From bb6fc620c2ed972f58a0174f64f8dbd22a5911b1 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 8 Aug 2016 05:59:56 +0000 Subject: ASoC: simple-card-utils: add asoc_simple_card_parse_clk() Current simple-card can get clock via DT clocks or "system-clock-frequency" property. This patch makes it simple style standard Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'include') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 86088aed9002..1392eb56cf0e 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -33,4 +33,12 @@ int asoc_simple_card_set_dailink_name(struct device *dev, 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); + #endif /* __SIMPLE_CARD_CORE_H */ -- cgit v1.2.3 From ae30a694da4c37b4d0c8b750c9a4104d8da749b3 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 8 Aug 2016 06:01:43 +0000 Subject: ASoC: simple-card-utils: add asoc_simple_card_parse_dai() simple-card needs to get its dai name and endpoint node. This patch makes it simple style standard Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 17 ++++++++++++++++ sound/soc/generic/simple-card-utils.c | 37 +++++++++++++++++++++++++++++++++++ 2 files changed, 54 insertions(+) (limited to 'include') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 1392eb56cf0e..62b392695d2d 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -41,4 +41,21 @@ 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); + #endif /* __SIMPLE_CARD_CORE_H */ diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 16f65f972d04..27e6d038a902 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -127,6 +127,43 @@ int asoc_simple_card_parse_clk(struct device_node *node, } EXPORT_SYMBOL_GPL(asoc_simple_card_parse_clk); +int asoc_simple_card_parse_dai(struct device_node *node, + struct device_node **dai_of_node, + const char **dai_name, + const char *list_name, + const char *cells_name, + int *is_single_link) +{ + struct of_phandle_args args; + int ret; + + if (!node) + return 0; + + /* + * Get node via "sound-dai = <&phandle port>" + * it will be used as xxx_of_node on soc_bind_dai_link() + */ + ret = of_parse_phandle_with_args(node, list_name, cells_name, 0, &args); + if (ret) + return ret; + + /* Get dai->name */ + if (dai_name) { + ret = snd_soc_of_get_dai_name(node, dai_name); + if (ret < 0) + return ret; + } + + *dai_of_node = args.np; + + if (is_single_link) + *is_single_link = !args.args_count; + + return 0; +} +EXPORT_SYMBOL_GPL(asoc_simple_card_parse_dai); + /* Module information */ MODULE_AUTHOR("Kuninori Morimoto "); MODULE_DESCRIPTION("ALSA SoC Simple Card Utils"); -- cgit v1.2.3 From 0306741004fdfc2bc515b4b129b1f86881c5fcf5 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Thu, 4 Aug 2016 15:38:41 +0200 Subject: ASoC: L3 bus: Add default gpio ops This adds aptional GPIO bit-bang based callback implementations for setting CLK, DATA and MODE L3 bus lines. It is added here to avoid possible duplicate implementations across users of the bus. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- include/sound/l3.h | 15 ++++++++--- sound/soc/codecs/l3.c | 71 ++++++++++++++++++++++++++++++++++++++++++--------- 2 files changed, 71 insertions(+), 15 deletions(-) (limited to 'include') 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/sound/soc/codecs/l3.c b/sound/soc/codecs/l3.c index 5353af58862c..a10ea3c716c6 100644 --- a/sound/soc/codecs/l3.c +++ b/sound/soc/codecs/l3.c @@ -20,6 +20,8 @@ #include #include #include +#include +#include #include @@ -32,11 +34,11 @@ static void sendbyte(struct l3_pins *adap, unsigned int byte) int i; for (i = 0; i < 8; i++) { - adap->setclk(0); + adap->setclk(adap, 0); udelay(adap->data_hold); - adap->setdat(byte & 1); + adap->setdat(adap, byte & 1); udelay(adap->data_setup); - adap->setclk(1); + adap->setclk(adap, 1); udelay(adap->clock_high); byte >>= 1; } @@ -55,10 +57,10 @@ static void sendbytes(struct l3_pins *adap, const u8 *buf, for (i = 0; i < len; i++) { if (i) { udelay(adap->mode_hold); - adap->setmode(0); + adap->setmode(adap, 0); udelay(adap->mode); } - adap->setmode(1); + adap->setmode(adap, 1); udelay(adap->mode_setup); sendbyte(adap, buf[i]); } @@ -66,26 +68,71 @@ static void sendbytes(struct l3_pins *adap, const u8 *buf, int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len) { - adap->setclk(1); - adap->setdat(1); - adap->setmode(1); + adap->setclk(adap, 1); + adap->setdat(adap, 1); + adap->setmode(adap, 1); udelay(adap->mode); - adap->setmode(0); + adap->setmode(adap, 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); + adap->setclk(adap, 1); + adap->setdat(adap, 1); + adap->setmode(adap, 0); return len; } EXPORT_SYMBOL_GPL(l3_write); + +static void l3_set_clk(struct l3_pins *adap, int val) +{ + gpio_set_value(adap->gpio_clk, val); +} + +static void l3_set_data(struct l3_pins *adap, int val) +{ + gpio_set_value(adap->gpio_data, val); +} + +static void l3_set_mode(struct l3_pins *adap, int val) +{ + gpio_set_value(adap->gpio_mode, val); +} + +int l3_set_gpio_ops(struct device *dev, struct l3_pins *adap) +{ + int ret; + + if (!adap->use_gpios) + return -EINVAL; + + ret = devm_gpio_request_one(dev, adap->gpio_data, + GPIOF_OUT_INIT_LOW, "l3_data"); + if (ret < 0) + return ret; + adap->setdat = l3_set_data; + + ret = devm_gpio_request_one(dev, adap->gpio_clk, + GPIOF_OUT_INIT_LOW, "l3_clk"); + if (ret < 0) + return ret; + adap->setclk = l3_set_clk; + + ret = devm_gpio_request_one(dev, adap->gpio_mode, + GPIOF_OUT_INIT_LOW, "l3_mode"); + if (ret < 0) + return ret; + adap->setmode = l3_set_mode; + + return 0; +} +EXPORT_SYMBOL_GPL(l3_set_gpio_ops); + MODULE_DESCRIPTION("L3 bit-banging driver"); MODULE_AUTHOR("Christian Pellegrin "); MODULE_LICENSE("GPL"); -- cgit v1.2.3 From 84c5c20395a8e4ca2043136e1f0d128cf758244b Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Thu, 4 Aug 2016 15:38:44 +0200 Subject: ASoC: s3c24xx_uda134x: Remove unused power() callback The power() callback has always been empty so just remove it. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- include/sound/s3c24xx_uda134x.h | 1 - sound/soc/samsung/s3c24xx_uda134x.c | 1 - 2 files changed, 2 deletions(-) (limited to 'include') 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/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 50849e137fc0..33479e7a0d3e 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -281,7 +281,6 @@ static int s3c24xx_uda134x_probe(struct platform_device *pdev) "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, -- cgit v1.2.3 From dc31e741db49e35e8b99d293dcc7afbbe9418fa7 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Tue, 26 Jul 2016 14:32:18 +0800 Subject: ASoC: topology: ABI - Add the types for BE DAI Define the type and ABI struct for Backend DAIs. Add the number of BE DAIs to manifest, and some reserved fields for future extensions. Pump the version number to 5. Topology core will check size of ABI objects to detect version mismatch between user space and kernel. Signed-off-by: Guneshwor Singh Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 34 ++++++++++++++++++++++++++++++++-- 1 file changed, 32 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index e4701a3c6331..f734bea9a032 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. @@ -285,6 +291,8 @@ 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 +458,26 @@ 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 -- cgit v1.2.3 From 8073aefa60823acf205a1e6a5ea118297179d766 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 8 Aug 2016 09:36:49 +0000 Subject: ASoC: remove codec duplicated callback function codec driver and component driver has duplicated callback functions, and codec side functions are just copied to component side when register timing. This was quick-hack, but no longer needed. This patch removes codec side duplicated callback function. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 8 -------- sound/soc/soc-core.c | 13 ------------- 2 files changed, 21 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 6144882cc96a..5eb2b38c3437 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); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 16369cad4803..edba975d893e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3333,19 +3333,6 @@ int snd_soc_register_codec(struct device *dev, if (ret) goto err_free; - if (codec_drv->controls) { - codec->component.controls = codec_drv->controls; - codec->component.num_controls = codec_drv->num_controls; - } - if (codec_drv->dapm_widgets) { - codec->component.dapm_widgets = codec_drv->dapm_widgets; - codec->component.num_dapm_widgets = codec_drv->num_dapm_widgets; - } - if (codec_drv->dapm_routes) { - codec->component.dapm_routes = codec_drv->dapm_routes; - codec->component.num_dapm_routes = codec_drv->num_dapm_routes; - } - if (codec_drv->probe) codec->component.probe = snd_soc_codec_drv_probe; if (codec_drv->remove) -- cgit v1.2.3 From 6720b38420a01d40dbeb8ee575eb601d612de691 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 4 Aug 2016 15:46:00 +0530 Subject: ALSA: hda - move bus_parse_capabilities to core HDA capability introduced recently are move to hdac core so that it can be used by legacy driver as well. Also move the capability pointers up to hdac_bus object. Signed-off-by: Vinod Koul Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 13 ++++++++ sound/hda/hdac_controller.c | 75 +++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 88 insertions(+) (limited to 'include') 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/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 9fee464e5d49..043065867656 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -255,6 +255,81 @@ int snd_hdac_bus_get_response(struct hdac_bus *bus, unsigned int addr, } EXPORT_SYMBOL_GPL(snd_hdac_bus_get_response); +#define HDAC_MAX_CAPS 10 +/** + * snd_hdac_bus_parse_capabilities - parse capability structure + * @bus: the pointer to bus object + * + * Returns 0 if successful, or a negative error code. + */ +int snd_hdac_bus_parse_capabilities(struct hdac_bus *bus) +{ + unsigned int cur_cap; + unsigned int offset; + unsigned int counter = 0; + + offset = snd_hdac_chip_readl(bus, LLCH); + + /* Lets walk the linked capabilities list */ + do { + cur_cap = _snd_hdac_chip_read(l, bus, offset); + + dev_dbg(bus->dev, "Capability version: 0x%x\n", + (cur_cap & AZX_CAP_HDR_VER_MASK) >> AZX_CAP_HDR_VER_OFF); + + dev_dbg(bus->dev, "HDA capability ID: 0x%x\n", + (cur_cap & AZX_CAP_HDR_ID_MASK) >> AZX_CAP_HDR_ID_OFF); + + switch ((cur_cap & AZX_CAP_HDR_ID_MASK) >> AZX_CAP_HDR_ID_OFF) { + case AZX_ML_CAP_ID: + dev_dbg(bus->dev, "Found ML capability\n"); + bus->mlcap = bus->remap_addr + offset; + break; + + case AZX_GTS_CAP_ID: + dev_dbg(bus->dev, "Found GTS capability offset=%x\n", offset); + bus->gtscap = bus->remap_addr + offset; + break; + + case AZX_PP_CAP_ID: + /* PP capability found, the Audio DSP is present */ + dev_dbg(bus->dev, "Found PP capability offset=%x\n", offset); + bus->ppcap = bus->remap_addr + offset; + break; + + case AZX_SPB_CAP_ID: + /* SPIB capability found, handler function */ + dev_dbg(bus->dev, "Found SPB capability\n"); + bus->spbcap = bus->remap_addr + offset; + break; + + case AZX_DRSM_CAP_ID: + /* DMA resume capability found, handler function */ + dev_dbg(bus->dev, "Found DRSM capability\n"); + bus->drsmcap = bus->remap_addr + offset; + break; + + default: + dev_dbg(bus->dev, "Unknown capability %d\n", cur_cap); + break; + } + + counter++; + + if (counter > HDAC_MAX_CAPS) { + dev_err(bus->dev, "We exceeded HDAC capabilities!!!