUT1_VOLUME,
+ WM8350_ROUT1_VOLUME,
+ 14, 1, 1),
+ SOC_DOUBLE_R("Out2 Playback Switch",
+ WM8350_LOUT2_VOLUME,
+ WM8350_ROUT2_VOLUME,
+ 14, 1, 1),
+};
+
+/*
+ * DAPM Controls
+ */
+
+/* Left Playback Mixer */
+static const struct snd_kcontrol_new wm8350_left_play_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Playback Switch",
+ WM8350_LEFT_MIXER_CONTROL, 11, 1, 0),
+ SOC_DAPM_SINGLE("Left Bypass Switch",
+ WM8350_LEFT_MIXER_CONTROL, 2, 1, 0),
+ SOC_DAPM_SINGLE("Right Playback Switch",
+ WM8350_LEFT_MIXER_CONTROL, 12, 1, 0),
+ SOC_DAPM_SINGLE("Left Sidetone Switch",
+ WM8350_LEFT_MIXER_CONTROL, 0, 1, 0),
+ SOC_DAPM_SINGLE("Right Sidetone Switch",
+ WM8350_LEFT_MIXER_CONTROL, 1, 1, 0),
+};
+
+/* Right Playback Mixer */
+static const struct snd_kcontrol_new wm8350_right_play_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Playback Switch",
+ WM8350_RIGHT_MIXER_CONTROL, 12, 1, 0),
+ SOC_DAPM_SINGLE("Right Bypass Switch",
+ WM8350_RIGHT_MIXER_CONTROL, 3, 1, 0),
+ SOC_DAPM_SINGLE("Left Playback Switch",
+ WM8350_RIGHT_MIXER_CONTROL, 11, 1, 0),
+ SOC_DAPM_SINGLE("Left Sidetone Switch",
+ WM8350_RIGHT_MIXER_CONTROL, 0, 1, 0),
+ SOC_DAPM_SINGLE("Right Sidetone Switch",
+ WM8350_RIGHT_MIXER_CONTROL, 1, 1, 0),
+};
+
+/* Out4 Mixer */
+static const struct snd_kcontrol_new wm8350_out4_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Right Playback Switch",
+ WM8350_OUT4_MIXER_CONTROL, 12, 1, 0),
+ SOC_DAPM_SINGLE("Left Playback Switch",
+ WM8350_OUT4_MIXER_CONTROL, 11, 1, 0),
+ SOC_DAPM_SINGLE("Right Capture Switch",
+ WM8350_OUT4_MIXER_CONTROL, 9, 1, 0),
+ SOC_DAPM_SINGLE("Out3 Playback Switch",
+ WM8350_OUT4_MIXER_CONTROL, 2, 1, 0),
+ SOC_DAPM_SINGLE("Right Mixer Switch",
+ WM8350_OUT4_MIXER_CONTROL, 1, 1, 0),
+ SOC_DAPM_SINGLE("Left Mixer Switch",
+ WM8350_OUT4_MIXER_CONTROL, 0, 1, 0),
+};
+
+/* Out3 Mixer */
+static const struct snd_kcontrol_new wm8350_out3_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Left Playback Switch",
+ WM8350_OUT3_MIXER_CONTROL, 11, 1, 0),
+ SOC_DAPM_SINGLE("Left Capture Switch",
+ WM8350_OUT3_MIXER_CONTROL, 8, 1, 0),
+ SOC_DAPM_SINGLE("Out4 Playback Switch",
+ WM8350_OUT3_MIXER_CONTROL, 3, 1, 0),
+ SOC_DAPM_SINGLE("Left Mixer Switch",
+ WM8350_OUT3_MIXER_CONTROL, 0, 1, 0),
+};
+
+/* Left Input Mixer */
+static const struct snd_kcontrol_new wm8350_left_capt_mixer_controls[] = {
+ SOC_DAPM_SINGLE_TLV("L2 Capture Volume",
+ WM8350_INPUT_MIXER_VOLUME_L, 1, 7, 0, out_mix_tlv),
+ SOC_DAPM_SINGLE_TLV("L3 Capture Volume",
+ WM8350_INPUT_MIXER_VOLUME_L, 9, 7, 0, out_mix_tlv),
+ SOC_DAPM_SINGLE("PGA Capture Switch",
+ WM8350_LEFT_INPUT_VOLUME, 14, 1, 0),
+};
+
+/* Right Input Mixer */
+static const struct snd_kcontrol_new wm8350_right_capt_mixer_controls[] = {
+ SOC_DAPM_SINGLE_TLV("L2 Capture Volume",
+ WM8350_INPUT_MIXER_VOLUME_R, 5, 7, 0, out_mix_tlv),
+ SOC_DAPM_SINGLE_TLV("L3 Capture Volume",
+ WM8350_INPUT_MIXER_VOLUME_R, 13, 7, 0, out_mix_tlv),
+ SOC_DAPM_SINGLE("PGA Capture Switch",
+ WM8350_RIGHT_INPUT_VOLUME, 14, 1, 0),
+};
+
+/* Left Mic Mixer */
+static const struct snd_kcontrol_new wm8350_left_mic_mixer_controls[] = {
+ SOC_DAPM_SINGLE("INN Capture Switch", WM8350_INPUT_CONTROL, 1, 1, 0),
+ SOC_DAPM_SINGLE("INP Capture Switch", WM8350_INPUT_CONTROL, 0, 1, 0),
+ SOC_DAPM_SINGLE("IN2 Capture Switch", WM8350_INPUT_CONTROL, 2, 1, 0),
+};
+
+/* Right Mic Mixer */
+static const struct snd_kcontrol_new wm8350_right_mic_mixer_controls[] = {
+ SOC_DAPM_SINGLE("INN Capture Switch", WM8350_INPUT_CONTROL, 9, 1, 0),
+ SOC_DAPM_SINGLE("INP Capture Switch", WM8350_INPUT_CONTROL, 8, 1, 0),
+ SOC_DAPM_SINGLE("IN2 Capture Switch", WM8350_INPUT_CONTROL, 10, 1, 0),
+};
+
+/* Beep Switch */
+static const struct snd_kcontrol_new wm8350_beep_switch_controls =
+SOC_DAPM_SINGLE("Switch", WM8350_BEEP_VOLUME, 15, 1, 1);
+
+/* Out4 Capture Mux */
+static const struct snd_kcontrol_new wm8350_out4_capture_controls =
+SOC_DAPM_ENUM("Route", wm8350_enum[8]);
+
+static const struct snd_soc_dapm_widget wm8350_dapm_widgets[] = {
+
+ SND_SOC_DAPM_PGA("IN3R PGA", WM8350_POWER_MGMT_2, 11, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IN3L PGA", WM8350_POWER_MGMT_2, 10, 0, NULL, 0),
+ SND_SOC_DAPM_PGA_E("Right Out2 PGA", WM8350_POWER_MGMT_3, 3, 0, NULL,
+ 0, pga_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_PGA_E("Left Out2 PGA", WM8350_POWER_MGMT_3, 2, 0, NULL, 0,
+ pga_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_PGA_E("Right Out1 PGA", WM8350_POWER_MGMT_3, 1, 0, NULL,
+ 0, pga_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_PGA_E("Left Out1 PGA", WM8350_POWER_MGMT_3, 0, 0, NULL, 0,
+ pga_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+
+ SND_SOC_DAPM_MIXER("Right Capture Mixer", WM8350_POWER_MGMT_2,
+ 7, 0, &wm8350_right_capt_mixer_controls[0],
+ ARRAY_SIZE(wm8350_right_capt_mixer_controls)),
+
+ SND_SOC_DAPM_MIXER("Left Capture Mixer", WM8350_POWER_MGMT_2,
+ 6, 0, &wm8350_left_capt_mixer_controls[0],
+ ARRAY_SIZE(wm8350_left_capt_mixer_controls)),
+
+ SND_SOC_DAPM_MIXER("Out4 Mixer", WM8350_POWER_MGMT_2, 5, 0,
+ &wm8350_out4_mixer_controls[0],
+ ARRAY_SIZE(wm8350_out4_mixer_controls)),
+
+ SND_SOC_DAPM_MIXER("Out3 Mixer", WM8350_POWER_MGMT_2, 4, 0,
+ &wm8350_out3_mixer_controls[0],
+ ARRAY_SIZE(wm8350_out3_mixer_controls)),
+
+ SND_SOC_DAPM_MIXER("Right Playback Mixer", WM8350_POWER_MGMT_2, 1, 0,
+ &wm8350_right_play_mixer_controls[0],
+ ARRAY_SIZE(wm8350_right_play_mixer_controls)),
+
+ SND_SOC_DAPM_MIXER("Left Playback Mixer", WM8350_POWER_MGMT_2, 0, 0,
+ &wm8350_left_play_mixer_controls[0],
+ ARRAY_SIZE(wm8350_left_play_mixer_controls)),
+
+ SND_SOC_DAPM_MIXER("Left Mic Mixer", WM8350_POWER_MGMT_2, 8, 0,
+ &wm8350_left_mic_mixer_controls[0],
+ ARRAY_SIZE(wm8350_left_mic_mixer_controls)),
+
+ SND_SOC_DAPM_MIXER("Right Mic Mixer", WM8350_POWER_MGMT_2, 9, 0,
+ &wm8350_right_mic_mixer_controls[0],
+ ARRAY_SIZE(wm8350_right_mic_mixer_controls)),
+
+ /* virtual mixer for Beep and Out2R */
+ SND_SOC_DAPM_MIXER("Out2 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ SND_SOC_DAPM_SWITCH("Beep", WM8350_POWER_MGMT_3, 7, 0,
+ &wm8350_beep_switch_controls),
+
+ SND_SOC_DAPM_ADC("Right ADC", "Right Capture",
+ WM8350_POWER_MGMT_4, 3, 0),
+ SND_SOC_DAPM_ADC("Left ADC", "Left Capture",
+ WM8350_POWER_MGMT_4, 2, 0),
+ SND_SOC_DAPM_DAC("Right DAC", "Right Playback",
+ WM8350_POWER_MGMT_4, 5, 0),
+ SND_SOC_DAPM_DAC("Left DAC", "Left Playback",
+ WM8350_POWER_MGMT_4, 4, 0),
+
+ SND_SOC_DAPM_MICBIAS("Mic Bias", WM8350_POWER_MGMT_1, 4, 0),
+
+ SND_SOC_DAPM_MUX("Out4 Capture Channel", SND_SOC_NOPM, 0, 0,
+ &wm8350_out4_capture_controls),
+
+ SND_SOC_DAPM_OUTPUT("OUT1R"),
+ SND_SOC_DAPM_OUTPUT("OUT1L"),
+ SND_SOC_DAPM_OUTPUT("OUT2R"),
+ SND_SOC_DAPM_OUTPUT("OUT2L"),
+ SND_SOC_DAPM_OUTPUT("OUT3"),
+ SND_SOC_DAPM_OUTPUT("OUT4"),
+
+ SND_SOC_DAPM_INPUT("IN1RN"),
+ SND_SOC_DAPM_INPUT("IN1RP"),
+ SND_SOC_DAPM_INPUT("IN2R"),
+ SND_SOC_DAPM_INPUT("IN1LP"),
+ SND_SOC_DAPM_INPUT("IN1LN"),
+ SND_SOC_DAPM_INPUT("IN2L"),
+ SND_SOC_DAPM_INPUT("IN3R"),
+ SND_SOC_DAPM_INPUT("IN3L"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+
+ /* left playback mixer */
+ {"Left Playback Mixer", "Playback Switch", "Left DAC"},
+ {"Left Playback Mixer", "Left Bypass Switch", "IN3L PGA"},
+ {"Left Playback Mixer", "Right Playback Switch", "Right DAC"},
+ {"Left Playback Mixer", "Left Sidetone Switch", "Left Mic Mixer"},
+ {"Left Playback Mixer", "Right Sidetone Switch", "Right Mic Mixer"},
+
+ /* right playback mixer */
+ {"Right Playback Mixer", "Playback Switch", "Right DAC"},
+ {"Right Playback Mixer", "Right Bypass Switch", "IN3R PGA"},
+ {"Right Playback Mixer", "Left Playback Switch", "Left DAC"},
+ {"Right Playback Mixer", "Left Sidetone Switch", "Left Mic Mixer"},
+ {"Right Playback Mixer", "Right Sidetone Switch", "Right Mic Mixer"},
+
+ /* out4 playback mixer */
+ {"Out4 Mixer", "Right Playback Switch", "Right DAC"},
+ {"Out4 Mixer", "Left Playback Switch", "Left DAC"},
+ {"Out4 Mixer", "Right Capture Switch", "Right Capture Mixer"},
+ {"Out4 Mixer", "Out3 Playback Switch", "Out3 Mixer"},
+ {"Out4 Mixer", "Right Mixer Switch", "Right Playback Mixer"},
+ {"Out4 Mixer", "Left Mixer Switch", "Left Playback Mixer"},
+ {"OUT4", NULL, "Out4 Mixer"},
+
+ /* out3 playback mixer */
+ {"Out3 Mixer", "Left Playback Switch", "Left DAC"},
+ {"Out3 Mixer", "Left Capture Switch", "Left Capture Mixer"},
+ {"Out3 Mixer", "Left Mixer Switch", "Left Playback Mixer"},
+ {"Out3 Mixer", "Out4 Playback Switch", "Out4 Mixer"},
+ {"OUT3", NULL, "Out3 Mixer"},
+
+ /* out2 */
+ {"Right Out2 PGA", NULL, "Right Playback Mixer"},
+ {"Left Out2 PGA", NULL, "Left Playback Mixer"},
+ {"OUT2L", NULL, "Left Out2 PGA"},
+ {"OUT2R", NULL, "Right Out2 PGA"},
+
+ /* out1 */
+ {"Right Out1 PGA", NULL, "Right Playback Mixer"},
+ {"Left Out1 PGA", NULL, "Left Playback Mixer"},
+ {"OUT1L", NULL, "Left Out1 PGA"},
+ {"OUT1R", NULL, "Right Out1 PGA"},
+
+ /* ADCs */
+ {"Left ADC", NULL, "Left Capture Mixer"},
+ {"Right ADC", NULL, "Right Capture Mixer"},
+
+ /* Left capture mixer */
+ {"Left Capture Mixer", "L2 Capture Volume", "IN2L"},
+ {"Left Capture Mixer", "L3 Capture Volume", "IN3L PGA"},
+ {"Left Capture Mixer", "PGA Capture Switch", "Left Mic Mixer"},
+ {"Left Capture Mixer", NULL, "Out4 Capture Channel"},
+
+ /* Right capture mixer */
+ {"Right Capture Mixer", "L2 Capture Volume", "IN2R"},
+ {"Right Capture Mixer", "L3 Capture Volume", "IN3R PGA"},
+ {"Right Capture Mixer", "PGA Capture Switch", "Right Mic Mixer"},
+ {"Right Capture Mixer", NULL, "Out4 Capture Channel"},
+
+ /* L3 Inputs */
+ {"IN3L PGA", NULL, "IN3L"},
+ {"IN3R PGA", NULL, "IN3R"},
+
+ /* Left Mic mixer */
+ {"Left Mic Mixer", "INN Capture Switch", "IN1LN"},
+ {"Left Mic Mixer", "INP Capture Switch", "IN1LP"},
+ {"Left Mic Mixer", "IN2 Capture Switch", "IN2L"},
+
+ /* Right Mic mixer */
+ {"Right Mic Mixer", "INN Capture Switch", "IN1RN"},
+ {"Right Mic Mixer", "INP Capture Switch", "IN1RP"},
+ {"Right Mic Mixer", "IN2 Capture Switch", "IN2R"},
+
+ /* out 4 capture */
+ {"Out4 Capture Channel", NULL, "Out4 Mixer"},
+
+ /* Beep */
+ {"Beep", NULL, "IN3R PGA"},
+};
+
+static int wm8350_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm8350_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8350_snd_controls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+static int wm8350_add_widgets(struct snd_soc_codec *codec)
+{
+ int ret;
+
+ ret = snd_soc_dapm_new_controls(codec,
+ wm8350_dapm_widgets,
+ ARRAY_SIZE(wm8350_dapm_widgets));
+ if (ret != 0) {
+ dev_err(codec->dev, "dapm control register failed\n");
+ return ret;
+ }
+
+ /* set up audio paths */
+ ret = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ if (ret != 0) {
+ dev_err(codec->dev, "DAPM route register failed\n");
+ return ret;
+ }
+
+ return snd_soc_dapm_new_widgets(codec);
+}
+
+static int wm8350_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct wm8350 *wm8350 = codec->control_data;
+ u16 fll_4;
+
+ switch (clk_id) {
+ case WM8350_MCLK_SEL_MCLK:
+ wm8350_clear_bits(wm8350, WM8350_CLOCK_CONTROL_1,
+ WM8350_MCLK_SEL);
+ break;
+ case WM8350_MCLK_SEL_PLL_MCLK:
+ case WM8350_MCLK_SEL_PLL_DAC:
+ case WM8350_MCLK_SEL_PLL_ADC:
+ case WM8350_MCLK_SEL_PLL_32K:
+ wm8350_set_bits(wm8350, WM8350_CLOCK_CONTROL_1,
+ WM8350_MCLK_SEL);
+ fll_4 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_4) &
+ ~WM8350_FLL_CLK_SRC_MASK;
+ wm8350_codec_write(codec, WM8350_FLL_CONTROL_4, fll_4 | clk_id);
+ break;
+ }
+
+ /* MCLK direction */
+ if (dir == WM8350_MCLK_DIR_OUT)
+ wm8350_set_bits(wm8350, WM8350_CLOCK_CONTROL_2,
+ WM8350_MCLK_DIR);
+ else
+ wm8350_clear_bits(wm8350, WM8350_CLOCK_CONTROL_2,
+ WM8350_MCLK_DIR);
+
+ return 0;
+}
+
+static int wm8350_set_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 val;
+
+ switch (div_id) {
+ case WM8350_ADC_CLKDIV:
+ val = wm8350_codec_read(codec, WM8350_ADC_DIVIDER) &
+ ~WM8350_ADC_CLKDIV_MASK;
+ wm8350_codec_write(codec, WM8350_ADC_DIVIDER, val | div);
+ break;
+ case WM8350_DAC_CLKDIV:
+ val = wm8350_codec_read(codec, WM8350_DAC_CLOCK_CONTROL) &
+ ~WM8350_DAC_CLKDIV_MASK;
+ wm8350_codec_write(codec, WM8350_DAC_CLOCK_CONTROL, val | div);
+ break;
+ case WM8350_BCLK_CLKDIV:
+ val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) &
+ ~WM8350_BCLK_DIV_MASK;
+ wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
+ break;
+ case WM8350_OPCLK_CLKDIV:
+ val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) &
+ ~WM8350_OPCLK_DIV_MASK;
+ wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
+ break;
+ case WM8350_SYS_CLKDIV:
+ val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) &
+ ~WM8350_MCLK_DIV_MASK;
+ wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
+ break;
+ case WM8350_DACLR_CLKDIV:
+ val = wm8350_codec_read(codec, WM8350_DAC_LR_RATE) &
+ ~WM8350_DACLRC_RATE_MASK;
+ wm8350_codec_write(codec, WM8350_DAC_LR_RATE, val | div);
+ break;
+ case WM8350_ADCLR_CLKDIV:
+ val = wm8350_codec_read(codec, WM8350_ADC_LR_RATE) &
+ ~WM8350_ADCLRC_RATE_MASK;
+ wm8350_codec_write(codec, WM8350_ADC_LR_RATE, val | div);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) &
+ ~(WM8350_AIF_BCLK_INV | WM8350_AIF_LRCLK_INV | WM8350_AIF_FMT_MASK);
+ u16 master = wm8350_codec_read(codec, WM8350_AI_DAC_CONTROL) &
+ ~WM8350_BCLK_MSTR;
+ u16 dac_lrc = wm8350_codec_read(codec, WM8350_DAC_LR_RATE) &
+ ~WM8350_DACLRC_ENA;
+ u16 adc_lrc = wm8350_codec_read(codec, WM8350_ADC_LR_RATE) &
+ ~WM8350_ADCLRC_ENA;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ master |= WM8350_BCLK_MSTR;
+ dac_lrc |= WM8350_DACLRC_ENA;
+ adc_lrc |= WM8350_ADCLRC_ENA;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= 0x2 << 8;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= 0x1 << 8;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface |= 0x3 << 8;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ iface |= 0x3 << 8; /* lg not sure which mode */
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface |= WM8350_AIF_LRCLK_INV | WM8350_AIF_BCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= WM8350_AIF_BCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ iface |= WM8350_AIF_LRCLK_INV;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm8350_codec_write(codec, WM8350_AI_FORMATING, iface);
+ wm8350_codec_write(codec, WM8350_AI_DAC_CONTROL, master);
+ wm8350_codec_write(codec, WM8350_DAC_LR_RATE, dac_lrc);
+ wm8350_codec_write(codec, WM8350_ADC_LR_RATE, adc_lrc);
+ return 0;
+}
+
+static int wm8350_pcm_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *codec_dai)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ int master = wm8350_codec_cache_read(codec, WM8350_AI_DAC_CONTROL) &
+ WM8350_BCLK_MSTR;
+ int enabled = 0;
+
+ /* Check that the DACs or ADCs are enabled since they are
+ * required for LRC in master mode. The DACs or ADCs need a
+ * valid audio path i.e. pin -> ADC or DAC -> pin before
+ * the LRC will be enabled in master mode. */
+ if (!master && cmd != SNDRV_PCM_TRIGGER_START)
+ return 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ enabled = wm8350_codec_cache_read(codec, WM8350_POWER_MGMT_4) &
+ (WM8350_ADCR_ENA | WM8350_ADCL_ENA);
+ } else {
+ enabled = wm8350_codec_cache_read(codec, WM8350_POWER_MGMT_4) &
+ (WM8350_DACR_ENA | WM8350_DACL_ENA);
+ }
+
+ if (!enabled) {
+ dev_err(codec->dev,
+ "%s: invalid audio path - no clocks available\n",
+ __func__);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *codec_dai)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) &
+ ~WM8350_AIF_WL_MASK;
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ iface |= 0x1 << 10;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iface |= 0x2 << 10;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ iface |= 0x3 << 10;
+ break;
+ }
+
+ wm8350_codec_write(codec, WM8350_AI_FORMATING, iface);
+ return 0;
+}
+
+static int wm8350_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct wm8350 *wm8350 = codec->control_data;
+
+ if (mute)
+ wm8350_set_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA);
+ else
+ wm8350_clear_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA);
+ return 0;
+}
+
+/* FLL divisors */
+struct _fll_div {
+ int div; /* FLL_OUTDIV */
+ int n;
+ int k;
+ int ratio; /* FLL_FRATIO */
+};
+
+/* The size in bits of the fll divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_FLL_SIZE ((1 << 16) * 10)
+
+static inline int fll_factors(struct _fll_div *fll_div, unsigned int input,
+ unsigned int output)
+{
+ u64 Kpart;
+ unsigned int t1, t2, K, Nmod;
+
+ if (output >= 2815250 && output <= 3125000)
+ fll_div->div = 0x4;
+ else if (output >= 5625000 && output <= 6250000)
+ fll_div->div = 0x3;
+ else if (output >= 11250000 && output <= 12500000)
+ fll_div->div = 0x2;
+ else if (output >= 22500000 && output <= 25000000)
+ fll_div->div = 0x1;
+ else {
+ printk(KERN_ERR "wm8350: fll freq %d out of range\n", output);
+ return -EINVAL;
+ }
+
+ if (input > 48000)
+ fll_div->ratio = 1;
+ else
+ fll_div->ratio = 8;
+
+ t1 = output * (1 << (fll_div->div + 1));
+ t2 = input * fll_div->ratio;
+
+ fll_div->n = t1 / t2;
+ Nmod = t1 % t2;
+
+ if (Nmod) {
+ Kpart = FIXED_FLL_SIZE * (long long)Nmod;
+ do_div(Kpart, t2);
+ K = Kpart & 0xFFFFFFFF;
+
+ /* Check if we need to round */
+ if ((K % 10) >= 5)
+ K += 5;
+
+ /* Move down to proper range now rounding is done */
+ K /= 10;
+ fll_div->k = K;
+ } else
+ fll_div->k = 0;
+
+ return 0;
+}
+
+static int wm8350_set_fll(struct snd_soc_dai *codec_dai,
+ int pll_id, unsigned int freq_in,
+ unsigned int freq_out)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct wm8350 *wm8350 = codec->control_data;
+ struct _fll_div fll_div;
+ int ret = 0;
+ u16 fll_1, fll_4;
+
+ /* power down FLL - we need to do this for reconfiguration */
+ wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4,
+ WM8350_FLL_ENA | WM8350_FLL_OSC_ENA);
+
+ if (freq_out == 0 || freq_in == 0)
+ return ret;
+
+ ret = fll_factors(&fll_div, freq_in, freq_out);
+ if (ret < 0)
+ return ret;
+ dev_dbg(wm8350->dev,
+ "FLL in %d FLL out %d N 0x%x K 0x%x div %d ratio %d",
+ freq_in, freq_out, fll_div.n, fll_div.k, fll_div.div,
+ fll_div.ratio);
+
+ /* set up N.K & dividers */
+ fll_1 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_1) &
+ ~(WM8350_FLL_OUTDIV_MASK | WM8350_FLL_RSP_RATE_MASK | 0xc000);
+ wm8350_codec_write(codec, WM8350_FLL_CONTROL_1,
+ fll_1 | (fll_div.div << 8) | 0x50);
+ wm8350_codec_write(codec, WM8350_FLL_CONTROL_2,
+ (fll_div.ratio << 11) | (fll_div.
+ n & WM8350_FLL_N_MASK));
+ wm8350_codec_write(codec, WM8350_FLL_CONTROL_3, fll_div.k);
+ fll_4 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_4) &
+ ~(WM8350_FLL_FRAC | WM8350_FLL_SLOW_LOCK_REF);
+ wm8350_codec_write(codec, WM8350_FLL_CONTROL_4,
+ fll_4 | (fll_div.k ? WM8350_FLL_FRAC : 0) |
+ (fll_div.ratio == 8 ? WM8350_FLL_SLOW_LOCK_REF : 0));
+
+ /* power FLL on */
+ wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_FLL_OSC_ENA);
+ wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_FLL_ENA);
+
+ return 0;
+}
+
+static int wm8350_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct wm8350 *wm8350 = codec->control_data;
+ struct wm8350_data *priv = codec->private_data;
+ struct wm8350_audio_platform_data *platform =
+ wm8350->codec.platform_data;
+ u16 pm1;
+ int ret;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
+ ~(WM8350_VMID_MASK | WM8350_CODEC_ISEL_MASK);
+ wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+ pm1 | WM8350_VMID_50K |
+ platform->codec_current_on << 14);
+ break;
+
+ case SND_SOC_BIAS_PREPARE:
+ pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1);
+ pm1 &= ~WM8350_VMID_MASK;
+ wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+ pm1 | WM8350_VMID_50K);
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies),
+ priv->supplies);
+ if (ret != 0)
+ return ret;
+
+ /* Enable the system clock */
+ wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4,
+ WM8350_SYSCLK_ENA);
+
+ /* mute DAC & outputs */
+ wm8350_set_bits(wm8350, WM8350_DAC_MUTE,
+ WM8350_DAC_MUTE_ENA);
+
+ /* discharge cap memory */
+ wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL,
+ platform->dis_out1 |
+ (platform->dis_out2 << 2) |
+ (platform->dis_out3 << 4) |
+ (platform->dis_out4 << 6));
+
+ /* wait for discharge */
+ schedule_timeout_interruptible(msecs_to_jiffies
+ (platform->
+ cap_discharge_msecs));
+
+ /* enable antipop */
+ wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL,
+ (platform->vmid_s_curve << 8));
+
+ /* ramp up vmid */
+ wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+ (platform->
+ codec_current_charge << 14) |
+ WM8350_VMID_5K | WM8350_VMIDEN |
+ WM8350_VBUFEN);
+
+ /* wait for vmid */
+ schedule_timeout_interruptible(msecs_to_jiffies
+ (platform->
+ vmid_charge_msecs));
+
+ /* turn on vmid 300k */
+ pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
+ ~(WM8350_VMID_MASK | WM8350_CODEC_ISEL_MASK);
+ pm1 |= WM8350_VMID_300K |
+ (platform->codec_current_standby << 14);
+ wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+ pm1);
+
+
+ /* enable analogue bias */
+ pm1 |= WM8350_BIASEN;
+ wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, pm1);
+
+ /* disable antipop */
+ wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL, 0);
+
+ } else {
+ /* turn on vmid 300k and reduce current */
+ pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
+ ~(WM8350_VMID_MASK | WM8350_CODEC_ISEL_MASK);
+ wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+ pm1 | WM8350_VMID_300K |
+ (platform->
+ codec_current_standby << 14));
+
+ }
+ break;
+
+ case SND_SOC_BIAS_OFF:
+
+ /* mute DAC & enable outputs */
+ wm8350_set_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA);
+
+ wm8350_set_bits(wm8350, WM8350_POWER_MGMT_3,
+ WM8350_OUT1L_ENA | WM8350_OUT1R_ENA |
+ WM8350_OUT2L_ENA | WM8350_OUT2R_ENA);
+
+ /* enable anti pop S curve */
+ wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL,
+ (platform->vmid_s_curve << 8));
+
+ /* turn off vmid */
+ pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
+ ~WM8350_VMIDEN;
+ wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, pm1);
+
+ /* wait */
+ schedule_timeout_interruptible(msecs_to_jiffies
+ (platform->
+ vmid_discharge_msecs));
+
+ wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL,
+ (platform->vmid_s_curve << 8) |
+ platform->dis_out1 |
+ (platform->dis_out2 << 2) |
+ (platform->dis_out3 << 4) |
+ (platform->dis_out4 << 6));
+
+ /* turn off VBuf and drain */
+ pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
+ ~(WM8350_VBUFEN | WM8350_VMID_MASK);
+ wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+ pm1 | WM8350_OUTPUT_DRAIN_EN);
+
+ /* wait */
+ schedule_timeout_interruptible(msecs_to_jiffies
+ (platform->drain_msecs));
+
+ pm1 &= ~WM8350_BIASEN;
+ wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, pm1);
+
+ /* disable anti-pop */
+ wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL, 0);
+
+ wm8350_clear_bits(wm8350, WM8350_LOUT1_VOLUME,
+ WM8350_OUT1L_ENA);
+ wm8350_clear_bits(wm8350, WM8350_ROUT1_VOLUME,
+ WM8350_OUT1R_ENA);
+ wm8350_clear_bits(wm8350, WM8350_LOUT2_VOLUME,
+ WM8350_OUT2L_ENA);
+ wm8350_clear_bits(wm8350, WM8350_ROUT2_VOLUME,
+ WM8350_OUT2R_ENA);
+
+ /* disable clock gen */
+ wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4,
+ WM8350_SYSCLK_ENA);
+
+ regulator_bulk_disable(ARRAY_SIZE(priv->supplies),
+ priv->supplies);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+static int wm8350_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int wm8350_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
+ wm8350_set_bias_level(codec, SND_SOC_BIAS_ON);
+
+ return 0;
+}
+
+static struct snd_soc_codec *wm8350_codec;
+
+static int wm8350_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ struct wm8350 *wm8350;
+ struct wm8350_data *priv;
+ int ret;
+ struct wm8350_output *out1;
+ struct wm8350_output *out2;
+
+ BUG_ON(!wm8350_codec);
+
+ socdev->codec = wm8350_codec;
+ codec = socdev->codec;
+ wm8350 = codec->control_data;
+ priv = codec->private_data;
+
+ /* Enable the codec */
+ wm8350_set_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
+
+ /* Enable robust clocking mode in ADC */
+ wm8350_codec_write(codec, WM8350_SECURITY, 0xa7);
+ wm8350_codec_write(codec, 0xde, 0x13);
+ wm8350_codec_write(codec, WM8350_SECURITY, 0);
+
+ /* read OUT1 & OUT2 volumes */
+ out1 = &priv->out1;
+ out2 = &priv->out2;
+ out1->left_vol = (wm8350_reg_read(wm8350, WM8350_LOUT1_VOLUME) &
+ WM8350_OUT1L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT;
+ out1->right_vol = (wm8350_reg_read(wm8350, WM8350_ROUT1_VOLUME) &
+ WM8350_OUT1R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT;
+ out2->left_vol = (wm8350_reg_read(wm8350, WM8350_LOUT2_VOLUME) &
+ WM8350_OUT2L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT;
+ out2->right_vol = (wm8350_reg_read(wm8350, WM8350_ROUT2_VOLUME) &
+ WM8350_OUT2R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT;
+ wm8350_reg_write(wm8350, WM8350_LOUT1_VOLUME, 0);
+ wm8350_reg_write(wm8350, WM8350_ROUT1_VOLUME, 0);
+ wm8350_reg_write(wm8350, WM8350_LOUT2_VOLUME, 0);
+ wm8350_reg_write(wm8350, WM8350_ROUT2_VOLUME, 0);
+
+ /* Latch VU bits & mute */
+ wm8350_set_bits(wm8350, WM8350_LOUT1_VOLUME,
+ WM8350_OUT1_VU | WM8350_OUT1L_MUTE);
+ wm8350_set_bits(wm8350, WM8350_LOUT2_VOLUME,
+ WM8350_OUT2_VU | WM8350_OUT2L_MUTE);
+ wm8350_set_bits(wm8350, WM8350_ROUT1_VOLUME,
+ WM8350_OUT1_VU | WM8350_OUT1R_MUTE);
+ wm8350_set_bits(wm8350, WM8350_ROUT2_VOLUME,
+ WM8350_OUT2_VU | WM8350_OUT2R_MUTE);
+
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(&pdev->dev, "failed to create pcms\n");
+ return ret;
+ }
+
+ wm8350_add_controls(codec);
+ wm8350_add_widgets(codec);
+
+ wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ dev_err(&pdev->dev, "failed to register card\n");
+ goto card_err;
+ }
+
+ return 0;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+ return ret;
+}
+
+static int wm8350_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ struct wm8350 *wm8350 = codec->control_data;
+ int ret;
+
+ /* cancel any work waiting to be queued. */
+ ret = cancel_delayed_work(&codec->delayed_work);
+
+ /* if there was any work waiting then we run it now and
+ * wait for its completion */
+ if (ret) {
+ schedule_delayed_work(&codec->delayed_work, 0);
+ flush_scheduled_work();
+ }
+
+ wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
+
+ return 0;
+}
+
+#define WM8350_RATES (SNDRV_PCM_RATE_8000_96000)
+
+#define WM8350_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+struct snd_soc_dai wm8350_dai = {
+ .name = "WM8350",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8350_RATES,
+ .formats = WM8350_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8350_RATES,
+ .formats = WM8350_FORMATS,
+ },
+ .ops = {
+ .hw_params = wm8350_pcm_hw_params,
+ .digital_mute = wm8350_mute,
+ .trigger = wm8350_pcm_trigger,
+ .set_fmt = wm8350_set_dai_fmt,
+ .set_sysclk = wm8350_set_dai_sysclk,
+ .set_pll = wm8350_set_fll,
+ .set_clkdiv = wm8350_set_clkdiv,
+ },
+};
+EXPORT_SYMBOL_GPL(wm8350_dai);
+
+struct snd_soc_codec_device soc_codec_dev_wm8350 = {
+ .probe = wm8350_probe,
+ .remove = wm8350_remove,
+ .suspend = wm8350_suspend,
+ .resume = wm8350_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8350);
+
+static int wm8350_codec_probe(struct platform_device *pdev)
+{
+ struct wm8350 *wm8350 = platform_get_drvdata(pdev);
+ struct wm8350_data *priv;
+ struct snd_soc_codec *codec;
+ int ret, i;
+
+ if (wm8350->codec.platform_data == NULL) {
+ dev_err(&pdev->dev, "No audio platform data supplied\n");
+ return -EINVAL;
+ }
+
+ priv = kzalloc(sizeof(struct wm8350_data), GFP_KERNEL);
+ if (priv == NULL)
+ return -ENOMEM;
+
+ for (i = 0; i < ARRAY_SIZE(supply_names); i++)
+ priv->supplies[i].supply = supply_names[i];
+
+ ret = regulator_bulk_get(wm8350->dev, ARRAY_SIZE(priv->supplies),
+ priv->supplies);
+ if (ret != 0)
+ goto err_priv;
+
+ codec = &priv->codec;
+ wm8350->codec.codec = codec;
+
+ wm8350_dai.dev = &pdev->dev;
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+ codec->dev = &pdev->dev;
+ codec->name = "WM8350";
+ codec->owner = THIS_MODULE;
+ codec->read = wm8350_codec_read;
+ codec->write = wm8350_codec_write;
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ codec->set_bias_level = wm8350_set_bias_level;
+ codec->dai = &wm8350_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = WM8350_MAX_REGISTER;
+ codec->private_data = priv;
+ codec->control_data = wm8350;
+
+ /* Put the codec into reset if it wasn't already */
+ wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
+
+ INIT_DELAYED_WORK(&codec->delayed_work, wm8350_pga_work);
+ ret = snd_soc_register_codec(codec);
+ if (ret != 0)
+ goto err_supply;
+
+ wm8350_codec = codec;
+
+ ret = snd_soc_register_dai(&wm8350_dai);
+ if (ret != 0)
+ goto err_codec;
+ return 0;
+
+err_codec:
+ snd_soc_unregister_codec(codec);
+err_supply:
+ regulator_bulk_free(ARRAY_SIZE(priv->supplies), priv->supplies);
+err_priv:
+ kfree(priv);
+ wm8350_codec = NULL;
+ return ret;
+}
+
+static int __devexit wm8350_codec_remove(struct platform_device *pdev)
+{
+ struct wm8350 *wm8350 = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = wm8350->codec.codec;
+ struct wm8350_data *priv = codec->private_data;
+
+ snd_soc_unregister_dai(&wm8350_dai);
+ snd_soc_unregister_codec(codec);
+ regulator_bulk_free(ARRAY_SIZE(priv->supplies), priv->supplies);
+ kfree(priv);
+ wm8350_codec = NULL;
+ return 0;
+}
+
+static struct platform_driver wm8350_codec_driver = {
+ .driver = {
+ .name = "wm8350-codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8350_codec_probe,
+ .remove = __devexit_p(wm8350_codec_remove),
+};
+
+static __init int wm8350_init(void)
+{
+ return platform_driver_register(&wm8350_codec_driver);
+}
+module_init(wm8350_init);
+
+static __exit void wm8350_exit(void)
+{
+ platform_driver_unregister(&wm8350_codec_driver);
+}
+module_exit(wm8350_exit);
+
+MODULE_DESCRIPTION("ASoC WM8350 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:wm8350-codec");
diff --git a/sound/soc/codecs/wm8350.h b/sound/soc/codecs/wm8350.h
new file mode 100644
index 0000000..cc2887a
--- /dev/null
+++ b/sound/soc/codecs/wm8350.h
@@ -0,0 +1,20 @@
+/*
+ * wm8350.h - WM8903 audio codec interface
+ *
+ * Copyright 2008 Wolfson Microelectronics PLC.
