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-rw-r--r--include/sound/da7219.h2
-rw-r--r--include/sound/hda_register.h36
-rw-r--r--include/sound/hdaudio.h13
-rw-r--r--include/sound/hdaudio_ext.h12
-rw-r--r--include/sound/l3.h15
-rw-r--r--include/sound/rt5660.h31
-rw-r--r--include/sound/s3c24xx_uda134x.h1
-rw-r--r--include/sound/simple_card_utils.h35
-rw-r--r--include/sound/soc.h19
-rw-r--r--include/sound/tlv.h78
-rw-r--r--include/uapi/sound/Kbuild1
-rw-r--r--include/uapi/sound/asoc.h35
-rw-r--r--include/uapi/sound/asound.h3
-rw-r--r--include/uapi/sound/snd_sst_tokens.h214
-rw-r--r--include/uapi/sound/tlv.h69
15 files changed, 473 insertions, 91 deletions
diff --git a/include/sound/da7219.h b/include/sound/da7219.h
index 02876acdc840..409ef1397fd3 100644
--- a/include/sound/da7219.h
+++ b/include/sound/da7219.h
@@ -34,6 +34,8 @@ enum da7219_mic_amp_in_sel {
struct da7219_aad_pdata;
struct da7219_pdata {
+ bool wakeup_source;
+
/* Mic */
enum da7219_micbias_voltage micbias_lvl;
enum da7219_mic_amp_in_sel mic_amp_in_sel;
diff --git a/include/sound/hda_register.h b/include/sound/hda_register.h
index ff1aecf325e8..0013063db7f2 100644
--- a/include/sound/hda_register.h
+++ b/include/sound/hda_register.h
@@ -89,6 +89,19 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
#define AZX_REG_SD_BDLPL 0x18
#define AZX_REG_SD_BDLPU 0x1c
+/* GTS registers */
+#define AZX_REG_LLCH 0x14
+
+#define AZX_REG_GTS_BASE 0x520
+
+#define AZX_REG_GTSCC (AZX_REG_GTS_BASE + 0x00)
+#define AZX_REG_WALFCC (AZX_REG_GTS_BASE + 0x04)
+#define AZX_REG_TSCCL (AZX_REG_GTS_BASE + 0x08)
+#define AZX_REG_TSCCU (AZX_REG_GTS_BASE + 0x0C)
+#define AZX_REG_LLPFOC (AZX_REG_GTS_BASE + 0x14)
+#define AZX_REG_LLPCL (AZX_REG_GTS_BASE + 0x18)
+#define AZX_REG_LLPCU (AZX_REG_GTS_BASE + 0x1C)
+
/* Haswell/Broadwell display HD-A controller Extended Mode registers */
#define AZX_REG_HSW_EM4 0x100c
#define AZX_REG_HSW_EM5 0x1010
@@ -242,6 +255,29 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
/* Interval used to calculate the iterating register offset */
#define AZX_DRSM_INTERVAL 0x08
+/* Global time synchronization registers */
+#define GTSCC_TSCCD_MASK 0x80000000
+#define GTSCC_TSCCD_SHIFT BIT(31)
+#define GTSCC_TSCCI_MASK 0x20
+#define GTSCC_CDMAS_DMA_DIR_SHIFT 4
+
+#define WALFCC_CIF_MASK 0x1FF
+#define WALFCC_FN_SHIFT 9
+#define HDA_CLK_CYCLES_PER_FRAME 512
+
+/*
+ * An error occurs near frame "rollover". The clocks in frame value indicates
+ * whether this error may have occurred. Here we use the value of 10. Please
+ * see the errata for the right number [<10]
+ */
+#define HDA_MAX_CYCLE_VALUE 499
+#define HDA_MAX_CYCLE_OFFSET 10
+#define HDA_MAX_CYCLE_READ_RETRY 10
+
+#define TSCCU_CCU_SHIFT 32
+#define LLPC_CCU_SHIFT 32
+
+
/*
* helpers to read the stream position
*/
diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h
index 93e63c56f48f..56004ec8d441 100644
--- a/include/sound/hdaudio.h
+++ b/include/sound/hdaudio.h
@@ -245,6 +245,12 @@ struct hdac_rb {
/*
* HD-audio bus base driver
+ *
+ * @ppcap: pp capabilities pointer
+ * @spbcap: SPIB capabilities pointer
+ * @mlcap: MultiLink capabilities pointer
+ * @gtscap: gts capabilities pointer
+ * @drsmcap: dma resume capabilities pointer
*/
struct hdac_bus {
struct device *dev;
@@ -256,6 +262,12 @@ struct hdac_bus {
void __iomem *remap_addr;
int irq;
+ void __iomem *ppcap;
+ void __iomem *spbcap;
+ void __iomem *mlcap;
+ void __iomem *gtscap;
+ void __iomem *drsmcap;
+
/* codec linked list */
struct list_head codec_list;
unsigned int num_codecs;
@@ -335,6 +347,7 @@ static inline void snd_hdac_codec_link_down(struct hdac_device *codec)