\n"); + break; + } + + /* read the offset of next capability */ + offset = cur_cap & AZX_CAP_HDR_NXT_PTR_MASK; + + } while (offset); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_bus_parse_capabilities); + /* * Lowlevel interface */ -- cgit v1.2.3 From 404735c9fd8adff8e5ad11e1f9f8db069d865698 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 4 Aug 2016 15:46:02 +0530 Subject: ALSA - Ext hda: remove bus_parse_capabilities Remove the unused one as we have moved it up to hdac core. Signed-off-by: Vinod Koul Signed-off-by: Takashi Iwai --- include/sound/hdaudio_ext.h | 12 ------ sound/hda/ext/hdac_ext_controller.c | 75 ------------------------------------- 2 files changed, 87 deletions(-) (limited to 'include') 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/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index cd65e007e864..261469188566 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -29,81 +29,6 @@ */ #define HDAC_MAX_CAPS 10 -/** - * snd_hdac_ext_bus_parse_capabilities - parse capablity structure - * @ebus: the pointer to extended bus object - * - * Returns 0 if successful, or a negative error code. - */ -int snd_hdac_ext_bus_parse_capabilities(struct hdac_ext_bus *ebus) -{ - unsigned int cur_cap; - unsigned int offset; - struct hdac_bus *bus = &ebus->bus; - unsigned int counter = 0; - - offset = snd_hdac_chip_readl(bus, LLCH); - - /* Lets walk the linked capabilities list */ - do { - cur_cap = _snd_hdac_chip_read(l, bus, offset); - - dev_dbg(bus->dev, "Capability version: 0x%x\n", - ((cur_cap & AZX_CAP_HDR_VER_MASK) >> AZX_CAP_HDR_VER_OFF)); - - dev_dbg(bus->dev, "HDA capability ID: 0x%x\n", - (cur_cap & AZX_CAP_HDR_ID_MASK) >> AZX_CAP_HDR_ID_OFF); - - switch ((cur_cap & AZX_CAP_HDR_ID_MASK) >> AZX_CAP_HDR_ID_OFF) { - case AZX_ML_CAP_ID: - dev_dbg(bus->dev, "Found ML capability\n"); - ebus->mlcap = bus->remap_addr + offset; - break; - - case AZX_GTS_CAP_ID: - dev_dbg(bus->dev, "Found GTS capability offset=%x\n", offset); - ebus->gtscap = bus->remap_addr + offset; - break; - - case AZX_PP_CAP_ID: - /* PP capability found, the Audio DSP is present */ - dev_dbg(bus->dev, "Found PP capability offset=%x\n", offset); - ebus->ppcap = bus->remap_addr + offset; - break; - - case AZX_SPB_CAP_ID: - /* SPIB capability found, handler function */ - dev_dbg(bus->dev, "Found SPB capability\n"); - ebus->spbcap = bus->remap_addr + offset; - break; - - case AZX_DRSM_CAP_ID: - /* DMA resume capability found, handler function */ - dev_dbg(bus->dev, "Found DRSM capability\n"); - ebus->drsmcap = bus->remap_addr + offset; - break; - - default: - dev_dbg(bus->dev, "Unknown capability %d\n", cur_cap); - break; - } - - counter++; - - if (counter > HDAC_MAX_CAPS) { - dev_err(bus->dev, "We exceeded HDAC Ext capablities!!!\n"); - break; - } - - /* read the offset of next capabiity */ - offset = cur_cap & AZX_CAP_HDR_NXT_PTR_MASK; - - } while (offset); - - return 0; -} -EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_parse_capabilities); - /* * processing pipe helpers - these helpers are useful for dealing with HDA * new capability of processing pipelines -- cgit v1.2.3 From 50279d9b5facde811280afe13dd8c79f0e7b21ed Mon Sep 17 00:00:00 2001 From: Guneshwor Singh Date: Thu, 4 Aug 2016 15:46:03 +0530 Subject: ALSA - hda: Add support for parsing new HDA capabilities Skylake onwards HDA controller supports new capabilities like Global Time Stamping (GTS) capability. So add support to parse these new capabilities. Signed-off-by: Guneshwor Singh Signed-off-by: Hardik T Shah Signed-off-by: Vinod Koul Signed-off-by: Takashi Iwai --- include/sound/hda_register.h | 36 ++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_controller.c | 5 +++++ sound/pci/hda/hda_controller.h | 3 +++ sound/pci/hda/hda_intel.c | 17 +++++++++++++++++ 4 files changed, 61 insertions(+) (limited to 'include') 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/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 27de8015717d..1567fe209e01 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -412,6 +412,11 