+ *
+ * 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 _WM8350_H
+#define _WM8350_H
+
+#include <sound/soc.h>
+
+extern struct snd_soc_dai wm8350_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8350;
+
+#endif
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index d8ca2da..40f8238 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -463,7 +463,8 @@
}
static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -585,8 +586,6 @@
.formats = WM8510_FORMATS,},
.ops = {
.hw_params = wm8510_pcm_hw_params,
- },
- .dai_ops = {
.digital_mute = wm8510_mute,
.set_fmt = wm8510_set_dai_fmt,
.set_clkdiv = wm8510_set_dai_clkdiv,
@@ -659,7 +658,7 @@
wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
wm8510_add_controls(codec);
wm8510_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "wm8510: failed to register card\n");
goto card_err;
@@ -890,6 +889,18 @@
};
EXPORT_SYMBOL_GPL(soc_codec_dev_wm8510);
+static int __init wm8510_modinit(void)
+{
+ return snd_soc_register_dai(&wm8510_dai);
+}
+module_init(wm8510_modinit);
+
+static void __exit wm8510_exit(void)
+{
+ snd_soc_unregister_dai(&wm8510_dai);
+}
+module_exit(wm8510_exit);
+
MODULE_DESCRIPTION("ASoC WM8510 driver");
MODULE_AUTHOR("Liam Girdwood");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 627ebfb..d004e58 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -548,13 +548,13 @@
* Set PCM DAI bit size and sample rate.
*/
static int wm8580_paif_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai_link *dai = rtd->dai;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
- u16 paifb = wm8580_read(codec, WM8580_PAIF3 + dai->codec_dai->id);
+ u16 paifb = wm8580_read(codec, WM8580_PAIF3 + dai->id);
paifb &= ~WM8580_AIF_LENGTH_MASK;
/* bit size */
@@ -574,7 +574,7 @@
return -EINVAL;
}
- wm8580_write(codec, WM8580_PAIF3 + dai->codec_dai->id, paifb);
+ wm8580_write(codec, WM8580_PAIF3 + dai->id, paifb);
return 0;
}
@@ -798,8 +798,6 @@
},
.ops = {
.hw_params = wm8580_paif_hw_params,
- },
- .dai_ops = {
.set_fmt = wm8580_set_paif_dai_fmt,
.set_clkdiv = wm8580_set_dai_clkdiv,
.set_pll = wm8580_set_dai_pll,
@@ -818,8 +816,6 @@
},
.ops = {
.hw_params = wm8580_paif_hw_params,
- },
- .dai_ops = {
.set_fmt = wm8580_set_paif_dai_fmt,
.set_clkdiv = wm8580_set_dai_clkdiv,
.set_pll = wm8580_set_dai_pll,
@@ -873,7 +869,7 @@
wm8580_add_controls(codec);
wm8580_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "wm8580: failed to register card\n");
goto card_err;
@@ -900,85 +896,85 @@
* low = 0x1a
* high = 0x1b
*/
-static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
-/* Magic definition of all other variables and things */
-I2C_CLIENT_INSMOD;
-
-static struct i2c_driver wm8580_i2c_driver;
-static struct i2c_client client_template;
-
-static int wm8580_codec_probe(struct i2c_adapter *adap, int addr, int kind)
+static int wm8580_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
{
struct snd_soc_device *socdev = wm8580_socdev;
- struct wm8580_setup_data *setup = socdev->codec_data;
struct snd_soc_codec *codec = socdev->codec;
- struct i2c_client *i2c;
int ret;
- if (addr != setup->i2c_address)
- return -ENODEV;
-
- client_template.adapter = adap;
- client_template.addr = addr;
-
- i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
- if (i2c == NULL) {
- kfree(codec);
- return -ENOMEM;
- }
i2c_set_clientdata(i2c, codec);
codec->control_data = i2c;
- ret = i2c_attach_client(i2c);
- if (ret < 0) {
- dev_err(&i2c->dev, "failed to attach codec at addr %x\n", addr);
- goto err;
- }
-
ret = wm8580_init(socdev);
- if (ret < 0) {
+ if (ret < 0)
dev_err(&i2c->dev, "failed to initialise WM8580\n");
- goto err;
- }
-
- return ret;
-
-err:
- kfree(codec);
- kfree(i2c);
return ret;
}
-static int wm8580_i2c_detach(struct i2c_client *client)
+static int wm8580_i2c_remove(struct i2c_client *client)
{
struct snd_soc_codec *codec = i2c_get_clientdata(client);
- i2c_detach_client(client);
kfree(codec->reg_cache);
- kfree(client);
return 0;
}
-static int wm8580_i2c_attach(struct i2c_adapter *adap)
-{
- return i2c_probe(adap, &addr_data, wm8580_codec_probe);
-}
+static const struct i2c_device_id wm8580_i2c_id[] = {
+ { "wm8580", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8580_i2c_id);
-/* corgi i2c codec control layer */
static struct i2c_driver wm8580_i2c_driver = {
.driver = {
.name = "WM8580 I2C Codec",
.owner = THIS_MODULE,
},
- .attach_adapter = wm8580_i2c_attach,
- .detach_client = wm8580_i2c_detach,
- .command = NULL,
+ .probe = wm8580_i2c_probe,
+ .remove = wm8580_i2c_remove,
+ .id_table = wm8580_i2c_id,
};
-static struct i2c_client client_template = {
- .name = "WM8580",
- .driver = &wm8580_i2c_driver,
-};
+static int wm8580_add_i2c_device(struct platform_device *pdev,
+ const struct wm8580_setup_data *setup)
+{
+ struct i2c_board_info info;
+ struct i2c_adapter *adapter;
+ struct i2c_client *client;
+ int ret;
+
+ ret = i2c_add_driver(&wm8580_i2c_driver);
+ if (ret != 0) {
+ dev_err(&pdev->dev, "can't add i2c driver\n");
+ return ret;
+ }
+
+ memset(&info, 0, sizeof(struct i2c_board_info));
+ info.addr = setup->i2c_address;
+ strlcpy(info.type, "wm8580", I2C_NAME_SIZE);
+
+ adapter = i2c_get_adapter(setup->i2c_bus);
+ if (!adapter) {
+ dev_err(&pdev->dev, "can't get i2c adapter %d\n",
+ setup->i2c_bus);
+ goto err_driver;
+ }
+
+ client = i2c_new_device(adapter, &info);
+ i2c_put_adapter(adapter);
+ if (!client) {
+ dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
+ (unsigned int)info.addr);
+ goto err_driver;
+ }
+
+ return 0;
+
+err_driver:
+ i2c_del_driver(&wm8580_i2c_driver);
+ return -ENODEV;
+}
#endif
static int wm8580_probe(struct platform_device *pdev)
@@ -1011,11 +1007,8 @@
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
if (setup->i2c_address) {
- normal_i2c[0] = setup->i2c_address;
codec->hw_write = (hw_write_t)i2c_master_send;
- ret = i2c_add_driver(&wm8580_i2c_driver);
- if (ret != 0)
- printk(KERN_ERR "can't add i2c driver");
+ ret = wm8580_add_i2c_device(pdev, setup);
}
#else
/* Add other interfaces here */
@@ -1034,6 +1027,7 @@
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_unregister_device(codec->control_data);
i2c_del_driver(&wm8580_i2c_driver);
#endif
kfree(codec->private_data);
@@ -1048,6 +1042,18 @@
};
EXPORT_SYMBOL_GPL(soc_codec_dev_wm8580);
+static int __init wm8580_modinit(void)
+{
+ return snd_soc_register_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai));
+}
+module_init(wm8580_modinit);
+
+static void __exit wm8580_exit(void)
+{
+ snd_soc_unregister_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai));
+}
+module_exit(wm8580_exit);
+
MODULE_DESCRIPTION("ASoC WM8580 driver");
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8580.h b/sound/soc/codecs/wm8580.h
index 589ddab..09e4422 100644
--- a/sound/soc/codecs/wm8580.h
+++ b/sound/soc/codecs/wm8580.h
@@ -29,6 +29,7 @@
#define WM8580_CLKSRC_NONE 5
struct wm8580_setup_data {
+ int i2c_bus;
unsigned short i2c_address;
};
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
new file mode 100644
index 0000000..80b1198
--- /dev/null
+++ b/sound/soc/codecs/wm8728.c
@@ -0,0 +1,585 @@
+/*
+ * wm8728.c -- WM8728 ALSA SoC Audio driver
+ *
+ * Copyright 2008 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@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/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/spi/spi.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "wm8728.h"
+
+struct snd_soc_codec_device soc_codec_dev_wm8728;
+
+/*
+ * We can't read the WM8728 register space so we cache them instead.
+ * Note that the defaults here aren't the physical defaults, we latch
+ * the volume update bits, mute the output and enable infinite zero
+ * detect.
+ */
+static const u16 wm8728_reg_defaults[] = {
+ 0x1ff,
+ 0x1ff,
+ 0x001,
+ 0x100,
+};
+
+static inline unsigned int wm8728_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ BUG_ON(reg > ARRAY_SIZE(wm8728_reg_defaults));
+ return cache[reg];
+}
+
+static inline void wm8728_write_reg_cache(struct snd_soc_codec *codec,
+ u16 reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ BUG_ON(reg > ARRAY_SIZE(wm8728_reg_defaults));
+ cache[reg] = value;
+}
+
+/*
+ * write to the WM8728 register space
+ */
+static int wm8728_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[2];
+
+ /* data is
+ * D15..D9 WM8728 register offset
+ * D8...D0 register data
+ */
+ data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+ data[1] = value & 0x00ff;
+
+ wm8728_write_reg_cache(codec, reg, value);
+
+ if (codec->hw_write(codec->control_data, data, 2) == 2)
+ return 0;
+ else
+ return -EIO;
+}
+
+static const DECLARE_TLV_DB_SCALE(wm8728_tlv, -12750, 50, 1);
+
+static const struct snd_kcontrol_new wm8728_snd_controls[] = {
+
+SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8728_DACLVOL, WM8728_DACRVOL,
+ 0, 255, 0, wm8728_tlv),
+
+SOC_SINGLE("Deemphasis", WM8728_DACCTL, 1, 1, 0),
+};
+
+static int wm8728_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm8728_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8728_snd_controls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+/*
+ * DAPM controls.
+ */
+static const struct snd_soc_dapm_widget wm8728_dapm_widgets[] = {
+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_OUTPUT("VOUTL"),
+SND_SOC_DAPM_OUTPUT("VOUTR"),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ {"VOUTL", NULL, "DAC"},
+ {"VOUTR", NULL, "DAC"},
+};
+
+static int wm8728_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, wm8728_dapm_widgets,
+ ARRAY_SIZE(wm8728_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+ snd_soc_dapm_new_widgets(codec);
+
+ return 0;
+}
+
+static int wm8728_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 mute_reg = wm8728_read_reg_cache(codec, WM8728_DACCTL);
+
+ if (mute)
+ wm8728_write(codec, WM8728_DACCTL, mute_reg | 1);
+ else
+ wm8728_write(codec, WM8728_DACCTL, mute_reg & ~1);
+
+ return 0;
+}
+
+static int wm8728_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 dac = wm8728_read_reg_cache(codec, WM8728_DACCTL);
+
+ dac &= ~0x18;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ dac |= 0x10;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ dac |= 0x08;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm8728_write(codec, WM8728_DACCTL, dac);
+
+ return 0;
+}
+
+static int wm8728_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface = wm8728_read_reg_cache(codec, WM8728_IFCTL);
+
+ /* Currently only I2S is supported by the driver, though the
+ * hardware is more flexible.
+ */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= 1;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* The hardware only support full slave mode */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ iface &= ~0x22;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= 0x20;
+ iface &= ~0x02;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ iface |= 0x02;
+ iface &= ~0x20;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface |= 0x22;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm8728_write(codec, WM8728_IFCTL, iface);
+ return 0;
+}
+
+static int wm8728_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 reg;
+ int i;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ /* Power everything up... */
+ reg = wm8728_read_reg_cache(codec, WM8728_DACCTL);
+ wm8728_write(codec, WM8728_DACCTL, reg & ~0x4);
+
+ /* ..then sync in the register cache. */
+ for (i = 0; i < ARRAY_SIZE(wm8728_reg_defaults); i++)
+ wm8728_write(codec, i,
+ wm8728_read_reg_cache(codec, i));
+ }
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ reg = wm8728_read_reg_cache(codec, WM8728_DACCTL);
+ wm8728_write(codec, WM8728_DACCTL, reg | 0x4);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+#define WM8728_RATES (SNDRV_PCM_RATE_8000_192000)
+
+#define WM8728_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+struct snd_soc_dai wm8728_dai = {
+ .name = "WM8728",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = WM8728_RATES,
+ .formats = WM8728_FORMATS,
+ },
+ .ops = {
+ .hw_params = wm8728_hw_params,
+ .digital_mute = wm8728_mute,
+ .set_fmt = wm8728_set_dai_fmt,
+ }
+};
+EXPORT_SYMBOL_GPL(wm8728_dai);
+
+static int wm8728_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ return 0;
+}
+
+static int wm8728_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ wm8728_set_bias_level(codec, codec->suspend_bias_level);
+
+ return 0;
+}
+
+/*
+ * initialise the WM8728 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int wm8728_init(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret = 0;
+
+ codec->name = "WM8728";
+ codec->owner = THIS_MODULE;
+ codec->read = wm8728_read_reg_cache;
+ codec->write = wm8728_write;
+ codec->set_bias_level = wm8728_set_bias_level;
+ codec->dai = &wm8728_dai;
+ codec->num_dai = 1;
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ codec->reg_cache_size = ARRAY_SIZE(wm8728_reg_defaults);
+ codec->reg_cache = kmemdup(wm8728_reg_defaults,
+ sizeof(wm8728_reg_defaults),
+ GFP_KERNEL);
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8728: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ /* power on device */
+ wm8728_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ wm8728_add_controls(codec);
+ wm8728_add_widgets(codec);
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8728: failed to register card\n");
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ kfree(codec->reg_cache);
+ return ret;
+}
+
+static struct snd_soc_device *wm8728_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+/*
+ * WM8728 2 wire address is determined by GPIO5
+ * state during powerup.
+ * low = 0x1a
+ * high = 0x1b
+ */
+
+static int wm8728_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct snd_soc_device *socdev = wm8728_socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret;
+
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
+ ret = wm8728_init(socdev);
+ if (ret < 0)
+ pr_err("failed to initialise WM8728\n");
+
+ return ret;
+}
+
+static int wm8728_i2c_remove(struct i2c_client *client)
+{
+ struct snd_soc_codec *codec = i2c_get_clientdata(client);
+ kfree(codec->reg_cache);
+ return 0;
+}
+
+static const struct i2c_device_id wm8728_i2c_id[] = {
+ { "wm8728", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8728_i2c_id);
+
+static struct i2c_driver wm8728_i2c_driver = {
+ .driver = {
+ .name = "WM8728 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8728_i2c_probe,
+ .remove = wm8728_i2c_remove,
+ .id_table = wm8728_i2c_id,
+};
+
+static int wm8728_add_i2c_device(struct platform_device *pdev,
+ const struct wm8728_setup_data *setup)
+{
+ struct i2c_board_info info;
+ struct i2c_adapter *adapter;
+ struct i2c_client *client;
+ int ret;
+
+ ret = i2c_add_driver(&wm8728_i2c_driver);
+ if (ret != 0) {
+ dev_err(&pdev->dev, "can't add i2c driver\n");
+ return ret;
+ }
+
+ memset(&info, 0, sizeof(struct i2c_board_info));
+ info.addr = setup->i2c_address;
+ strlcpy(info.type, "wm8728", I2C_NAME_SIZE);
+
+ adapter = i2c_get_adapter(setup->i2c_bus);
+ if (!adapter) {
+ dev_err(&pdev->dev, "can't get i2c adapter %d\n",
+ setup->i2c_bus);
+ goto err_driver;
+ }
+
+ client = i2c_new_device(adapter, &info);
+ i2c_put_adapter(adapter);
+ if (!client) {
+ dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
+ (unsigned int)info.addr);
+ goto err_driver;
+ }
+
+ return 0;
+
+err_driver:
+ i2c_del_driver(&wm8728_i2c_driver);
+ return -ENODEV;
+}
+#endif
+
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit wm8728_spi_probe(struct spi_device *spi)
+{
+ struct snd_soc_device *socdev = wm8728_socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret;
+
+ codec->control_data = spi;
+
+ ret = wm8728_init(socdev);
+ if (ret < 0)
+ dev_err(&spi->dev, "failed to initialise WM8728\n");
+
+ return ret;
+}
+
+static int __devexit wm8728_spi_remove(struct spi_device *spi)
+{
+ return 0;
+}
+
+static struct spi_driver wm8728_spi_driver = {
+ .driver = {
+ .name = "wm8728",
+ .bus = &spi_bus_type,
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8728_spi_probe,
+ .remove = __devexit_p(wm8728_spi_remove),
+};
+
+static int wm8728_spi_write(struct spi_device *spi, const char *data, int len)
+{
+ struct spi_transfer t;
+ struct spi_message m;
+ u8 msg[2];
+
+ if (len <= 0)
+ return 0;
+
+ msg[0] = data[0];
+ msg[1] = data[1];
+
+ spi_message_init(&m);
+ memset(&t, 0, (sizeof t));
+
+ t.tx_buf = &msg[0];
+ t.len = len;
+
+ spi_message_add_tail(&t, &m);
+ spi_sync(spi, &m);
+
+ return len;
+}
+#endif /* CONFIG_SPI_MASTER */
+
+static int wm8728_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct wm8728_setup_data *setup;
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ setup = socdev->codec_data;
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ wm8728_socdev = socdev;
+ ret = -ENODEV;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ if (setup->i2c_address) {
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ ret = wm8728_add_i2c_device(pdev, setup);
+ }
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ if (setup->spi) {
+ codec->hw_write = (hw_write_t)wm8728_spi_write;
+ ret = spi_register_driver(&wm8728_spi_driver);
+ if (ret != 0)
+ printk(KERN_ERR "can't add spi driver");
+ }
+#endif
+
+ if (ret != 0)
+ kfree(codec);
+
+ return ret;
+}
+
+/* power down chip */
+static int wm8728_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec->control_data)
+ wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_unregister_device(codec->control_data);
+ i2c_del_driver(&wm8728_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ spi_unregister_driver(&wm8728_spi_driver);
+#endif
+ kfree(codec);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8728 = {
+ .probe = wm8728_probe,
+ .remove = wm8728_remove,
+ .suspend = wm8728_suspend,
+ .resume = wm8728_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8728);
+
+static int __init wm8728_modinit(void)
+{
+ return snd_soc_register_dai(&wm8728_dai);
+}
+module_init(wm8728_modinit);
+
+static void __exit wm8728_exit(void)
+{
+ snd_soc_unregister_dai(&wm8728_dai);
+}
+module_exit(wm8728_exit);
+
+MODULE_DESCRIPTION("ASoC WM8728 driver");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8728.h b/sound/soc/codecs/wm8728.h
new file mode 100644
index 0000000..d269c13
--- /dev/null
+++ b/sound/soc/codecs/wm8728.h
@@ -0,0 +1,30 @@
+/*
+ * wm8728.h -- WM8728 ASoC codec driver
+ *
+ * Copyright 2008 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@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.
+ */
+
+#ifndef _WM8728_H
+#define _WM8728_H
+
+#define WM8728_DACLVOL 0x00
+#define WM8728_DACRVOL 0x01
+#define WM8728_DACCTL 0x02
+#define WM8728_IFCTL 0x03
+
+struct wm8728_setup_data {
+ int spi;
+ int i2c_bus;
+ unsigned short i2c_address;
+};
+
+extern struct snd_soc_dai wm8728_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8728;
+
+#endif
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 7f8a7e3..c444b9f 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -264,7 +264,8 @@
}
static int wm8731_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -293,7 +294,8 @@
return 0;
}
-static int wm8731_pcm_prepare(struct snd_pcm_substream *substream)
+static int wm8731_pcm_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -305,7 +307,8 @@
return 0;
}
-static void wm8731_shutdown(struct snd_pcm_substream *substream)
+static void wm8731_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -461,8 +464,6 @@
.prepare = wm8731_pcm_prepare,
.hw_params = wm8731_hw_params,
.shutdown = wm8731_shutdown,
- },
- .dai_ops = {
.digital_mute = wm8731_mute,
.set_sysclk = wm8731_set_dai_sysclk,
.set_fmt = wm8731_set_dai_fmt,
@@ -544,7 +545,7 @@
wm8731_add_controls(codec);
wm8731_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "wm8731: failed to register card\n");
goto card_err;
@@ -792,6 +793,18 @@
};
EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731);
+static int __init wm8731_modinit(void)
+{
+ return snd_soc_register_dai(&wm8731_dai);
+}
+module_init(wm8731_modinit);
+
+static void __exit wm8731_exit(void)
+{
+ snd_soc_unregister_dai(&wm8731_dai);
+}
+module_exit(wm8731_exit);
+
MODULE_DESCRIPTION("ASoC WM8731 driver");
MODULE_AUTHOR("Richard Purdie");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 9b7296e..5997fa6 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -614,7 +614,8 @@
}
static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -709,8 +710,6 @@
.formats = WM8750_FORMATS,},
.ops = {
.hw_params = wm8750_pcm_hw_params,
- },
- .dai_ops = {
.digital_mute = wm8750_mute,
.set_fmt = wm8750_set_dai_fmt,
.set_sysclk = wm8750_set_dai_sysclk,
@@ -819,7 +818,7 @@
wm8750_add_controls(codec);
wm8750_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "wm8750: failed to register card\n");
goto card_err;
@@ -1086,6 +1085,18 @@
};
EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750);
+static int __init wm8750_modinit(void)
+{
+ return snd_soc_register_dai(&wm8750_dai);
+}
+module_init(wm8750_modinit);
+
+static void __exit wm8750_exit(void)
+{
+ snd_soc_unregister_dai(&wm8750_dai);
+}
+module_exit(wm8750_exit);
+
MODULE_DESCRIPTION("ASoC WM8750 driver");
MODULE_AUTHOR("Liam Girdwood");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index d426eaa..6c21b50 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -922,7 +922,8 @@
* Set PCM DAI bit size and sample rate.
*/
static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -1155,7 +1156,8 @@
* Set PCM DAI bit size and sample rate.