int snd_hdac_bus_send_cmd(struct hdac_bus *bus, unsigned int val);
int snd_hdac_bus_get_response(struct hdac_bus *bus, unsigned int addr,
unsigned int *res);
+int snd_hdac_bus_parse_capabilities(struct hdac_bus *bus);
int snd_hdac_link_power(struct hdac_device *codec, bool enable);
bool snd_hdac_bus_init_chip(struct hdac_bus *bus, bool full_reset);
diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h
index b9593b201599..8660a7f10851 100644
--- a/include/sound/hdaudio_ext.h
+++ b/include/sound/hdaudio_ext.h
@@ -8,11 +8,6 @@
*
* @bus: hdac bus
* @num_streams: streams supported
- * @ppcap: pp capabilities pointer
- * @spbcap: SPIB capabilities pointer
- * @mlcap: MultiLink capabilities pointer
- * @gtscap: gts capabilities pointer
- * @drsmcap: dma resume capabilities pointer
* @hlink_list: link list of HDA links
* @lock: lock for link mgmt
* @cmd_dma_state: state of cmd DMAs: CORB and RIRB
@@ -22,12 +17,6 @@ struct hdac_ext_bus {
int num_streams;
int idx;
- void __iomem *ppcap;
- void __iomem *spbcap;
- void __iomem *mlcap;
- void __iomem *gtscap;
- void __iomem *drsmcap;
-
struct list_head hlink_list;
struct mutex lock;
@@ -54,7 +43,6 @@ void snd_hdac_ext_bus_device_remove(struct hdac_ext_bus *ebus);
#define HDA_CODEC_EXT_ENTRY(_vid, _revid, _name, _drv_data) \
HDA_CODEC_REV_EXT_ENTRY(_vid, _revid, _name, _drv_data)
-int snd_hdac_ext_bus_parse_capabilities(struct hdac_ext_bus *sbus);
void snd_hdac_ext_bus_ppcap_enable(struct hdac_ext_bus *chip, bool enable);
void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_ext_bus *chip, bool enable);
diff --git a/include/sound/l3.h b/include/sound/l3.h
index 423a08f0f1b0..1471da22adad 100644
--- a/include/sound/l3.h
+++ b/include/sound/l3.h
@@ -2,9 +2,15 @@
#define _L3_H_ 1
struct l3_pins {
- void (*setdat)(int);
- void (*setclk)(int);
- void (*setmode)(int);
+ void (*setdat)(struct l3_pins *, int);
+ void (*setclk)(struct l3_pins *, int);
+ void (*setmode)(struct l3_pins *, int);
+
+ int gpio_data;
+ int gpio_clk;
+ int gpio_mode;
+ int use_gpios;
+
int data_hold;
int data_setup;
int clock_high;
@@ -13,6 +19,9 @@ struct l3_pins {
int mode_setup;
};
+struct device;
+
int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len);
+int l3_set_gpio_ops(struct device *dev, struct l3_pins *adap);
#endif
diff --git a/include/sound/rt5660.h b/include/sound/rt5660.h
new file mode 100644
index 000000000000..065f83a24db6
--- /dev/null
+++ b/include/sound/rt5660.h
@@ -0,0 +1,31 @@
+/*
+ * linux/sound/rt5660.h -- Platform data for RT5660
+ *
+ * Copyright 2016 Realtek Semiconductor Corp.
+ * Author: Oder Chiou <oder_chiou@realtek.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 __LINUX_SND_RT5660_H
+#define __LINUX_SND_RT5660_H
+
+enum rt5660_dmic1_data_pin {
+ RT5660_DMIC1_NULL,
+ RT5660_DMIC1_DATA_GPIO2,
+ RT5660_DMIC1_DATA_IN1P,
+};
+
+struct rt5660_platform_data {
+ /* IN1 & IN3 can optionally be differential */
+ bool in1_diff;
+ bool in3_diff;
+ bool use_ldo2;
+ bool poweroff_codec_in_suspend;
+
+ enum rt5660_dmic1_data_pin dmic1_data_pin;
+};
+
+#endif
diff --git a/include/sound/s3c24xx_uda134x.h b/include/sound/s3c24xx_uda134x.h
index 33df4cb909d3..ffaf1f098c8e 100644
--- a/include/sound/s3c24xx_uda134x.h
+++ b/include/sound/s3c24xx_uda134x.h
@@ -7,7 +7,6 @@ struct s3c24xx_uda134x_platform_data {
int l3_clk;
int l3_mode;
int l3_data;
- void (*power) (int);
int model;
};
diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h
index 86088aed9002..fd6412551145 100644
--- a/include/sound/simple_card_utils.h
+++ b/include/sound/simple_card_utils.h
