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authorLinus Torvalds <torvalds@linux-foundation.org>2016-10-05 22:13:14 +0300
committerLinus Torvalds <torvalds@linux-foundation.org>2016-10-05 22:13:14 +0300
commit5691f0e9a3e7855832d5fd094801bf600347c2d0 (patch)
tree4cd4538de2bde556be944e984a3882a64b25dfcb /include
parent878fb5dc96b9dfae1de45be1b85aba40aca3356e (diff)
parenteeea8b40cd2866ca24f25e5ef09225edb076ae45 (diff)
downloadlinux-5691f0e9a3e7855832d5fd094801bf600347c2d0.tar.xz
Merge tag 'sound-4.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai: "Again the diffstat shows a widely distributed pattern at this cycle, as there've been many code cleanups and refactoring allover the places. Other than that, the development was relatively calm, and no big surprise shouldn't be expected. Here are some highlights: Core: - Sequencer code refactoring / documentation updates - TLV code moved to uapi, following some relevant cleanups USB-Audio: - Lots of LINE6 driver fixes / updates - DragonFly and TEAC device quirk updates HD-audio: - Usual fixupes for Dell, Lenovo and HP machines - Link-audio time reporting capability ASoC: - Large refactoring of simple-card code to be shared with rcar driver - Removal of some duplicated ops over lots of CODEC drivers - Again quite a few Intel SKL driver updates - New drivers for Nuvoton NAU88C10, Realtek RT5660 and RT5663" * tag 'sound-4.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (465 commits) ASoC: fsl: Fix lockups with recent cache changes ASoC: Intel: Skylake: fix memory leak of module on error exit path ASoC: rsnd: add SNDRV_PCM_TRIGGER_SUSPEND/RESUME ASoC: wm8960: remove usage of obsoleted TLV-related macro ASoC: rt5616: remove usage of obsoleted TLV-related macro ASoC: max9867: remove usage of obsoleted TLV-related macro ASoC: trivial: system spelling fix ASoC: da7219: fix inappropriate condition statement ASoC: tlv320aic31xx: do not declare support for mono DAI ASoC: stac9766: fix wrong usage of DECLARE_TLV_DB_LINEAR() ASoC: wm8991: remove unused variable ASoC: wm8991: fix wrong usage of DECLARE_TLV_DB_LINEAR() ASOC: tpa6130a2: add static qualifier for file local symbols ASoC: sst-bxt-rt298: fix obsoleted initializers for array ASoC: sst-bxt-da7219_max98357a: fix obsoleted initializers for array ASoC: rt5616: add static qualifier for file local symbols ASoC: arizona: Add output power up/down delays for speaker path ASoC: arizona: Add debug prints for output power up/down times ALSA: hda - Add the top speaker pin config for HP Spectre x360 ASoC: Intel: Add DMIC channel constraint for bxt machine ...
Diffstat (limited to 'include')
-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