*/
static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -1323,16 +1325,15 @@
.channels_min = 1,
.channels_max = 2,
.rates = WM8753_RATES,
- .formats = WM8753_FORMATS,},
+ .formats = WM8753_FORMATS},
.capture = { /* dummy for fast DAI switching */
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM8753_RATES,
- .formats = WM8753_FORMATS,},
+ .formats = WM8753_FORMATS},
.ops = {
- .hw_params = wm8753_i2s_hw_params,},
- .dai_ops = {
+ .hw_params = wm8753_i2s_hw_params,
.digital_mute = wm8753_mute,
.set_fmt = wm8753_mode1h_set_dai_fmt,
.set_clkdiv = wm8753_set_dai_clkdiv,
@@ -1356,8 +1357,7 @@
.rates = WM8753_RATES,
.formats = WM8753_FORMATS,},
.ops = {
- .hw_params = wm8753_pcm_hw_params,},
- .dai_ops = {
+ .hw_params = wm8753_pcm_hw_params,
.digital_mute = wm8753_mute,
.set_fmt = wm8753_mode1v_set_dai_fmt,
.set_clkdiv = wm8753_set_dai_clkdiv,
@@ -1385,8 +1385,7 @@
.rates = WM8753_RATES,
.formats = WM8753_FORMATS,},
.ops = {
- .hw_params = wm8753_pcm_hw_params,},
- .dai_ops = {
+ .hw_params = wm8753_pcm_hw_params,
.digital_mute = wm8753_mute,
.set_fmt = wm8753_mode2_set_dai_fmt,
.set_clkdiv = wm8753_set_dai_clkdiv,
@@ -1410,8 +1409,7 @@
.rates = WM8753_RATES,
.formats = WM8753_FORMATS,},
.ops = {
- .hw_params = wm8753_i2s_hw_params,},
- .dai_ops = {
+ .hw_params = wm8753_i2s_hw_params,
.digital_mute = wm8753_mute,
.set_fmt = wm8753_mode3_4_set_dai_fmt,
.set_clkdiv = wm8753_set_dai_clkdiv,
@@ -1439,8 +1437,7 @@
.rates = WM8753_RATES,
.formats = WM8753_FORMATS,},
.ops = {
- .hw_params = wm8753_i2s_hw_params,},
- .dai_ops = {
+ .hw_params = wm8753_i2s_hw_params,
.digital_mute = wm8753_mute,
.set_fmt = wm8753_mode3_4_set_dai_fmt,
.set_clkdiv = wm8753_set_dai_clkdiv,
@@ -1608,7 +1605,7 @@
wm8753_add_controls(codec);
wm8753_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "wm8753: failed to register card\n");
goto card_err;
@@ -1877,6 +1874,18 @@
};
EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753);
+static int __init wm8753_modinit(void)
+{
+ return snd_soc_register_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai));
+}
+module_init(wm8753_modinit);
+
+static void __exit wm8753_exit(void)
+{
+ snd_soc_unregister_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai));
+}
+module_exit(wm8753_exit);
+
MODULE_DESCRIPTION("ASoC WM8753 driver");
MODULE_AUTHOR("Liam Girdwood");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 3b326c9..6767de1 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -138,6 +138,10 @@
struct snd_soc_codec_device soc_codec_dev_wm8900;
struct wm8900_priv {
+ struct snd_soc_codec codec;
+
+ u16 reg_cache[WM8900_MAXREG];
+
u32 fll_in; /* FLL input frequency */
u32 fll_out; /* FLL output frequency */
};
@@ -727,7 +731,8 @@
}
static int wm8900_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -1117,8 +1122,6 @@
},
.ops = {
.hw_params = wm8900_hw_params,
- },
- .dai_ops = {
.set_clkdiv = wm8900_set_dai_clkdiv,
.set_pll = wm8900_set_dai_pll,
.set_fmt = wm8900_set_dai_fmt,
@@ -1283,16 +1286,28 @@
return 0;
}
-/*
- * initialise the WM8900 driver
- * register the mixer and dsp interfaces with the kernel
- */
-static int wm8900_init(struct snd_soc_device *socdev)
+static struct snd_soc_codec *wm8900_codec;
+
+static int wm8900_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
{
- struct snd_soc_codec *codec = socdev->codec;
- int ret = 0;
+ struct wm8900_priv *wm8900;
+ struct snd_soc_codec *codec;
unsigned int reg;
- struct i2c_client *i2c_client = socdev->codec->control_data;
+ int ret;
+
+ wm8900 = kzalloc(sizeof(struct wm8900_priv), GFP_KERNEL);
+ if (wm8900 == NULL)
+ return -ENOMEM;
+
+ codec = &wm8900->codec;
+ codec->private_data = wm8900;
+ codec->reg_cache = &wm8900->reg_cache[0];
+ codec->reg_cache_size = WM8900_MAXREG;
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
codec->name = "WM8900";
codec->owner = THIS_MODULE;
@@ -1300,33 +1315,28 @@
codec->write = wm8900_write;
codec->dai = &wm8900_dai;
codec->num_dai = 1;
- codec->reg_cache_size = WM8900_MAXREG;
- codec->reg_cache = kmemdup(wm8900_reg_defaults,
- sizeof(wm8900_reg_defaults), GFP_KERNEL);
-
- if (codec->reg_cache == NULL)
- return -ENOMEM;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ codec->control_data = i2c;
+ codec->set_bias_level = wm8900_set_bias_level;
+ codec->dev = &i2c->dev;
reg = wm8900_read(codec, WM8900_REG_ID);
if (reg != 0x8900) {
- dev_err(&i2c_client->dev, "Device is not a WM8900 - ID %x\n",
- reg);
- return -ENODEV;
- }
-
- codec->private_data = kzalloc(sizeof(struct wm8900_priv), GFP_KERNEL);
- if (codec->private_data == NULL) {
- ret = -ENOMEM;
- goto priv_err;
+ dev_err(&i2c->dev, "Device is not a WM8900 - ID %x\n", reg);
+ ret = -ENODEV;
+ goto err;
}
/* Read back from the chip */
reg = wm8900_chip_read(codec, WM8900_REG_POWER1);
reg = (reg >> 12) & 0xf;
- dev_info(&i2c_client->dev, "WM8900 revision %d\n", reg);
+ dev_info(&i2c->dev, "WM8900 revision %d\n", reg);
wm8900_reset(codec);
+ /* Turn the chip on */
+ wm8900_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
/* Latch the volume update bits */
wm8900_write(codec, WM8900_REG_LINVOL,
wm8900_read(codec, WM8900_REG_LINVOL) | 0x100);
@@ -1352,160 +1362,98 @@
/* Set the DAC and mixer output bias */
wm8900_write(codec, WM8900_REG_OUTBIASCTL, 0x81);
+ wm8900_dai.dev = &i2c->dev;
+
+ wm8900_codec = codec;
+
+ ret = snd_soc_register_codec(codec);
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Failed to register codec: %d\n", ret);
+ goto err;
+ }
+
+ ret = snd_soc_register_dai(&wm8900_dai);
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Failed to register DAI: %d\n", ret);
+ goto err_codec;
+ }
+
+ return ret;
+
+err_codec:
+ snd_soc_unregister_codec(codec);
+err:
+ kfree(wm8900);
+ wm8900_codec = NULL;
+ return ret;
+}
+
+static int wm8900_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_dai(&wm8900_dai);
+ snd_soc_unregister_codec(wm8900_codec);
+
+ wm8900_set_bias_level(wm8900_codec, SND_SOC_BIAS_OFF);
+
+ wm8900_dai.dev = NULL;
+ kfree(wm8900_codec->private_data);
+ wm8900_codec = NULL;
+
+ return 0;
+}
+
+static const struct i2c_device_id wm8900_i2c_id[] = {
+ { "wm8900", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8900_i2c_id);
+
+static struct i2c_driver wm8900_i2c_driver = {
+ .driver = {
+ .name = "WM8900",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8900_i2c_probe,
+ .remove = wm8900_i2c_remove,
+ .id_table = wm8900_i2c_id,
+};
+
+static int wm8900_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ if (!wm8900_codec) {
+ dev_err(&pdev->dev, "I2C client not yet instantiated\n");
+ return -ENODEV;
+ }
+
+ codec = wm8900_codec;
+ socdev->codec = codec;
+
/* Register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0) {
- dev_err(&i2c_client->dev, "Failed to register new PCMs\n");
+ dev_err(&pdev->dev, "Failed to register new PCMs\n");
goto pcm_err;
}
- /* Turn the chip on */
- codec->bias_level = SND_SOC_BIAS_OFF;
- wm8900_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
wm8900_add_controls(codec);
wm8900_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
- dev_err(&i2c_client->dev, "Failed to register card\n");
+ dev_err(&pdev->dev, "Failed to register card\n");
goto card_err;
}
+
return ret;
card_err:
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
pcm_err:
- kfree(codec->reg_cache);
-priv_err:
- kfree(codec->private_data);
- return ret;
-}
-
-static struct snd_soc_device *wm8900_socdev;
-
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-
-static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
-
-/* Magic definition of all other variables and things */
-I2C_CLIENT_INSMOD;
-
-static struct i2c_driver wm8900_i2c_driver;
-static struct i2c_client client_template;
-
-/* If the i2c layer weren't so broken, we could pass this kind of data
- around */
-static int wm8900_codec_probe(struct i2c_adapter *adap, int addr, int kind)
-{
- struct snd_soc_device *socdev = wm8900_socdev;
- struct wm8900_setup_data *setup = socdev->codec_data;
- struct snd_soc_codec *codec = socdev->codec;
- struct i2c_client *i2c;
- int ret;
-
- if (addr != setup->i2c_address)
- return -ENODEV;
-
- dev_err(&adap->dev, "Probe on %x\n", addr);
-
- client_template.adapter = adap;
- client_template.addr = addr;
-
- i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
- if (i2c == NULL) {
- kfree(codec);
- return -ENOMEM;
- }
- i2c_set_clientdata(i2c, codec);
- codec->control_data = i2c;
-
- ret = i2c_attach_client(i2c);
- if (ret < 0) {
- dev_err(&adap->dev,
- "failed to attach codec at addr %x\n", addr);
- goto err;
- }
-
- ret = wm8900_init(socdev);
- if (ret < 0) {
- dev_err(&adap->dev, "failed to initialise WM8900\n");
- goto err;
- }
- return ret;
-
-err:
- kfree(codec);
- kfree(i2c);
- return ret;
-}
-
-static int wm8900_i2c_detach(struct i2c_client *client)
-{
- struct snd_soc_codec *codec = i2c_get_clientdata(client);
- i2c_detach_client(client);
- kfree(codec->reg_cache);
- kfree(client);
- return 0;
-}
-
-static int wm8900_i2c_attach(struct i2c_adapter *adap)
-{
- return i2c_probe(adap, &addr_data, wm8900_codec_probe);
-}
-
-/* corgi i2c codec control layer */
-static struct i2c_driver wm8900_i2c_driver = {
- .driver = {
- .name = "WM8900 I2C codec",
- .owner = THIS_MODULE,
- },
- .attach_adapter = wm8900_i2c_attach,
- .detach_client = wm8900_i2c_detach,
- .command = NULL,
-};
-
-static struct i2c_client client_template = {
- .name = "WM8900",
- .driver = &wm8900_i2c_driver,
-};
-#endif
-
-static int wm8900_probe(struct platform_device *pdev)
-{
- struct snd_soc_device *socdev = platform_get_drvdata(pdev);
- struct wm8900_setup_data *setup;
- struct snd_soc_codec *codec;
- int ret = 0;
-
- dev_info(&pdev->dev, "WM8900 Audio Codec\n");
-
- setup = socdev->codec_data;
- codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
- if (codec == NULL)
- return -ENOMEM;
-
- mutex_init(&codec->mutex);
- INIT_LIST_HEAD(&codec->dapm_widgets);
- INIT_LIST_HEAD(&codec->dapm_paths);
-
- socdev->codec = codec;
-
- codec->set_bias_level = wm8900_set_bias_level;
-
- wm8900_socdev = socdev;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
- if (setup->i2c_address) {
- normal_i2c[0] = setup->i2c_address;
- codec->hw_write = (hw_write_t)i2c_master_send;
- ret = i2c_add_driver(&wm8900_i2c_driver);
- if (ret != 0)
- printk(KERN_ERR "can't add i2c driver");
- }
-#else
-#error Non-I2C interfaces not yet supported
-#endif
return ret;
}
@@ -1513,17 +1461,9 @@
static int wm8900_remove(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
- struct snd_soc_codec *codec = socdev->codec;
-
- if (codec->control_data)
- wm8900_set_bias_level(codec, SND_SOC_BIAS_OFF);
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
- i2c_del_driver(&wm8900_i2c_driver);
-#endif
- kfree(codec);
return 0;
}
@@ -1536,6 +1476,18 @@
};
EXPORT_SYMBOL_GPL(soc_codec_dev_wm8900);
+static int __init wm8900_modinit(void)
+{
+ return i2c_add_driver(&wm8900_i2c_driver);
+}
+module_init(wm8900_modinit);
+
+static void __exit wm8900_exit(void)
+{
+ i2c_del_driver(&wm8900_i2c_driver);
+}
+module_exit(wm8900_exit);
+
MODULE_DESCRIPTION("ASoC WM8900 driver");
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfonmicro.com>");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8900.h b/sound/soc/codecs/wm8900.h
index ba450d9..fd15007 100644
--- a/sound/soc/codecs/wm8900.h
+++ b/sound/soc/codecs/wm8900.h
@@ -52,12 +52,6 @@
#define WM8900_DAC_CLKDIV_5_5 0x14
#define WM8900_DAC_CLKDIV_6 0x18
-#define WM8900_
-
-struct wm8900_setup_data {
- unsigned short i2c_address;
-};
-
extern struct snd_soc_dai wm8900_dai;
extern struct snd_soc_codec_device soc_codec_dev_wm8900;
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index ce40d78..bde7454 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -33,19 +33,6 @@
#include "wm8903.h"
-struct wm8903_priv {
- int sysclk;
-
- /* Reference counts */
- int charge_pump_users;
- int class_w_users;
- int playback_active;
- int capture_active;
-
- struct snd_pcm_substream *master_substream;
- struct snd_pcm_substream *slave_substream;
-};
-
/* Register defaults at reset */
static u16 wm8903_reg_defaults[] = {
0x8903, /* R0 - SW Reset and ID */
@@ -223,6 +210,23 @@
0x0000, /* R172 - Analogue Output Bias 0 */
};
+struct wm8903_priv {
+ struct snd_soc_codec codec;
+ u16 reg_cache[ARRAY_SIZE(wm8903_reg_defaults)];
+
+ int sysclk;
+
+ /* Reference counts */
+ int charge_pump_users;
+ int class_w_users;
+ int playback_active;
+ int capture_active;
+
+ struct snd_pcm_substream *master_substream;
+ struct snd_pcm_substream *slave_substream;
+};
+
+
static unsigned int wm8903_read_reg_cache(struct snd_soc_codec *codec,
unsigned int reg)
{
@@ -360,6 +364,8 @@
static void wm8903_reset(struct snd_soc_codec *codec)
{
wm8903_write(codec, WM8903_SW_RESET_AND_ID, 0);
+ memcpy(codec->reg_cache, wm8903_reg_defaults,
+ sizeof(wm8903_reg_defaults));
}
#define WM8903_OUTPUT_SHORT 0x8
@@ -392,6 +398,7 @@
break;
default:
BUG();
+ return -EINVAL; /* Spurious warning from some compilers */
}
switch (w->shift) {
@@ -403,6 +410,7 @@
break;
default:
BUG();
+ return -EINVAL; /* Spurious warning from some compilers */
}
if (event & SND_SOC_DAPM_PRE_PMU) {
@@ -773,14 +781,14 @@
SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_LEFT_MIX_0, 3, 1, 0),
SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_LEFT_MIX_0, 2, 1, 0),
SOC_DAPM_SINGLE_W("Left Bypass Switch", WM8903_ANALOGUE_LEFT_MIX_0, 1, 1, 0),
-SOC_DAPM_SINGLE_W("Right Bypass Switch", WM8903_ANALOGUE_LEFT_MIX_0, 1, 1, 0),
+SOC_DAPM_SINGLE_W("Right Bypass Switch", WM8903_ANALOGUE_LEFT_MIX_0, 0, 1, 0),
};
static const struct snd_kcontrol_new right_output_mixer[] = {
SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 3, 1, 0),
SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 2, 1, 0),
SOC_DAPM_SINGLE_W("Left Bypass Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 1, 1, 0),
-SOC_DAPM_SINGLE_W("Right Bypass Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 1, 1, 0),
+SOC_DAPM_SINGLE_W("Right Bypass Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 0, 1, 0),
};
static const struct snd_kcontrol_new left_speaker_mixer[] = {
@@ -788,7 +796,7 @@
SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0, 2, 1, 0),
SOC_DAPM_SINGLE("Left Bypass Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0, 1, 1, 0),
SOC_DAPM_SINGLE("Right Bypass Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0,
- 1, 1, 0),
+ 0, 1, 0),
};
static const struct snd_kcontrol_new right_speaker_mixer[] = {
@@ -797,7 +805,7 @@
SOC_DAPM_SINGLE("Left Bypass Switch", WM8903_ANALOGUE_SPK_MIX_RIGHT_0,
1, 1, 0),
SOC_DAPM_SINGLE("Right Bypass Switch", WM8903_ANALOGUE_SPK_MIX_RIGHT_0,
- 1, 1, 0),
+ 0, 1, 0),
};
static const struct snd_soc_dapm_widget wm8903_dapm_widgets[] = {
@@ -989,6 +997,9 @@
case SND_SOC_BIAS_STANDBY:
if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ wm8903_write(codec, WM8903_CLOCK_RATES_2,
+ WM8903_CLK_SYS_ENA);
+
wm8903_run_sequence(codec, 0);
wm8903_sync_reg_cache(codec, codec->reg_cache);
@@ -1019,6 +1030,9 @@
case SND_SOC_BIAS_OFF:
wm8903_run_sequence(codec, 32);
+ reg = wm8903_read(codec, WM8903_CLOCK_RATES_2);
+ reg &= ~WM8903_CLK_SYS_ENA;
+ wm8903_write(codec, WM8903_CLOCK_RATES_2, reg);
break;
}
@@ -1257,7 +1271,8 @@
{ 0, 0 },
};
-static int wm8903_startup(struct snd_pcm_substream *substream)
+static int wm8903_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -1298,7 +1313,8 @@
return 0;
}
-static void wm8903_shutdown(struct snd_pcm_substream *substream)
+static void wm8903_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -1317,7 +1333,8 @@
}
static int wm8903_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -1515,8 +1532,6 @@
.startup = wm8903_startup,
.shutdown = wm8903_shutdown,
.hw_params = wm8903_hw_params,
- },
- .dai_ops = {
.digital_mute = wm8903_digital_mute,
.set_fmt = wm8903_set_dai_fmt,
.set_sysclk = wm8903_set_dai_sysclk
@@ -1560,17 +1575,43 @@
return 0;
}
-/*
- * initialise the WM8903 driver
- * register the mixer and dsp interfaces with the kernel
- */
-static int wm8903_init(struct snd_soc_device *socdev)
+static struct snd_soc_codec *wm8903_codec;
+
+static int wm8903_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
{
- struct snd_soc_codec *codec = socdev->codec;
- struct i2c_client *i2c = codec->control_data;
- int ret = 0;
+ struct wm8903_priv *wm8903;
+ struct snd_soc_codec *codec;
+ int ret;
u16 val;
+ wm8903 = kzalloc(sizeof(struct wm8903_priv), GFP_KERNEL);
+ if (wm8903 == NULL)
+ return -ENOMEM;
+
+ codec = &wm8903->codec;
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->dev = &i2c->dev;
+ codec->name = "WM8903";
+ codec->owner = THIS_MODULE;
+ codec->read = wm8903_read;
+ codec->write = wm8903_write;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ codec->set_bias_level = wm8903_set_bias_level;
+ codec->dai = &wm8903_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = ARRAY_SIZE(wm8903->reg_cache);
+ codec->reg_cache = &wm8903->reg_cache[0];
+ codec->private_data = wm8903;
+
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
val = wm8903_hw_read(codec, WM8903_SW_RESET_AND_ID);
if (val != wm8903_reg_defaults[WM8903_SW_RESET_AND_ID]) {
dev_err(&i2c->dev,
@@ -1578,39 +1619,12 @@
return -ENODEV;
}
- codec->name = "WM8903";
- codec->owner = THIS_MODULE;
- codec->read = wm8903_read;
- codec->write = wm8903_write;
- codec->bias_level = SND_SOC_BIAS_OFF;
- codec->set_bias_level = wm8903_set_bias_level;
- codec->dai = &wm8903_dai;
- codec->num_dai = 1;
- codec->reg_cache_size = ARRAY_SIZE(wm8903_reg_defaults);
- codec->reg_cache = kmemdup(wm8903_reg_defaults,
- sizeof(wm8903_reg_defaults),
- GFP_KERNEL);
- if (codec->reg_cache == NULL) {
- dev_err(&i2c->dev, "Failed to allocate register cache\n");
- return -ENOMEM;
- }
-
val = wm8903_read(codec, WM8903_REVISION_NUMBER);
dev_info(&i2c->dev, "WM8903 revision %d\n",
val & WM8903_CHIP_REV_MASK);
wm8903_reset(codec);
- /* register pcms */
- ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
- if (ret < 0) {
- dev_err(&i2c->dev, "failed to create pcms\n");
- goto pcm_err;
- }
-
- /* SYSCLK is required for pretty much anything */
- wm8903_write(codec, WM8903_CLOCK_RATES_2, WM8903_CLK_SYS_ENA);
-
/* power on device */
wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
@@ -1645,47 +1659,45 @@
val |= WM8903_DAC_MUTEMODE;
wm8903_write(codec, WM8903_DAC_DIGITAL_1, val);
- wm8903_add_controls(codec);
- wm8903_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
- if (ret < 0) {
- dev_err(&i2c->dev, "wm8903: failed to register card\n");
- goto card_err;
+ wm8903_dai.dev = &i2c->dev;
+ wm8903_codec = codec;
+
+ ret = snd_soc_register_codec(codec);
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Failed to register codec: %d\n", ret);
+ goto err;
+ }
+
+ ret = snd_soc_register_dai(&wm8903_dai);
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Failed to register DAI: %d\n", ret);
+ goto err_codec;
}
return ret;
-card_err:
- snd_soc_free_pcms(socdev);
- snd_soc_dapm_free(socdev);
-pcm_err:
- kfree(codec->reg_cache);
- return ret;
-}
-
-static struct snd_soc_device *wm8903_socdev;
-
-static int wm8903_i2c_probe(struct i2c_client *i2c,
- const struct i2c_device_id *id)
-{
- struct snd_soc_device *socdev = wm8903_socdev;
- struct snd_soc_codec *codec = socdev->codec;
- int ret;
-
- i2c_set_clientdata(i2c, codec);
- codec->control_data = i2c;
-
- ret = wm8903_init(socdev);
- if (ret < 0)
- dev_err(&i2c->dev, "Device initialisation failed\n");
-
+err_codec:
+ snd_soc_unregister_codec(codec);
+err:
+ wm8903_codec = NULL;
+ kfree(wm8903);
return ret;
}
static int wm8903_i2c_remove(struct i2c_client *client)
{
struct snd_soc_codec *codec = i2c_get_clientdata(client);
- kfree(codec->reg_cache);
+
+ snd_soc_unregister_dai(&wm8903_dai);
+ snd_soc_unregister_codec(codec);
+
+ wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ kfree(codec->private_data);
+
+ wm8903_codec = NULL;
+ wm8903_dai.dev = NULL;
+
return 0;
}
@@ -1709,75 +1721,37 @@
static int wm8903_probe(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
- struct wm8903_setup_data *setup;
- struct snd_soc_codec *codec;
- struct wm8903_priv *wm8903;
- struct i2c_board_info board_info;
- struct i2c_adapter *adapter;
- struct i2c_client *i2c_client;
int ret = 0;
- setup = socdev->codec_data;
-
- if (!setup->i2c_address) {
- dev_err(&pdev->dev, "No codec address provided\n");
- return -ENODEV;
+ if (!wm8903_codec) {
+ dev_err(&pdev->dev, "I2C device not yet probed\n");
+ goto err;
}
- codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
- if (codec == NULL)
- return -ENOMEM;
+ socdev->codec = wm8903_codec;
- wm8903 = kzalloc(sizeof(struct wm8903_priv), GFP_KERNEL);
- if (wm8903 == NULL) {
- ret = -ENOMEM;
- goto err_codec;
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(&pdev->dev, "failed to create pcms\n");
+ goto err;
}
- codec->private_data = wm8903;
- socdev->codec = codec;
- mutex_init(&codec->mutex);
- INIT_LIST_HEAD(&codec->dapm_widgets);
- INIT_LIST_HEAD(&codec->dapm_paths);
+ wm8903_add_controls(socdev->codec);
+ wm8903_add_widgets(socdev->codec);
- wm8903_socdev = socdev;
-
- codec->hw_write = (hw_write_t)i2c_master_send;
- ret = i2c_add_driver(&wm8903_i2c_driver);
- if (ret != 0) {
- dev_err(&pdev->dev, "can't add i2c driver\n");
- goto err_priv;
- } else {
- memset(&board_info, 0, sizeof(board_info));
- strlcpy(board_info.type, "wm8903", I2C_NAME_SIZE);
- board_info.addr = setup->i2c_address;
-
- adapter = i2c_get_adapter(setup->i2c_bus);
- if (!adapter) {
- dev_err(&pdev->dev, "Can't get I2C bus %d\n",
- setup->i2c_bus);
- ret = -ENODEV;
- goto err_adapter;
- }
-
- i2c_client = i2c_new_device(adapter, &board_info);
- i2c_put_adapter(adapter);
- if (i2c_client == NULL) {
- dev_err(&pdev->dev,
- "I2C driver registration failed\n");
- ret = -ENODEV;
- goto err_adapter;
- }
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ dev_err(&pdev->dev, "wm8903: failed to register card\n");
+ goto card_err;
}
return ret;
-err_adapter:
- i2c_del_driver(&wm8903_i2c_driver);
-err_priv:
- kfree(codec->private_data);
-err_codec:
- kfree(codec);
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+err:
return ret;
}
@@ -1792,10 +1766,6 @@
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
- i2c_unregister_device(socdev->codec->control_data);
- i2c_del_driver(&wm8903_i2c_driver);
- kfree(codec->private_data);
- kfree(codec);
return 0;
}
@@ -1808,6 +1778,18 @@
};
EXPORT_SYMBOL_GPL(soc_codec_dev_wm8903);
+static int __init wm8903_modinit(void)
+{
+ return i2c_add_driver(&wm8903_i2c_driver);
+}
+module_init(wm8903_modinit);
+
+static void __exit wm8903_exit(void)
+{
+ i2c_del_driver(&wm8903_i2c_driver);
+}
+module_exit(wm8903_exit);
+
MODULE_DESCRIPTION("ASoC WM8903 driver");
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.cm>");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8903.h b/sound/soc/codecs/wm8903.h
index cec622f..0ea27e2 100644
--- a/sound/soc/codecs/wm8903.h
+++ b/sound/soc/codecs/wm8903.h
@@ -18,11 +18,6 @@
extern struct snd_soc_dai wm8903_dai;
extern struct snd_soc_codec_device soc_codec_dev_wm8903;
-struct wm8903_setup_data {
- int i2c_bus;
- int i2c_address;
-};
-
#define WM8903_MCLK_DIV_2 1
#define WM8903_CLK_SYS 2
#define WM8903_BCLK 3
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index f41a578..88ead7f 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -541,7 +541,8 @@
}
static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -634,8 +635,6 @@
.formats = WM8971_FORMATS,},
.ops = {
.hw_params = wm8971_pcm_hw_params,
- },
- .dai_ops = {
.digital_mute = wm8971_mute,
.set_fmt = wm8971_set_dai_fmt,
.set_sysclk = wm8971_set_dai_sysclk,
@@ -748,7 +747,7 @@
wm8971_add_controls(codec);
wm8971_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "wm8971: failed to register card\n");
goto card_err;
@@ -936,6 +935,18 @@
EXPORT_SYMBOL_GPL(soc_codec_dev_wm8971);
+static int __init wm8971_modinit(void)
+{
+ return snd_soc_register_dai(&wm8971_dai);
+}
+module_init(wm8971_modinit);
+
+static void __exit wm8971_exit(void)
+{
+ snd_soc_unregister_dai(&wm8971_dai);
+}
+module_exit(wm8971_exit);
+
MODULE_DESCRIPTION("ASoC WM8971 driver");
MODULE_AUTHOR("Lab126");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 572d22b..5b5afc1 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -106,6 +106,7 @@
0x0008, /* R60 - PLL1 */
0x0031, /* R61 - PLL2 */
0x0026, /* R62 - PLL3 */
+ 0x0000, /* R63 - Driver internal */
};
/*
@@ -126,10 +127,9 @@
unsigned int reg, unsigned int value)
{
u16 *cache = codec->reg_cache;
- BUG_ON(reg > (ARRAY_SIZE(wm8990_reg)) - 1);
- /* Reset register is uncached */
- if (reg == 0)
+ /* Reset register and reserved registers are uncached */
+ if (reg == 0 || reg > ARRAY_SIZE(wm8990_reg) - 1)
return;
cache[reg] = value;
@@ -1172,7 +1172,8 @@
* Set PCM DAI bit size and sample rate.
*/
static int wm8990_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
@@ -1222,8 +1223,14 @@
switch (level) {
case SND_SOC_BIAS_ON:
break;
+
case SND_SOC_BIAS_PREPARE:
+ /* VMID=2*50k */
+ val = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_1) &
+ ~WM8990_VMID_MODE_MASK;
+ wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x2);
break;
+
case SND_SOC_BIAS_STANDBY:
if (codec->bias_level == SND_SOC_BIAS_OFF) {
/* Enable all output discharge bits */
@@ -1272,10 +1279,17 @@
/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
wm8990_write(codec, WM8990_ANTIPOP2, WM8990_BUFIOEN);
- } else {
- /* ON -> standby */
+ /* Enable workaround for ADC clocking issue. */
+ wm8990_write(codec, WM8990_EXT_ACCESS_ENA, 0x2);
+ wm8990_write(codec, WM8990_EXT_CTL1, 0xa003);
+ wm8990_write(codec, WM8990_EXT_ACCESS_ENA, 0);
}
+
+ /* VMID=2*250k */
+ val = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_1) &
+ ~WM8990_VMID_MODE_MASK;
+ wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x4);
break;
case SND_SOC_BIAS_OFF:
@@ -1349,8 +1363,7 @@
.rates = WM8990_RATES,
.formats = WM8990_FORMATS,},
.ops = {
- .hw_params = wm8990_hw_params,},
- .dai_ops = {
+ .hw_params = wm8990_hw_params,
.digital_mute = wm8990_mute,
.set_fmt = wm8990_set_dai_fmt,
.set_clkdiv = wm8990_set_dai_clkdiv,
@@ -1449,7 +1462,7 @@
wm8990_add_controls(codec);
wm8990_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "wm8990: failed to register card\n");
goto card_err;
@@ -1630,6 +1643,18 @@
};
EXPORT_SYMBOL_GPL(soc_codec_dev_wm8990);
+static int __init wm8990_modinit(void)
+{
+ return snd_soc_register_dai(&wm8990_dai);
+}
+module_init(wm8990_modinit);
+
+static void __exit wm8990_exit(void)
+{
+ snd_soc_unregister_dai(&wm8990_dai);
+}
+module_exit(wm8990_exit);
+
MODULE_DESCRIPTION("ASoC WM8990 driver");
MODULE_AUTHOR("Liam Girdwood");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8990.h b/sound/soc/codecs/wm8990.h
index 0e192f3..7114ddc 100644
--- a/sound/soc/codecs/wm8990.h
+++ b/sound/soc/codecs/wm8990.h
@@ -80,8 +80,8 @@
#define WM8990_PLL3 0x3E
#define WM8990_INTDRIVBITS 0x3F
-#define WM8990_REGISTER_COUNT 60
-#define WM8990_MAX_REGISTER 0x3F
+#define WM8990_EXT_ACCESS_ENA 0x75
+#define WM8990_EXT_CTL1 0x7a
/*
* Field Definitions.
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index ffb471e..af83d62 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -487,7 +487,8 @@
return 0;
}
-static int ac97_prepare(struct snd_pcm_substream *substream)
+static int ac97_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -507,7 +508,8 @@
return ac97_write(codec, reg, runtime->rate);
}
-static int ac97_aux_prepare(struct snd_pcm_substream *substream)
+static int ac97_aux_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -533,7 +535,7 @@
struct snd_soc_dai wm9712_dai[] = {
{
.name = "AC97 HiFi",
- .type = SND_SOC_DAI_AC97_BUS,
+ .ac97_control = 1,
.playback = {
.stream_name = "HiFi Playback",
.channels_min = 1,
@@ -688,7 +690,7 @@
ret = wm9712_reset(codec, 0);
if (ret < 0) {
- printk(KERN_ERR "AC97 link error\n");
+ printk(KERN_ERR "Failed to reset WM9712: AC97 link error\n");
goto reset_err;
}
@@ -698,7 +700,7 @@
wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
wm9712_add_controls(codec);
wm9712_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "wm9712: failed to register card\n");
goto reset_err;
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 945b32e..f3ca8aa 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -928,11 +928,10 @@
}
static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_device *socdev = rtd->socdev;
- struct snd_soc_codec *codec = socdev->codec;
+ struct snd_soc_codec *codec = dai->codec;
u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xfff3;
switch (params_format(params)) {
@@ -954,11 +953,10 @@
return 0;
}
-static void wm9713_voiceshutdown(struct snd_pcm_substream *substream)
+static void wm9713_voiceshutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_device *socdev = rtd->socdev;
- struct snd_soc_codec *codec = socdev->codec;
+ struct snd_soc_codec *codec = dai->codec;
u16 status;
/* Gracefully shut down the voice interface. */
@@ -969,12 +967,11 @@
ac97_write(codec, AC97_EXTENDED_MID, status);
}
-static int ac97_hifi_prepare(struct snd_pcm_substream *substream)
+static int ac97_hifi_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
+ struct snd_soc_codec *codec = dai->codec;
struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_device *socdev = rtd->socdev;
- struct snd_soc_codec *codec = socdev->codec;
int reg;
u16 vra;
@@ -989,12 +986,11 @@
return ac97_write(codec, reg, runtime->rate);
}
-static int ac97_aux_prepare(struct snd_pcm_substream *substream)
+static int ac97_aux_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
+ struct snd_soc_codec *codec = dai->codec;
struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_device *socdev = rtd->socdev;
- struct snd_soc_codec *codec = socdev->codec;
u16 vra, xsle;
vra = ac97_read(codec, AC97_EXTENDED_STATUS);
@@ -1028,7 +1024,7 @@
struct snd_soc_dai wm9713_dai[] = {
{
.name = "AC97 HiFi",
- .type = SND_SOC_DAI_AC97_BUS,
+ .ac97_control = 1,
.playback = {
.stream_name = "HiFi Playback",
.channels_min = 1,
@@ -1042,8 +1038,7 @@
.rates = WM9713_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
.ops = {
- .prepare = ac97_hifi_prepare,},
- .dai_ops = {
+ .prepare = ac97_hifi_prepare,
.set_clkdiv = wm9713_set_dai_clkdiv,
.set_pll = wm9713_set_dai_pll,},
},
@@ -1056,8 +1051,7 @@
.rates = WM9713_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
.ops = {
- .prepare = ac97_aux_prepare,},
- .dai_ops = {
+ .prepare = ac97_aux_prepare,
.set_clkdiv = wm9713_set_dai_clkdiv,
.set_pll = wm9713_set_dai_pll,},
},
@@ -1077,8 +1071,7 @@
.formats = WM9713_PCM_FORMATS,},
.ops = {
.hw_params = wm9713_pcm_hw_params,
- .shutdown = wm9713_voiceshutdown,},
- .dai_ops = {
+ .shutdown = wm9713_voiceshutdown,
.set_clkdiv = wm9713_set_dai_clkdiv,
.set_pll = wm9713_set_dai_pll,
.set_fmt = wm9713_set_dai_fmt,
@@ -1097,6 +1090,8 @@
}
soc_ac97_ops.reset(codec->ac97);
+ if (soc_ac97_ops.warm_reset)
+ soc_ac97_ops.warm_reset(codec->ac97);
if (ac97_read(codec, 0) != wm9713_reg[0])
return -EIO;
return 0;
@@ -1240,7 +1235,7 @@
wm9713_reset(codec, 0);
ret = wm9713_reset(codec, 1);
if (ret < 0) {
- printk(KERN_ERR "AC97 link error\n");
+ printk(KERN_ERR "Failed to reset WM9713: AC97 link error\n");
goto reset_err;
}
@@ -1252,7 +1247,7 @@
wm9713_add_controls(codec);
wm9713_add_widgets(codec);
- ret = snd_soc_register_card(socdev);
+ ret = snd_soc_init_card(socdev);
if (ret < 0)
goto reset_err;
return 0;
@@ -1288,7 +1283,6 @@
snd_soc_free_ac97_codec(codec);
kfree(codec->private_data);
kfree(codec->reg_cache);
- kfree(codec->dai);
kfree(codec);
return 0;
}
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index 8f7e338..b502741 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -17,3 +17,13 @@
help
Say Y if you want to add support for SoC audio on TI
DaVinci EVM platform.
+
+config SND_DAVINCI_SOC_SFFSDR
+ tristate "SoC Audio support for SFFSDR"
+ depends on SND_DAVINCI_SOC && MACH_DAVINCI_SFFSDR
+ select SND_DAVINCI_SOC_I2S
+ select SND_SOC_PCM3008
+ select SFFSDR_FPGA
+ help
+ Say Y if you want to add support for SoC audio on
+ Lyrtech SFFSDR board.
diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile
index ca772e5..ca8bae1 100644
--- a/sound/soc/davinci/Makefile
+++ b/sound/soc/davinci/Makefile
@@ -7,5 +7,7 @@
# DAVINCI Machine Support
snd-soc-evm-objs := davinci-evm.o
+snd-soc-sffsdr-objs := davinci-sffsdr.o
obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o
+obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 9e6062c..01b948b 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -28,6 +28,8 @@
#define EVM_CODEC_CLOCK 22579200
+#define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \
+ SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF)
static int evm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
@@ -37,14 +39,12 @@
int ret = 0;
/* set codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_CBM_CFM);
+ ret = snd_soc_dai_set_fmt(codec_dai, AUDIO_FORMAT);
if (ret < 0)
return ret;
/* set cpu DAI configuration */
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM |
- SND_SOC_DAIFMT_IB_NF);
+ ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT);
if (ret < 0)
return ret;
@@ -128,8 +128,9 @@
};
/* davinci-evm audio machine driver */
-static struct snd_soc_machine snd_soc_machine_evm = {
+static struct snd_soc_card snd_soc_card_evm = {
.name = "DaVinci EVM",
+ .platform = &davinci_soc_platform,
.dai_link = &evm_dai,
.num_links = 1,
};
@@ -142,8 +143,7 @@
/* evm audio subsystem */
static struct snd_soc_device evm_snd_devdata = {
- .machine = &snd_soc_machine_evm,
- .platform = &davinci_soc_platform,
+ .card = &snd_soc_card_evm,
.codec_dev = &soc_codec_dev_aic3x,
.codec_data = &evm_aic3x_setup,
};
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index abb5fed..0fee779 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -59,6 +59,7 @@
#define DAVINCI_MCBSP_PCR_CLKXP (1 << 1)
#define DAVINCI_MCBSP_PCR_FSRP (1 << 2)
#define DAVINCI_MCBSP_PCR_FSXP (1 << 3)
+#define DAVINCI_MCBSP_PCR_SCLKME (1 << 7)
#define DAVINCI_MCBSP_PCR_CLKRM (1 << 8)
#define DAVINCI_MCBSP_PCR_CLKXM (1 << 9)
#define DAVINCI_MCBSP_PCR_FSRM (1 << 10)
@@ -110,17 +111,60 @@
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_platform *platform = socdev->card->platform;
u32 w;
+ int ret;
/* Start the sample generator and enable transmitter/receiver */
w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_GRST, 1);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 1);
- else
- MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_RRST, 1);
davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ /* Stop the DMA to avoid data loss */
+ /* while the transmitter is out of reset to handle XSYNCERR */
+ if (platform->pcm_ops->trigger) {
+ ret = platform->pcm_ops->trigger(substream,
+ SNDRV_PCM_TRIGGER_STOP);
+ if (ret < 0)
+ printk(KERN_DEBUG "Playback DMA stop failed\n");
+ }
+
+ /* Enable the transmitter */
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 1);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+
+ /* wait for any unexpected frame sync error to occur */
+ udelay(100);
+
+ /* Disable the transmitter to clear any outstanding XSYNCERR */
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 0);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+
+ /* Restart the DMA */
+ if (platform->pcm_ops->trigger) {
+ ret = platform->pcm_ops->trigger(substream,
+ SNDRV_PCM_TRIGGER_START);
+ if (ret < 0)
+ printk(KERN_DEBUG "Playback DMA start failed\n");
+ }
+ /* Enable the transmitter */
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 1);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+
+ } else {
+
+ /* Enable the reciever */
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_RRST, 1);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+ }
+
+
/* Start frame sync */
w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_FRST, 1);
@@ -144,7 +188,8 @@
davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
}
-static int davinci_i2s_startup(struct snd_pcm_substream *substream)
+static int davinci_i2s_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -155,61 +200,138 @@
return 0;
}
+#define DEFAULT_BITPERSAMPLE 16
+
static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
unsigned int fmt)
{
struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
- u32 w;
+ unsigned int pcr;
+ unsigned int srgr;
+ unsigned int rcr;
+ unsigned int xcr;
+ srgr = DAVINCI_MCBSP_SRGR_FSGM |
+ DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) |
+ DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1);
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG,
- DAVINCI_MCBSP_PCR_FSXM |
- DAVINCI_MCBSP_PCR_FSRM |
- DAVINCI_MCBSP_PCR_CLKXM |
- DAVINCI_MCBSP_PCR_CLKRM);
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG,
- DAVINCI_MCBSP_SRGR_FSGM);
+ /* cpu is master */
+ pcr = DAVINCI_MCBSP_PCR_FSXM |
+ DAVINCI_MCBSP_PCR_FSRM |
+ DAVINCI_MCBSP_PCR_CLKXM |
+ DAVINCI_MCBSP_PCR_CLKRM;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ /* McBSP CLKR pin is the input for the Sample Rate Generator.
+ * McBSP FSR and FSX are driven by the Sample Rate Generator. */
+ pcr = DAVINCI_MCBSP_PCR_SCLKME |
+ DAVINCI_MCBSP_PCR_FSXM |
+ DAVINCI_MCBSP_PCR_FSRM;
break;
case SND_SOC_DAIFMT_CBM_CFM:
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, 0);
+ /* codec is master */
+ pcr = 0;
break;
default:
+ printk(KERN_ERR "%s:bad master\n", __func__);
+ return -EINVAL;
+ }
+
+ rcr = DAVINCI_MCBSP_RCR_RFRLEN1(1);
+ xcr = DAVINCI_MCBSP_XCR_XFIG | DAVINCI_MCBSP_XCR_XFRLEN1(1);
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_DSP_B:
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ /* Davinci doesn't support TRUE I2S, but some codecs will have
+ * the left and right channels contiguous. This allows
+ * dsp_a mode to be used with an inverted normal frame clk.
+ * If your codec is master and does not have contiguous
+ * channels, then you will have sound on only one channel.
+ * Try using a different mode, or codec as slave.
+ *
+ * The TLV320AIC33 is an example of a codec where this works.
+ * It has a variable bit clock frequency allowing it to have
+ * valid data on every bit clock.
+ *
+ * The TLV320AIC23 is an example of a codec where this does not
+ * work. It has a fixed bit clock frequency with progressively
+ * more empty bit clock slots between channels as the sample
+ * rate is lowered.