@@ -27,10 +27,45 @@ int asoc_simple_card_parse_daifmt(struct device *dev,
struct device_node *codec,
char *prefix,
unsigned int *retfmt);
+__printf(3, 4)
int asoc_simple_card_set_dailink_name(struct device *dev,
struct snd_soc_dai_link *dai_link,
const char *fmt, ...);
int asoc_simple_card_parse_card_name(struct snd_soc_card *card,
char *prefix);
+#define asoc_simple_card_parse_clk_cpu(node, dai_link, simple_dai) \
+ asoc_simple_card_parse_clk(node, dai_link->cpu_of_node, simple_dai)
+#define asoc_simple_card_parse_clk_codec(node, dai_link, simple_dai) \
+ asoc_simple_card_parse_clk(node, dai_link->codec_of_node, simple_dai)
+int asoc_simple_card_parse_clk(struct device_node *node,
+ struct device_node *dai_of_node,
+ struct asoc_simple_dai *simple_dai);
+
+#define asoc_simple_card_parse_cpu(node, dai_link, \
+ list_name, cells_name, is_single_link) \
+ asoc_simple_card_parse_dai(node, &dai_link->cpu_of_node, \
+ &dai_link->cpu_dai_name, list_name, cells_name, is_single_link)
+#define asoc_simple_card_parse_codec(node, dai_link, list_name, cells_name) \
+ asoc_simple_card_parse_dai(node, &dai_link->codec_of_node, \
+ &dai_link->codec_dai_name, list_name, cells_name, NULL)
+#define asoc_simple_card_parse_platform(node, dai_link, list_name, cells_name) \
+ asoc_simple_card_parse_dai(node, &dai_link->platform_of_node, \
+ NULL, list_name, cells_name, NULL)
+int asoc_simple_card_parse_dai(struct device_node *node,
+ struct device_node **endpoint_np,
+ const char **dai_name,
+ const char *list_name,
+ const char *cells_name,
+ int *is_single_links);
+
+int asoc_simple_card_init_dai(struct snd_soc_dai *dai,
+ struct asoc_simple_dai *simple_dai);
+
+int asoc_simple_card_canonicalize_dailink(struct snd_soc_dai_link *dai_link);
+void asoc_simple_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link,
+ int is_single_links);
+
+int asoc_simple_card_clean_reference(struct snd_soc_card *card);
+
#endif /* __SIMPLE_CARD_CORE_H */
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 6144882cc96a..4f1c784e44f6 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -898,14 +898,6 @@ struct snd_soc_codec_driver {
int (*resume)(struct snd_soc_codec *);
struct snd_soc_component_driver component_driver;
- /* Default control and setup, added after probe() is run */
- const struct snd_kcontrol_new *controls;
- int num_controls;
- const struct snd_soc_dapm_widget *dapm_widgets;
- int num_dapm_widgets;
- const struct snd_soc_dapm_route *dapm_routes;
- int num_dapm_routes;
-
/* codec wide operations */
int (*set_sysclk)(struct snd_soc_codec *codec,
int clk_id, int source, unsigned int freq, int dir);
@@ -1547,17 +1539,6 @@ static inline void *snd_soc_platform_get_drvdata(struct snd_soc_platform *platfo
return snd_soc_component_get_drvdata(&platform->component);
}
-static inline void snd_soc_pcm_set_drvdata(struct snd_soc_pcm_runtime *rtd,
- void *data)
-{
- dev_set_drvdata(rtd->dev, data);
-}
-
-static inline void *snd_soc_pcm_get_drvdata(struct snd_soc_pcm_runtime *rtd)
-{
- return dev_get_drvdata(rtd->dev);
-}
-
static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card)
{
INIT_LIST_HEAD(&card->codec_dev_list);
diff --git a/include/sound/tlv.h b/include/sound/tlv.h
index df97d1966468..3677ebb928d5 100644
--- a/include/sound/tlv.h
+++ b/include/sound/tlv.h
@@ -22,67 +22,39 @@
*
*/
-/*
- * TLV structure is right behind the struct snd_ctl_tlv:
- * unsigned int type - see SNDRV_CTL_TLVT_*
- * unsigned int length
- * .... data aligned to sizeof(unsigned int), use
- * block_length = (length + (sizeof(unsigned int) - 1)) &
- * ~(sizeof(unsigned int) - 1)) ....