+ */
+ fmt ^= SND_SOC_DAIFMT_NB_IF;
+ case SND_SOC_DAIFMT_DSP_A:
+ rcr |= DAVINCI_MCBSP_RCR_RDATDLY(1);
+ xcr |= DAVINCI_MCBSP_XCR_XDATDLY(1);
+ break;
+ default:
+ printk(KERN_ERR "%s:bad format\n", __func__);
return -EINVAL;
}
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
- case SND_SOC_DAIFMT_IB_NF:
- w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG);
- MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_CLKXP |
- DAVINCI_MCBSP_PCR_CLKRP, 1);
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w);
- break;
- case SND_SOC_DAIFMT_NB_IF:
- w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG);
- MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_FSXP |
- DAVINCI_MCBSP_PCR_FSRP, 1);
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w);
+ case SND_SOC_DAIFMT_NB_NF:
+ /* CLKRP Receive clock polarity,
+ * 1 - sampled on rising edge of CLKR
+ * valid on rising edge
+ * CLKXP Transmit clock polarity,
+ * 1 - clocked on falling edge of CLKX
+ * valid on rising edge
+ * FSRP Receive frame sync pol, 0 - active high
+ * FSXP Transmit frame sync pol, 0 - active high
+ */
+ pcr |= (DAVINCI_MCBSP_PCR_CLKXP | DAVINCI_MCBSP_PCR_CLKRP);
break;
case SND_SOC_DAIFMT_IB_IF:
- w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG);
- MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_CLKXP |
- DAVINCI_MCBSP_PCR_CLKRP |
- DAVINCI_MCBSP_PCR_FSXP |
- DAVINCI_MCBSP_PCR_FSRP, 1);
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w);
+ /* CLKRP Receive clock polarity,
+ * 0 - sampled on falling edge of CLKR
+ * valid on falling edge
+ * CLKXP Transmit clock polarity,
+ * 0 - clocked on rising edge of CLKX
+ * valid on falling edge
+ * FSRP Receive frame sync pol, 1 - active low
+ * FSXP Transmit frame sync pol, 1 - active low
+ */
+ pcr |= (DAVINCI_MCBSP_PCR_FSXP | DAVINCI_MCBSP_PCR_FSRP);
break;
- case SND_SOC_DAIFMT_NB_NF:
+ case SND_SOC_DAIFMT_NB_IF:
+ /* CLKRP Receive clock polarity,
+ * 1 - sampled on rising edge of CLKR
+ * valid on rising edge
+ * CLKXP Transmit clock polarity,
+ * 1 - clocked on falling edge of CLKX
+ * valid on rising edge
+ * FSRP Receive frame sync pol, 1 - active low
+ * FSXP Transmit frame sync pol, 1 - active low
+ */
+ pcr |= (DAVINCI_MCBSP_PCR_CLKXP | DAVINCI_MCBSP_PCR_CLKRP |
+ DAVINCI_MCBSP_PCR_FSXP | DAVINCI_MCBSP_PCR_FSRP);
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ /* CLKRP Receive clock polarity,
+ * 0 - sampled on falling edge of CLKR
+ * valid on falling edge
+ * CLKXP Transmit clock polarity,
+ * 0 - clocked on rising edge of CLKX
+ * valid on falling edge
+ * FSRP Receive frame sync pol, 0 - active high
+ * FSXP Transmit frame sync pol, 0 - active high
+ */
break;
default:
return -EINVAL;
}
-
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, pcr);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, rcr);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, xcr);
return 0;
}
static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct davinci_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data;
@@ -219,25 +341,20 @@
u32 w;
/* general line settings */
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG,
- DAVINCI_MCBSP_SPCR_RINTM(3) |
- DAVINCI_MCBSP_SPCR_XINTM(3) |
- DAVINCI_MCBSP_SPCR_FREE);
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG,
- DAVINCI_MCBSP_RCR_RFRLEN1(1) |
- DAVINCI_MCBSP_RCR_RDATDLY(1));
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG,
- DAVINCI_MCBSP_XCR_XFRLEN1(1) |
- DAVINCI_MCBSP_XCR_XDATDLY(1) |
- DAVINCI_MCBSP_XCR_XFIG);
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ w |= DAVINCI_MCBSP_SPCR_RINTM(3) | DAVINCI_MCBSP_SPCR_FREE;
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+ } else {
+ w |= DAVINCI_MCBSP_SPCR_XINTM(3) | DAVINCI_MCBSP_SPCR_FREE;
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+ }
i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS);
- w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SRGR_REG);
+ w = DAVINCI_MCBSP_SRGR_FSGM;
MOD_REG_BIT(w, DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1), 1);
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, w);
i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS);
- w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SRGR_REG);
MOD_REG_BIT(w, DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1), 1);
davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, w);
@@ -260,20 +377,24 @@
return -EINVAL;
}
- w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_RCR_REG);
- MOD_REG_BIT(w, DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) |
- DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length), 1);
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, w);
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_RCR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) |
+ DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length), 1);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, w);
- w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_XCR_REG);
- MOD_REG_BIT(w, DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) |
- DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length), 1);
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, w);
+ } else {
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_XCR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) |
+ DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length), 1);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, w);
+ }
return 0;
}
-static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
{
int ret = 0;
@@ -299,8 +420,8 @@
struct snd_soc_dai *dai)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
- struct snd_soc_machine *machine = socdev->machine;
- struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai;
+ struct snd_soc_card *card = socdev->card;
+ struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai;
struct davinci_mcbsp_dev *dev;
struct resource *mem, *ioarea;
struct evm_snd_platform_data *pdata;
@@ -361,8 +482,8 @@
struct snd_soc_dai *dai)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
- struct snd_soc_machine *machine = socdev->machine;
- struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai;
+ struct snd_soc_card *card = socdev->card;
+ struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai;
struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
struct resource *mem;
@@ -381,7 +502,6 @@
struct snd_soc_dai davinci_i2s_dai = {
.name = "davinci-i2s",
.id = 0,
- .type = SND_SOC_DAI_I2S,
.probe = davinci_i2s_probe,
.remove = davinci_i2s_remove,
.playback = {
@@ -397,13 +517,24 @@
.ops = {
.startup = davinci_i2s_startup,
.trigger = davinci_i2s_trigger,
- .hw_params = davinci_i2s_hw_params,},
- .dai_ops = {
+ .hw_params = davinci_i2s_hw_params,
.set_fmt = davinci_i2s_set_dai_fmt,
},
};
EXPORT_SYMBOL_GPL(davinci_i2s_dai);
+static int __init davinci_i2s_init(void)
+{
+ return snd_soc_register_dai(&davinci_i2s_dai);
+}
+module_init(davinci_i2s_init);
+
+static void __exit davinci_i2s_exit(void)
+{
+ snd_soc_unregister_dai(&davinci_i2s_dai);
+}
+module_exit(davinci_i2s_exit);
+
MODULE_AUTHOR("Vladimir Barinov");
MODULE_DESCRIPTION("TI DAVINCI I2S (McBSP) SoC Interface");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 76feaa6..74abc9b 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -14,6 +14,7 @@
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <linux/dma-mapping.h>
+#include <linux/kernel.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -24,13 +25,6 @@
#include "davinci-pcm.h"
-#define DAVINCI_PCM_DEBUG 0
-#if DAVINCI_PCM_DEBUG
-#define DPRINTK(x...) printk(KERN_DEBUG x)
-#else
-#define DPRINTK(x...)
-#endif
-
static struct snd_pcm_hardware davinci_pcm_hardware = {
.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
@@ -78,8 +72,8 @@
dma_offset = prtd->period * period_size;
dma_pos = runtime->dma_addr + dma_offset;
- DPRINTK("audio_set_dma_params_play channel = %d dma_ptr = %x "
- "period_size=%x\n", lch, dma_pos, period_size);
+ pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d "
+ "dma_ptr = %x period_size=%x\n", lch, dma_pos, period_size);
data_type = prtd->params->data_type;
count = period_size / data_type;
@@ -112,7 +106,7 @@
struct snd_pcm_substream *substream = data;
struct davinci_runtime_data *prtd = substream->runtime->private_data;
- DPRINTK("lch=%d, status=0x%x\n", lch, ch_status);
+ pr_debug("davinci_pcm: lch=%d, status=0x%x\n", lch, ch_status);
if (unlikely(ch_status != DMA_COMPLETE))
return;
@@ -316,8 +310,8 @@
buf->area = dma_alloc_writecombine(pcm->card->dev, size,
&buf->addr, GFP_KERNEL);
- DPRINTK("preallocate_dma_buffer: area=%p, addr=%p, size=%d\n",
- (void *) buf->area, (void *) buf->addr, size);
+ pr_debug("davinci_pcm: preallocate_dma_buffer: area=%p, addr=%p, "
+ "size=%d\n", (void *) buf->area, (void *) buf->addr, size);
if (!buf->area)
return -ENOMEM;
@@ -384,6 +378,18 @@
};
EXPORT_SYMBOL_GPL(davinci_soc_platform);
+static int __init davinci_soc_platform_init(void)
+{
+ return snd_soc_register_platform(&davinci_soc_platform);
+}
+module_init(davinci_soc_platform_init);
+
+static void __exit davinci_soc_platform_exit(void)
+{
+ snd_soc_unregister_platform(&davinci_soc_platform);
+}
+module_exit(davinci_soc_platform_exit);
+
MODULE_AUTHOR("Vladimir Barinov");
MODULE_DESCRIPTION("TI DAVINCI PCM DMA module");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c
new file mode 100644
index 0000000..f67579d
--- /dev/null
+++ b/sound/soc/davinci/davinci-sffsdr.c
@@ -0,0 +1,157 @@
+/*
+ * ASoC driver for Lyrtech SFFSDR board.
+ *
+ * Author: Hugo Villeneuve
+ * Copyright (C) 2008 Lyrtech inc
+ *
+ * Based on ASoC driver for TI DAVINCI EVM platform, original copyright follow:
+ * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.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/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/dma.h>
+#include <asm/plat-sffsdr/sffsdr-fpga.h>
+
+#include <mach/mcbsp.h>
+#include <mach/edma.h>
+
+#include "../codecs/pcm3008.h"
+#include "davinci-pcm.h"
+#include "davinci-i2s.h"
+
+static int sffsdr_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int fs;
+ int ret = 0;
+
+ /* Set cpu DAI configuration:
+ * CLKX and CLKR are the inputs for the Sample Rate Generator.
+ * FSX and FSR are outputs, driven by the sample Rate Generator. */
+ ret = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_RIGHT_J |
+ SND_SOC_DAIFMT_CBM_CFS |
+ SND_SOC_DAIFMT_IB_NF);
+ if (ret < 0)
+ return ret;
+
+ /* Fsref can be 32000, 44100 or 48000. */
+ fs = params_rate(params);
+
+ pr_debug("sffsdr_hw_params: rate = %d Hz\n", fs);
+
+ return sffsdr_fpga_set_codec_fs(fs);
+}
+
+static struct snd_soc_ops sffsdr_ops = {
+ .hw_params = sffsdr_hw_params,
+};
+
+/* davinci-sffsdr digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link sffsdr_dai = {
+ .name = "PCM3008", /* Codec name */
+ .stream_name = "PCM3008 HiFi",
+ .cpu_dai = &davinci_i2s_dai,
+ .codec_dai = &pcm3008_dai,
+ .ops = &sffsdr_ops,
+};
+
+/* davinci-sffsdr audio machine driver */
+static struct snd_soc_card snd_soc_sffsdr = {
+ .name = "DaVinci SFFSDR",
+ .platform = &davinci_soc_platform,
+ .dai_link = &sffsdr_dai,
+ .num_links = 1,
+};
+
+/* sffsdr audio private data */
+static struct pcm3008_setup_data sffsdr_pcm3008_setup = {
+ .dem0_pin = GPIO(45),
+ .dem1_pin = GPIO(46),
+ .pdad_pin = GPIO(47),
+ .pdda_pin = GPIO(38),
+};
+
+/* sffsdr audio subsystem */
+static struct snd_soc_device sffsdr_snd_devdata = {
+ .card = &snd_soc_sffsdr,
+ .codec_dev = &soc_codec_dev_pcm3008,
+ .codec_data = &sffsdr_pcm3008_setup,
+};
+
+static struct resource sffsdr_snd_resources[] = {
+ {
+ .start = DAVINCI_MCBSP_BASE,
+ .end = DAVINCI_MCBSP_BASE + SZ_8K - 1,
+ .flags = IORESOURCE_MEM,
+ },
+};
+
+static struct evm_snd_platform_data sffsdr_snd_data = {
+ .tx_dma_ch = DAVINCI_DMA_MCBSP_TX,
+ .rx_dma_ch = DAVINCI_DMA_MCBSP_RX,
+};
+
+static struct platform_device *sffsdr_snd_device;
+
+static int __init sffsdr_init(void)
+{
+ int ret;
+
+ sffsdr_snd_device = platform_device_alloc("soc-audio", 0);
+ if (!sffsdr_snd_device) {
+ printk(KERN_ERR "platform device allocation failed\n");
+ return -ENOMEM;
+ }
+
+ platform_set_drvdata(sffsdr_snd_device, &sffsdr_snd_devdata);
+ sffsdr_snd_devdata.dev = &sffsdr_snd_device->dev;
+ sffsdr_snd_device->dev.platform_data = &sffsdr_snd_data;
+
+ ret = platform_device_add_resources(sffsdr_snd_device,
+ sffsdr_snd_resources,
+ ARRAY_SIZE(sffsdr_snd_resources));
+ if (ret) {
+ printk(KERN_ERR "platform device add ressources failed\n");
+ goto error;
+ }
+
+ ret = platform_device_add(sffsdr_snd_device);
+ if (ret)
+ goto error;
+
+ return ret;
+
+error:
+ platform_device_put(sffsdr_snd_device);
+ return ret;
+}
+
+static void __exit sffsdr_exit(void)
+{
+ platform_device_unregister(sffsdr_snd_device);
+}
+
+module_init(sffsdr_init);
+module_exit(sffsdr_exit);
+
+MODULE_AUTHOR("Hugo Villeneuve");
+MODULE_DESCRIPTION("Lyrtech SFFSDR ASoC driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 8d73edc..95c12b2 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -20,7 +20,7 @@
config SND_SOC_MPC5200_I2S
tristate "Freescale MPC5200 PSC in I2S mode driver"
- depends on SND_SOC && PPC_MPC52xx && PPC_BESTCOMM
+ depends on PPC_MPC52xx && PPC_BESTCOMM
select SND_SOC_OF_SIMPLE
select PPC_BESTCOMM_GEN_BD
help
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index d2d3da9..64993ed 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -284,7 +284,7 @@
* fsl_dma_new: initialize this PCM driver.
*
* This function is called when the codec driver calls snd_soc_new_pcms(),
- * once for each .dai_link in the machine driver's snd_soc_machine
+ * once for each .dai_link in the machine driver's snd_soc_card
* structure.
*/
static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
@@ -853,6 +853,18 @@
}
EXPORT_SYMBOL_GPL(fsl_dma_configure);
+static int __init fsl_soc_platform_init(void)
+{
+ return snd_soc_register_platform(&fsl_soc_platform);
+}
+module_init(fsl_soc_platform_init);
+
+static void __exit fsl_soc_platform_exit(void)
+{
+ snd_soc_unregister_platform(&fsl_soc_platform);
+}
+module_exit(fsl_soc_platform_exit);
+
MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
MODULE_DESCRIPTION("Freescale Elo DMA ASoC PCM module");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 157a789..c6d6eb7 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -266,7 +266,8 @@
* If this is the first stream open, then grab the IRQ and program most of
* the SSI registers.
*/
-static int fsl_ssi_startup(struct snd_pcm_substream *substream)
+static int fsl_ssi_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
@@ -411,7 +412,8 @@
* Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the
* clock master.
*/
-static int fsl_ssi_prepare(struct snd_pcm_substream *substream)
+static int fsl_ssi_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -441,7 +443,8 @@
* The DMA channel is in external master start and pause mode, which
* means the SSI completely controls the flow of data.
*/
-static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd)
+static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
@@ -490,7 +493,8 @@
*
* Shutdown the SSI if there are no other substreams open.
*/
-static void fsl_ssi_shutdown(struct snd_pcm_substream *substream)
+static void fsl_ssi_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
@@ -578,8 +582,6 @@
.prepare = fsl_ssi_prepare,
.shutdown = fsl_ssi_shutdown,
.trigger = fsl_ssi_trigger,
- },
- .dai_ops = {
.set_sysclk = fsl_ssi_set_sysclk,
.set_fmt = fsl_ssi_set_fmt,
},
@@ -671,6 +673,14 @@
fsl_ssi_dai->private_data = ssi_private;
fsl_ssi_dai->name = ssi_private->name;
fsl_ssi_dai->id = ssi_info->id;
+ fsl_ssi_dai->dev = ssi_info->dev;
+
+ ret = snd_soc_register_dai(fsl_ssi_dai);
+ if (ret != 0) {
+ dev_err(ssi_info->dev, "failed to register DAI: %d\n", ret);
+ kfree(fsl_ssi_dai);
+ return NULL;
+ }
return fsl_ssi_dai;
}
@@ -688,6 +698,8 @@
device_remove_file(ssi_private->dev, &ssi_private->dev_attr);
+ snd_soc_unregister_dai(&ssi_private->cpu_dai);
+
kfree(ssi_private);
}
EXPORT_SYMBOL_GPL(fsl_ssi_destroy_dai);
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 94a02ea..9eb1ce1 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -187,7 +187,8 @@
* If this is the first stream open, then grab the IRQ and program most of
* the PSC registers.
*/
-static int psc_i2s_startup(struct snd_pcm_substream *substream)
+static int psc_i2s_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
@@ -220,7 +221,8 @@
}
static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
@@ -256,7 +258,8 @@
return 0;
}
-static int psc_i2s_hw_free(struct snd_pcm_substream *substream)
+static int psc_i2s_hw_free(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
snd_pcm_set_runtime_buffer(substream, NULL);
return 0;
@@ -268,7 +271,8 @@
* This function is called by ALSA to start, stop, pause, and resume the DMA
* transfer of data.
*/
-static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
@@ -383,7 +387,8 @@
*
* Shutdown the PSC if there are no other substreams open.
*/
-static void psc_i2s_shutdown(struct snd_pcm_substream *substream)
+static void psc_i2s_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
@@ -464,7 +469,6 @@
* psc_i2s_dai_template: template CPU Digital Audio Interface
*/
static struct snd_soc_dai psc_i2s_dai_template = {
- .type = SND_SOC_DAI_I2S,
.playback = {
.channels_min = 2,
.channels_max = 2,
@@ -483,8 +487,6 @@
.hw_free = psc_i2s_hw_free,
.shutdown = psc_i2s_shutdown,
.trigger = psc_i2s_trigger,
- },
- .dai_ops = {
.set_sysclk = psc_i2s_set_sysclk,
.set_fmt = psc_i2s_set_fmt,
},
@@ -826,6 +828,8 @@
if (rc)
dev_info(psc_i2s->dev, "error creating sysfs files\n");
+ snd_soc_register_platform(&psc_i2s_pcm_soc_platform);
+
/* Tell the ASoC OF helpers about it */
of_snd_soc_register_platform(&psc_i2s_pcm_soc_platform, op->node,
&psc_i2s->dai);
@@ -839,6 +843,8 @@
dev_dbg(&op->dev, "psc_i2s_remove()\n");
+ snd_soc_unregister_platform(&psc_i2s_pcm_soc_platform);
+
bcom_gen_bd_rx_release(psc_i2s->capture.bcom_task);
bcom_gen_bd_tx_release(psc_i2s->playback.bcom_task);
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index 94f89de..bcec3f6 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -29,7 +29,7 @@
struct mpc8610_hpcd_data {
struct snd_soc_device sound_devdata;
struct snd_soc_dai_link dai;
- struct snd_soc_machine machine;
+ struct snd_soc_card machine;
unsigned int dai_format;
unsigned int codec_clk_direction;
unsigned int cpu_clk_direction;
@@ -185,7 +185,7 @@
/**
* mpc8610_hpcd_machine: ASoC machine data
*/
-static struct snd_soc_machine mpc8610_hpcd_machine = {
+static struct snd_soc_card mpc8610_hpcd_machine = {
.probe = mpc8610_hpcd_machine_probe,
.remove = mpc8610_hpcd_machine_remove,
.name = "MPC8610 HPCD",
@@ -465,9 +465,9 @@
goto error;
}
- machine_data->sound_devdata.machine = &mpc8610_hpcd_machine;
+ machine_data->sound_devdata.card = &mpc8610_hpcd_machine;
machine_data->sound_devdata.codec_dev = &soc_codec_device_cs4270;
- machine_data->sound_devdata.platform = &fsl_soc_platform;
+ machine_data->machine.platform = &fsl_soc_platform;
sound_device->dev.platform_data = machine_data;
diff --git a/sound/soc/fsl/soc-of-simple.c b/sound/soc/fsl/soc-of-simple.c
index 0382fda..8bc5cd9 100644
--- a/sound/soc/fsl/soc-of-simple.c
+++ b/sound/soc/fsl/soc-of-simple.c
@@ -31,7 +31,7 @@
int id;
struct list_head list;
struct snd_soc_device device;
- struct snd_soc_machine machine;
+ struct snd_soc_card card;
struct snd_soc_dai_link dai_link;
struct platform_device *pdev;
struct device_node *platform_node;
@@ -58,9 +58,9 @@
/* Initialize the structure and add it to the global list */
of_soc->codec_node = codec_node;
of_soc->id = of_snd_soc_next_index++;
- of_soc->machine.dai_link = &of_soc->dai_link;
- of_soc->machine.num_links = 1;
- of_soc->device.machine = &of_soc->machine;
+ of_soc->card.dai_link = &of_soc->dai_link;
+ of_soc->card.num_links = 1;
+ of_soc->device.card = &of_soc->card;
of_soc->dai_link.ops = &of_snd_soc_ops;
list_add(&of_soc->list, &of_snd_soc_device_list);
@@ -158,8 +158,8 @@
of_soc->platform_node = node;
of_soc->dai_link.cpu_dai = cpu_dai;
- of_soc->device.platform = platform;
- of_soc->machine.name = of_soc->dai_link.cpu_dai->name;
+ of_soc->card.platform = platform;
+ of_soc->card.name = of_soc->dai_link.cpu_dai->name;
/* Now try to register the SoC device */
of_snd_soc_register_device(of_soc);
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 8b7766b..a7b1d77 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -1,6 +1,6 @@
config SND_OMAP_SOC
tristate "SoC Audio for the Texas Instruments OMAP chips"
- depends on ARCH_OMAP && SND_SOC
+ depends on ARCH_OMAP
config SND_OMAP_SOC_MCBSP
tristate
@@ -21,3 +21,36 @@
select SND_SOC_TLV320AIC23
help
Say Y if you want to add support for SoC audio on osk5912.
+
+config SND_OMAP_SOC_OVERO
+ tristate "SoC Audio support for Gumstix Overo"
+ depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OVERO
+ select SND_OMAP_SOC_MCBSP
+ select SND_SOC_TWL4030
+ help
+ Say Y if you want to add support for SoC audio on the Gumstix Overo.
+
+config SND_OMAP_SOC_OMAP2EVM
+ tristate "SoC Audio support for OMAP2EVM board"
+ depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP2EVM
+ select SND_OMAP_SOC_MCBSP
+ select SND_SOC_TWL4030
+ help
+ Say Y if you want to add support for SoC audio on the omap2evm board.
+
+config SND_OMAP_SOC_SDP3430
+ tristate "SoC Audio support for Texas Instruments SDP3430"
+ depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP
+ select SND_OMAP_SOC_MCBSP
+ select SND_SOC_TWL4030
+ help
+ Say Y if you want to add support for SoC audio on Texas Instruments
+ SDP3430.
+
+config SND_OMAP_SOC_OMAP3_PANDORA
+ tristate "SoC Audio support for OMAP3 Pandora"
+ depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_PANDORA
+ select SND_OMAP_SOC_MCBSP
+ select SND_SOC_TWL4030
+ help
+ Say Y if you want to add support for SoC audio on the OMAP3 Pandora.
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index e09d1f2..76fedd9 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -8,6 +8,14 @@
# OMAP Machine Support
snd-soc-n810-objs := n810.o
snd-soc-osk5912-objs := osk5912.o
+snd-soc-overo-objs := overo.o
+snd-soc-omap2evm-objs := omap2evm.o
+snd-soc-sdp3430-objs := sdp3430.o
+snd-soc-omap3pandora-objs := omap3pandora.o
obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
+obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
+obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o
+obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
+obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index fae3ad3..25593fe 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -70,9 +70,13 @@
static int n810_startup(struct snd_pcm_substream *substream)
{
+ struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->socdev->codec;
+ snd_pcm_hw_constraint_minmax(runtime,
+ SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
+
n810_ext_control(codec);
return clk_enable(sys_clkout2);
}
@@ -282,8 +286,9 @@
};
/* Audio machine driver */
-static struct snd_soc_machine snd_soc_machine_n810 = {
+static struct snd_soc_card snd_soc_n810 = {
.name = "N810",
+ .platform = &omap_soc_platform,
.dai_link = &n810_dai,
.num_links = 1,
};
@@ -298,8 +303,7 @@
/* Audio subsystem */
static struct snd_soc_device n810_snd_devdata = {
- .machine = &snd_soc_machine_n810,
- .platform = &omap_soc_platform,
+ .card = &snd_soc_n810,
.codec_dev = &soc_codec_dev_aic3x,
.codec_data = &n810_aic33_setup,
};
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 8485a8a..ec5e18a 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -36,9 +36,7 @@
#include "omap-mcbsp.h"
#include "omap-pcm.h"
-#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_44100 | \
- SNDRV_PCM_RATE_48000 | \
- SNDRV_PCM_RATE_KNOT)
+#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_96000)
struct omap_mcbsp_data {
unsigned int bus_id;
@@ -140,7 +138,8 @@
static const unsigned long omap34xx_mcbsp_port[][2] = {};
#endif
-static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream)
+static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -153,7 +152,8 @@
return err;
}
-static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream)
+static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -165,7 +165,8 @@
}
}
-static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd)
+static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -194,14 +195,15 @@
}
static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id;
- int wlen;
+ int wlen, channels;
unsigned long port;
if (cpu_class_is_omap1()) {
@@ -230,12 +232,17 @@
return 0;
}
- switch (params_channels(params)) {
+ channels = params_channels(params);
+ switch (channels) {
case 2:
- /* Set 1 word per (McBPSP) frame and use dual-phase frames */
- regs->rcr2 |= RFRLEN2(1 - 1) | RPHASE;
+ /* Use dual-phase frames */
+ regs->rcr2 |= RPHASE;
+ regs->xcr2 |= XPHASE;
+ case 1:
+ /* Set 1 word per (McBSP) frame */
+ regs->rcr2 |= RFRLEN2(1 - 1);
regs->rcr1 |= RFRLEN1(1 - 1);
- regs->xcr2 |= XFRLEN2(1 - 1) | XPHASE;
+ regs->xcr2 |= XFRLEN2(1 - 1);
regs->xcr1 |= XFRLEN1(1 - 1);
break;
default:
@@ -263,9 +270,9 @@
regs->srgr2 |= FPER(wlen * 2 - 1);
regs->srgr1 |= FWID(wlen - 1);
break;
- case SND_SOC_DAIFMT_DSP_A:
- regs->srgr2 |= FPER(wlen * 2 - 1);
- regs->srgr1 |= FWID(wlen * 2 - 2);
+ case SND_SOC_DAIFMT_DSP_B:
+ regs->srgr2 |= FPER(wlen * channels - 1);
+ regs->srgr1 |= FWID(wlen * channels - 2);
break;
}
@@ -302,7 +309,7 @@
regs->rcr2 |= RDATDLY(1);
regs->xcr2 |= XDATDLY(1);
break;
- case SND_SOC_DAIFMT_DSP_A:
+ case SND_SOC_DAIFMT_DSP_B:
/* 0-bit data delay */
regs->rcr2 |= RDATDLY(0);
regs->xcr2 |= XDATDLY(0);
@@ -452,17 +459,16 @@
#define OMAP_MCBSP_DAI_BUILDER(link_id) \
{ \
- .name = "omap-mcbsp-dai-(link_id)", \
+ .name = "omap-mcbsp-dai-"#link_id, \
.id = (link_id), \
- .type = SND_SOC_DAI_I2S, \
.playback = { \
- .channels_min = 2, \
+ .channels_min = 1, \
.channels_max = 2, \
.rates = OMAP_MCBSP_RATES, \
.formats = SNDRV_PCM_FMTBIT_S16_LE, \
}, \
.capture = { \
- .channels_min = 2, \
+ .channels_min = 1, \
.channels_max = 2, \
.rates = OMAP_MCBSP_RATES, \
.formats = SNDRV_PCM_FMTBIT_S16_LE, \
@@ -472,8 +478,6 @@
.shutdown = omap_mcbsp_dai_shutdown, \
.trigger = omap_mcbsp_dai_trigger, \
.hw_params = omap_mcbsp_dai_hw_params, \
- }, \
- .dai_ops = { \
.set_fmt = omap_mcbsp_dai_set_dai_fmt, \
.set_clkdiv = omap_mcbsp_dai_set_clkdiv, \
.set_sysclk = omap_mcbsp_dai_set_dai_sysclk, \
@@ -495,6 +499,19 @@
EXPORT_SYMBOL_GPL(omap_mcbsp_dai);
+static int __init snd_omap_mcbsp_init(void)
+{
+ return snd_soc_register_dais(omap_mcbsp_dai,
+ ARRAY_SIZE(omap_mcbsp_dai));
+}
+module_init(snd_omap_mcbsp_init);
+
+static void __exit snd_omap_mcbsp_exit(void)
+{
+ snd_soc_unregister_dais(omap_mcbsp_dai, ARRAY_SIZE(omap_mcbsp_dai));
+}
+module_exit(snd_omap_mcbsp_exit);
+
MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
MODULE_DESCRIPTION("OMAP I2S SoC Interface");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index acd68ef..b0362df 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -97,7 +97,7 @@
prtd->dma_data = dma_data;
err = omap_request_dma(dma_data->dma_req, dma_data->name,
omap_pcm_dma_irq, substream, &prtd->dma_ch);
- if (!err & !cpu_is_omap1510()) {
+ if (!err && !cpu_is_omap1510()) {
/*
* Link channel with itself so DMA doesn't need any
* reprogramming while looping the buffer
@@ -354,6 +354,18 @@
};
EXPORT_SYMBOL_GPL(omap_soc_platform);
+static int __init omap_soc_platform_init(void)
+{
+ return snd_soc_register_platform(&omap_soc_platform);
+}
+module_init(omap_soc_platform_init);
+
+static void __exit omap_soc_platform_exit(void)
+{
+ snd_soc_unregister_platform(&omap_soc_platform);
+}
+module_exit(omap_soc_platform_exit);
+
MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
MODULE_DESCRIPTION("OMAP PCM DMA module");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c
new file mode 100644
index 0000000..0c2322d
--- /dev/null
+++ b/sound/soc/omap/omap2evm.c
@@ -0,0 +1,151 @@
+/*
+ * omap2evm.c -- SoC audio machine driver for omap2evm board
+ *
+ * Author: Arun KS <arunks@mistralsolutions.com>
+ *
+ * Based on sound/soc/omap/overo.c by Steve Sakoman
+ *
+ * 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.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+static int omap2evm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int ret;
+
+ /* Set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set codec DAI configuration\n");
+ return ret;
+ }
+
+ /* Set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set cpu DAI configuration\n");
+ return ret;
+ }
+
+ /* Set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set codec system clock\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops omap2evm_ops = {
+ .hw_params = omap2evm_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link omap2evm_dai = {
+ .name = "TWL4030",
+ .stream_name = "TWL4030",
+ .cpu_dai = &omap_mcbsp_dai[0],
+ .codec_dai = &twl4030_dai,
+ .ops = &omap2evm_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_omap2evm = {
+ .name = "omap2evm",
+ .platform = &omap_soc_platform,
+ .dai_link = &omap2evm_dai,
+ .num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device omap2evm_snd_devdata = {
+ .card = &snd_soc_omap2evm,
+ .codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *omap2evm_snd_device;
+
+static int __init omap2evm_soc_init(void)
+{
+ int ret;
+
+ if (!machine_is_omap2evm()) {
+ pr_debug("Not omap2evm!\n");
+ return -ENODEV;
+ }
+ printk(KERN_INFO "omap2evm SoC init\n");
+
+ omap2evm_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!omap2evm_snd_device) {
+ printk(KERN_ERR "Platform device allocation failed\n");
+ return -ENOMEM;
+ }
+
+ platform_set_drvdata(omap2evm_snd_device, &omap2evm_snd_devdata);
+ omap2evm_snd_devdata.dev = &omap2evm_snd_device->dev;
+ *(unsigned int *)omap2evm_dai.cpu_dai->private_data = 1; /* McBSP2 */
+
+ ret = platform_device_add(omap2evm_snd_device);
+ if (ret)
+ goto err1;
+
+ return 0;
+
+err1:
+ printk(KERN_ERR "Unable to add platform device\n");
+ platform_device_put(omap2evm_snd_device);
+
+ return ret;
+}
+module_init(omap2evm_soc_init);
+
+static void __exit omap2evm_soc_exit(void)
+{
+ platform_device_unregister(omap2evm_snd_device);
+}
+module_exit(omap2evm_soc_exit);
+
+MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
+MODULE_DESCRIPTION("ALSA SoC omap2evm");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c
new file mode 100644
index 0000000..fd24a4a
--- /dev/null
+++ b/sound/soc/omap/omap3beagle.c
@@ -0,0 +1,149 @@
+/*
+ * omap3beagle.c -- SoC audio for OMAP3 Beagle
+ *
+ * Author: Steve Sakoman <steve@sakoman.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.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+static int omap3beagle_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int ret;
+
+ /* Set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set codec DAI configuration\n");
+ return ret;
+ }
+
+ /* Set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set cpu DAI configuration\n");
+ return ret;
+ }
+
+ /* Set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set codec system clock\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops omap3beagle_ops = {
+ .hw_params = omap3beagle_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link omap3beagle_dai = {
+ .name = "TWL4030",
+ .stream_name = "TWL4030",
+ .cpu_dai = &omap_mcbsp_dai[0],
+ .codec_dai = &twl4030_dai,
+ .ops = &omap3beagle_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_omap3beagle = {
+ .name = "omap3beagle",
+ .platform = &omap_soc_platform,
+ .dai_link = &omap3beagle_dai,
+ .num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device omap3beagle_snd_devdata = {
+ .card = &snd_soc_omap3beagle,
+ .codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *omap3beagle_snd_device;
+
+static int __init omap3beagle_soc_init(void)
+{
+ int ret;
+
+ if (!machine_is_omap3_beagle()) {
+ pr_debug("Not OMAP3 Beagle!\n");
+ return -ENODEV;
+ }
+ pr_info("OMAP3 Beagle SoC init\n");
+
+ omap3beagle_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!omap3beagle_snd_device) {
+ printk(KERN_ERR "Platform device allocation failed\n");
+ return -ENOMEM;
+ }
+
+ platform_set_drvdata(omap3beagle_snd_device, &omap3beagle_snd_devdata);
+ omap3beagle_snd_devdata.dev = &omap3beagle_snd_device->dev;
+ *(unsigned int *)omap3beagle_dai.cpu_dai->private_data = 1; /* McBSP2 */
+
+ ret = platform_device_add(omap3beagle_snd_device);
+ if (ret)
+ goto err1;
+
+ return 0;
+
+err1:
+ printk(KERN_ERR "Unable to add platform device\n");
+ platform_device_put(omap3beagle_snd_device);
+
+ return ret;
+}
+
+static void __exit omap3beagle_soc_exit(void)
+{
+ platform_device_unregister(omap3beagle_snd_device);
+}
+
+module_init(omap3beagle_soc_init);
+module_exit(omap3beagle_soc_exit);
+
+MODULE_AUTHOR("Steve Sakoman <steve@sakoman.com>");
+MODULE_DESCRIPTION("ALSA SoC OMAP3 Beagle");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
new file mode 100644
index 0000000..bd91594
--- /dev/null
+++ b/sound/soc/omap/omap3pandora.c
@@ -0,0 +1,311 @@
+/*
+ * omap3pandora.c -- SoC audio for Pandora Handheld Console
+ *
+ * Author: Gražvydas Ignotas <notasas@gmail.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.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+#include <linux/delay.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+#define OMAP3_PANDORA_DAC_POWER_GPIO 118
+#define OMAP3_PANDORA_AMP_POWER_GPIO 14
+
+#define PREFIX "ASoC omap3pandora: "
+
+static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai,
+ struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+ int ret;
+
+ /* Set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, fmt);
+ if (ret < 0) {
+ pr_err(PREFIX "can't set codec DAI configuration\n");
+ return ret;
+ }
+
+ /* Set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
+ if (ret < 0) {
+ pr_err(PREFIX "can't set cpu DAI configuration\n");
+ return ret;
+ }
+
+ /* Set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ pr_err(PREFIX "can't set codec system clock\n");
+ return ret;
+ }
+
+ /* Set McBSP clock to external */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, 0,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ pr_err(PREFIX "can't set cpu system clock\n");
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_MCBSP_CLKGDV, 8);
+ if (ret < 0) {
+ pr_err(PREFIX "can't set SRG clock divider\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static int omap3pandora_out_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ return omap3pandora_cmn_hw_params(codec_dai, cpu_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_IB_NF |
+ SND_SOC_DAIFMT_CBS_CFS);
+}
+
+static int omap3pandora_in_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ return omap3pandora_cmn_hw_params(codec_dai, cpu_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS);
+}
+
+static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ if (SND_SOC_DAPM_EVENT_ON(event)) {
+ gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 1);
+ gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 1);
+ } else {
+ gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 0);
+ mdelay(1);
+ gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 0);
+ }
+
+ return 0;
+}
+
+/*
+ * Audio paths on Pandora board:
+ *
+ * |O| ---> PCM DAC +-> AMP -> Headphone Jack
+ * |M| A +--------> Line Out
+ * |A| <~~clk~~+
+ * |P| <--- TWL4030 <--------- Line In and MICs
+ */
+static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = {
+ SND_SOC_DAPM_DAC("PCM DAC", "Playback", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_PGA_E("Headphone Amplifier", SND_SOC_NOPM,
+ 0, 0, NULL, 0, omap3pandora_hp_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_LINE("Line Out", NULL),
+};
+
+static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = {
+ SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
+ SND_SOC_DAPM_MIC("Mic (external)", NULL),
+ SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route omap3pandora_out_map[] = {
+ {"Headphone Amplifier", NULL, "PCM DAC"},
+ {"Line Out", NULL, "PCM DAC"},
+ {"Headphone Jack", NULL, "Headphone Amplifier"},
+};
+
+static const struct snd_soc_dapm_route omap3pandora_in_map[] = {
+ {"INL", NULL, "Line In"},
+ {"INR", NULL, "Line In"},
+ {"INL", NULL, "Mic (Internal)"},
+ {"INR", NULL, "Mic (external)"},
+};
+
+static int omap3pandora_out_init(struct snd_soc_codec *codec)
+{
+ int ret;
+
+ ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets,
+ ARRAY_SIZE(omap3pandora_out_dapm_widgets));
+ if (ret < 0)
+ return ret;
+
+ snd_soc_dapm_add_routes(codec, omap3pandora_out_map,
+ ARRAY_SIZE(omap3pandora_out_map));
+
+ return snd_soc_dapm_sync(codec);
+}
+
+static int omap3pandora_in_init(struct snd_soc_codec *codec)
+{
+ int ret;
+
+ ret = snd_soc_dapm_new_controls(codec, omap3pandora_in_dapm_widgets,
+ ARRAY_SIZE(omap3pandora_in_dapm_widgets));
+ if (ret < 0)
+ return ret;
+
+ snd_soc_dapm_add_routes(codec, omap3pandora_in_map,
+ ARRAY_SIZE(omap3pandora_in_map));
+
+ return snd_soc_dapm_sync(codec);
+}
+
+static struct snd_soc_ops omap3pandora_out_ops = {
+ .hw_params = omap3pandora_out_hw_params,
+};
+
+static struct snd_soc_ops omap3pandora_in_ops = {
+ .hw_params = omap3pandora_in_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link omap3pandora_dai[] = {
+ {
+ .name = "PCM1773",
+ .stream_name = "HiFi Out",
+ .cpu_dai = &omap_mcbsp_dai[0],
+ .codec_dai = &twl4030_dai,
+ .ops = &omap3pandora_out_ops,
+ .init = omap3pandora_out_init,
+ }, {
+ .name = "TWL4030",
+ .stream_name = "Line/Mic In",
+ .cpu_dai = &omap_mcbsp_dai[1],
+ .codec_dai = &twl4030_dai,
+ .ops = &omap3pandora_in_ops,
+ .init = omap3pandora_in_init,
+ }
+};
+
+/* SoC card */
+static struct snd_soc_card snd_soc_card_omap3pandora = {
+ .name = "omap3pandora",
+ .platform = &omap_soc_platform,
+ .dai_link = omap3pandora_dai,
+ .num_links = ARRAY_SIZE(omap3pandora_dai),
+};
+
+/* Audio subsystem */
+static struct snd_soc_device omap3pandora_snd_data = {
+ .card = &snd_soc_card_omap3pandora,
+ .codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *omap3pandora_snd_device;
+
+static int __init omap3pandora_soc_init(void)
+{
+ int ret;
+
+ if (!machine_is_omap3_pandora()) {
+ pr_debug(PREFIX "Not OMAP3 Pandora\n");
+ return -ENODEV;
+ }
+ pr_info("OMAP3 Pandora SoC init\n");
+
+ ret = gpio_request(OMAP3_PANDORA_DAC_POWER_GPIO, "dac_power");
+ if (ret) {
+ pr_err(PREFIX "Failed to get DAC power GPIO\n");
+ return ret;
+ }
+
+ ret = gpio_direction_output(OMAP3_PANDORA_DAC_POWER_GPIO, 0);
+ if (ret) {
+ pr_err(PREFIX "Failed to set DAC power GPIO direction\n");
+ goto fail0;
+ }
+
+ ret = gpio_request(OMAP3_PANDORA_AMP_POWER_GPIO, "amp_power");
+ if (ret) {
+ pr_err(PREFIX "Failed to get amp power GPIO\n");
+ goto fail0;
+ }
+
+ ret = gpio_direction_output(OMAP3_PANDORA_AMP_POWER_GPIO, 0);
+ if (ret) {
+ pr_err(PREFIX "Failed to set amp power GPIO direction\n");
+ goto fail1;
+ }
+
+ omap3pandora_snd_device = platform_device_alloc("soc-audio", -1);
+ if (omap3pandora_snd_device == NULL) {
+ pr_err(PREFIX "Platform device allocation failed\n");
+ ret = -ENOMEM;
+ goto fail1;
+ }
+
+ platform_set_drvdata(omap3pandora_snd_device, &omap3pandora_snd_data);
+ omap3pandora_snd_data.dev = &omap3pandora_snd_device->dev;
+ *(unsigned int *)omap_mcbsp_dai[0].private_data = 1; /* McBSP2 */
+ *(unsigned int *)omap_mcbsp_dai[1].private_data = 3; /* McBSP4 */
+
+ ret = platform_device_add(omap3pandora_snd_device);
+ if (ret) {
+ pr_err(PREFIX "Unable to add platform device\n");
+ goto fail2;
+ }
+
+ return 0;
+
+fail2:
+ platform_device_put(omap3pandora_snd_device);
+fail1:
+ gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO);
+fail0:
+ gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO);
+ return ret;
+}
+module_init(omap3pandora_soc_init);
+
+static void __exit omap3pandora_soc_exit(void)
+{
+ platform_device_unregister(omap3pandora_snd_device);
+ gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO);
+ gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO);
+}
+module_exit(omap3pandora_soc_exit);
+
+MODULE_AUTHOR("Grazvydas Ignotas <notasas@gmail.com>");
+MODULE_DESCRIPTION("ALSA SoC OMAP3 Pandora");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
index 0fe7337..cd41a94 100644
--- a/sound/soc/omap/osk5912.c
+++ b/sound/soc/omap/osk5912.c
@@ -61,7 +61,7 @@
/* Set codec DAI configuration */
err = snd_soc_dai_set_fmt(codec_dai,
- SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_DSP_B |
SND_SOC_DAIFMT_NB_IF |
SND_SOC_DAIFMT_CBM_CFM);
if (err < 0) {
@@ -71,7 +71,7 @@
/* Set cpu DAI configuration */
err = snd_soc_dai_set_fmt(cpu_dai,
- SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_DSP_B |
SND_SOC_DAIFMT_NB_IF |
SND_SOC_DAIFMT_CBM_CFM);
if (err < 0) {
@@ -143,16 +143,16 @@
};
/* Audio machine driver */
-static struct snd_soc_machine snd_soc_machine_osk = {
+static struct snd_soc_card snd_soc_card_osk = {
.name = "OSK5912",
+ .platform = &omap_soc_platform,
.dai_link = &osk_dai,
.num_links = 1,
};
/* Audio subsystem */
static struct snd_soc_device osk_snd_devdata = {
- .machine = &snd_soc_machine_osk,
- .platform = &omap_soc_platform,
+ .card = &snd_soc_card_osk,
.codec_dev = &soc_codec_dev_tlv320aic23,
};
diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c
new file mode 100644
index 0000000..a72dc4e
--- /dev/null
+++ b/sound/soc/omap/overo.c
@@ -0,0 +1,148 @@
+/*
+ * overo.c -- SoC audio for Gumstix Overo
+ *
+ * Author: Steve Sakoman <steve@sakoman.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.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+static int overo_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int ret;
+
+ /* Set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set codec DAI configuration\n");
+ return ret;
+ }
+
+ /* Set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set cpu DAI configuration\n");
+ return ret;
+ }
+
+ /* Set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set codec system clock\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops overo_ops = {
+ .hw_params = overo_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link overo_dai = {
+ .name = "TWL4030",
+ .stream_name = "TWL4030",
+ .cpu_dai = &omap_mcbsp_dai[0],
+ .codec_dai = &twl4030_dai,
+ .ops = &overo_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_card_overo = {
+ .name = "overo",
+ .platform = &omap_soc_platform,
+ .dai_link = &overo_dai,
+ .num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device overo_snd_devdata = {
+ .card = &snd_soc_card_overo,
+ .codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *overo_snd_device;
+
+static int __init overo_soc_init(void)
+{
+ int ret;
+
+ if (!machine_is_overo()) {
+ pr_debug("Not Overo!\n");
+ return -ENODEV;
+ }
+ printk(KERN_INFO "overo SoC init\n");
+
+ overo_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!overo_snd_device) {
+ printk(KERN_ERR "Platform device allocation failed\n");
+ return -ENOMEM;
+ }
+
+ platform_set_drvdata(overo_snd_device, &overo_snd_devdata);
+ overo_snd_devdata.dev = &overo_snd_device->dev;
+ *(unsigned int *)overo_dai.cpu_dai->private_data = 1; /* McBSP2 */
+
+ ret = platform_device_add(overo_snd_device);
+ if (ret)
+ goto err1;
+
+ return 0;
+
+err1:
+ printk(KERN_ERR "Unable to add platform device\n");
+ platform_device_put(overo_snd_device);
+
+ return ret;
+}
+module_init(overo_soc_init);
+
+static void __exit overo_soc_exit(void)
+{
+ platform_device_unregister(overo_snd_device);
+}
+module_exit(overo_soc_exit);
+
+MODULE_AUTHOR("Steve Sakoman <steve@sakoman.com>");
+MODULE_DESCRIPTION("ALSA SoC overo");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
new file mode 100644
index 0000000..ad97836
--- /dev/null
+++ b/sound/soc/omap/sdp3430.c
@@ -0,0 +1,152 @@
+/*
+ * sdp3430.c -- SoC audio for TI OMAP3430 SDP
+ *
+ * Author: Misael Lopez Cruz <x0052729@ti.com>
+ *
+ * Based on:
+ * Author: Steve Sakoman <steve@sakoman.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.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+static int sdp3430_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int ret;
+
+ /* Set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set codec DAI configuration\n");
+ return ret;
+ }
+
+ /* Set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set cpu DAI configuration\n");
+ return ret;
+ }
+
+ /* Set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set codec system clock\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops sdp3430_ops = {
+ .hw_params = sdp3430_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link sdp3430_dai = {
+ .name = "TWL4030",
+ .stream_name = "TWL4030",
+ .cpu_dai = &omap_mcbsp_dai[0],
+ .codec_dai = &twl4030_dai,
+ .ops = &sdp3430_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_machine snd_soc_machine_sdp3430 = {
+ .name = "SDP3430",
+ .platform = &omap_soc_platform,
+ .dai_link = &sdp3430_dai,
+ .num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device sdp3430_snd_devdata = {
+ .machine = &snd_soc_machine_sdp3430,
+ .codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *sdp3430_snd_device;
+
+static int __init sdp3430_soc_init(void)
+{
+ int ret;
+
+ if (!machine_is_omap_3430sdp()) {
+ pr_debug("Not SDP3430!\n");
+ return -ENODEV;
+ }
+ printk(KERN_INFO "SDP3430 SoC init\n");
+
+ sdp3430_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!sdp3430_snd_device) {
+ printk(KERN_ERR "Platform device allocation failed\n");
+ return -ENOMEM;
+ }
+
+ platform_set_drvdata(sdp3430_snd_device, &sdp3430_snd_devdata);
+ sdp3430_snd_devdata.dev = &sdp3430_snd_device->dev;
+ *(unsigned int *)sdp3430_dai.cpu_dai->private_data = 1; /* McBSP2 */
+
+ ret = platform_device_add(sdp3430_snd_device);
+ if (ret)
+ goto err1;
+
+ return 0;
+
+err1:
+ printk(KERN_ERR "Unable to add platform device\n");
+ platform_device_put(sdp3430_snd_device);
+
+ return ret;
+}
+module_init(sdp3430_soc_init);
+
+static void __exit sdp3430_soc_exit(void)
+{
+ platform_device_unregister(sdp3430_snd_device);
+}
+module_exit(sdp3430_soc_exit);
+
+MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC SDP3430");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index f8c1cdd..f82e106 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -21,6 +21,9 @@
config SND_PXA2XX_SOC_I2S
tristate
+config SND_PXA_SOC_SSP
+ tristate
+
config SND_PXA2XX_SOC_CORGI
tristate "SoC Audio support for Sharp Zaurus SL-C7x0"
depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx
@@ -75,3 +78,22 @@
help
Say Y if you want to add support for SoC audio on
CompuLab EM-x270.
+
+config SND_PXA2XX_SOC_PALM27X
+ bool "SoC Audio support for Palm T|X, T5 and LifeDrive"
+ depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || MACH_PALMT5)
+ select SND_PXA2XX_SOC_AC97
+ select SND_SOC_WM9712
+ help
+ Say Y if you want to add support for SoC audio on
+ Palm T|X, T5 or LifeDrive handheld computer.
+
+config SND_SOC_ZYLONITE
+ tristate "SoC Audio support for Marvell Zylonite"
+ depends on SND_PXA2XX_SOC && MACH_ZYLONITE
+ select SND_PXA2XX_SOC_AC97
+ select SND_PXA_SOC_SSP
+ select SND_SOC_WM9713
+ help
+ Say Y if you want to add support for SoC audio on the
+ Marvell Zylonite reference platform.
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index 5bc8edf..08a9f27 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -2,10 +2,12 @@
snd-soc-pxa2xx-objs := pxa2xx-pcm.o
snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o
snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o
+snd-soc-pxa-ssp-objs := pxa-ssp.o
obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o
obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o
obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o
+obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o
# PXA Machine Support
snd-soc-corgi-objs := corgi.o
@@ -14,6 +16,8 @@
snd-soc-e800-objs := e800_wm9712.o
snd-soc-spitz-objs := spitz.o
snd-soc-em-x270-objs := em-x270.o
+snd-soc-palm27x-objs := palm27x.o
+snd-soc-zylonite-objs := zylonite.o
obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
@@ -21,3 +25,5 @@
obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o
obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o
+obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
+obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 2718eaf..1ba25a5 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -108,15 +108,11 @@
}
/* we need to unmute the HP at shutdown as the mute burns power on corgi */
-static int corgi_shutdown(struct snd_pcm_substream *substream)
+static void corgi_shutdown(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->socdev->codec;
-
/* set = unmute headphone */
gpio_set_value(CORGI_GPIO_MUTE_L, 1);
gpio_set_value(CORGI_GPIO_MUTE_R, 1);
- return 0;
}
static int corgi_hw_params(struct snd_pcm_substream *substream,
@@ -314,8 +310,9 @@
};
/* corgi audio machine driver */
-static struct snd_soc_machine snd_soc_machine_corgi = {
+static struct snd_soc_card snd_soc_corgi = {
.name = "Corgi",
+ .platform = &pxa2xx_soc_platform,
.dai_link = &corgi_dai,
.num_links = 1,
};
@@ -328,8 +325,7 @@
/* corgi audio subsystem */
static struct snd_soc_device corgi_snd_devdata = {
- .machine = &snd_soc_machine_corgi,
- .platform = &pxa2xx_soc_platform,
+ .card = &snd_soc_corgi,
.codec_dev = &soc_codec_dev_wm8731,
.codec_data = &corgi_wm8731_setup,
};
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index 6781c5b..2e3386d 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -29,7 +29,7 @@
#include "pxa2xx-pcm.h"
#include "pxa2xx-ac97.h"
-static struct snd_soc_machine e800;
+static struct snd_soc_card e800;
static struct snd_soc_dai_link e800_dai[] = {
{
@@ -40,15 +40,15 @@
},
};
-static struct snd_soc_machine e800 = {
+static struct snd_soc_card e800 = {
.name = "Toshiba e800",
+ .platform = &pxa2xx_soc_platform,
.dai_link = e800_dai,
.num_links = ARRAY_SIZE(e800_dai),
};
static struct snd_soc_device e800_snd_devdata = {
- .machine = &e800,
- .platform = &pxa2xx_soc_platform,
+ .card = &e800,
.codec_dev = &soc_codec_dev_wm9712,
};
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
index e6ff692..fe4a729 100644
--- a/sound/soc/pxa/em-x270.c
+++ b/sound/soc/pxa/em-x270.c
@@ -23,7 +23,6 @@
#include <linux/moduleparam.h>
#include <linux/device.h>
-#include <sound/driver.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
@@ -53,15 +52,15 @@
},
};
-static struct snd_soc_machine em_x270 = {
+static struct snd_soc_card em_x270 = {
.name = "EM-X270",
+ .platform = &pxa2xx_soc_platform,
.dai_link = em_x270_dai,
.num_links = ARRAY_SIZE(em_x270_dai),
};
static struct snd_soc_device em_x270_snd_devdata = {
- .machine = &em_x270,
- .platform = &pxa2xx_soc_platform,
+ .card = &em_x270,
.codec_dev = &soc_codec_dev_wm9712,
};
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
new file mode 100644
index 0000000..4a9cf30
--- /dev/null
+++ b/sound/soc/pxa/palm27x.c
@@ -0,0 +1,269 @@
+/*
+ * linux/sound/soc/pxa/palm27x.c
+ *
+ * SoC Audio driver for Palm T|X, T5 and LifeDrive
+ *
+ * based on tosa.c
+ *
+ * Copyright (C) 2008 Marek Vasut <marek.vasut@gmail.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/device.h>
+#include <linux/gpio.h>
+#include <linux/interrupt.h>
+#include <linux/irq.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/audio.h>
+#include <mach/palmasoc.h>
+
+#include "../codecs/wm9712.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+
+static int palm27x_jack_func = 1;
+static int palm27x_spk_func = 1;
+static int palm27x_ep_gpio = -1;
+
+static void palm27x_ext_control(struct snd_soc_codec *codec)
+{
+ if (!palm27x_spk_func)
+ snd_soc_dapm_enable_pin(codec, "Speaker");
+ else
+ snd_soc_dapm_disable_pin(codec, "Speaker");
+
+ if (!palm27x_jack_func)
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ else
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+
+ snd_soc_dapm_sync(codec);
+}
+
+static int palm27x_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->socdev->codec;
+
+ /* check the jack status at stream startup */
+ palm27x_ext_control(codec);
+ return 0;
+}
+
+static struct snd_soc_ops palm27x_ops = {
+ .startup = palm27x_startup,
+};
+
+static irqreturn_t palm27x_interrupt(int irq, void *v)
+{
+ palm27x_spk_func = gpio_get_value(palm27x_ep_gpio);
+ palm27x_jack_func = !palm27x_spk_func;
+ return IRQ_HANDLED;
+}
+
+static int palm27x_get_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = palm27x_jack_func;
+ return 0;
+}
+
+static int palm27x_set_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (palm27x_jack_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ palm27x_jack_func = ucontrol->value.integer.value[0];
+ palm27x_ext_control(codec);
+ return 1;
+}
+
+static int palm27x_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = palm27x_spk_func;
+ return 0;
+}
+
+static int palm27x_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (palm27x_spk_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ palm27x_spk_func = ucontrol->value.integer.value[0];
+ palm27x_ext_control(codec);
+ return 1;
+}
+
+/* PalmTX machine dapm widgets */
+static const struct snd_soc_dapm_widget palm27x_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+/* PalmTX audio map */
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* headphone connected to HPOUTL, HPOUTR */
+ {"Headphone Jack", NULL, "HPOUTL"},
+ {"Headphone Jack", NULL, "HPOUTR"},
+
+ /* ext speaker connected to ROUT2, LOUT2 */
+ {"Speaker", NULL, "LOUT2"},
+ {"Speaker", NULL, "ROUT2"},
+};
+
+static const char *jack_function[] = {"Headphone", "Off"};
+static const char *spk_function[] = {"On", "Off"};
+static const struct soc_enum palm27x_enum[] = {
+ SOC_ENUM_SINGLE_EXT(2, jack_function),
+ SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const struct snd_kcontrol_new palm27x_controls[] = {
+ SOC_ENUM_EXT("Jack Function", palm27x_enum[0], palm27x_get_jack,
+ palm27x_set_jack),
+ SOC_ENUM_EXT("Speaker Function", palm27x_enum[1], palm27x_get_spk,
+ palm27x_set_spk),
+};
+
+static int palm27x_ac97_init(struct snd_soc_codec *codec)
+{
+ int i, err;
+
+ snd_soc_dapm_nc_pin(codec, "OUT3");
+ snd_soc_dapm_nc_pin(codec, "MONOOUT");
+
+ /* add palm27x specific controls */
+ for (i = 0; i < ARRAY_SIZE(palm27x_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&palm27x_controls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ /* add palm27x specific widgets */
+ snd_soc_dapm_new_controls(codec, palm27x_dapm_widgets,
+ ARRAY_SIZE(palm27x_dapm_widgets));
+
+ /* set up palm27x specific audio path audio_map */
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_sync(codec);
+ return 0;
+}
+
+static struct snd_soc_dai_link palm27x_dai[] = {
+{
+ .name = "AC97 HiFi",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
+ .init = palm27x_ac97_init,
+ .ops = &palm27x_ops,
+},
+{
+ .name = "AC97 Aux",
+ .stream_name = "AC97 Aux",
+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
+ .ops = &palm27x_ops,
+},
+};
+
+static struct snd_soc_card palm27x_asoc = {
+ .name = "Palm/PXA27x",
+ .platform = &pxa2xx_soc_platform,
+ .dai_link = palm27x_dai,
+ .num_links = ARRAY_SIZE(palm27x_dai),
+};
+
+static struct snd_soc_device palm27x_snd_devdata = {
+ .card = &palm27x_asoc,
+ .codec_dev = &soc_codec_dev_wm9712,
+};
+
+static struct platform_device *palm27x_snd_device;
+
+static int __init palm27x_asoc_init(void)
+{
+ int ret;
+
+ if (!(machine_is_palmtx() || machine_is_palmt5() ||
+ machine_is_palmld()))
+ return -ENODEV;
+
+ ret = gpio_request(palm27x_ep_gpio, "Headphone Jack");
+ if (ret)
+ return ret;
+ ret = gpio_direction_input(palm27x_ep_gpio);
+ if (ret)
+ goto err_alloc;
+
+ if (request_irq(gpio_to_irq(palm27x_ep_gpio), palm27x_interrupt,
+ IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING,
+ "Headphone jack", NULL))
+ goto err_alloc;
+
+ palm27x_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!palm27x_snd_device) {
+ ret = -ENOMEM;
+ goto err_dev;
+ }
+
+ platform_set_drvdata(palm27x_snd_device, &palm27x_snd_devdata);
+ palm27x_snd_devdata.dev = &palm27x_snd_device->dev;
+ ret = platform_device_add(palm27x_snd_device);
+
+ if (ret != 0)
+ goto put_device;
+
+ return 0;
+
+put_device:
+ platform_device_put(palm27x_snd_device);
+err_dev:
+ free_irq(gpio_to_irq(palm27x_ep_gpio), NULL);
+err_alloc:
+ gpio_free(palm27x_ep_gpio);
+
+ return ret;
+}
+
+static void __exit palm27x_asoc_exit(void)
+{
+ free_irq(gpio_to_irq(palm27x_ep_gpio), NULL);
+ gpio_free(palm27x_ep_gpio);
+ platform_device_unregister(palm27x_snd_device);
+}
+
+void __init palm27x_asoc_set_pdata(struct palm27x_asoc_info *data)
+{
+ palm27x_ep_gpio = data->jack_gpio;
+}
+
+module_init(palm27x_asoc_init);
+module_exit(palm27x_asoc_exit);
+
+/* Module information */
+MODULE_AUTHOR("Marek Vasut <marek.vasut@gmail.com>");
+MODULE_DESCRIPTION("ALSA SoC Palm T|X, T5 and LifeDrive");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 4d9930c..6e98271 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -276,8 +276,9 @@
};
/* poodle audio machine driver */
-static struct snd_soc_machine snd_soc_machine_poodle = {
+static struct snd_soc_card snd_soc_poodle = {
.name = "Poodle",
+ .platform = &pxa2xx_soc_platform,
.dai_link = &poodle_dai,
.num_links = 1,
};
@@ -290,8 +291,7 @@
/* poodle audio subsystem */
static struct snd_soc_device poodle_snd_devdata = {
- .machine = &snd_soc_machine_poodle,
- .platform = &pxa2xx_soc_platform,
+ .card = &snd_soc_poodle,
.codec_dev = &soc_codec_dev_wm8731,
.codec_data = &poodle_wm8731_setup,
};
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
new file mode 100644
index 0000000..73cb6b4
--- /dev/null
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -0,0 +1,931 @@
+#define DEBUG
+/*
+ * pxa-ssp.c -- ALSA Soc Audio Layer
+ *
+ * Copyright 2005,2008 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood
+ * Mark Brown <broonie@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.
+ *
+ * TODO:
+ * o Test network mode for > 16bit sample size
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/clk.h>
+#include <linux/io.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/pxa2xx-lib.h>
+
+#include <mach/hardware.h>
+#include <mach/pxa-regs.h>
+#include <mach/regs-ssp.h>
+#include <mach/audio.h>
+#include <mach/ssp.h>
+
+#include "pxa2xx-pcm.h"
+#include "pxa-ssp.h"
+
+/*
+ * SSP audio private data
+ */
+struct ssp_priv {
+ struct ssp_dev dev;
+ unsigned int sysclk;
+ int dai_fmt;
+#ifdef CONFIG_PM
+ struct ssp_state state;
+#endif
+};
+
+#define PXA2xx_SSP1_BASE 0x41000000
+#define PXA27x_SSP2_BASE 0x41700000
+#define PXA27x_SSP3_BASE 0x41900000
+#define PXA3xx_SSP4_BASE 0x41a00000
+
+static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_out = {
+ .name = "SSP1 PCM Mono out",
+ .dev_addr = PXA2xx_SSP1_BASE + SSDR,
+ .drcmr = &DRCMR(14),
+ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
+ DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_in = {
+ .name = "SSP1 PCM Mono in",
+ .dev_addr = PXA2xx_SSP1_BASE + SSDR,
+ .drcmr = &DRCMR(13),
+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
+ DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_out = {
+ .name = "SSP1 PCM Stereo out",
+ .dev_addr = PXA2xx_SSP1_BASE + SSDR,
+ .drcmr = &DRCMR(14),
+ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
+ DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_in = {
+ .name = "SSP1 PCM Stereo in",
+ .dev_addr = PXA2xx_SSP1_BASE + SSDR,
+ .drcmr = &DRCMR(13),
+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
+ DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_out = {
+ .name = "SSP2 PCM Mono out",
+ .dev_addr = PXA27x_SSP2_BASE + SSDR,
+ .drcmr = &DRCMR(16),
+ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
+ DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_in = {
+ .name = "SSP2 PCM Mono in",
+ .dev_addr = PXA27x_SSP2_BASE + SSDR,
+ .drcmr = &DRCMR(15),
+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
+ DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_out = {
+ .name = "SSP2 PCM Stereo out",
+ .dev_addr = PXA27x_SSP2_BASE + SSDR,
+ .drcmr = &DRCMR(16),
+ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
+ DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_in = {
+ .name = "SSP2 PCM Stereo in",
+ .dev_addr = PXA27x_SSP2_BASE + SSDR,
+ .drcmr = &DRCMR(15),
+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
+ DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_out = {
+ .name = "SSP3 PCM Mono out",
+ .dev_addr = PXA27x_SSP3_BASE + SSDR,
+ .drcmr = &DRCMR(67),
+ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
+ DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_in = {
+ .name = "SSP3 PCM Mono in",
+ .dev_addr = PXA27x_SSP3_BASE + SSDR,
+ .drcmr = &DRCMR(66),
+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
+ DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_out = {
+ .name = "SSP3 PCM Stereo out",
+ .dev_addr = PXA27x_SSP3_BASE + SSDR,
+ .drcmr = &DRCMR(67),
+ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
+ DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_in = {
+ .name = "SSP3 PCM Stereo in",
+ .dev_addr = PXA27x_SSP3_BASE + SSDR,
+ .drcmr = &DRCMR(66),
+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
+ DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_out = {
+ .name = "SSP4 PCM Mono out",
+ .dev_addr = PXA3xx_SSP4_BASE + SSDR,
+ .drcmr = &DRCMR(67),
+ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
+ DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_in = {
+ .name = "SSP4 PCM Mono in",
+ .dev_addr = PXA3xx_SSP4_BASE + SSDR,
+ .drcmr = &DRCMR(66),
+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
+ DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_out = {
+ .name = "SSP4 PCM Stereo out",
+ .dev_addr = PXA3xx_SSP4_BASE + SSDR,
+ .drcmr = &DRCMR(67),
+ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
+ DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_in = {
+ .name = "SSP4 PCM Stereo in",
+ .dev_addr = PXA3xx_SSP4_BASE + SSDR,
+ .drcmr = &DRCMR(66),
+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
+ DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static void dump_registers(struct ssp_device *ssp)
+{
+ dev_dbg(&ssp->pdev->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n",
+ ssp_read_reg(ssp, SSCR0), ssp_read_reg(ssp, SSCR1),
+ ssp_read_reg(ssp, SSTO));
+
+ dev_dbg(&ssp->pdev->dev, "SSPSP 0x%08x SSSR 0x%08x SSACD 0x%08x\n",
+ ssp_read_reg(ssp, SSPSP), ssp_read_reg(ssp, SSSR),
+ ssp_read_reg(ssp, SSACD));
+}
+
+static struct pxa2xx_pcm_dma_params *ssp_dma_params[4][4] = {
+ {
+ &pxa_ssp1_pcm_mono_out, &pxa_ssp1_pcm_mono_in,
+ &pxa_ssp1_pcm_stereo_out, &pxa_ssp1_pcm_stereo_in,
+ },
+ {
+ &pxa_ssp2_pcm_mono_out, &pxa_ssp2_pcm_mono_in,
+ &pxa_ssp2_pcm_stereo_out, &pxa_ssp2_pcm_stereo_in,
+ },
+ {
+ &pxa_ssp3_pcm_mono_out, &pxa_ssp3_pcm_mono_in,
+ &pxa_ssp3_pcm_stereo_out, &pxa_ssp3_pcm_stereo_in,
+ },
+ {
+ &pxa_ssp4_pcm_mono_out, &pxa_ssp4_pcm_mono_in,
+ &pxa_ssp4_pcm_stereo_out, &pxa_ssp4_pcm_stereo_in,
+ },
+};
+
+static int pxa_ssp_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct ssp_priv *priv = cpu_dai->private_data;
+ int ret = 0;
+
+ if (!cpu_dai->active) {
+ ret = ssp_init(&priv->dev, cpu_dai->id + 1, SSP_NO_IRQ);
+ if (ret < 0)
+ return ret;
+ ssp_disable(&priv->dev);
+ }
+ return ret;
+}
+
+static void pxa_ssp_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct ssp_priv *priv = cpu_dai->private_data;
+
+ if (!cpu_dai->active) {
+ ssp_disable(&priv->dev);
+ ssp_exit(&priv->dev);
+ }
+}
+
+#ifdef CONFIG_PM
+
+static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai)
+{
+ struct ssp_priv *priv = cpu_dai->private_data;
+
+ if (!cpu_dai->active)
+ return 0;
+
+ ssp_save_state(&priv->dev, &priv->state);
+ clk_disable(priv->dev.ssp->clk);
+ return 0;
+}
+
+static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai)
+{
+ struct ssp_priv *priv = cpu_dai->private_data;
+
+ if (!cpu_dai->active)
+ return 0;
+
+ clk_enable(priv->dev.ssp->clk);
+ ssp_restore_state(&priv->dev, &priv->state);
+ ssp_enable(&priv->dev);
+
+ return 0;
+}
+
+#else
+#define pxa_ssp_suspend NULL
+#define pxa_ssp_resume NULL
+#endif
+
+/**
+ * ssp_set_clkdiv - set SSP clock divider
+ * @div: serial clock rate divider
+ */
+static void ssp_set_scr(struct ssp_dev *dev, u32 div)
+{
+ struct ssp_device *ssp = dev->ssp;
+ u32 sscr0 = ssp_read_reg(dev->ssp, SSCR0) & ~SSCR0_SCR;
+
+ ssp_write_reg(ssp, SSCR0, (sscr0 | SSCR0_SerClkDiv(div)));
+}
+
+/*
+ * Set the SSP ports SYSCLK.
+ */
+static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct ssp_priv *priv = cpu_dai->private_data;
+ struct ssp_device *ssp = priv->dev.ssp;
+ int val;
+
+ u32 sscr0 = ssp_read_reg(ssp, SSCR0) &
+ ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ADC);
+
+ dev_dbg(&ssp->pdev->dev,
+ "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %d\n",
+ cpu_dai->id, clk_id, freq);
+
+ switch (clk_id) {
+ case PXA_SSP_CLK_NET_PLL:
+ sscr0 |= SSCR0_MOD;
+ break;
+ case PXA_SSP_CLK_PLL:
+ /* Internal PLL is fixed */
+ if (cpu_is_pxa25x())
+ priv->sysclk = 1843200;
+ else
+ priv->sysclk = 13000000;
+ break;
+ case PXA_SSP_CLK_EXT:
+ priv->sysclk = freq;
+ sscr0 |= SSCR0_ECS;
+ break;
+ case PXA_SSP_CLK_NET:
+ priv->sysclk = freq;
+ sscr0 |= SSCR0_NCS | SSCR0_MOD;
+ break;
+ case PXA_SSP_CLK_AUDIO:
+ priv->sysclk = 0;
+ ssp_set_scr(&priv->dev, 1);
+ sscr0 |= SSCR0_ADC;
+ break;
+ default:
+ return -ENODEV;
+ }
+
+ /* The SSP clock must be disabled when changing SSP clock mode
+ * on PXA2xx. On PXA3xx it must be enabled when doing so. */
+ if (!cpu_is_pxa3xx())
+ clk_disable(priv->dev.ssp->clk);
+ val = ssp_read_reg(ssp, SSCR0) | sscr0;
+ ssp_write_reg(ssp, SSCR0, val);
+ if (!cpu_is_pxa3xx())
+ clk_enable(priv->dev.ssp->clk);
+
+ return 0;
+}
+
+/*
+ * Set the SSP clock dividers.