- */
-
#include <uapi/sound/tlv.h>
-#define TLV_ITEM(type, ...) \
- (type), TLV_LENGTH(__VA_ARGS__), __VA_ARGS__
-#define TLV_LENGTH(...) \
- ((unsigned int)sizeof((const unsigned int[]) { __VA_ARGS__ }))
+/* For historical reasons, these macros are aliases to the ones in UAPI. */
+#define TLV_ITEM SNDRV_CTL_TLVD_ITEM
+#define TLV_LENGTH SNDRV_CTL_TLVD_LENGTH
+
+#define TLV_CONTAINER_ITEM SNDRV_CTL_TLVD_CONTAINER_ITEM
+#define DECLARE_TLV_CONTAINER SNDRV_CTL_TLVD_DECLARE_CONTAINER
-#define TLV_CONTAINER_ITEM(...) \
- TLV_ITEM(SNDRV_CTL_TLVT_CONTAINER, __VA_ARGS__)
-#define DECLARE_TLV_CONTAINER(name, ...) \
- unsigned int name[] = { TLV_CONTAINER_ITEM(__VA_ARGS__) }
+#define TLV_DB_SCALE_MASK SNDRV_CTL_TLVD_DB_SCALE_MASK
+#define TLV_DB_SCALE_MUTE SNDRV_CTL_TLVD_DB_SCALE_MUTE
+#define TLV_DB_SCALE_ITEM SNDRV_CTL_TLVD_DB_SCALE_ITEM
+#define DECLARE_TLV_DB_SCALE SNDRV_CTL_TLVD_DECLARE_DB_SCALE
-#define TLV_DB_SCALE_MASK 0xffff
-#define TLV_DB_SCALE_MUTE 0x10000
-#define TLV_DB_SCALE_ITEM(min, step, mute) \
- TLV_ITEM(SNDRV_CTL_TLVT_DB_SCALE, \
- (min), \
- ((step) & TLV_DB_SCALE_MASK) | \
- ((mute) ? TLV_DB_SCALE_MUTE : 0))
-#define DECLARE_TLV_DB_SCALE(name, min, step, mute) \
- unsigned int name[] = { TLV_DB_SCALE_ITEM(min, step, mute) }
+#define TLV_DB_MINMAX_ITEM SNDRV_CTL_TLVD_DB_MINMAX_ITEM
+#define TLV_DB_MINMAX_MUTE_ITEM SNDRV_CTL_TLVD_DB_MINMAX_MUTE_ITEM
+#define DECLARE_TLV_DB_MINMAX SNDRV_CTL_TLVD_DECLARE_DB_MINMAX
+#define DECLARE_TLV_DB_MINMAX_MUTE SNDRV_CTL_TLVD_DECLARE_DB_MINMAX_MUTE
-/* dB scale specified with min/max values instead of step */
-#define TLV_DB_MINMAX_ITEM(min_dB, max_dB) \
- TLV_ITEM(SNDRV_CTL_TLVT_DB_MINMAX, (min_dB), (max_dB))
-#define TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) \
- TLV_ITEM(SNDRV_CTL_TLVT_DB_MINMAX_MUTE, (min_dB), (max_dB))
-#define DECLARE_TLV_DB_MINMAX(name, min_dB, max_dB) \
- unsigned int name[] = { TLV_DB_MINMAX_ITEM(min_dB, max_dB) }
-#define DECLARE_TLV_DB_MINMAX_MUTE(name, min_dB, max_dB) \
- unsigned int name[] = { TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) }
+#define TLV_DB_LINEAR_ITEM SNDRV_CTL_TLVD_DB_LINEAR_ITEM
+#define DECLARE_TLV_DB_LINEAR SNDRV_CTL_TLVD_DECLARE_DB_LINEAR
-/* linear volume between min_dB and max_dB (.01dB unit) */
-#define TLV_DB_LINEAR_ITEM(min_dB, max_dB) \
- TLV_ITEM(SNDRV_CTL_TLVT_DB_LINEAR, (min_dB), (max_dB))
-#define DECLARE_TLV_DB_LINEAR(name, min_dB, max_dB) \
- unsigned int name[] = { TLV_DB_LINEAR_ITEM(min_dB, max_dB) }
+#define TLV_DB_RANGE_ITEM SNDRV_CTL_TLVD_DB_RANGE_ITEM
+#define DECLARE_TLV_DB_RANGE SNDRV_CTL_TLVD_DECLARE_DB_RANGE
-/* dB range container:
- * Items in dB range container must be ordered by their values and by their
- * dB values. This implies that larger values must correspond with larger
- * dB values (which is also required for all other mixer controls).
+#define TLV_DB_GAIN_MUTE SNDRV_CTL_TLVD_DB_GAIN_MUTE
+
+/*
+ * The below assumes that each item TLV is 4 words like DB_SCALE or LINEAR.
+ * This is an old fasion and obsoleted by commit bf1d1c9b6179("ALSA: tlv: add
+ * DECLARE_TLV_DB_RANGE()").