+ */
+static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
+ int div_id, int div)
+{
+ struct ssp_priv *priv = cpu_dai->private_data;
+ struct ssp_device *ssp = priv->dev.ssp;
+ int val;
+
+ switch (div_id) {
+ case PXA_SSP_AUDIO_DIV_ACDS:
+ val = (ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div);
+ ssp_write_reg(ssp, SSACD, val);
+ break;
+ case PXA_SSP_AUDIO_DIV_SCDB:
+ val = ssp_read_reg(ssp, SSACD);
+ val &= ~SSACD_SCDB;
+#if defined(CONFIG_PXA3xx)
+ if (cpu_is_pxa3xx())
+ val &= ~SSACD_SCDX8;
+#endif
+ switch (div) {
+ case PXA_SSP_CLK_SCDB_1:
+ val |= SSACD_SCDB;
+ break;
+ case PXA_SSP_CLK_SCDB_4:
+ break;
+#if defined(CONFIG_PXA3xx)
+ case PXA_SSP_CLK_SCDB_8:
+ if (cpu_is_pxa3xx())
+ val |= SSACD_SCDX8;
+ else
+ return -EINVAL;
+ break;
+#endif
+ default:
+ return -EINVAL;
+ }
+ ssp_write_reg(ssp, SSACD, val);
+ break;
+ case PXA_SSP_DIV_SCR:
+ ssp_set_scr(&priv->dev, div);
+ break;
+ default:
+ return -ENODEV;
+ }
+
+ return 0;
+}
+
+/*
+ * Configure the PLL frequency pxa27x and (afaik - pxa320 only)
+ */
+static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai,
+ int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+ struct ssp_priv *priv = cpu_dai->private_data;
+ struct ssp_device *ssp = priv->dev.ssp;
+ u32 ssacd = ssp_read_reg(ssp, SSACD) & ~0x70;
+
+#if defined(CONFIG_PXA3xx)
+ if (cpu_is_pxa3xx())
+ ssp_write_reg(ssp, SSACDD, 0);
+#endif
+
+ switch (freq_out) {
+ case 5622000:
+ break;
+ case 11345000:
+ ssacd |= (0x1 << 4);
+ break;
+ case 12235000:
+ ssacd |= (0x2 << 4);
+ break;
+ case 14857000:
+ ssacd |= (0x3 << 4);
+ break;
+ case 32842000:
+ ssacd |= (0x4 << 4);
+ break;
+ case 48000000:
+ ssacd |= (0x5 << 4);
+ break;
+ case 0:
+ /* Disable */
+ break;
+
+ default:
+#ifdef CONFIG_PXA3xx
+ /* PXA3xx has a clock ditherer which can be used to generate
+ * a wider range of frequencies - calculate a value for it.
+ */
+ if (cpu_is_pxa3xx()) {
+ u32 val;
+ u64 tmp = 19968;
+ tmp *= 1000000;
+ do_div(tmp, freq_out);
+ val = tmp;
+
+ val = (val << 16) | 64;;
+ ssp_write_reg(ssp, SSACDD, val);
+
+ ssacd |= (0x6 << 4);
+
+ dev_dbg(&ssp->pdev->dev,
+ "Using SSACDD %x to supply %dHz\n",
+ val, freq_out);
+ break;
+ }
+#endif
+
+ return -EINVAL;
+ }
+
+ ssp_write_reg(ssp, SSACD, ssacd);
+
+ return 0;
+}
+
+/*
+ * Set the active slots in TDM/Network mode
+ */
+static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai,
+ unsigned int mask, int slots)
+{
+ struct ssp_priv *priv = cpu_dai->private_data;
+ struct ssp_device *ssp = priv->dev.ssp;
+ u32 sscr0;
+
+ sscr0 = ssp_read_reg(ssp, SSCR0) & ~SSCR0_SlotsPerFrm(7);
+
+ /* set number of active slots */
+ sscr0 |= SSCR0_SlotsPerFrm(slots);
+ ssp_write_reg(ssp, SSCR0, sscr0);
+
+ /* set active slot mask */
+ ssp_write_reg(ssp, SSTSA, mask);
+ ssp_write_reg(ssp, SSRSA, mask);
+ return 0;
+}
+
+/*
+ * Tristate the SSP DAI lines
+ */
+static int pxa_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai,
+ int tristate)
+{
+ struct ssp_priv *priv = cpu_dai->private_data;
+ struct ssp_device *ssp = priv->dev.ssp;
+ u32 sscr1;
+
+ sscr1 = ssp_read_reg(ssp, SSCR1);
+ if (tristate)
+ sscr1 &= ~SSCR1_TTE;
+ else
+ sscr1 |= SSCR1_TTE;
+ ssp_write_reg(ssp, SSCR1, sscr1);
+
+ return 0;
+}
+
+/*
+ * Set up the SSP DAI format.
+ * The SSP Port must be inactive before calling this function as the
+ * physical interface format is changed.
+ */
+static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+ unsigned int fmt)
+{
+ struct ssp_priv *priv = cpu_dai->private_data;
+ struct ssp_device *ssp = priv->dev.ssp;
+ u32 sscr0;
+ u32 sscr1;
+ u32 sspsp;
+
+ /* reset port settings */
+ sscr0 = ssp_read_reg(ssp, SSCR0) &
+ (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ADC);
+ sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7);
+ sspsp = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ sscr1 |= SSCR1_SCLKDIR | SSCR1_SFRMDIR;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ sscr1 |= SSCR1_SCLKDIR;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ ssp_write_reg(ssp, SSCR0, sscr0);
+ ssp_write_reg(ssp, SSCR1, sscr1);
+ ssp_write_reg(ssp, SSPSP, sspsp);
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ sscr0 |= SSCR0_MOD | SSCR0_PSP;
+ sscr1 |= SSCR1_RWOT | SSCR1_TRAIL;
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ sspsp |= SSPSP_FSRT;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ sspsp |= SSPSP_SFRMP | SSPSP_FSRT;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ sspsp |= SSPSP_SFRMP;
+ break;
+ default:
+ return -EINVAL;
+ }
+ break;
+
+ case SND_SOC_DAIFMT_DSP_A:
+ sspsp |= SSPSP_FSRT;
+ case SND_SOC_DAIFMT_DSP_B:
+ sscr0 |= SSCR0_MOD | SSCR0_PSP;
+ sscr1 |= SSCR1_TRAIL | SSCR1_RWOT;
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ sspsp |= SSPSP_SFRMP;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ break;
+ default:
+ return -EINVAL;
+ }
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ ssp_write_reg(ssp, SSCR0, sscr0);
+ ssp_write_reg(ssp, SSCR1, sscr1);
+ ssp_write_reg(ssp, SSPSP, sspsp);
+
+ dump_registers(ssp);
+
+ /* Since we are configuring the timings for the format by hand
+ * we have to defer some things until hw_params() where we
+ * know parameters like the sample size.
+ */
+ priv->dai_fmt = fmt;
+
+ return 0;
+}
+
+/*
+ * Set the SSP audio DMA parameters and sample size.
+ * Can be called multiple times by oss emulation.
+ */
+static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct ssp_priv *priv = cpu_dai->private_data;
+ struct ssp_device *ssp = priv->dev.ssp;
+ int dma = 0, chn = params_channels(params);
+ u32 sscr0;
+ u32 sspsp;
+ int width = snd_pcm_format_physical_width(params_format(params));
+
+ /* select correct DMA params */
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ dma = 1; /* capture DMA offset is 1,3 */
+ if (chn == 2)
+ dma += 2; /* stereo DMA offset is 2, mono is 0 */
+ cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma];
+
+ dev_dbg(&ssp->pdev->dev, "pxa_ssp_hw_params: dma %d\n", dma);
+
+ /* we can only change the settings if the port is not in use */
+ if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE)
+ return 0;
+
+ /* clear selected SSP bits */
+ sscr0 = ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS);
+ ssp_write_reg(ssp, SSCR0, sscr0);
+
+ /* bit size */
+ sscr0 = ssp_read_reg(ssp, SSCR0);
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+#ifdef CONFIG_PXA3xx
+ if (cpu_is_pxa3xx())
+ sscr0 |= SSCR0_FPCKE;
+#endif
+ sscr0 |= SSCR0_DataSize(16);
+ if (params_channels(params) > 1)
+ sscr0 |= SSCR0_EDSS;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8));
+ /* we must be in network mode (2 slots) for 24 bit stereo */
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16));
+ /* we must be in network mode (2 slots) for 32 bit stereo */
+ break;
+ }
+ ssp_write_reg(ssp, SSCR0, sscr0);
+
+ switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ /* Cleared when the DAI format is set */
+ sspsp = ssp_read_reg(ssp, SSPSP) | SSPSP_SFRMWDTH(width);
+ ssp_write_reg(ssp, SSPSP, sspsp);
+ break;
+ default:
+ break;
+ }
+
+ /* We always use a network mode so we always require TDM slots
+ * - complain loudly and fail if they've not been set up yet.
+ */
+ if (!(ssp_read_reg(ssp, SSTSA) & 0xf)) {
+ dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n");
+ return -EINVAL;
+ }
+
+ dump_registers(ssp);
+
+ return 0;
+}
+
+static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int ret = 0;
+ struct ssp_priv *priv = cpu_dai->private_data;
+ struct ssp_device *ssp = priv->dev.ssp;
+ int val;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_RESUME:
+ ssp_enable(&priv->dev);
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ val = ssp_read_reg(ssp, SSCR1);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ val |= SSCR1_TSRE;
+ else
+ val |= SSCR1_RSRE;
+ ssp_write_reg(ssp, SSCR1, val);
+ val = ssp_read_reg(ssp, SSSR);
+ ssp_write_reg(ssp, SSSR, val);
+ break;
+ case SNDRV_PCM_TRIGGER_START:
+ val = ssp_read_reg(ssp, SSCR1);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ val |= SSCR1_TSRE;
+ else
+ val |= SSCR1_RSRE;
+ ssp_write_reg(ssp, SSCR1, val);
+ ssp_enable(&priv->dev);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ val = ssp_read_reg(ssp, SSCR1);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ val &= ~SSCR1_TSRE;
+ else
+ val &= ~SSCR1_RSRE;
+ ssp_write_reg(ssp, SSCR1, val);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ ssp_disable(&priv->dev);
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ val = ssp_read_reg(ssp, SSCR1);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ val &= ~SSCR1_TSRE;
+ else
+ val &= ~SSCR1_RSRE;
+ ssp_write_reg(ssp, SSCR1, val);
+ break;
+
+ default:
+ ret = -EINVAL;
+ }
+
+ dump_registers(ssp);
+
+ return ret;
+}
+
+static int pxa_ssp_probe(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ struct ssp_priv *priv;
+ int ret;
+
+ priv = kzalloc(sizeof(struct ssp_priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ priv->dev.ssp = ssp_request(dai->id, "SoC audio");
+ if (priv->dev.ssp == NULL) {
+ ret = -ENODEV;
+ goto err_priv;
+ }
+
+ dai->private_data = priv;
+
+ return 0;
+
+err_priv:
+ kfree(priv);
+ return ret;
+}
+
+static void pxa_ssp_remove(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ struct ssp_priv *priv = dai->private_data;
+ ssp_free(priv->dev.ssp);
+}
+
+#define PXA_SSP_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+struct snd_soc_dai pxa_ssp_dai[] = {
+ {
+ .name = "pxa2xx-ssp1",
+ .id = 0,
+ .probe = pxa_ssp_probe,
+ .remove = pxa_ssp_remove,
+ .suspend = pxa_ssp_suspend,
+ .resume = pxa_ssp_resume,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = PXA_SSP_RATES,
+ .formats = PXA_SSP_FORMATS,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = PXA_SSP_RATES,
+ .formats = PXA_SSP_FORMATS,
+ },
+ .ops = {
+ .startup = pxa_ssp_startup,
+ .shutdown = pxa_ssp_shutdown,
+ .trigger = pxa_ssp_trigger,
+ .hw_params = pxa_ssp_hw_params,
+ .set_sysclk = pxa_ssp_set_dai_sysclk,
+ .set_clkdiv = pxa_ssp_set_dai_clkdiv,
+ .set_pll = pxa_ssp_set_dai_pll,
+ .set_fmt = pxa_ssp_set_dai_fmt,
+ .set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
+ .set_tristate = pxa_ssp_set_dai_tristate,
+ },
+ },
+ { .name = "pxa2xx-ssp2",
+ .id = 1,
+ .probe = pxa_ssp_probe,
+ .remove = pxa_ssp_remove,
+ .suspend = pxa_ssp_suspend,
+ .resume = pxa_ssp_resume,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = PXA_SSP_RATES,
+ .formats = PXA_SSP_FORMATS,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = PXA_SSP_RATES,
+ .formats = PXA_SSP_FORMATS,
+ },
+ .ops = {
+ .startup = pxa_ssp_startup,
+ .shutdown = pxa_ssp_shutdown,
+ .trigger = pxa_ssp_trigger,
+ .hw_params = pxa_ssp_hw_params,
+ .set_sysclk = pxa_ssp_set_dai_sysclk,
+ .set_clkdiv = pxa_ssp_set_dai_clkdiv,
+ .set_pll = pxa_ssp_set_dai_pll,
+ .set_fmt = pxa_ssp_set_dai_fmt,
+ .set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
+ .set_tristate = pxa_ssp_set_dai_tristate,
+ },
+ },
+ {
+ .name = "pxa2xx-ssp3",
+ .id = 2,
+ .probe = pxa_ssp_probe,
+ .remove = pxa_ssp_remove,
+ .suspend = pxa_ssp_suspend,
+ .resume = pxa_ssp_resume,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = PXA_SSP_RATES,
+ .formats = PXA_SSP_FORMATS,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = PXA_SSP_RATES,
+ .formats = PXA_SSP_FORMATS,
+ },
+ .ops = {
+ .startup = pxa_ssp_startup,
+ .shutdown = pxa_ssp_shutdown,
+ .trigger = pxa_ssp_trigger,
+ .hw_params = pxa_ssp_hw_params,
+ .set_sysclk = pxa_ssp_set_dai_sysclk,
+ .set_clkdiv = pxa_ssp_set_dai_clkdiv,
+ .set_pll = pxa_ssp_set_dai_pll,
+ .set_fmt = pxa_ssp_set_dai_fmt,
+ .set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
+ .set_tristate = pxa_ssp_set_dai_tristate,
+ },
+ },
+ {
+ .name = "pxa2xx-ssp4",
+ .id = 3,
+ .probe = pxa_ssp_probe,
+ .remove = pxa_ssp_remove,
+ .suspend = pxa_ssp_suspend,
+ .resume = pxa_ssp_resume,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = PXA_SSP_RATES,
+ .formats = PXA_SSP_FORMATS,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = PXA_SSP_RATES,
+ .formats = PXA_SSP_FORMATS,
+ },
+ .ops = {
+ .startup = pxa_ssp_startup,
+ .shutdown = pxa_ssp_shutdown,
+ .trigger = pxa_ssp_trigger,
+ .hw_params = pxa_ssp_hw_params,
+ .set_sysclk = pxa_ssp_set_dai_sysclk,
+ .set_clkdiv = pxa_ssp_set_dai_clkdiv,
+ .set_pll = pxa_ssp_set_dai_pll,
+ .set_fmt = pxa_ssp_set_dai_fmt,
+ .set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
+ .set_tristate = pxa_ssp_set_dai_tristate,
+ },
+ },
+};
+EXPORT_SYMBOL_GPL(pxa_ssp_dai);
+
+static int __init pxa_ssp_init(void)
+{
+ return snd_soc_register_dais(pxa_ssp_dai, ARRAY_SIZE(pxa_ssp_dai));
+}
+module_init(pxa_ssp_init);
+
+static void __exit pxa_ssp_exit(void)
+{
+ snd_soc_unregister_dais(pxa_ssp_dai, ARRAY_SIZE(pxa_ssp_dai));
+}
+module_exit(pxa_ssp_exit);
+
+/* Module information */
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_DESCRIPTION("PXA SSP/PCM SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa-ssp.h b/sound/soc/pxa/pxa-ssp.h
new file mode 100644
index 0000000..91deadd
--- /dev/null
+++ b/sound/soc/pxa/pxa-ssp.h
@@ -0,0 +1,47 @@
+/*
+ * ASoC PXA SSP port support
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _PXA_SSP_H
+#define _PXA_SSP_H
+
+/* pxa DAI SSP IDs */
+#define PXA_DAI_SSP1 0
+#define PXA_DAI_SSP2 1
+#define PXA_DAI_SSP3 2
+#define PXA_DAI_SSP4 3
+
+/* SSP clock sources */
+#define PXA_SSP_CLK_PLL 0
+#define PXA_SSP_CLK_EXT 1
+#define PXA_SSP_CLK_NET 2
+#define PXA_SSP_CLK_AUDIO 3
+#define PXA_SSP_CLK_NET_PLL 4
+
+/* SSP audio dividers */
+#define PXA_SSP_AUDIO_DIV_ACDS 0
+#define PXA_SSP_AUDIO_DIV_SCDB 1
+#define PXA_SSP_DIV_SCR 2
+
+/* SSP ACDS audio dividers values */
+#define PXA_SSP_CLK_AUDIO_DIV_1 0
+#define PXA_SSP_CLK_AUDIO_DIV_2 1
+#define PXA_SSP_CLK_AUDIO_DIV_4 2
+#define PXA_SSP_CLK_AUDIO_DIV_8 3
+#define PXA_SSP_CLK_AUDIO_DIV_16 4
+#define PXA_SSP_CLK_AUDIO_DIV_32 5
+
+/* SSP divider bypass */
+#define PXA_SSP_CLK_SCDB_4 0
+#define PXA_SSP_CLK_SCDB_1 1
+#define PXA_SSP_CLK_SCDB_8 2
+
+#define PXA_SSP_PLL_OUT 0
+
+extern struct snd_soc_dai pxa_ssp_dai[4];
+
+#endif
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index a7a3a9c..780db67 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -87,14 +87,12 @@
};
#ifdef CONFIG_PM
-static int pxa2xx_ac97_suspend(struct platform_device *pdev,
- struct snd_soc_dai *dai)
+static int pxa2xx_ac97_suspend(struct snd_soc_dai *dai)
{
return pxa2xx_ac97_hw_suspend();
}
-static int pxa2xx_ac97_resume(struct platform_device *pdev,
- struct snd_soc_dai *dai)
+static int pxa2xx_ac97_resume(struct snd_soc_dai *dai)
{
return pxa2xx_ac97_hw_resume();
}
@@ -117,7 +115,8 @@
}
static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -131,7 +130,8 @@
}
static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -145,7 +145,8 @@
}
static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -170,7 +171,7 @@
{
.name = "pxa2xx-ac97",
.id = 0,
- .type = SND_SOC_DAI_AC97,
+ .ac97_control = 1,
.probe = pxa2xx_ac97_probe,
.remove = pxa2xx_ac97_remove,
.suspend = pxa2xx_ac97_suspend,
@@ -193,7 +194,7 @@
{
.name = "pxa2xx-ac97-aux",
.id = 1,
- .type = SND_SOC_DAI_AC97,
+ .ac97_control = 1,
.playback = {
.stream_name = "AC97 Aux Playback",
.channels_min = 1,
@@ -212,7 +213,7 @@
{
.name = "pxa2xx-ac97-mic",
.id = 2,
- .type = SND_SOC_DAI_AC97,
+ .ac97_control = 1,
.capture = {
.stream_name = "AC97 Mic Capture",
.channels_min = 1,
@@ -227,6 +228,18 @@
EXPORT_SYMBOL_GPL(pxa_ac97_dai);
EXPORT_SYMBOL_GPL(soc_ac97_ops);
+static int __init pxa_ac97_init(void)
+{
+ return snd_soc_register_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
+}
+module_init(pxa_ac97_init);
+
+static void __exit pxa_ac97_exit(void)
+{
+ snd_soc_unregister_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
+}
+module_exit(pxa_ac97_exit);
+
MODULE_AUTHOR("Nicolas Pitre");
MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index e758034..517991f 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -121,7 +121,8 @@
},
};
-static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream)
+static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -187,7 +188,8 @@
}
static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -248,7 +250,8 @@
return 0;
}
-static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
{
int ret = 0;
@@ -269,7 +272,8 @@
return ret;
}
-static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream)
+static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
SACR1 |= SACR1_DRPL;
@@ -289,8 +293,7 @@
}
#ifdef CONFIG_PM
-static int pxa2xx_i2s_suspend(struct platform_device *dev,
- struct snd_soc_dai *dai)
+static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai)
{
if (!dai->active)
return 0;
@@ -307,8 +310,7 @@
return 0;
}
-static int pxa2xx_i2s_resume(struct platform_device *pdev,
- struct snd_soc_dai *dai)
+static int pxa2xx_i2s_resume(struct snd_soc_dai *dai)
{
if (!dai->active)
return 0;
@@ -336,7 +338,6 @@
struct snd_soc_dai pxa_i2s_dai = {
.name = "pxa2xx-i2s",
.id = 0,
- .type = SND_SOC_DAI_I2S,
.suspend = pxa2xx_i2s_suspend,
.resume = pxa2xx_i2s_resume,
.playback = {
@@ -353,8 +354,7 @@
.startup = pxa2xx_i2s_startup,
.shutdown = pxa2xx_i2s_shutdown,
.trigger = pxa2xx_i2s_trigger,
- .hw_params = pxa2xx_i2s_hw_params,},
- .dai_ops = {
+ .hw_params = pxa2xx_i2s_hw_params,
.set_fmt = pxa2xx_i2s_set_dai_fmt,
.set_sysclk = pxa2xx_i2s_set_dai_sysclk,
},
@@ -364,12 +364,23 @@
static int pxa2xx_i2s_probe(struct platform_device *dev)
{
+ int ret;
+
clk_i2s = clk_get(&dev->dev, "I2SCLK");
- return IS_ERR(clk_i2s) ? PTR_ERR(clk_i2s) : 0;
+ if (IS_ERR(clk_i2s))
+ return PTR_ERR(clk_i2s);
+
+ pxa_i2s_dai.dev = &dev->dev;
+ ret = snd_soc_register_dai(&pxa_i2s_dai);
+ if (ret != 0)
+ clk_put(clk_i2s);
+
+ return ret;
}
static int __devexit pxa2xx_i2s_remove(struct platform_device *dev)
{
+ snd_soc_unregister_dai(&pxa_i2s_dai);
clk_put(clk_i2s);
clk_i2s = ERR_PTR(-ENOENT);
return 0;
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index afcd892..c670d08 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -69,7 +69,7 @@
return 0;
}
-struct snd_pcm_ops pxa2xx_pcm_ops = {
+static struct snd_pcm_ops pxa2xx_pcm_ops = {
.open = __pxa2xx_pcm_open,
.close = __pxa2xx_pcm_close,
.ioctl = snd_pcm_lib_ioctl,
@@ -118,6 +118,18 @@
};
EXPORT_SYMBOL_GPL(pxa2xx_soc_platform);
+static int __init pxa2xx_soc_platform_init(void)
+{
+ return snd_soc_register_platform(&pxa2xx_soc_platform);
+}
+module_init(pxa2xx_soc_platform_init);
+
+static void __exit pxa2xx_soc_platform_exit(void)
+{
+ snd_soc_unregister_platform(&pxa2xx_soc_platform);
+}
+module_exit(pxa2xx_soc_platform_exit);
+
MODULE_AUTHOR("Nicolas Pitre");
MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index d307b67..a3b9e6b 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -319,8 +319,9 @@
};
/* spitz audio machine driver */
-static struct snd_soc_machine snd_soc_machine_spitz = {
+static struct snd_soc_card snd_soc_spitz = {
.name = "Spitz",
+ .platform = &pxa2xx_soc_platform,
.dai_link = &spitz_dai,
.num_links = 1,
};
@@ -333,8 +334,7 @@
/* spitz audio subsystem */
static struct snd_soc_device spitz_snd_devdata = {
- .machine = &snd_soc_machine_spitz,
- .platform = &pxa2xx_soc_platform,
+ .card = &snd_soc_spitz,
.codec_dev = &soc_codec_dev_wm8750,
.codec_data = &spitz_wm8750_setup,
};
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index afefe41..c77194f 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -38,7 +38,7 @@
#include "pxa2xx-pcm.h"
#include "pxa2xx-ac97.h"
-static struct snd_soc_machine tosa;
+static struct snd_soc_card tosa;
#define TOSA_HP 0
#define TOSA_MIC_INT 1
@@ -230,15 +230,37 @@
},
};
-static struct snd_soc_machine tosa = {
+static int tosa_probe(struct platform_device *dev)
+{
+ int ret;
+
+ ret = gpio_request(TOSA_GPIO_L_MUTE, "Headphone Jack");
+ if (ret)
+ return ret;
+ ret = gpio_direction_output(TOSA_GPIO_L_MUTE, 0);
+ if (ret)
+ gpio_free(TOSA_GPIO_L_MUTE);
+
+ return ret;
+}
+
+static int tosa_remove(struct platform_device *dev)
+{
+ gpio_free(TOSA_GPIO_L_MUTE);
+ return 0;
+}
+
+static struct snd_soc_card tosa = {
.name = "Tosa",
+ .platform = &pxa2xx_soc_platform,
.dai_link = tosa_dai,
.num_links = ARRAY_SIZE(tosa_dai),
+ .probe = tosa_probe,
+ .remove = tosa_remove,
};
static struct snd_soc_device tosa_snd_devdata = {
- .machine = &tosa,
- .platform = &pxa2xx_soc_platform,
+ .card = &tosa,
.codec_dev = &soc_codec_dev_wm9712,
};
@@ -251,11 +273,6 @@
if (!machine_is_tosa())
return -ENODEV;
- ret = gpio_request(TOSA_GPIO_L_MUTE, "Headphone Jack");
- if (ret)
- return ret;
- gpio_direction_output(TOSA_GPIO_L_MUTE, 0);
-
tosa_snd_device = platform_device_alloc("soc-audio", -1);
if (!tosa_snd_device) {
ret = -ENOMEM;
@@ -272,15 +289,12 @@
platform_device_put(tosa_snd_device);
err_alloc:
- gpio_free(TOSA_GPIO_L_MUTE);
-
return ret;
}
static void __exit tosa_exit(void)
{
platform_device_unregister(tosa_snd_device);
- gpio_free(TOSA_GPIO_L_MUTE);
}
module_init(tosa_init);
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
new file mode 100644
index 0000000..f8e9ecd
--- /dev/null
+++ b/sound/soc/pxa/zylonite.c
@@ -0,0 +1,219 @@
+/*
+ * zylonite.c -- SoC audio for Zylonite
+ *
+ * Copyright 2008 Wolfson Microelectronics PLC.
+ * Author: Mark Brown <broonie@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.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/i2c.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include "../codecs/wm9713.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+#include "pxa-ssp.h"
+
+static struct snd_soc_card zylonite;
+
+static const struct snd_soc_dapm_widget zylonite_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_MIC("Headset Microphone", NULL),
+ SND_SOC_DAPM_MIC("Handset Microphone", NULL),
+ SND_SOC_DAPM_SPK("Multiactor", NULL),
+ SND_SOC_DAPM_SPK("Headset Earpiece", NULL),
+};
+
+/* Currently supported audio map */
+static const struct snd_soc_dapm_route audio_map[] = {
+
+ /* Headphone output connected to HPL/HPR */
+ { "Headphone", NULL, "HPL" },
+ { "Headphone", NULL, "HPR" },
+
+ /* On-board earpiece */
+ { "Headset Earpiece", NULL, "OUT3" },
+
+ /* Headphone mic */
+ { "MIC2A", NULL, "Mic Bias" },
+ { "Mic Bias", NULL, "Headset Microphone" },
+
+ /* On-board mic */
+ { "MIC1", NULL, "Mic Bias" },
+ { "Mic Bias", NULL, "Handset Microphone" },
+
+ /* Multiactor differentially connected over SPKL/SPKR */
+ { "Multiactor", NULL, "SPKL" },
+ { "Multiactor", NULL, "SPKR" },
+};
+
+static int zylonite_wm9713_init(struct snd_soc_codec *codec)
+{
+ /* Currently we only support use of the AC97 clock here. If
+ * CLK_POUT is selected by SW15 then the clock API will need
+ * to be used to request and enable it here.
+ */
+
+ snd_soc_dapm_new_controls(codec, zylonite_dapm_widgets,
+ ARRAY_SIZE(zylonite_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ /* Static setup for now */
+ snd_soc_dapm_enable_pin(codec, "Headphone");
+ snd_soc_dapm_enable_pin(codec, "Headset Earpiece");
+
+ snd_soc_dapm_sync(codec);
+ return 0;
+}
+
+static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ unsigned int pll_out = 0;
+ unsigned int acds = 0;
+ unsigned int wm9713_div = 0;
+ int ret = 0;
+
+ switch (params_rate(params)) {
+ case 8000:
+ wm9713_div = 12;
+ pll_out = 2048000;
+ break;
+ case 16000:
+ wm9713_div = 6;
+ pll_out = 4096000;
+ break;
+ case 48000:
+ default:
+ wm9713_div = 2;
+ pll_out = 12288000;
+ acds = 1;
+ break;
+ }
+
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_tdm_slot(cpu_dai,
+ params_channels(params),
+ params_channels(params));
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, pll_out);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_AUDIO_DIV_ACDS, acds);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1);
+ if (ret < 0)
+ return ret;
+
+ /* Note that if the PLL is in use the WM9713_PCMCLK_PLL_DIV needs
+ * to be set instead.
+ */
+ ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_DIV,
+ WM9713_PCMDIV(wm9713_div));
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops zylonite_voice_ops = {
+ .hw_params = zylonite_voice_hw_params,
+};
+
+static struct snd_soc_dai_link zylonite_dai[] = {
+{
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+ .codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI],
+ .init = zylonite_wm9713_init,
+},
+{
+ .name = "AC97 Aux",
+ .stream_name = "AC97 Aux",
+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+ .codec_dai = &wm9713_dai[WM9713_DAI_AC97_AUX],
+},
+{
+ .name = "WM9713 Voice",
+ .stream_name = "WM9713 Voice",
+ .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP3],
+ .codec_dai = &wm9713_dai[WM9713_DAI_PCM_VOICE],
+ .ops = &zylonite_voice_ops,
+},
+};
+
+static struct snd_soc_card zylonite = {
+ .name = "Zylonite",
+ .platform = &pxa2xx_soc_platform,
+ .dai_link = zylonite_dai,
+ .num_links = ARRAY_SIZE(zylonite_dai),
+};
+
+static struct snd_soc_device zylonite_snd_ac97_devdata = {
+ .card = &zylonite,
+ .codec_dev = &soc_codec_dev_wm9713,
+};
+
+static struct platform_device *zylonite_snd_ac97_device;
+
+static int __init zylonite_init(void)
+{
+ int ret;
+
+ zylonite_snd_ac97_device = platform_device_alloc("soc-audio", -1);
+ if (!zylonite_snd_ac97_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(zylonite_snd_ac97_device,
+ &zylonite_snd_ac97_devdata);
+ zylonite_snd_ac97_devdata.dev = &zylonite_snd_ac97_device->dev;
+
+ ret = platform_device_add(zylonite_snd_ac97_device);
+ if (ret != 0)
+ platform_device_put(zylonite_snd_ac97_device);
+
+ return ret;
+}
+
+static void __exit zylonite_exit(void)
+{
+ platform_device_unregister(zylonite_snd_ac97_device);
+}
+
+module_init(zylonite_init);
+module_exit(zylonite_exit);
+
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_DESCRIPTION("ALSA SoC WM9713 Zylonite");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index b9f2353..fcd03ac 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -44,3 +44,8 @@
Say Y if you want to add support for SoC audio on ln2440sbc
with the ALC650.
+config SND_S3C24XX_SOC_S3C24XX_UDA134X
+ tristate "SoC I2S Audio support UDA134X wired to a S3C24XX"
+ depends on SND_S3C24XX_SOC
+ select SND_S3C24XX_SOC_I2S
+ select SND_SOC_UDA134X
diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile
index 0aa5fb0..96b3f3f 100644
--- a/sound/soc/s3c24xx/Makefile
+++ b/sound/soc/s3c24xx/Makefile
@@ -13,7 +13,9 @@
snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o
snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o
snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o
+snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o
obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o
obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o
+obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o
diff --git a/sound/soc/s3c24xx/ln2440sbc_alc650.c b/sound/soc/s3c24xx/ln2440sbc_alc650.c
index 4eab2c1..12c7148 100644
--- a/sound/soc/s3c24xx/ln2440sbc_alc650.c
+++ b/sound/soc/s3c24xx/ln2440sbc_alc650.c
@@ -27,7 +27,7 @@
#include "s3c24xx-pcm.h"
#include "s3c24xx-ac97.h"
-static struct snd_soc_machine ln2440sbc;
+static struct snd_soc_card ln2440sbc;
static struct snd_soc_dai_link ln2440sbc_dai[] = {
{
@@ -38,15 +38,15 @@
},
};
-static struct snd_soc_machine ln2440sbc = {
+static struct snd_soc_card ln2440sbc = {
.name = "LN2440SBC",
+ .platform = &s3c24xx_soc_platform,
.dai_link = ln2440sbc_dai,
.num_links = ARRAY_SIZE(ln2440sbc_dai),
};
static struct snd_soc_device ln2440sbc_snd_ac97_devdata = {
- .machine = &ln2440sbc,
- .platform = &s3c24xx_soc_platform,
+ .card = &ln2440sbc,
.codec_dev = &soc_codec_dev_ac97,
};
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index 87ddfef..45bb12e 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -59,7 +59,7 @@
#define NEO_CAPTURE_HEADSET 7
#define NEO_CAPTURE_BLUETOOTH 8
-static struct snd_soc_machine neo1973;
+static struct snd_soc_card neo1973;
static struct i2c_client *i2c;
static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
@@ -548,7 +548,6 @@
static struct snd_soc_dai bt_dai = {
.name = "Bluetooth",
.id = 0,
- .type = SND_SOC_DAI_PCM,
.playback = {
.channels_min = 1,
.channels_max = 1,
@@ -579,8 +578,9 @@
},
};
-static struct snd_soc_machine neo1973 = {
+static struct snd_soc_card neo1973 = {
.name = "neo1973",
+ .platform = &s3c24xx_soc_platform,
.dai_link = neo1973_dai,
.num_links = ARRAY_SIZE(neo1973_dai),
};
@@ -591,8 +591,7 @@
};
static struct snd_soc_device neo1973_snd_devdata = {
- .machine = &neo1973,
- .platform = &s3c24xx_soc_platform,
+ .card = &neo1973,
.codec_dev = &soc_codec_dev_wm8753,
.codec_data = &neo1973_wm8753_setup,
};
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index ded7d99..f3fc0ab 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -343,7 +343,8 @@
}
static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
u32 iismod;
@@ -373,7 +374,8 @@
return 0;
}
-static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
{
int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
unsigned long irqs;
@@ -647,8 +649,7 @@
}
#ifdef CONFIG_PM
-static int s3c2412_i2s_suspend(struct platform_device *dev,
- struct snd_soc_dai *dai)
+static int s3c2412_i2s_suspend(struct snd_soc_dai *dai)
{
struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
u32 iismod;
@@ -663,25 +664,24 @@
iismod = readl(i2s->regs + S3C2412_IISMOD);
if (iismod & S3C2412_IISCON_RXDMA_ACTIVE)
- dev_warn(&dev->dev, "%s: RXDMA active?\n", __func__);
+ pr_warning("%s: RXDMA active?\n", __func__);
if (iismod & S3C2412_IISCON_TXDMA_ACTIVE)
- dev_warn(&dev->dev, "%s: TXDMA active?\n", __func__);
+ pr_warning("%s: TXDMA active?\n", __func__);
if (iismod & S3C2412_IISCON_IIS_ACTIVE)
- dev_warn(&dev->dev, "%s: IIS active\n", __func__);
+ pr_warning("%s: IIS active\n", __func__);
}
return 0;
}
-static int s3c2412_i2s_resume(struct platform_device *pdev,
- struct snd_soc_dai *dai)
+static int s3c2412_i2s_resume(struct snd_soc_dai *dai)
{
struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
- dev_info(&pdev->dev, "dai_active %d, IISMOD %08x, IISCON %08x\n",
- dai->active, i2s->suspend_iismod, i2s->suspend_iiscon);
+ pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n",
+ dai->active, i2s->suspend_iismod, i2s->suspend_iiscon);
if (dai->active) {
writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON);
@@ -711,7 +711,6 @@
struct snd_soc_dai s3c2412_i2s_dai = {
.name = "s3c2412-i2s",
.id = 0,
- .type = SND_SOC_DAI_I2S,
.probe = s3c2412_i2s_probe,
.suspend = s3c2412_i2s_suspend,
.resume = s3c2412_i2s_resume,
@@ -730,8 +729,6 @@
.ops = {
.trigger = s3c2412_i2s_trigger,
.hw_params = s3c2412_i2s_hw_params,
- },
- .dai_ops = {
.set_fmt = s3c2412_i2s_set_fmt,
.set_clkdiv = s3c2412_i2s_set_clkdiv,
.set_sysclk = s3c2412_i2s_set_sysclk,
@@ -739,6 +736,19 @@
};
EXPORT_SYMBOL_GPL(s3c2412_i2s_dai);
+static int __init s3c2412_i2s_init(void)
+{
+ return snd_soc_register_dai(&s3c2412_i2s_dai);
+}
+module_init(s3c2412_i2s_init);
+
+static void __exit s3c2412_i2s_exit(void)
+{
+ snd_soc_unregister_dai(&s3c2412_i2s_dai);
+}
+module_exit(s3c2412_i2s_exit);
+
+
/* Module information */
MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
MODULE_DESCRIPTION("S3C2412 I2S SoC Interface");
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
index 19c5c3c..1bfce40 100644
--- a/sound/soc/s3c24xx/s3c2443-ac97.c
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -271,7 +271,8 @@
}
static int s3c2443_ac97_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -284,7 +285,8 @@
return 0;
}
-static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd)
+static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
{
u32 ac_glbctrl;
@@ -313,7 +315,8 @@
}
static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -327,7 +330,7 @@
}
static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream,
- int cmd)
+ int cmd, struct snd_soc_dai *dai)
{
u32 ac_glbctrl;
@@ -356,7 +359,7 @@
{
.name = "s3c2443-ac97",
.id = 0,
- .type = SND_SOC_DAI_AC97,
+ .ac97_control = 1,
.probe = s3c2443_ac97_probe,
.remove = s3c2443_ac97_remove,
.playback = {
@@ -378,7 +381,7 @@
{
.name = "pxa2xx-ac97-mic",
.id = 1,
- .type = SND_SOC_DAI_AC97,
+ .ac97_control = 1,
.capture = {
.stream_name = "AC97 Mic Capture",
.channels_min = 1,
@@ -393,6 +396,21 @@
EXPORT_SYMBOL_GPL(s3c2443_ac97_dai);
EXPORT_SYMBOL_GPL(soc_ac97_ops);
+static int __init s3c2443_ac97_init(void)
+{
+ return snd_soc_register_dais(s3c2443_ac97_dai,
+ ARRAY_SIZE(s3c2443_ac97_dai));
+}
+module_init(s3c2443_ac97_init);
+
+static void __exit s3c2443_ac97_exit(void)
+{
+ snd_soc_unregister_dais(s3c2443_ac97_dai,
+ ARRAY_SIZE(s3c2443_ac97_dai));
+}
+module_exit(s3c2443_ac97_exit);
+
+
MODULE_AUTHOR("Graeme Gregory");
MODULE_DESCRIPTION("AC97 driver for the Samsung s3c2443 chip");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index ba4476b..6f4d439 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -243,7 +243,8 @@
}
static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
u32 iismod;
@@ -261,10 +262,17 @@
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S8:
+ iismod &= ~S3C2410_IISMOD_16BIT;
+ ((struct s3c24xx_pcm_dma_params *)
+ rtd->dai->cpu_dai->dma_data)->dma_size = 1;
break;
case SNDRV_PCM_FORMAT_S16_LE:
iismod |= S3C2410_IISMOD_16BIT;
+ ((struct s3c24xx_pcm_dma_params *)
+ rtd->dai->cpu_dai->dma_data)->dma_size = 2;
break;
+ default:
+ return -EINVAL;
}
writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
@@ -272,7 +280,8 @@
return 0;
}
-static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
{
int ret = 0;
@@ -410,8 +419,7 @@
}
#ifdef CONFIG_PM
-static int s3c24xx_i2s_suspend(struct platform_device *pdev,
- struct snd_soc_dai *cpu_dai)
+static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai)
{
DBG("Entered %s\n", __func__);
@@ -425,8 +433,7 @@
return 0;
}
-static int s3c24xx_i2s_resume(struct platform_device *pdev,
- struct snd_soc_dai *cpu_dai)
+static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai)
{
DBG("Entered %s\n", __func__);
clk_enable(s3c24xx_i2s.iis_clk);
@@ -452,7 +459,6 @@
struct snd_soc_dai s3c24xx_i2s_dai = {
.name = "s3c24xx-i2s",
.id = 0,
- .type = SND_SOC_DAI_I2S,
.probe = s3c24xx_i2s_probe,
.suspend = s3c24xx_i2s_suspend,
.resume = s3c24xx_i2s_resume,
@@ -468,8 +474,7 @@
.formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,},
.ops = {
.trigger = s3c24xx_i2s_trigger,
- .hw_params = s3c24xx_i2s_hw_params,},
- .dai_ops = {
+ .hw_params = s3c24xx_i2s_hw_params,
.set_fmt = s3c24xx_i2s_set_fmt,
.set_clkdiv = s3c24xx_i2s_set_clkdiv,
.set_sysclk = s3c24xx_i2s_set_sysclk,
@@ -477,6 +482,18 @@
};
EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai);
+static int __init s3c24xx_i2s_init(void)
+{
+ return snd_soc_register_dai(&s3c24xx_i2s_dai);
+}
+module_init(s3c24xx_i2s_init);
+
+static void __exit s3c24xx_i2s_exit(void)
+{
+ snd_soc_unregister_dai(&s3c24xx_i2s_dai);
+}
+module_exit(s3c24xx_i2s_exit);
+
/* Module information */
MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
MODULE_DESCRIPTION("s3c24xx I2S SoC Interface");
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index e13e614..7c64d31 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -465,6 +465,18 @@
};
EXPORT_SYMBOL_GPL(s3c24xx_soc_platform);
+static int __init s3c24xx_soc_platform_init(void)
+{
+ return snd_soc_register_platform(&s3c24xx_soc_platform);
+}
+module_init(s3c24xx_soc_platform_init);
+
+static void __exit s3c24xx_soc_platform_exit(void)
+{
+ snd_soc_unregister_platform(&s3c24xx_soc_platform);
+}
+module_exit(s3c24xx_soc_platform_exit);
+
MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
MODULE_DESCRIPTION("Samsung S3C24XX PCM DMA module");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c
new file mode 100644
index 0000000..a0a4d18
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c
@@ -0,0 +1,373 @@
+/*
+ * Modifications by Christian Pellegrin <chripell@evolware.org>
+ *
+ * s3c24xx_uda134x.c -- S3C24XX_UDA134X ALSA SoC Audio board driver
+ *
+ * Copyright 2007 Dension Audio Systems Ltd.
+ * Author: Zoltan Devai
+ *
+ * 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/clk.h>
+#include <linux/mutex.h>
+#include <linux/gpio.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/s3c24xx_uda134x.h>
+#include <sound/uda134x.h>
+
+#include <asm/plat-s3c24xx/regs-iis.h>
+
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-i2s.h"
+#include "../codecs/uda134x.h"
+
+
+/* #define ENFORCE_RATES 1 */
+/*
+ Unfortunately the S3C24XX in master mode has a limited capacity of
+ generating the clock for the codec. If you define this only rates
+ that are really available will be enforced. But be careful, most
+ user level application just want the usual sampling frequencies (8,
+ 11.025, 22.050, 44.1 kHz) and anyway resampling is a costly
+ operation for embedded systems. So if you aren't very lucky or your
+ hardware engineer wasn't very forward-looking it's better to leave
+ this undefined. If you do so an approximate value for the requested
+ sampling rate in the range -/+ 5% will be chosen. If this in not
+ possible an error will be returned.
+*/
+
+static struct clk *xtal;
+static struct clk *pclk;
+/* this is need because we don't have a place where to keep the
+ * pointers to the clocks in each substream. We get the clocks only
+ * when we are actually using them so we don't block stuff like
+ * frequency change or oscillator power-off */
+static int clk_users;
+static DEFINE_MUTEX(clk_lock);
+
+static unsigned int rates[33 * 2];
+#ifdef ENFORCE_RATES
+static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+};
+#endif
+
+static struct platform_device *s3c24xx_uda134x_snd_device;
+
+static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+#ifdef ENFORCE_RATES
+ struct snd_pcm_runtime *runtime = substream->runtime;;
+#endif
+
+ mutex_lock(&clk_lock);
+ pr_debug("%s %d\n", __func__, clk_users);
+ if (clk_users == 0) {
+ xtal = clk_get(&s3c24xx_uda134x_snd_device->dev, "xtal");
+ if (!xtal) {
+ printk(KERN_ERR "%s cannot get xtal\n", __func__);
+ ret = -EBUSY;
+ } else {
+ pclk = clk_get(&s3c24xx_uda134x_snd_device->dev,
+ "pclk");
+ if (!pclk) {
+ printk(KERN_ERR "%s cannot get pclk\n",
+ __func__);
+ clk_put(xtal);
+ ret = -EBUSY;
+ }
+ }
+ if (!ret) {
+ int i, j;
+
+ for (i = 0; i < 2; i++) {
+ int fs = i ? 256 : 384;
+
+ rates[i*33] = clk_get_rate(xtal) / fs;
+ for (j = 1; j < 33; j++)
+ rates[i*33 + j] = clk_get_rate(pclk) /
+ (j * fs);
+ }
+ }
+ }
+ clk_users += 1;
+ mutex_unlock(&clk_lock);
+ if (!ret) {
+#ifdef ENFORCE_RATES
+ ret = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &hw_constraints_rates);
+ if (ret < 0)
+ printk(KERN_ERR "%s cannot set constraints\n",
+ __func__);
+#endif
+ }
+ return ret;
+}
+
+static void s3c24xx_uda134x_shutdown(struct snd_pcm_substream *substream)
+{
+ mutex_lock(&clk_lock);
+ pr_debug("%s %d\n", __func__, clk_users);
+ clk_users -= 1;
+ if (clk_users == 0) {
+ clk_put(xtal);
+ xtal = NULL;
+ clk_put(pclk);
+ pclk = NULL;
+ }
+ mutex_unlock(&clk_lock);
+}
+
+static int s3c24xx_uda134x_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ unsigned int clk = 0;
+ int ret = 0;
+ int clk_source, fs_mode;
+ unsigned long rate = params_rate(params);
+ long err, cerr;
+ unsigned int div;
+ int i, bi;
+
+ err = 999999;
+ bi = 0;
+ for (i = 0; i < 2*33; i++) {
+ cerr = rates[i] - rate;
+ if (cerr < 0)
+ cerr = -cerr;
+ if (cerr < err) {
+ err = cerr;
+ bi = i;
+ }
+ }
+ if (bi / 33 == 1)
+ fs_mode = S3C2410_IISMOD_256FS;
+ else
+ fs_mode = S3C2410_IISMOD_384FS;
+ if (bi % 33 == 0) {
+ clk_source = S3C24XX_CLKSRC_MPLL;
+ div = 1;
+ } else {
+ clk_source = S3C24XX_CLKSRC_PCLK;
+ div = bi % 33;
+ }
+ pr_debug("%s desired rate %lu, %d\n", __func__, rate, bi);
+
+ clk = (fs_mode == S3C2410_IISMOD_384FS ? 384 : 256) * rate;
+ pr_debug("%s will use: %s %s %d sysclk %d err %ld\n", __func__,
+ fs_mode == S3C2410_IISMOD_384FS ? "384FS" : "256FS",
+ clk_source == S3C24XX_CLKSRC_MPLL ? "MPLLin" : "PCLK",
+ div, clk, err);
+
+ if ((err * 100 / rate) > 5) {
+ printk(KERN_ERR "S3C24XX_UDA134X: effective frequency "
+ "too different from desired (%ld%%)\n",
+ err * 100 / rate);
+ return -EINVAL;
+ }
+
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_sysclk(cpu_dai, clk_source , clk,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, fs_mode);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK,
+ S3C2410_IISMOD_32FS);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
+ S3C24XX_PRESCALE(div, div));
+ if (ret < 0)
+ return ret;
+
+ /* set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk,
+ SND_SOC_CLOCK_OUT);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops s3c24xx_uda134x_ops = {
+ .startup = s3c24xx_uda134x_startup,
+ .shutdown = s3c24xx_uda134x_shutdown,
+ .hw_params = s3c24xx_uda134x_hw_params,
+};
+
+static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = {
+ .name = "UDA134X",
+ .stream_name = "UDA134X",
+ .codec_dai = &uda134x_dai,
+ .cpu_dai = &s3c24xx_i2s_dai,
+ .ops = &s3c24xx_uda134x_ops,
+};
+
+static struct snd_soc_card snd_soc_s3c24xx_uda134x = {
+ .name = "S3C24XX_UDA134X",
+ .platform = &s3c24xx_soc_platform,
+ .dai_link = &s3c24xx_uda134x_dai_link,
+ .num_links = 1,
+};
+
+static struct s3c24xx_uda134x_platform_data *s3c24xx_uda134x_l3_pins;
+
+static void setdat(int v)
+{
+ gpio_set_value(s3c24xx_uda134x_l3_pins->l3_data, v > 0);
+}
+
+static void setclk(int v)
+{
+ gpio_set_value(s3c24xx_uda134x_l3_pins->l3_clk, v > 0);
+}
+
+static void setmode(int v)
+{
+ gpio_set_value(s3c24xx_uda134x_l3_pins->l3_mode, v > 0);
+}
+
+static struct uda134x_platform_data s3c24xx_uda134x = {
+ .l3 = {
+ .setdat = setdat,
+ .setclk = setclk,
+ .setmode = setmode,
+ .data_hold = 1,
+ .data_setup = 1,
+ .clock_high = 1,
+ .mode_hold = 1,
+ .mode = 1,
+ .mode_setup = 1,
+ },
+};
+
+static struct snd_soc_device s3c24xx_uda134x_snd_devdata = {
+ .card = &snd_soc_s3c24xx_uda134x,
+ .codec_dev = &soc_codec_dev_uda134x,
+ .codec_data = &s3c24xx_uda134x,
+};
+
+static int s3c24xx_uda134x_setup_pin(int pin, char *fun)
+{
+ if (gpio_request(pin, "s3c24xx_uda134x") < 0) {
+ printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: "
+ "l3 %s pin already in use", fun);
+ return -EBUSY;
+ }
+ gpio_direction_output(pin, 0);
+ return 0;
+}
+
+static int s3c24xx_uda134x_probe(struct platform_device *pdev)
+{
+ int ret;
+
+ printk(KERN_INFO "S3C24XX_UDA134X SoC Audio driver\n");
+
+ s3c24xx_uda134x_l3_pins = pdev->dev.platform_data;
+ if (s3c24xx_uda134x_l3_pins == NULL) {
+ printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: "
+ "unable to find platform data\n");
+ return -ENODEV;
+ }
+ s3c24xx_uda134x.power = s3c24xx_uda134x_l3_pins->power;
+ s3c24xx_uda134x.model = s3c24xx_uda134x_l3_pins->model;
+
+ if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_data,
+ "data") < 0)
+ return -EBUSY;
+ if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_clk,
+ "clk") < 0) {
+ gpio_free(s3c24xx_uda134x_l3_pins->l3_data);
+ return -EBUSY;
+ }
+ if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_mode,
+ "mode") < 0) {
+ gpio_free(s3c24xx_uda134x_l3_pins->l3_data);
+ gpio_free(s3c24xx_uda134x_l3_pins->l3_clk);
+ return -EBUSY;
+ }
+
+ s3c24xx_uda134x_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!s3c24xx_uda134x_snd_device) {
+ printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: "
+ "Unable to register\n");
+ return -ENOMEM;
+ }
+
+ platform_set_drvdata(s3c24xx_uda134x_snd_device,
+ &s3c24xx_uda134x_snd_devdata);
+ s3c24xx_uda134x_snd_devdata.dev = &s3c24xx_uda134x_snd_device->dev;
+ ret = platform_device_add(s3c24xx_uda134x_snd_device);
+ if (ret) {
+ printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: Unable to add\n");
+ platform_device_put(s3c24xx_uda134x_snd_device);
+ }
+
+ return ret;
+}
+
+static int s3c24xx_uda134x_remove(struct platform_device *pdev)
+{
+ platform_device_unregister(s3c24xx_uda134x_snd_device);
+ gpio_free(s3c24xx_uda134x_l3_pins->l3_data);
+ gpio_free(s3c24xx_uda134x_l3_pins->l3_clk);
+ gpio_free(s3c24xx_uda134x_l3_pins->l3_mode);
+ return 0;
+}
+
+static struct platform_driver s3c24xx_uda134x_driver = {
+ .probe = s3c24xx_uda134x_probe,
+ .remove = s3c24xx_uda134x_remove,
+ .driver = {
+ .name = "s3c24xx_uda134x",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init s3c24xx_uda134x_init(void)
+{
+ return platform_driver_register(&s3c24xx_uda134x_driver);
+}
+
+static void __exit s3c24xx_uda134x_exit(void)
+{
+ platform_driver_unregister(&s3c24xx_uda134x_driver);
+}
+
+
+module_init(s3c24xx_uda134x_init);
+module_exit(s3c24xx_uda134x_exit);
+
+MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell@evolware.org>");
+MODULE_DESCRIPTION("S3C24XX_UDA134X ALSA SoC audio driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c
index 8515d6f..a2a4f53 100644
--- a/sound/soc/s3c24xx/smdk2443_wm9710.c
+++ b/sound/soc/s3c24xx/smdk2443_wm9710.c
@@ -23,7 +23,7 @@
#include "s3c24xx-pcm.h"
#include "s3c24xx-ac97.h"
-static struct snd_soc_machine smdk2443;
+static struct snd_soc_card smdk2443;
static struct snd_soc_dai_link smdk2443_dai[] = {
{
@@ -34,15 +34,15 @@
},
};
-static struct snd_soc_machine smdk2443 = {
+static struct snd_soc_card smdk2443 = {
.name = "SMDK2443",
+ .platform = &s3c24xx_soc_platform,
.dai_link = smdk2443_dai,
.num_links = ARRAY_SIZE(smdk2443_dai),
};
static struct snd_soc_device smdk2443_snd_ac97_devdata = {
- .machine = &smdk2443,
- .platform = &s3c24xx_soc_platform,
+ .card = &smdk2443,
.codec_dev = &soc_codec_dev_ac97,
};
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 9faa126..0dad3a0 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -348,6 +348,18 @@
};
EXPORT_SYMBOL_GPL(sh7760_soc_platform);
+static int __init sh7760_soc_platform_init(void)
+{
+ return snd_soc_register_platform(&sh7760_soc_platform);
+}
+module_init(sh7760_soc_platform_init);
+
+static void __exit sh7760_soc_platform_exit(void)
+{
+ snd_soc_unregister_platform(&sh7760_soc_platform);
+}
+module_exit(sh7760_soc_platform_exit);
+
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("SH7760 Audio DMA (DMABRG) driver");
MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c
index df7bc34..eab3183 100644
--- a/sound/soc/sh/hac.c
+++ b/sound/soc/sh/hac.c
@@ -236,7 +236,8 @@
EXPORT_SYMBOL_GPL(soc_ac97_ops);
static int hac_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct hac_priv *hac = &hac_cpu_data[rtd->dai->cpu_dai->id];
@@ -270,7 +271,7 @@
{
.name = "HAC0",
.id = 0,
- .type = SND_SOC_DAI_AC97,
+ .ac97_control = 1,
.playback = {
.rates = AC97_RATES,
.formats = AC97_FMTS,
@@ -290,8 +291,8 @@
#ifdef CONFIG_CPU_SUBTYPE_SH7760
{
.name = "HAC1",
+ .ac97_control = 1,
.id = 1,
- .type = SND_SOC_DAI_AC97,
.playback = {
.rates = AC97_RATES,
.formats = AC97_FMTS,
@@ -313,6 +314,18 @@
};
EXPORT_SYMBOL_GPL(sh4_hac_dai);
+static int __init sh4_hac_init(void)
+{
+ return snd_soc_register_dais(sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai));
+}
+module_init(sh4_hac_init);
+
+static void __exit sh4_hac_exit(void)
+{
+ snd_soc_unregister_dais(sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai));
+}
+module_exit(sh4_hac_exit);
+
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("SuperH onchip HAC (AC97) audio driver");
MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c
index 92bfaf4..ce7f95b 100644
--- a/sound/soc/sh/sh7760-ac97.c
+++ b/sound/soc/sh/sh7760-ac97.c
@@ -38,15 +38,15 @@
.ops = NULL,
};
-static struct snd_soc_machine sh7760_ac97_soc_machine = {
+static struct snd_soc_card sh7760_ac97_soc_machine = {
.name = "SH7760 AC97",
+ .platform = &sh7760_soc_platform,
.dai_link = &sh7760_ac97_dai,
.num_links = 1,
};
static struct snd_soc_device sh7760_ac97_snd_devdata = {
- .machine = &sh7760_ac97_soc_machine,
- .platform = &sh7760_soc_platform,
+ .card = &sh7760_ac97_soc_machine,
.codec_dev = &soc_codec_dev_ac97,
};
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
index 55c3464..d1e5390 100644
--- a/sound/soc/sh/ssi.c
+++ b/sound/soc/sh/ssi.c
@@ -89,7 +89,8 @@
* track usage of the SSI; it is simplex-only so prevent attempts of
* concurrent playback + capture. FIXME: any locking required?
*/
-static int ssi_startup(struct snd_pcm_substream *substream)
+static int ssi_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
@@ -101,7 +102,8 @@
return 0;
}
-static void ssi_shutdown(struct snd_pcm_substream *substream)
+static void ssi_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
@@ -109,7 +111,8 @@
ssi->inuse = 0;
}
-static int ssi_trigger(struct snd_pcm_substream *substream, int cmd)
+static int ssi_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
@@ -129,7 +132,8 @@
}
static int ssi_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
@@ -336,7 +340,6 @@
{
.name = "SSI0",
.id = 0,
- .type = SND_SOC_DAI_I2S,
.playback = {
.rates = SSI_RATES,
.formats = SSI_FMTS,
@@ -354,8 +357,6 @@
.shutdown = ssi_shutdown,
.trigger = ssi_trigger,
.hw_params = ssi_hw_params,
- },
- .dai_ops = {
.set_sysclk = ssi_set_sysclk,
.set_clkdiv = ssi_set_clkdiv,
.set_fmt = ssi_set_fmt,
@@ -365,7 +366,6 @@
{
.name = "SSI1",
.id = 1,
- .type = SND_SOC_DAI_I2S,
.playback = {
.rates = SSI_RATES,
.formats = SSI_FMTS,
@@ -383,8 +383,6 @@
.shutdown = ssi_shutdown,
.trigger = ssi_trigger,
.hw_params = ssi_hw_params,
- },
- .dai_ops = {
.set_sysclk = ssi_set_sysclk,
.set_clkdiv = ssi_set_clkdiv,
.set_fmt = ssi_set_fmt,
@@ -394,6 +392,18 @@
};
EXPORT_SYMBOL_GPL(sh4_ssi_dai);
+static int __init sh4_ssi_init(void)
+{
+ return snd_soc_register_dais(sh4_ssi_dai, ARRAY_SIZE(sh4_ssi_dai));
+}
+module_init(sh4_ssi_init);
+
+static void __exit sh4_ssi_exit(void)
+{
+ snd_soc_unregister_dais(sh4_ssi_dai, ARRAY_SIZE(sh4_ssi_dai));
+}
+module_exit(sh4_ssi_exit);
+
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("SuperH onchip SSI (I2S) audio driver");
MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 16c7453..b098c0b 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -26,6 +26,7 @@
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/bitops.h>
+#include <linux/debugfs.h>
#include <linux/platform_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -34,18 +35,23 @@
#include <sound/soc-dapm.h>
#include <sound/initval.h>
-/* debug */
-#define SOC_DEBUG 0
-#if SOC_DEBUG
-#define dbg(format, arg...) printk(format, ## arg)
-#else
-#define dbg(format, arg...)
-#endif
-
static DEFINE_MUTEX(pcm_mutex);
static DEFINE_MUTEX(io_mutex);
static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq);
+#ifdef CONFIG_DEBUG_FS
+static struct dentry *debugfs_root;
+#endif
+
+static DEFINE_MUTEX(client_mutex);
+static LIST_HEAD(card_list);
+static LIST_HEAD(dai_list);
+static LIST_HEAD(platform_list);
+static LIST_HEAD(codec_list);
+
+static int snd_soc_register_card(struct snd_soc_card *card);
+static int snd_soc_unregister_card(struct snd_soc_card *card);
+
/*
* This is a timeout to do a DAPM powerdown after a stream is closed().
* It can be used to eliminate pops between different playback streams, e.g.
@@ -107,20 +113,6 @@
}
#endif
-static inline const char *get_dai_name(int type)
-{
- switch (type) {
- case SND_SOC_DAI_AC97_BUS:
- case SND_SOC_DAI_AC97:
- return "AC97";
- case SND_SOC_DAI_I2S:
- return "I2S";
- case SND_SOC_DAI_PCM:
- return "PCM";
- }
- return NULL;
-}
-
/*
* Called by ALSA when a PCM substream is opened, the runtime->hw record is
* then initialized and any private data can be allocated. This also calls
@@ -130,9 +122,10 @@
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_card *card = socdev->card;
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_dai_link *machine = rtd->dai;
- struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_platform *platform = card->platform;
struct snd_soc_dai *cpu_dai = machine->cpu_dai;
struct snd_soc_dai *codec_dai = machine->codec_dai;
int ret = 0;
@@ -141,7 +134,7 @@
/* startup the audio subsystem */
if (cpu_dai->ops.startup) {
- ret = cpu_dai->ops.startup(substream);
+ ret = cpu_dai->ops.startup(substream, cpu_dai);
if (ret < 0) {
printk(KERN_ERR "asoc: can't open interface %s\n",
cpu_dai->name);
@@ -158,7 +151,7 @@
}
if (codec_dai->ops.startup) {
- ret = codec_dai->ops.startup(substream);
+ ret = codec_dai->ops.startup(substream, codec_dai);
if (ret < 0) {
printk(KERN_ERR "asoc: can't open codec %s\n",
codec_dai->name);
@@ -228,12 +221,12 @@
goto machine_err;
}
- dbg("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name);
- dbg("asoc: rate mask 0x%x\n", runtime->hw.rates);
- dbg("asoc: min ch %d max ch %d\n", runtime->hw.channels_min,
- runtime->hw.channels_max);
- dbg("asoc: min rate %d max rate %d\n", runtime->hw.rate_min,
- runtime->hw.rate_max);
+ pr_debug("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name);
+ pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates);
+ pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min,
+ runtime->hw.channels_max);
+ pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min,
+ runtime->hw.rate_max);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
cpu_dai->playback.active = codec_dai->playback.active = 1;
@@ -255,7 +248,7 @@
platform_err:
if (cpu_dai->ops.shutdown)
- cpu_dai->ops.shutdown(substream);
+ cpu_dai->ops.shutdown(substream, cpu_dai);
out:
mutex_unlock(&pcm_mutex);
return ret;
@@ -268,8 +261,9 @@
*/
static void close_delayed_work(struct work_struct *work)
{
- struct snd_soc_device *socdev =
- container_of(work, struct snd_soc_device, delayed_work.work);
+ struct snd_soc_card *card = container_of(work, struct snd_soc_card,
+ delayed_work.work);
+ struct snd_soc_device *socdev = card->socdev;
struct snd_soc_codec *codec = socdev->codec;
struct snd_soc_dai *codec_dai;
int i;
@@ -278,18 +272,18 @@
for (i = 0; i < codec->num_dai; i++) {
codec_dai = &codec->dai[i];
- dbg("pop wq checking: %s status: %s waiting: %s\n",
- codec_dai->playback.stream_name,
- codec_dai->playback.active ? "active" : "inactive",
- codec_dai->pop_wait ? "yes" : "no");
+ pr_debug("pop wq checking: %s status: %s waiting: %s\n",
+ codec_dai->playback.stream_name,
+ codec_dai->playback.active ? "active" : "inactive",
+ codec_dai->pop_wait ? "yes" : "no");
/* are we waiting on this codec DAI stream */
if (codec_dai->pop_wait == 1) {
/* Reduce power if no longer active */
if (codec->active == 0) {
- dbg("pop wq D1 %s %s\n", codec->name,
- codec_dai->playback.stream_name);
+ pr_debug("pop wq D1 %s %s\n", codec->name,
+ codec_dai->playback.stream_name);
snd_soc_dapm_set_bias_level(socdev,
SND_SOC_BIAS_PREPARE);
}
@@ -301,8 +295,8 @@
/* Fall into standby if no longer active */
if (codec->active == 0) {
- dbg("pop wq D3 %s %s\n", codec->name,
- codec_dai->playback.stream_name);
+ pr_debug("pop wq D3 %s %s\n", codec->name,
+ codec_dai->playback.stream_name);
snd_soc_dapm_set_bias_level(socdev,
SND_SOC_BIAS_STANDBY);
}
@@ -320,8 +314,9 @@
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_card *card = socdev->card;
struct snd_soc_dai_link *machine = rtd->dai;
- struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_platform *platform = card->platform;
struct snd_soc_dai *cpu_dai = machine->cpu_dai;
struct snd_soc_dai *codec_dai = machine->codec_dai;
struct snd_soc_codec *codec = socdev->codec;
@@ -346,10 +341,10 @@
snd_soc_dai_digital_mute(codec_dai, 1);
if (cpu_dai->ops.shutdown)
- cpu_dai->ops.shutdown(substream);
+ cpu_dai->ops.shutdown(substream, cpu_dai);
if (codec_dai->ops.shutdown)
- codec_dai->ops.shutdown(substream);
+ codec_dai->ops.shutdown(substream, codec_dai);
if (machine->ops && machine->ops->shutdown)
machine->ops->shutdown(substream);
@@ -361,7 +356,7 @@
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
/* start delayed pop wq here for playback streams */
codec_dai->pop_wait = 1;
- schedule_delayed_work(&socdev->delayed_work,
+ schedule_delayed_work(&card->delayed_work,
msecs_to_jiffies(pmdown_time));
} else {
/* capture streams can be powered down now */
@@ -387,8 +382,9 @@
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_card *card = socdev->card;
struct snd_soc_dai_link *machine = rtd->dai;
- struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_platform *platform = card->platform;
struct snd_soc_dai *cpu_dai = machine->cpu_dai;
struct snd_soc_dai *codec_dai = machine->codec_dai;
struct snd_soc_codec *codec = socdev->codec;
@@ -413,7 +409,7 @@
}
if (codec_dai->ops.prepare) {
- ret = codec_dai->ops.prepare(substream);
+ ret = codec_dai->ops.prepare(substream, codec_dai);
if (ret < 0) {
printk(KERN_ERR "asoc: codec DAI prepare error\n");
goto out;
@@ -421,58 +417,49 @@
}
if (cpu_dai->ops.prepare) {
- ret = cpu_dai->ops.prepare(substream);
+ ret = cpu_dai->ops.prepare(substream, cpu_dai);
if (ret < 0) {
printk(KERN_ERR "asoc: cpu DAI prepare error\n");
goto out;
}
}
- /* we only want to start a DAPM playback stream if we are not waiting
- * on an existing one stopping */
- if (codec_dai->pop_wait) {
- /* we are waiting for the delayed work to start */
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
- snd_soc_dapm_stream_event(socdev->codec,
+ /* cancel any delayed stream shutdown that is pending */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+ codec_dai->pop_wait) {
+ codec_dai->pop_wait = 0;
+ cancel_delayed_work(&card->delayed_work);
+ }
+
+ /* do we need to power up codec */
+ if (codec->bias_level != SND_SOC_BIAS_ON) {
+ snd_soc_dapm_set_bias_level(socdev,
+ SND_SOC_BIAS_PREPARE);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ snd_soc_dapm_stream_event(codec,
+ codec_dai->playback.stream_name,
+ SND_SOC_DAPM_STREAM_START);
+ else
+ snd_soc_dapm_stream_event(codec,
codec_dai->capture.stream_name,
SND_SOC_DAPM_STREAM_START);
- else {
- codec_dai->pop_wait = 0;
- cancel_delayed_work(&socdev->delayed_work);
- snd_soc_dai_digital_mute(codec_dai, 0);
- }
+
+ snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON);
+ snd_soc_dai_digital_mute(codec_dai, 0);
+
} else {
- /* no delayed work - do we need to power up codec */
- if (codec->bias_level != SND_SOC_BIAS_ON) {
-
- snd_soc_dapm_set_bias_level(socdev,
- SND_SOC_BIAS_PREPARE);
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- snd_soc_dapm_stream_event(codec,
+ /* codec already powered - power on widgets */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ snd_soc_dapm_stream_event(codec,
codec_dai->playback.stream_name,
SND_SOC_DAPM_STREAM_START);
- else
- snd_soc_dapm_stream_event(codec,
+ else
+ snd_soc_dapm_stream_event(codec,
codec_dai->capture.stream_name,
SND_SOC_DAPM_STREAM_START);
- snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON);
- snd_soc_dai_digital_mute(codec_dai, 0);
-
- } else {
- /* codec already powered - power on widgets */
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- snd_soc_dapm_stream_event(codec,
- codec_dai->playback.stream_name,
- SND_SOC_DAPM_STREAM_START);
- else
- snd_soc_dapm_stream_event(codec,
- codec_dai->capture.stream_name,
- SND_SOC_DAPM_STREAM_START);
-
- snd_soc_dai_digital_mute(codec_dai, 0);
- }
+ snd_soc_dai_digital_mute(codec_dai, 0);
}
out:
@@ -491,7 +478,8 @@
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_dai_link *machine = rtd->dai;
- struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_card *card = socdev->card;
+ struct snd_soc_platform *platform = card->platform;
struct snd_soc_dai *cpu_dai = machine->cpu_dai;
struct snd_soc_dai *codec_dai = machine->codec_dai;
int ret = 0;
@@ -507,7 +495,7 @@
}
if (codec_dai->ops.hw_params) {
- ret = codec_dai->ops.hw_params(substream, params);
+ ret = codec_dai->ops.hw_params(substream, params, codec_dai);
if (ret < 0) {
printk(KERN_ERR "asoc: can't set codec %s hw params\n",
codec_dai->name);
@@ -516,7 +504,7 @@
}
if (cpu_dai->ops.hw_params) {
- ret = cpu_dai->ops.hw_params(substream, params);
+ ret = cpu_dai->ops.hw_params(substream, params, cpu_dai);
if (ret < 0) {
printk(KERN_ERR "asoc: interface %s hw params failed\n",
cpu_dai->name);
@@ -539,11 +527,11 @@
platform_err:
if (cpu_dai->ops.hw_free)
- cpu_dai->ops.hw_free(substream);
+ cpu_dai->ops.hw_free(substream, cpu_dai);
interface_err:
if (codec_dai->ops.hw_free)
- codec_dai->ops.hw_free(substream);
+ codec_dai->ops.hw_free(substream, codec_dai);
codec_err:
if (machine->ops && machine->ops->hw_free)
@@ -561,7 +549,8 @@
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_dai_link *machine = rtd->dai;
- struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_card *card = socdev->card;
+ struct snd_soc_platform *platform = card->platform;
struct snd_soc_dai *cpu_dai = machine->cpu_dai;
struct snd_soc_dai *codec_dai = machine->codec_dai;
struct snd_soc_codec *codec = socdev->codec;
@@ -582,10 +571,10 @@
/* now free hw params for the DAI's */
if (codec_dai->ops.hw_free)
- codec_dai->ops.hw_free(substream);
+ codec_dai->ops.hw_free(substream, codec_dai);
if (cpu_dai->ops.hw_free)
- cpu_dai->ops.hw_free(substream);
+ cpu_dai->ops.hw_free(substream, cpu_dai);
mutex_unlock(&pcm_mutex);
return 0;
@@ -595,14 +584,15 @@
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_card *card= socdev->card;
struct snd_soc_dai_link *machine = rtd->dai;
- struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_platform *platform = card->platform;
struct snd_soc_dai *cpu_dai = machine->cpu_dai;
struct snd_soc_dai *codec_dai = machine->codec_dai;
int ret;
if (codec_dai->ops.trigger) {
- ret = codec_dai->ops.trigger(substream, cmd);
+ ret = codec_dai->ops.trigger(substream, cmd, codec_dai);
if (ret < 0)
return ret;
}
@@ -614,7 +604,7 @@
}
if (cpu_dai->ops.trigger) {
- ret = cpu_dai->ops.trigger(substream, cmd);
+ ret = cpu_dai->ops.trigger(substream, cmd, cpu_dai);
if (ret < 0)
return ret;
}
@@ -636,8 +626,8 @@
static int soc_suspend(struct platform_device *pdev, pm_message_t state)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
- struct snd_soc_machine *machine = socdev->machine;
- struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_card *card = socdev->card;
+ struct snd_soc_platform *platform = card->platform;
struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
struct snd_soc_codec *codec = socdev->codec;
int i;
@@ -653,29 +643,29 @@
snd_power_change_state(codec->card, SNDRV_CTL_POWER_D3hot);
/* mute any active DAC's */
- for (i = 0; i < machine->num_links; i++) {
- struct snd_soc_dai *dai = machine->dai_link[i].codec_dai;
- if (dai->dai_ops.digital_mute && dai->playback.active)
- dai->dai_ops.digital_mute(dai, 1);
+ for (i = 0; i < card->num_links; i++) {
+ struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
+ if (dai->ops.digital_mute && dai->playback.active)
+ dai->ops.digital_mute(dai, 1);
}
/* suspend all pcms */
- for (i = 0; i < machine->num_links; i++)
- snd_pcm_suspend_all(machine->dai_link[i].pcm);
+ for (i = 0; i < card->num_links; i++)
+ snd_pcm_suspend_all(card->dai_link[i].pcm);
- if (machine->suspend_pre)
- machine->suspend_pre(pdev, state);
+ if (card->suspend_pre)
+ card->suspend_pre(pdev, state);
- for (i = 0; i < machine->num_links; i++) {
- struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
- if (cpu_dai->suspend && cpu_dai->type != SND_SOC_DAI_AC97)
- cpu_dai->suspend(pdev, cpu_dai);
+ for (i = 0; i < card->num_links; i++) {
+ struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
+ if (cpu_dai->suspend && !cpu_dai->ac97_control)
+ cpu_dai->suspend(cpu_dai);
if (platform->suspend)
- platform->suspend(pdev, cpu_dai);
+ platform->suspend(cpu_dai);
}
/* close any waiting streams and save state */
- run_delayed_work(&socdev->delayed_work);
+ run_delayed_work(&card->delayed_work);
codec->suspend_bias_level = codec->bias_level;
for (i = 0; i < codec->num_dai; i++) {
@@ -692,14 +682,14 @@
if (codec_dev->suspend)
codec_dev->suspend(pdev, state);
- for (i = 0; i < machine->num_links; i++) {
- struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
- if (cpu_dai->suspend && cpu_dai->type == SND_SOC_DAI_AC97)
- cpu_dai->suspend(pdev, cpu_dai);
+ for (i = 0; i < card->num_links; i++) {
+ struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
+ if (cpu_dai->suspend && cpu_dai->ac97_control)
+ cpu_dai->suspend(cpu_dai);
}
- if (machine->suspend_post)
- machine->suspend_post(pdev, state);
+ if (card->suspend_post)
+ card->suspend_post(pdev, state);
return 0;
}
@@ -709,11 +699,11 @@
*/
static void soc_resume_deferred(struct work_struct *work)
{
- struct snd_soc_device *socdev = container_of(work,
- struct snd_soc_device,
- deferred_resume_work);
- struct snd_soc_machine *machine = socdev->machine;
- struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_card *card = container_of(work,
+ struct snd_soc_card,
+ deferred_resume_work);
+ struct snd_soc_device *socdev = card->socdev;
+ struct snd_soc_platform *platform = card->platform;
struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
struct snd_soc_codec *codec = socdev->codec;
struct platform_device *pdev = to_platform_device(socdev->dev);
@@ -723,15 +713,15 @@
* so userspace apps are blocked from touching us
*/
- dev_info(socdev->dev, "starting resume work\n");
+ dev_dbg(socdev->dev, "starting resume work\n");
- if (machine->resume_pre)
- machine->resume_pre(pdev);
+ if (card->resume_pre)
+ card->resume_pre(pdev);
- for (i = 0; i < machine->num_links; i++) {
- struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
- if (cpu_dai->resume && cpu_dai->type == SND_SOC_DAI_AC97)
- cpu_dai->resume(pdev, cpu_dai);
+ for (i = 0; i < card->num_links; i++) {
+ struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
+ if (cpu_dai->resume && cpu_dai->ac97_control)
+ cpu_dai->resume(cpu_dai);
}
if (codec_dev->resume)
@@ -749,24 +739,24 @@
}
/* unmute any active DACs */
- for (i = 0; i < machine->num_links; i++) {
- struct snd_soc_dai *dai = machine->dai_link[i].codec_dai;
- if (dai->dai_ops.digital_mute && dai->playback.active)
- dai->dai_ops.digital_mute(dai, 0);
+ for (i = 0; i < card->num_links; i++) {
+ struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
+ if (dai->ops.digital_mute && dai->playback.active)
+ dai->ops.digital_mute(dai, 0);
}
- for (i = 0; i < machine->num_links; i++) {
- struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
- if (cpu_dai->resume && cpu_dai->type != SND_SOC_DAI_AC97)
- cpu_dai->resume(pdev, cpu_dai);
+ for (i = 0; i < card->num_links; i++) {
+ struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
+ if (cpu_dai->resume && !cpu_dai->ac97_control)
+ cpu_dai->resume(cpu_dai);
if (platform->resume)
- platform->resume(pdev, cpu_dai);
+ platform->resume(cpu_dai);
}
- if (machine->resume_post)
- machine->resume_post(pdev);
+ if (card->resume_post)
+ card->resume_post(pdev);
- dev_info(socdev->dev, "resume work completed\n");
+ dev_dbg(socdev->dev, "resume work completed\n");
/* userspace can access us now we are back as we were before */
snd_power_change_state(codec->card, SNDRV_CTL_POWER_D0);
@@ -776,11 +766,12 @@
static int soc_resume(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_card *card = socdev->card;
- dev_info(socdev->dev, "scheduling resume work\n");
+ dev_dbg(socdev->dev, "scheduling resume work\n");
- if (!schedule_work(&socdev->deferred_resume_work))
- dev_err(socdev->dev, "work item may be lost\n");
+ if (!schedule_work(&card->deferred_resume_work))
+ dev_err(socdev->dev, "resume work item may be lost\n");
return 0;
}
@@ -790,23 +781,83 @@
#define soc_resume NULL
#endif
-/* probes a new socdev */
-static int soc_probe(struct platform_device *pdev)
+static void snd_soc_instantiate_card(struct snd_soc_card *card)
{
- int ret = 0, i;
- struct snd_soc_device *socdev = platform_get_drvdata(pdev);
- struct snd_soc_machine *machine = socdev->machine;
- struct snd_soc_platform *platform = socdev->platform;
- struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
+ struct platform_device *pdev = container_of(card->dev,
+ struct platform_device,
+ dev);
+ struct snd_soc_codec_device *codec_dev = card->socdev->codec_dev;
+ struct snd_soc_platform *platform;
+ struct snd_soc_dai *dai;
+ int i, found, ret, ac97;
- if (machine->probe) {
- ret = machine->probe(pdev);
- if (ret < 0)
- return ret;
+ if (card->instantiated)
+ return;
+
+ found = 0;
+ list_for_each_entry(platform, &platform_list, list)
+ if (card->platform == platform) {
+ found = 1;
+ break;
+ }
+ if (!found) {
+ dev_dbg(card->dev, "Platform %s not registered\n",
+ card->platform->name);
+ return;
}
- for (i = 0; i < machine->num_links; i++) {
- struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+ ac97 = 0;
+ for (i = 0; i < card->num_links; i++) {
+ found = 0;
+ list_for_each_entry(dai, &dai_list, list)
+ if (card->dai_link[i].cpu_dai == dai) {
+ found = 1;
+ break;
+ }
+ if (!found) {
+ dev_dbg(card->dev, "DAI %s not registered\n",
+ card->dai_link[i].cpu_dai->name);
+ return;
+ }
+
+ if (card->dai_link[i].cpu_dai->ac97_control)
+ ac97 = 1;
+ }
+
+ /* If we have AC97 in the system then don't wait for the
+ * codec. This will need revisiting if we have to handle
+ * systems with mixed AC97 and non-AC97 parts. Only check for
+ * DAIs currently; we can't do this per link since some AC97
+ * codecs have non-AC97 DAIs.