*/
-/* Each item is: <min> <max> <TLV> */
-#define TLV_DB_RANGE_ITEM(...) \
- TLV_ITEM(SNDRV_CTL_TLVT_DB_RANGE, __VA_ARGS__)
-#define DECLARE_TLV_DB_RANGE(name, ...) \
- unsigned int name[] = { TLV_DB_RANGE_ITEM(__VA_ARGS__) }
-/* The below assumes that each item TLV is 4 words like DB_SCALE or LINEAR */
-#define TLV_DB_RANGE_HEAD(num) \
+#define TLV_DB_RANGE_HEAD(num) \
SNDRV_CTL_TLVT_DB_RANGE, 6 * (num) * sizeof(unsigned int)
-#define TLV_DB_GAIN_MUTE -9999999
-
#endif /* __SOUND_TLV_H */
diff --git a/include/uapi/sound/Kbuild b/include/uapi/sound/Kbuild
index 691984cb0b91..9578d8bdbf31 100644
--- a/include/uapi/sound/Kbuild
+++ b/include/uapi/sound/Kbuild
@@ -13,3 +13,4 @@ header-y += sb16_csp.h
header-y += sfnt_info.h
header-y += tlv.h
header-y += usb_stream.h
+header-y += snd_sst_tokens.h
diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h
index e4701a3c6331..33d00a4ce656 100644
--- a/include/uapi/sound/asoc.h
+++ b/include/uapi/sound/asoc.h
@@ -83,7 +83,7 @@
#define SND_SOC_TPLG_NUM_TEXTS 16
/* ABI version */
-#define SND_SOC_TPLG_ABI_VERSION 0x4
+#define SND_SOC_TPLG_ABI_VERSION 0x5
/* Max size of TLV data */
#define SND_SOC_TPLG_TLV_SIZE 32
@@ -105,7 +105,8 @@
#define SND_SOC_TPLG_TYPE_CODEC_LINK 9
#define SND_SOC_TPLG_TYPE_BACKEND_LINK 10
#define SND_SOC_TPLG_TYPE_PDATA 11
-#define SND_SOC_TPLG_TYPE_MAX SND_SOC_TPLG_TYPE_PDATA
+#define SND_SOC_TPLG_TYPE_BE_DAI 12
+#define SND_SOC_TPLG_TYPE_MAX SND_SOC_TPLG_TYPE_BE_DAI
/* vendor block IDs - please add new vendor types to end */
#define SND_SOC_TPLG_TYPE_VENDOR_FW 1000
@@ -124,6 +125,11 @@
#define SND_SOC_TPLG_TUPLE_TYPE_WORD 4
#define SND_SOC_TPLG_TUPLE_TYPE_SHORT 5
+/* BE DAI flags */
+#define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_RATES (1 << 0)
+#define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_CHANNELS (1 << 1)
+#define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_SAMPLEBITS (1 << 2)
+
/*
* Block Header.
* This header precedes all object and object arrays below.
@@ -251,6 +257,7 @@ struct snd_soc_tplg_stream_caps {
__le32 period_size_max; /* max period size bytes */
__le32 buffer_size_min; /* min buffer size bytes */
__le32 buffer_size_max; /* max buffer size bytes */
+ __le32 sig_bits; /* number of bits of content */
} __attribute__((packed));
/*
@@ -285,6 +292,8 @@ struct snd_soc_tplg_manifest {
__le32 graph_elems; /* number of graph elements */
__le32 pcm_elems; /* number of PCM elements */
__le32 dai_link_elems; /* number of DAI link elements */
+ __le32 be_dai_elems; /* number of BE DAI elements */
+ __le32 reserved[20]; /* reserved for new ABI element types */
struct snd_soc_tplg_private priv;
} __attribute__((packed));
@@ -450,4 +459,26 @@ struct snd_soc_tplg_link_config {
struct snd_soc_tplg_stream stream[SND_SOC_TPLG_STREAM_CONFIG_MAX]; /* supported configs playback and captrure */
__le32 num_streams; /* number of streams */
} __attribute__((packed));
+
+/*
+ * Describes SW/FW specific features of BE DAI.
+ *
+ * File block representation for BE DAI :-
+ * +-----------------------------------+-----+
+ * | struct snd_soc_tplg_hdr | 1 |
+ * +-----------------------------------+-----+
+ * | struct snd_soc_tplg_be_dai | N |
+ * +-----------------------------------+-----+
+ */
+struct snd_soc_tplg_be_dai {
+ __le32 size; /* in bytes of this structure */
+ char dai_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; /* name - used to match */
+ __le32 dai_id; /* unique ID - used to match */
+ __le32 playback; /* supports playback mode */
+ __le32 capture; /* supports capture mode */
+ struct snd_soc_tplg_stream_caps caps[2]; /* playback and capture for DAI */
+ __le32 flag_mask; /* bitmask of flags to configure */
+ __le32 flags; /* SND_SOC_TPLG_DAI_FLGBIT_* */
+ struct snd_soc_tplg_private priv;
+} __attribute__((packed));
#endif
diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h
index 609cadb8739d..be353a78c303 100644
--- a/include/uapi/sound/asound.h
+++ b/include/uapi/sound/asound.h
@@ -106,9 +106,10 @@ enum {
SNDRV_HWDEP_IFACE_FW_OXFW, /* Oxford OXFW970/971 based device */
SNDRV_HWDEP_IFACE_FW_DIGI00X, /* Digidesign Digi 002/003 family */
SNDRV_HWDEP_IFACE_FW_TASCAM, /* TASCAM FireWire series */
+ SNDRV_HWDEP_IFACE_LINE6, /* Line6 USB processors */
/* Don't forget to change the following: */
- SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_TASCAM
+ SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_LINE6
};
struct snd_hwdep_info {
diff --git a/include/uapi/sound/snd_sst_tokens.h b/include/uapi/sound/snd_sst_tokens.h
new file mode 100644
index 000000000000..1ee2e943d66a
--- /dev/null
+++ b/include/uapi/sound/snd_sst_tokens.h
@@ -0,0 +1,214 @@
+/*
+ * snd_sst_tokens.h - Intel SST tokens definition
+ *
+ * Copyright (C) 2016 Intel Corp
+ * Author: Shreyas NC <shreyas.nc@intel.com>
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as 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.