+ */
+ if (!ac97)
+ for (i = 0; i < card->num_links; i++) {
+ found = 0;
+ list_for_each_entry(dai, &dai_list, list)
+ if (card->dai_link[i].codec_dai == dai) {
+ found = 1;
+ break;
+ }
+ if (!found) {
+ dev_dbg(card->dev, "DAI %s not registered\n",
+ card->dai_link[i].codec_dai->name);
+ return;
+ }
+ }
+
+ /* Note that we do not current check for codec components */
+
+ dev_dbg(card->dev, "All components present, instantiating\n");
+
+ /* Found everything, bring it up */
+ if (card->probe) {
+ ret = card->probe(pdev);
+ if (ret < 0)
+ return;
+ }
+
+ for (i = 0; i < card->num_links; i++) {
+ struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
if (cpu_dai->probe) {
ret = cpu_dai->probe(pdev, cpu_dai);
if (ret < 0)
@@ -827,13 +878,15 @@
}
/* DAPM stream work */
- INIT_DELAYED_WORK(&socdev->delayed_work, close_delayed_work);
+ INIT_DELAYED_WORK(&card->delayed_work, close_delayed_work);
#ifdef CONFIG_PM
/* deferred resume work */
- INIT_WORK(&socdev->deferred_resume_work, soc_resume_deferred);
+ INIT_WORK(&card->deferred_resume_work, soc_resume_deferred);
#endif
- return 0;
+ card->instantiated = 1;
+
+ return;
platform_err:
if (codec_dev->remove)
@@ -841,15 +894,45 @@
cpu_dai_err:
for (i--; i >= 0; i--) {
- struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+ struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
if (cpu_dai->remove)
cpu_dai->remove(pdev, cpu_dai);
}
- if (machine->remove)
- machine->remove(pdev);
+ if (card->remove)
+ card->remove(pdev);
+}
- return ret;
+/*
+ * Attempt to initialise any uninitalised cards. Must be called with
+ * client_mutex.
+ */
+static void snd_soc_instantiate_cards(void)
+{
+ struct snd_soc_card *card;
+ list_for_each_entry(card, &card_list, list)
+ snd_soc_instantiate_card(card);
+}
+
+/* probes a new socdev */
+static int soc_probe(struct platform_device *pdev)
+{
+ int ret = 0;
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_card *card = socdev->card;
+
+ /* Bodge while we push things out of socdev */
+ card->socdev = socdev;
+
+ /* 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;
}
/* removes a socdev */
@@ -857,11 +940,11 @@
{
int i;
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
- struct snd_soc_machine *machine = socdev->machine;
- struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_card *card = socdev->card;
+ struct snd_soc_platform *platform = card->platform;
struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
- run_delayed_work(&socdev->delayed_work);
+ run_delayed_work(&card->delayed_work);
if (platform->remove)
platform->remove(pdev);
@@ -869,14 +952,16 @@
if (codec_dev->remove)
codec_dev->remove(pdev);
- for (i = 0; i < machine->num_links; i++) {
- struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+ for (i = 0; i < card->num_links; i++) {
+ struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
if (cpu_dai->remove)
cpu_dai->remove(pdev, cpu_dai);
}
- if (machine->remove)
- machine->remove(pdev);
+ if (card->remove)
+ card->remove(pdev);
+
+ snd_soc_unregister_card(card);
return 0;
}
@@ -898,6 +983,8 @@
struct snd_soc_dai_link *dai_link, int num)
{
struct snd_soc_codec *codec = socdev->codec;
+ struct snd_soc_card *card = socdev->card;
+ struct snd_soc_platform *platform = card->platform;
struct snd_soc_dai *codec_dai = dai_link->codec_dai;
struct snd_soc_dai *cpu_dai = dai_link->cpu_dai;
struct snd_soc_pcm_runtime *rtd;
@@ -914,8 +1001,8 @@
codec_dai->codec = socdev->codec;
/* check client and interface hw capabilities */
- sprintf(new_name, "%s %s-%s-%d", dai_link->stream_name, codec_dai->name,
- get_dai_name(cpu_dai->type), num);
+ sprintf(new_name, "%s %s-%d", dai_link->stream_name, codec_dai->name,
+ num);
if (codec_dai->playback.channels_min)
playback = 1;
@@ -933,13 +1020,13 @@
dai_link->pcm = pcm;
pcm->private_data = rtd;
- soc_pcm_ops.mmap = socdev->platform->pcm_ops->mmap;
- soc_pcm_ops.pointer = socdev->platform->pcm_ops->pointer;
- soc_pcm_ops.ioctl = socdev->platform->pcm_ops->ioctl;
- soc_pcm_ops.copy = socdev->platform->pcm_ops->copy;
- soc_pcm_ops.silence = socdev->platform->pcm_ops->silence;
- soc_pcm_ops.ack = socdev->platform->pcm_ops->ack;
- soc_pcm_ops.page = socdev->platform->pcm_ops->page;
+ soc_pcm_ops.mmap = platform->pcm_ops->mmap;
+ soc_pcm_ops.pointer = platform->pcm_ops->pointer;
+ soc_pcm_ops.ioctl = platform->pcm_ops->ioctl;
+ soc_pcm_ops.copy = platform->pcm_ops->copy;
+ soc_pcm_ops.silence = platform->pcm_ops->silence;
+ soc_pcm_ops.ack = platform->pcm_ops->ack;
+ soc_pcm_ops.page = platform->pcm_ops->page;
if (playback)
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops);
@@ -947,24 +1034,22 @@
if (capture)
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops);
- ret = socdev->platform->pcm_new(codec->card, codec_dai, pcm);
+ ret = platform->pcm_new(codec->card, codec_dai, pcm);
if (ret < 0) {
printk(KERN_ERR "asoc: platform pcm constructor failed\n");
kfree(rtd);
return ret;
}
- pcm->private_free = socdev->platform->pcm_free;
+ pcm->private_free = platform->pcm_free;
printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
cpu_dai->name);
return ret;
}
/* codec register dump */
-static ssize_t codec_reg_show(struct device *dev,
- struct device_attribute *attr, char *buf)
+static ssize_t soc_codec_reg_show(struct snd_soc_device *devdata, char *buf)
{
- struct snd_soc_device *devdata = dev_get_drvdata(dev);
struct snd_soc_codec *codec = devdata->codec;
int i, step = 1, count = 0;
@@ -1001,8 +1086,110 @@
return count;
}
+static ssize_t codec_reg_show(struct device *dev,
+ struct device_attribute *attr, char *buf)
+{
+ struct snd_soc_device *devdata = dev_get_drvdata(dev);
+ return soc_codec_reg_show(devdata, buf);
+}
+
static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL);
+#ifdef CONFIG_DEBUG_FS
+static int codec_reg_open_file(struct inode *inode, struct file *file)
+{
+ file->private_data = inode->i_private;
+ return 0;
+}
+
+static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf,
+ size_t count, loff_t *ppos)
+{
+ ssize_t ret;
+ struct snd_soc_codec *codec = file->private_data;
+ struct device *card_dev = codec->card->dev;
+ struct snd_soc_device *devdata = card_dev->driver_data;
+ char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
+ if (!buf)
+ return -ENOMEM;
+ ret = soc_codec_reg_show(devdata, buf);
+ if (ret >= 0)
+ ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
+ kfree(buf);
+ return ret;
+}
+
+static ssize_t codec_reg_write_file(struct file *file,
+ const char __user *user_buf, size_t count, loff_t *ppos)
+{
+ char buf[32];
+ int buf_size;
+ char *start = buf;
+ unsigned long reg, value;
+ int step = 1;
+ struct snd_soc_codec *codec = file->private_data;
+
+ buf_size = min(count, (sizeof(buf)-1));
+ if (copy_from_user(buf, user_buf, buf_size))
+ return -EFAULT;
+ buf[buf_size] = 0;
+
+ if (codec->reg_cache_step)
+ step = codec->reg_cache_step;
+
+ while (*start == ' ')
+ start++;
+ reg = simple_strtoul(start, &start, 16);
+ if ((reg >= codec->reg_cache_size) || (reg % step))
+ return -EINVAL;
+ while (*start == ' ')
+ start++;
+ if (strict_strtoul(start, 16, &value))
+ return -EINVAL;
+ codec->write(codec, reg, value);
+ return buf_size;
+}
+
+static const struct file_operations codec_reg_fops = {
+ .open = codec_reg_open_file,
+ .read = codec_reg_read_file,
+ .write = codec_reg_write_file,
+};
+
+static void soc_init_codec_debugfs(struct snd_soc_codec *codec)
+{
+ codec->debugfs_reg = debugfs_create_file("codec_reg", 0644,
+ debugfs_root, codec,
+ &codec_reg_fops);
+ if (!codec->debugfs_reg)
+ printk(KERN_WARNING
+ "ASoC: Failed to create codec register debugfs file\n");
+
+ codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0744,
+ debugfs_root,
+ &codec->pop_time);
+ if (!codec->debugfs_pop_time)
+ printk(KERN_WARNING
+ "Failed to create pop time debugfs file\n");
+}
+
+static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
+{
+ debugfs_remove(codec->debugfs_pop_time);
+ debugfs_remove(codec->debugfs_reg);
+}
+
+#else
+
+static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec)
+{
+}
+
+static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
+{
+}
+#endif
+
/**
* snd_soc_new_ac97_codec - initailise AC97 device
* @codec: audio codec
@@ -1121,7 +1308,7 @@
int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
{
struct snd_soc_codec *codec = socdev->codec;
- struct snd_soc_machine *machine = socdev->machine;
+ struct snd_soc_card *card = socdev->card;
int ret = 0, i;
mutex_lock(&codec->mutex);
@@ -1140,11 +1327,11 @@
strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver));
/* create the pcms */
- for (i = 0; i < machine->num_links; i++) {
- ret = soc_new_pcm(socdev, &machine->dai_link[i], i);
+ for (i = 0; i < card->num_links; i++) {
+ ret = soc_new_pcm(socdev, &card->dai_link[i], i);
if (ret < 0) {
printk(KERN_ERR "asoc: can't create pcm %s\n",
- machine->dai_link[i].stream_name);
+ card->dai_link[i].stream_name);
mutex_unlock(&codec->mutex);
return ret;
}
@@ -1156,7 +1343,7 @@
EXPORT_SYMBOL_GPL(snd_soc_new_pcms);
/**
- * snd_soc_register_card - register sound card
+ * snd_soc_init_card - register sound card
* @socdev: the SoC audio device
*
* Register a SoC sound card. Also registers an AC97 device if the
@@ -1164,29 +1351,28 @@
*
* Returns 0 for success, else error.
*/
-int snd_soc_register_card(struct snd_soc_device *socdev)
+int snd_soc_init_card(struct snd_soc_device *socdev)
{
struct snd_soc_codec *codec = socdev->codec;
- struct snd_soc_machine *machine = socdev->machine;
+ struct snd_soc_card *card = socdev->card;
int ret = 0, i, ac97 = 0, err = 0;
- for (i = 0; i < machine->num_links; i++) {
- if (socdev->machine->dai_link[i].init) {
- err = socdev->machine->dai_link[i].init(codec);
+ for (i = 0; i < card->num_links; i++) {
+ if (card->dai_link[i].init) {
+ err = card->dai_link[i].init(codec);
if (err < 0) {
printk(KERN_ERR "asoc: failed to init %s\n",
- socdev->machine->dai_link[i].stream_name);
+ card->dai_link[i].stream_name);
continue;
}
}
- if (socdev->machine->dai_link[i].codec_dai->type ==
- SND_SOC_DAI_AC97_BUS)
+ if (card->dai_link[i].codec_dai->ac97_control)
ac97 = 1;
}
snprintf(codec->card->shortname, sizeof(codec->card->shortname),
- "%s", machine->name);
+ "%s", card->name);
snprintf(codec->card->longname, sizeof(codec->card->longname),
- "%s (%s)", machine->name, codec->name);
+ "%s (%s)", card->name, codec->name);
ret = snd_card_register(codec->card);
if (ret < 0) {
@@ -1216,12 +1402,13 @@
if (err < 0)
printk(KERN_WARNING "asoc: failed to add codec sysfs files\n");
+ soc_init_codec_debugfs(socdev->codec);
mutex_unlock(&codec->mutex);
out:
return ret;
}
-EXPORT_SYMBOL_GPL(snd_soc_register_card);
+EXPORT_SYMBOL_GPL(snd_soc_init_card);
/**
* snd_soc_free_pcms - free sound card and pcms
@@ -1239,10 +1426,11 @@
#endif
mutex_lock(&codec->mutex);
+ soc_cleanup_codec_debugfs(socdev->codec);
#ifdef CONFIG_SND_SOC_AC97_BUS
for (i = 0; i < codec->num_dai; i++) {
codec_dai = &codec->dai[i];
- if (codec_dai->type == SND_SOC_DAI_AC97_BUS && codec->ac97) {
+ if (codec_dai->ac97_control && codec->ac97) {
soc_ac97_dev_unregister(codec);
goto free_card;
}
@@ -1756,8 +1944,8 @@
int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
unsigned int freq, int dir)
{
- if (dai->dai_ops.set_sysclk)
- return dai->dai_ops.set_sysclk(dai, clk_id, freq, dir);
+ if (dai->ops.set_sysclk)
+ return dai->ops.set_sysclk(dai, clk_id, freq, dir);
else
return -EINVAL;
}
@@ -1776,8 +1964,8 @@
int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
int div_id, int div)
{
- if (dai->dai_ops.set_clkdiv)
- return dai->dai_ops.set_clkdiv(dai, div_id, div);
+ if (dai->ops.set_clkdiv)
+ return dai->ops.set_clkdiv(dai, div_id, div);
else
return -EINVAL;
}
@@ -1795,8 +1983,8 @@
int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
int pll_id, unsigned int freq_in, unsigned int freq_out)
{
- if (dai->dai_ops.set_pll)
- return dai->dai_ops.set_pll(dai, pll_id, freq_in, freq_out);
+ if (dai->ops.set_pll)
+ return dai->ops.set_pll(dai, pll_id, freq_in, freq_out);
else
return -EINVAL;
}
@@ -1805,15 +1993,14 @@
/**
* snd_soc_dai_set_fmt - configure DAI hardware audio format.
* @dai: DAI
- * @clk_id: DAI specific clock ID
* @fmt: SND_SOC_DAIFMT_ format value.
*
* Configures the DAI hardware format and clocking.
*/
int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
{
- if (dai->dai_ops.set_fmt)
- return dai->dai_ops.set_fmt(dai, fmt);
+ if (dai->ops.set_fmt)
+ return dai->ops.set_fmt(dai, fmt);
else
return -EINVAL;
}
@@ -1831,8 +2018,8 @@
int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
unsigned int mask, int slots)
{
- if (dai->dai_ops.set_sysclk)
- return dai->dai_ops.set_tdm_slot(dai, mask, slots);
+ if (dai->ops.set_sysclk)
+ return dai->ops.set_tdm_slot(dai, mask, slots);
else
return -EINVAL;
}
@@ -1847,8 +2034,8 @@
*/
int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate)
{
- if (dai->dai_ops.set_sysclk)
- return dai->dai_ops.set_tristate(dai, tristate);
+ if (dai->ops.set_sysclk)
+ return dai->ops.set_tristate(dai, tristate);
else
return -EINVAL;
}
@@ -1863,21 +2050,242 @@
*/
int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute)
{
- if (dai->dai_ops.digital_mute)
- return dai->dai_ops.digital_mute(dai, mute);
+ if (dai->ops.digital_mute)
+ return dai->ops.digital_mute(dai, mute);
else
return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
-static int __devinit snd_soc_init(void)
+/**
+ * snd_soc_register_card - Register a card with the ASoC core
+ *
+ * @param card Card to register
+ *
+ * Note that currently this is an internal only function: it will be
+ * exposed to machine drivers after further backporting of ASoC v2
+ * registration APIs.
+ */
+static int snd_soc_register_card(struct snd_soc_card *card)
{
- printk(KERN_INFO "ASoC version %s\n", SND_SOC_VERSION);
+ if (!card->name || !card->dev)
+ return -EINVAL;
+
+ INIT_LIST_HEAD(&card->list);
+ card->instantiated = 0;
+
+ mutex_lock(&client_mutex);
+ list_add(&card->list, &card_list);
+ snd_soc_instantiate_cards();
+ mutex_unlock(&client_mutex);
+
+ dev_dbg(card->dev, "Registered card '%s'\n", card->name);
+
+ return 0;
+}
+
+/**
+ * snd_soc_unregister_card - Unregister a card with the ASoC core
+ *
+ * @param card Card to unregister
+ *
+ * Note that currently this is an internal only function: it will be
+ * exposed to machine drivers after further backporting of ASoC v2
+ * registration APIs.
+ */
+static int snd_soc_unregister_card(struct snd_soc_card *card)
+{
+ mutex_lock(&client_mutex);
+ list_del(&card->list);
+ mutex_unlock(&client_mutex);
+
+ dev_dbg(card->dev, "Unregistered card '%s'\n", card->name);
+
+ return 0;
+}
+
+/**
+ * snd_soc_register_dai - Register a DAI with the ASoC core
+ *
+ * @param dai DAI to register
+ */
+int snd_soc_register_dai(struct snd_soc_dai *dai)
+{
+ if (!dai->name)
+ return -EINVAL;
+
+ /* The device should become mandatory over time */
+ if (!dai->dev)
+ printk(KERN_WARNING "No device for DAI %s\n", dai->name);
+
+ INIT_LIST_HEAD(&dai->list);
+
+ mutex_lock(&client_mutex);
+ list_add(&dai->list, &dai_list);
+ snd_soc_instantiate_cards();
+ mutex_unlock(&client_mutex);
+
+ pr_debug("Registered DAI '%s'\n", dai->name);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_register_dai);
+
+/**
+ * snd_soc_unregister_dai - Unregister a DAI from the ASoC core
+ *
+ * @param dai DAI to unregister
+ */
+void snd_soc_unregister_dai(struct snd_soc_dai *dai)
+{
+ mutex_lock(&client_mutex);
+ list_del(&dai->list);
+ mutex_unlock(&client_mutex);
+
+ pr_debug("Unregistered DAI '%s'\n", dai->name);
+}
+EXPORT_SYMBOL_GPL(snd_soc_unregister_dai);
+
+/**
+ * snd_soc_register_dais - Register multiple DAIs with the ASoC core
+ *
+ * @param dai Array of DAIs to register
+ * @param count Number of DAIs
+ */
+int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count)
+{
+ int i, ret;
+
+ for (i = 0; i < count; i++) {
+ ret = snd_soc_register_dai(&dai[i]);
+ if (ret != 0)
+ goto err;
+ }
+
+ return 0;
+
+err:
+ for (i--; i >= 0; i--)
+ snd_soc_unregister_dai(&dai[i]);
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_register_dais);
+
+/**
+ * snd_soc_unregister_dais - Unregister multiple DAIs from the ASoC core
+ *
+ * @param dai Array of DAIs to unregister
+ * @param count Number of DAIs
+ */
+void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count)
+{
+ int i;
+
+ for (i = 0; i < count; i++)
+ snd_soc_unregister_dai(&dai[i]);
+}
+EXPORT_SYMBOL_GPL(snd_soc_unregister_dais);
+
+/**
+ * snd_soc_register_platform - Register a platform with the ASoC core
+ *
+ * @param platform platform to register
+ */
+int snd_soc_register_platform(struct snd_soc_platform *platform)
+{
+ if (!platform->name)
+ return -EINVAL;
+
+ INIT_LIST_HEAD(&platform->list);
+
+ mutex_lock(&client_mutex);
+ list_add(&platform->list, &platform_list);
+ snd_soc_instantiate_cards();
+ mutex_unlock(&client_mutex);
+
+ pr_debug("Registered platform '%s'\n", platform->name);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_register_platform);
+
+/**
+ * snd_soc_unregister_platform - Unregister a platform from the ASoC core
+ *
+ * @param platform platform to unregister
+ */
+void snd_soc_unregister_platform(struct snd_soc_platform *platform)
+{
+ mutex_lock(&client_mutex);
+ list_del(&platform->list);
+ mutex_unlock(&client_mutex);
+
+ pr_debug("Unregistered platform '%s'\n", platform->name);
+}
+EXPORT_SYMBOL_GPL(snd_soc_unregister_platform);
+
+/**
+ * snd_soc_register_codec - Register a codec with the ASoC core
+ *
+ * @param codec codec to register
+ */
+int snd_soc_register_codec(struct snd_soc_codec *codec)
+{
+ if (!codec->name)
+ return -EINVAL;
+
+ /* The device should become mandatory over time */
+ if (!codec->dev)
+ printk(KERN_WARNING "No device for codec %s\n", codec->name);
+
+ INIT_LIST_HEAD(&codec->list);
+
+ mutex_lock(&client_mutex);
+ list_add(&codec->list, &codec_list);
+ snd_soc_instantiate_cards();
+ mutex_unlock(&client_mutex);
+
+ pr_debug("Registered codec '%s'\n", codec->name);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_register_codec);
+
+/**
+ * snd_soc_unregister_codec - Unregister a codec from the ASoC core
+ *
+ * @param codec codec to unregister
+ */
+void snd_soc_unregister_codec(struct snd_soc_codec *codec)
+{
+ mutex_lock(&client_mutex);
+ list_del(&codec->list);
+ mutex_unlock(&client_mutex);
+
+ pr_debug("Unregistered codec '%s'\n", codec->name);
+}
+EXPORT_SYMBOL_GPL(snd_soc_unregister_codec);
+
+static int __init snd_soc_init(void)
+{
+#ifdef CONFIG_DEBUG_FS
+ debugfs_root = debugfs_create_dir("asoc", NULL);
+ if (IS_ERR(debugfs_root) || !debugfs_root) {
+ printk(KERN_WARNING
+ "ASoC: Failed to create debugfs directory\n");
+ debugfs_root = NULL;
+ }
+#endif
+
return platform_driver_register(&soc_driver);
}
-static void snd_soc_exit(void)
+static void __exit snd_soc_exit(void)
{
+#ifdef CONFIG_DEBUG_FS
+ debugfs_remove_recursive(debugfs_root);
+#endif
platform_driver_unregister(&soc_driver);
}
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 7351db9..8863edd 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -37,7 +37,6 @@
#include <linux/bitops.h>
#include <linux/platform_device.h>
#include <linux/jiffies.h>
-#include <linux/debugfs.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -67,17 +66,13 @@
module_param(dapm_status, int, 0);
MODULE_PARM_DESC(dapm_status, "enable DPM sysfs entries");
-static struct dentry *asoc_debugfs;
-
-static u32 pop_time;
-
-static void pop_wait(void)
+static void pop_wait(u32 pop_time)
{
if (pop_time)
schedule_timeout_uninterruptible(msecs_to_jiffies(pop_time));
}
-static void pop_dbg(const char *fmt, ...)
+static void pop_dbg(u32 pop_time, const char *fmt, ...)
{
va_list args;
@@ -85,7 +80,7 @@
if (pop_time) {
vprintk(fmt, args);
- pop_wait();
+ pop_wait(pop_time);
}
va_end(args);
@@ -230,10 +225,11 @@
change = old != new;
if (change) {
- pop_dbg("pop test %s : %s in %d ms\n", widget->name,
- widget->power ? "on" : "off", pop_time);
+ pop_dbg(codec->pop_time, "pop test %s : %s in %d ms\n",
+ widget->name, widget->power ? "on" : "off",
+ codec->pop_time);
snd_soc_write(codec, widget->reg, new);
- pop_wait();
+ pop_wait(codec->pop_time);
}
pr_debug("reg %x old %x new %x change %d\n", widget->reg,
old, new, change);
@@ -293,7 +289,7 @@
struct snd_soc_dapm_widget *w)
{
int i, ret = 0;
- char name[32];
+ size_t name_len;
struct snd_soc_dapm_path *path;
/* add kcontrol */
@@ -307,11 +303,16 @@
continue;
/* add dapm control with long name */
- snprintf(name, 32, "%s %s", w->name, w->kcontrols[i].name);
- path->long_name = kstrdup (name, GFP_KERNEL);
+ name_len = 2 + strlen(w->name)
+ + strlen(w->kcontrols[i].name);
+ path->long_name = kmalloc(name_len, GFP_KERNEL);
if (path->long_name == NULL)
return -ENOMEM;
+ snprintf(path->long_name, name_len, "%s %s",
+ w->name, w->kcontrols[i].name);
+ path->long_name[name_len - 1] = '\0';
+
path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w,
path->long_name);
ret = snd_ctl_add(codec->card, path->kcontrol);
@@ -821,23 +822,9 @@
int snd_soc_dapm_sys_add(struct device *dev)
{
- int ret = 0;
-
if (!dapm_status)
return 0;
-
- ret = device_create_file(dev, &dev_attr_dapm_widget);
- if (ret != 0)
- return ret;
-
- asoc_debugfs = debugfs_create_dir("asoc", NULL);
- if (!IS_ERR(asoc_debugfs) && asoc_debugfs)
- debugfs_create_u32("dapm_pop_time", 0744, asoc_debugfs,
- &pop_time);
- else
- asoc_debugfs = NULL;
-
- return 0;
+ return device_create_file(dev, &dev_attr_dapm_widget);
}
static void snd_soc_dapm_sys_remove(struct device *dev)
@@ -845,9 +832,6 @@
if (dapm_status) {
device_remove_file(dev, &dev_attr_dapm_widget);
}
-
- if (asoc_debugfs)
- debugfs_remove_recursive(asoc_debugfs);
}
/* free all dapm widgets and resources */
@@ -1007,28 +991,6 @@
}
/**
- * snd_soc_dapm_connect_input - connect dapm widgets
- * @codec: audio codec
- * @sink: name of target widget
- * @control: mixer control name
- * @source: name of source name
- *
- * Connects 2 dapm widgets together via a named audio path. The sink is
- * the widget receiving the audio signal, whilst the source is the sender
- * of the audio signal.
- *
- * This function has been deprecated in favour of snd_soc_dapm_add_routes().
- *
- * Returns 0 for success else error.
- */
-int snd_soc_dapm_connect_input(struct snd_soc_codec *codec, const char *sink,
- const char *control, const char *source)
-{
- return snd_soc_dapm_add_route(codec, sink, control, source);
-}
-EXPORT_SYMBOL_GPL(snd_soc_dapm_connect_input);
-
-/**
* snd_soc_dapm_add_routes - Add routes between DAPM widgets
* @codec: codec
* @route: audio routes
@@ -1358,8 +1320,12 @@
for (i = 0; i < num; i++) {
ret = snd_soc_dapm_new_control(codec, widget);
- if (ret < 0)
+ if (ret < 0) {
+ printk(KERN_ERR
+ "ASoC: Failed to create DAPM control %s: %d\n",
+ widget->name, ret);
return ret;
+ }
widget++;
}
return 0;
@@ -1440,11 +1406,11 @@
enum snd_soc_bias_level level)
{
struct snd_soc_codec *codec = socdev->codec;
- struct snd_soc_machine *machine = socdev->machine;
+ struct snd_soc_card *card = socdev->card;
int ret = 0;
- if (machine->set_bias_level)
- ret = machine->set_bias_level(machine, level);
+ if (card->set_bias_level)
+ ret = card->set_bias_level(card, level);
if (ret == 0 && codec->set_bias_level)
ret = codec->set_bias_level(codec, level);