+ */
+#ifndef __SND_SST_TOKENS_H__
+#define __SND_SST_TOKENS_H__
+
+/**
+ * %SKL_TKN_UUID: Module UUID
+ *
+ * %SKL_TKN_U8_BLOCK_TYPE: Type of the private data block.Can be:
+ * tuples, bytes, short and words
+ *
+ * %SKL_TKN_U8_IN_PIN_TYPE: Input pin type,
+ * homogenous=0, heterogenous=1
+ *
+ * %SKL_TKN_U8_OUT_PIN_TYPE: Output pin type,
+ * homogenous=0, heterogenous=1
+ * %SKL_TKN_U8_DYN_IN_PIN: Configure Input pin dynamically
+ * if true
+ *
+ * %SKL_TKN_U8_DYN_OUT_PIN: Configure Output pin dynamically
+ * if true
+ *
+ * %SKL_TKN_U8_IN_QUEUE_COUNT: Store the number of Input pins
+ *
+ * %SKL_TKN_U8_OUT_QUEUE_COUNT: Store the number of Output pins
+ *
+ * %SKL_TKN_U8_TIME_SLOT: TDM slot number
+ *
+ * %SKL_TKN_U8_CORE_ID: Stores module affinity value.Can take
+ * the values:
+ * SKL_AFFINITY_CORE_0 = 0,
+ * SKL_AFFINITY_CORE_1,
+ * SKL_AFFINITY_CORE_MAX
+ *
+ * %SKL_TKN_U8_MOD_TYPE: Module type value.
+ *
+ * %SKL_TKN_U8_CONN_TYPE: Module connection type can be a FE,
+ * BE or NONE as defined :
+ * SKL_PIPE_CONN_TYPE_NONE = 0,
+ * SKL_PIPE_CONN_TYPE_FE = 1 (HOST_DMA)
+ * SKL_PIPE_CONN_TYPE_BE = 2 (LINK_DMA)
+ *
+ * %SKL_TKN_U8_DEV_TYPE: Type of device to which the module is
+ * connected
+ * Can take the values:
+ * SKL_DEVICE_BT = 0x0,
+ * SKL_DEVICE_DMIC = 0x1,
+ * SKL_DEVICE_I2S = 0x2,
+ * SKL_DEVICE_SLIMBUS = 0x3,
+ * SKL_DEVICE_HDALINK = 0x4,
+ * SKL_DEVICE_HDAHOST = 0x5,
+ * SKL_DEVICE_NONE
+ *
+ * %SKL_TKN_U8_HW_CONN_TYPE: Connection type of the HW to which the
+ * module is connected
+ * SKL_CONN_NONE = 0,
+ * SKL_CONN_SOURCE = 1,
+ * SKL_CONN_SINK = 2
+ *
+ * %SKL_TKN_U16_PIN_INST_ID: Stores the pin instance id
+ *
+ * %SKL_TKN_U16_MOD_INST_ID: Stores the mdule instance id
+ *
+ * %SKL_TKN_U32_MAX_MCPS: Module max mcps value
+ *
+ * %SKL_TKN_U32_MEM_PAGES: Module resource pages
+ *
+ * %SKL_TKN_U32_OBS: Stores Output Buffer size
+ *
+ * %SKL_TKN_U32_IBS: Stores input buffer size
+ *
+ * %SKL_TKN_U32_VBUS_ID: Module VBUS_ID. PDM=0, SSP0=0,
+ * SSP1=1,SSP2=2,
+ * SSP3=3, SSP4=4,
+ * SSP5=5, SSP6=6,INVALID
+ *
+ * %SKL_TKN_U32_PARAMS_FIXUP: Module Params fixup mask
+ * %SKL_TKN_U32_CONVERTER: Module params converter mask
+ * %SKL_TKN_U32_PIPE_ID: Stores the pipe id
+ *
+ * %SKL_TKN_U32_PIPE_CONN_TYPE: Type of the token to which the pipe is
+ * connected to. It can be
+ * SKL_PIPE_CONN_TYPE_NONE = 0,
+ * SKL_PIPE_CONN_TYPE_FE = 1 (HOST_DMA),
+ * SKL_PIPE_CONN_TYPE_BE = 2 (LINK_DMA),
+ *
+ * %SKL_TKN_U32_PIPE_PRIORITY: Pipe priority value
+ * %SKL_TKN_U32_PIPE_MEM_PGS: Pipe resource pages
+ *
+ * %SKL_TKN_U32_DIR_PIN_COUNT: Value for the direction to set input/output
+ * formats and the pin count.
+ * The first 4 bits have the direction
+ * value and the next 4 have
+ * the pin count value.
+ * SKL_DIR_IN = 0, SKL_DIR_OUT = 1.
+ * The input and output formats
+ * share the same set of tokens
+ * with the distinction between input
+ * and output made by reading direction
+ * token.
+ *
+ * %SKL_TKN_U32_FMT_CH: Supported channel count
+ *
+ * %SKL_TKN_U32_FMT_FREQ: Supported frequency/sample rate
+ *
+ * %SKL_TKN_U32_FMT_BIT_DEPTH: Supported container size
+ *
+ * %SKL_TKN_U32_FMT_SAMPLE_SIZE:Number of samples in the container
+ *
+ * %SKL_TKN_U32_FMT_CH_CONFIG: Supported channel configurations for the
+ * input/output.
+ *
+ * %SKL_TKN_U32_FMT_INTERLEAVE: Interleaving style which can be per
+ * channel or per sample. The values can be :
+ * SKL_INTERLEAVING_PER_CHANNEL = 0,
+ * SKL_INTERLEAVING_PER_SAMPLE = 1,
+ *
+ * %SKL_TKN_U32_FMT_SAMPLE_TYPE:
+ * Specifies the sample type. Can take the
+ * values: SKL_SAMPLE_TYPE_INT_MSB = 0,
+ * SKL_SAMPLE_TYPE_INT_LSB = 1,
+ * SKL_SAMPLE_TYPE_INT_SIGNED = 2,
+ * SKL_SAMPLE_TYPE_INT_UNSIGNED = 3,
+ * SKL_SAMPLE_TYPE_FLOAT = 4
+ *
+ * %SKL_TKN_U32_CH_MAP: Channel map values
+ * %SKL_TKN_U32_MOD_SET_PARAMS: It can take these values:
+ * SKL_PARAM_DEFAULT, SKL_PARAM_INIT,
+ * SKL_PARAM_SET, SKL_PARAM_BIND
+ *
+ * %SKL_TKN_U32_MOD_PARAM_ID: ID of the module params
+ *
+ * %SKL_TKN_U32_CAPS_SET_PARAMS:
+ * Set params value
+ *
+ * %SKL_TKN_U32_CAPS_PARAMS_ID: Params ID
+ *
+ * %SKL_TKN_U32_CAPS_SIZE: Caps size
+ *
+ * %SKL_TKN_U32_PROC_DOMAIN: Specify processing domain
+ *
+ * %SKL_TKN_U32_LIB_COUNT: Specifies the number of libraries
+ *
+ * %SKL_TKN_STR_LIB_NAME: Specifies the library name
+ *
+ * module_id and loadable flags dont have tokens as these values will be
+ * read from the DSP FW manifest
+ */
+enum SKL_TKNS {
+ SKL_TKN_UUID = 1,
+ SKL_TKN_U8_NUM_BLOCKS,
+ SKL_TKN_U8_BLOCK_TYPE,
+ SKL_TKN_U8_IN_PIN_TYPE,
+ SKL_TKN_U8_OUT_PIN_TYPE,
+ SKL_TKN_U8_DYN_IN_PIN,
+ SKL_TKN_U8_DYN_OUT_PIN,
+ SKL_TKN_U8_IN_QUEUE_COUNT,
+ SKL_TKN_U8_OUT_QUEUE_COUNT,
+ SKL_TKN_U8_TIME_SLOT,
+ SKL_TKN_U8_CORE_ID,
+ SKL_TKN_U8_MOD_TYPE,
+ SKL_TKN_U8_CONN_TYPE,
+ SKL_TKN_U8_DEV_TYPE,
+ SKL_TKN_U8_HW_CONN_TYPE,
+ SKL_TKN_U16_MOD_INST_ID,
+ SKL_TKN_U16_BLOCK_SIZE,
+ SKL_TKN_U32_MAX_MCPS,
+ SKL_TKN_U32_MEM_PAGES,
+ SKL_TKN_U32_OBS,
+ SKL_TKN_U32_IBS,
+ SKL_TKN_U32_VBUS_ID,
+ SKL_TKN_U32_PARAMS_FIXUP,
+ SKL_TKN_U32_CONVERTER,
+ SKL_TKN_U32_PIPE_ID,
+ SKL_TKN_U32_PIPE_CONN_TYPE,
+ SKL_TKN_U32_PIPE_PRIORITY,
+ SKL_TKN_U32_PIPE_MEM_PGS,
+ SKL_TKN_U32_DIR_PIN_COUNT,
+ SKL_TKN_U32_FMT_CH,
+ SKL_TKN_U32_FMT_FREQ,
+ SKL_TKN_U32_FMT_BIT_DEPTH,
+ SKL_TKN_U32_FMT_SAMPLE_SIZE,
+ SKL_TKN_U32_FMT_CH_CONFIG,
+ SKL_TKN_U32_FMT_INTERLEAVE,
+ SKL_TKN_U32_FMT_SAMPLE_TYPE,
+ SKL_TKN_U32_FMT_CH_MAP,
+ SKL_TKN_U32_PIN_MOD_ID,
+ SKL_TKN_U32_PIN_INST_ID,
+ SKL_TKN_U32_MOD_SET_PARAMS,
+ SKL_TKN_U32_MOD_PARAM_ID,
+ SKL_TKN_U32_CAPS_SET_PARAMS,
+ SKL_TKN_U32_CAPS_PARAMS_ID,
+ SKL_TKN_U32_CAPS_SIZE,
+ SKL_TKN_U32_PROC_DOMAIN,
+ SKL_TKN_U32_LIB_COUNT,
+ SKL_TKN_STR_LIB_NAME,
+ SKL_TKN_MAX = SKL_TKN_STR_LIB_NAME,
+};
+
+#endif
diff --git a/include/uapi/sound/tlv.h b/include/uapi/sound/tlv.h
index ffc4f203146c..b4df440c015b 100644
--- a/include/uapi/sound/tlv.h
+++ b/include/uapi/sound/tlv.h
@@ -28,4 +28,73 @@
#define SNDRV_CTL_TLVT_CHMAP_VAR 0x102 /* channels freely swappable */
#define SNDRV_CTL_TLVT_CHMAP_PAIRED 0x103 /* pair-wise swappable */
+/*
+ * TLV structure is right behind the struct snd_ctl_tlv:
+ * unsigned int type - see SNDRV_CTL_TLVT_*
+ * unsigned int length
+ * .... data aligned to sizeof(unsigned int), use
+ * block_length = (length + (sizeof(unsigned int) - 1)) &
+ * ~(sizeof(unsigned int) - 1)) ....
+ */
+#define SNDRV_CTL_TLVD_ITEM(type, ...) \
+ (type), SNDRV_CTL_TLVD_LENGTH(__VA_ARGS__), __VA_ARGS__
+#define SNDRV_CTL_TLVD_LENGTH(...) \
+ ((unsigned int)sizeof((const unsigned int[]) { __VA_ARGS__ }))
+
+#define SNDRV_CTL_TLVD_CONTAINER_ITEM(...) \
+ SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_CONTAINER, __VA_ARGS__)
+#define SNDRV_CTL_TLVD_DECLARE_CONTAINER(name, ...) \
+ unsigned int name[] = { \
+ SNDRV_CTL_TLVD_CONTAINER_ITEM(__VA_ARGS__) \
+ }
+
+#define SNDRV_CTL_TLVD_DB_SCALE_MASK 0xffff
+#define SNDRV_CTL_TLVD_DB_SCALE_MUTE 0x10000
+#define SNDRV_CTL_TLVD_DB_SCALE_ITEM(min, step, mute) \
+ SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_DB_SCALE, \
+ (min), \
+ ((step) & SNDRV_CTL_TLVD_DB_SCALE_MASK) | \
+ ((mute) ? SNDRV_CTL_TLVD_DB_SCALE_MUTE : 0))
+#define SNDRV_CTL_TLVD_DECLARE_DB_SCALE(name, min, step, mute) \
+ unsigned int name[] = { \
+ SNDRV_CTL_TLVD_DB_SCALE_ITEM(min, step, mute) \
+ }
+
+/* dB scale specified with min/max values instead of step */
+#define SNDRV_CTL_TLVD_DB_MINMAX_ITEM(min_dB, max_dB) \
+ SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_DB_MINMAX, (min_dB), (max_dB))
+#define SNDRV_CTL_TLVD_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) \
+ SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_DB_MINMAX_MUTE, (min_dB), (max_dB))
+#define SNDRV_CTL_TLVD_DECLARE_DB_MINMAX(name, min_dB, max_dB) \
+ unsigned int name[] = { \
+ SNDRV_CTL_TLVD_DB_MINMAX_ITEM(min_dB, max_dB) \
+ }
+#define SNDRV_CTL_TLVD_DECLARE_DB_MINMAX_MUTE(name, min_dB, max_dB) \
+ unsigned int name[] = { \
+ SNDRV_CTL_TLVD_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) \
+ }
+
+/* linear volume between min_dB and max_dB (.01dB unit) */
+#define SNDRV_CTL_TLVD_DB_LINEAR_ITEM(min_dB, max_dB) \
+ SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_DB_LINEAR, (min_dB), (max_dB))
+#define SNDRV_CTL_TLVD_DECLARE_DB_LINEAR(name, min_dB, max_dB) \
+ unsigned int name[] = { \
+ SNDRV_CTL_TLVD_DB_LINEAR_ITEM(min_dB, max_dB) \
+ }
+
+/* dB range container:
+ * Items in dB range container must be ordered by their values and by their
+ * dB values. This implies that larger values must correspond with larger
+ * dB values (which is also required for all other mixer controls).
+ */
+/* Each item is: <min> <max> <TLV> */
+#define SNDRV_CTL_TLVD_DB_RANGE_ITEM(...) \
+ SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_DB_RANGE, __VA_ARGS__)
+#define SNDRV_CTL_TLVD_DECLARE_DB_RANGE(name, ...) \
+ unsigned int name[] = { \
+ SNDRV_CTL_TLVD_DB_RANGE_ITEM(__VA_ARGS__) \
+ }
+
+#define SNDRV_CTL_TLVD_DB_GAIN_MUTE -9999999
+
#endif
generated by cgit v1.2.3-70-g09d2 (git 2.47.1) at 2025-01-23 21:44:43 +0000