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) goto unlock; } runtime->private_data = azx_dev; + + if (chip->gts_present) + azx_pcm_hw.info = azx_pcm_hw.info | + SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME; + runtime->hw = azx_pcm_hw; runtime->hw.channels_min = hinfo->channels_min; runtime->hw.channels_max = hinfo->channels_max; diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index ec63bbf1ec6d..a50e0532622a 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -159,6 +159,9 @@ struct azx { unsigned int region_requested:1; unsigned int disabled:1; /* disabled by vga_switcheroo */ + /* GTS present */ + unsigned int gts_present:1; + #ifdef CONFIG_SND_HDA_DSP_LOADER struct azx_dev saved_azx_dev; #endif diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 89dacf9b4e6c..4786f435eb64 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -54,6 +54,7 @@ /* for snoop control */ #include #include +#include #endif #include #include @@ -1655,6 +1656,22 @@ static int azx_first_init(struct azx *chip) return -ENXIO; } + if (IS_SKL_PLUS(pci)) + snd_hdac_bus_parse_capabilities(bus); + + /* + * Some Intel CPUs has always running timer (ART) feature and + * controller may have Global time sync reporting capability, so + * check both of these before declaring synchronized time reporting + * capability SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME + */ + chip->gts_present = false; + +#ifdef CONFIG_X86 + if (bus->ppcap && boot_cpu_has(X86_FEATURE_ART)) + chip->gts_present = true; +#endif + if (chip->msi) { if (chip->driver_caps & AZX_DCAPS_NO_MSI64) { dev_dbg(card->dev, "Disabling 64bit MSI\n"); -- cgit v1.2.3 From 1e814030954015e42621191f3adc52df2241dc08 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 1 Aug 2016 06:11:36 +0000 Subject: ASoC: remove snd_soc_pcm_set/get_drvdata() snd_soc_pcm_set_drvdata() will set driver data to rtd->dev, but driver data of rtd->dev is already used as "rtd" on soc_post_component_init(). static int soc_post_component_init(xxx) { ... dev_set_drvdata(rtd->dev, rtd); ... } To remove confusion, this patch removes snd_soc_pcm_set/get_drvdata(). Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 11 ----------- 1 file changed, 11 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 6144882cc96a..bc953994acb9 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1547,17 +1547,6 @@ 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); -- cgit v1.2.3 From 21ba62f849a20accf7d41e3056ea8913bc23cfb1 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 9 Aug 2016 05:48:30 +0000 Subject: ASoC: simple-card-utils: add asoc_simple_card_init_dai() simple-card is supporting clock/tdm slot initialization. This patch makes this method simple style standard. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 2 ++ sound/soc/generic/simple-card-utils.c | 29 +++++++++++++++++++++++++++++ 2 files changed, 31 insertions(+) (limited to 'include') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 62b392695d2d..13168e029232 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -58,4 +58,6 @@ int asoc_simple_card_parse_dai(struct device_node *node, 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); #endif /* __SIMPLE_CARD_CORE_H */ diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 27e6d038a902..33abe1f7014e 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -164,6 +164,35 @@ int asoc_simple_card_parse_dai(struct device_node *node, } EXPORT_SYMBOL_GPL(asoc_simple_card_parse_dai); +int asoc_simple_card_init_dai(struct snd_soc_dai *dai, + struct asoc_simple_dai *simple_dai) +{ + int ret; + + if (simple_dai->sysclk) { + ret = snd_soc_dai_set_sysclk(dai, 0, simple_dai->sysclk, 0); + if (ret && ret != -ENOTSUPP) { + dev_err(dai->dev, "simple-card: set_sysclk error\n"); + return ret; + } + } + + if (simple_dai->slots) { + ret = snd_soc_dai_set_tdm_slot(dai, + simple_dai->tx_slot_mask, + simple_dai->rx_slot_mask, + simple_dai->slots, + simple_dai->slot_width); + if (ret && ret != -ENOTSUPP) { + dev_err(dai->dev, "simple-card: set_tdm_slot error\n"); + return ret; + } + } + + return 0; +} +EXPORT_SYMBOL_GPL(asoc_simple_card_init_dai); + /* Module information */ MODULE_AUTHOR("Kuninori Morimoto "); MODULE_DESCRIPTION("ALSA SoC Simple Card Utils"); -- cgit v1.2.3 From c262c9ab7a956b15c4c12b81472502a587d0dfcd Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 9 Aug 2016 05:49:41 +0000 Subject: ASoC: simple-card-utils: add asoc_simple_card_canonicalize_dailink() simple-card is assuming that sometimes platform and cpu are same. This patch makes this method simple style standard. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 3 +++ sound/soc/generic/simple-card-utils.c | 13 +++++++++++++ 2 files changed, 16 insertions(+) (limited to 'include') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 13168e029232..a71d46a95ca8 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -60,4 +60,7 @@ int asoc_simple_card_parse_dai(struct device_node *node, 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); + #endif /* __SIMPLE_CARD_CORE_H */ diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 33abe1f7014e..189eadddfad3 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -193,6 +193,19 @@ int asoc_simple_card_init_dai(struct snd_soc_dai *dai, } EXPORT_SYMBOL_GPL(asoc_simple_card_init_dai); +int asoc_simple_card_canonicalize_dailink(struct snd_soc_dai_link *dai_link) +{ + if (!dai_link->cpu_dai_name || !dai_link->codec_dai_name) + return -EINVAL; + + /* Assumes platform == cpu */ + if (!dai_link->platform_of_node) + dai_link->platform_of_node = dai_link->cpu_of_node; + + return 0; +} +EXPORT_SYMBOL_GPL(asoc_simple_card_canonicalize_dailink); + /* Module information */ MODULE_AUTHOR("Kuninori Morimoto "); MODULE_DESCRIPTION("ALSA SoC Simple Card Utils"); -- cgit v1.2.3 From 983cebd602af8c2bf9d5830f15fb4e18fb38f994 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 10 Aug 2016 02:20:19 +0000 Subject: ASoC: simple-card-utils: add asoc_simple_card_canonicalize_cpu() simple-card needs remove dai_link->cpu_dai_name if it CPU was single DAI. This patch makes this method simple style standard. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 2 ++ sound/soc/generic/simple-card-utils.c | 17 +++++++++++++++++ 2 files changed, 19 insertions(+) (limited to 'include') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index a71d46a95ca8..f760f559393e 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -62,5 +62,7 @@ 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); #endif /* __SIMPLE_CARD_CORE_H */ diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 189eadddfad3..c5d32dad48b2 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -206,6 +206,23 @@ int asoc_simple_card_canonicalize_dailink(struct snd_soc_dai_link *dai_link) } EXPORT_SYMBOL_GPL(asoc_simple_card_canonicalize_dailink); +void asoc_simple_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link, + int is_single_links) +{ + /* + * In soc_bind_dai_link() will check cpu name after + * of_node matching if dai_link has cpu_dai_name. + * but, it will never match if name was created by + * fmt_single_name() remove cpu_dai_name if cpu_args + * was 0. See: + * fmt_single_name() + * fmt_multiple_name() + */ + if (is_single_links) + dai_link->cpu_dai_name = NULL; +} +EXPORT_SYMBOL_GPL(asoc_simple_card_canonicalize_cpu); + /* Module information */ MODULE_AUTHOR("Kuninori Morimoto "); MODULE_DESCRIPTION("ALSA SoC Simple Card Utils"); -- cgit v1.2.3 From 0f4e0711b735a148cb6da0e6f91253b62fc2c5b2 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 10 Aug 2016 02:21:25 +0000 Subject: ASoC: simple-card-utils: add asoc_simple_card_clean_reference() simple-card needs to decrease the reference count of the device nodes. This patch makes this method simple style standard. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 2 ++ sound/soc/generic/simple-card-utils.c | 15 +++++++++++++++ 2 files changed, 17 insertions(+) (limited to 'include') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index f760f559393e..403ec92164fc 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -65,4 +65,6 @@ 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/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index c5d32dad48b2..1cb39309f5d5 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -223,6 +223,21 @@ void asoc_simple_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link, } EXPORT_SYMBOL_GPL(asoc_simple_card_canonicalize_cpu); +int asoc_simple_card_clean_reference(struct snd_soc_card *card) +{ + struct snd_soc_dai_link *dai_link; + int num_links; + + for (num_links = 0, dai_link = card->dai_link; + num_links < card->num_links; + num_links++, dai_link++) { + of_node_put(dai_link->cpu_of_node); + of_node_put(dai_link->codec_of_node); + } + return 0; +} +EXPORT_SYMBOL_GPL(asoc_simple_card_clean_reference); + /* Module information */ MODULE_AUTHOR("Kuninori Morimoto "); MODULE_DESCRIPTION("ALSA SoC Simple Card Utils"); -- cgit v1.2.3 From 15f0d4f531d84015511dbdc2512e5a77c0173d49 Mon Sep 17 00:00:00 2001 From: Shreyas NC Date: Fri, 12 Aug 2016 12:29:50 +0530 Subject: ASoC: uapi: Intel: Skylake: Define vendor specific tokens With recent topology changes in alsa-lib, driver data for modules can now be passed in topology conf file using tuples. This patch defines vendor specific tokens to describe private data with tuples. The allowed token types are UUID, string, bool, byte, short and word. These tokens will be referenced by the vendor tuples in the conf file. In the topology conf file, multiple data blocks can be defined for a widget which can be either tuple vendor array or blob. So, each data block will be preceded by a descriptor to identify size and type of block. These descriptors will be token value pairs. Tokens for module_id and loadable flag are not defined as these are read from the DSP FW manifest. Signed-off-by: Shreyas NC Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- include/uapi/sound/Kbuild | 1 + include/uapi/sound/snd_sst_tokens.h | 208 ++++++++++++++++++++++++++++++++++++ 2 files changed, 209 insertions(+) create mode 100644 include/uapi/sound/snd_sst_tokens.h (limited to 'include') 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/snd_sst_tokens.h b/include/uapi/sound/snd_sst_tokens.h new file mode 100644 index 000000000000..f56a932736ca --- /dev/null +++ b/include/uapi/sound/snd_sst_tokens.h @@ -0,0 +1,208 @@ +/* + * snd_sst_tokens.h - Intel SST tokens definition + * + * Copyright (C) 2016 Intel Corp + * Author: Shreyas NC + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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 + * + * 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_MAX = SKL_TKN_U32_PROC_DOMAIN, +}; + +#endif -- cgit v1.2.3 From f918e1697b1a8f2f26a4813db053cfbcafc48046 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Fri, 19 Aug 2016 18:12:46 +0800 Subject: ASoC: topology: ABI - Add sig_bits to stream caps Kernel struct snd_soc_pcm_stream, SoC PCM stream information, needs this field. Although current topology users don't configure this, we define it for future extension. Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 1 + sound/soc/soc-topology.c | 1 + 2 files changed, 2 insertions(+) (limited to 'include') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index f734bea9a032..33d00a4ce656 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -257,6 +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)); /* diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index a9e83a2dd91c..6b05047a4134 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1556,6 +1556,7 @@ static void set_stream_info(struct snd_soc_pcm_stream *stream, stream->rate_min = caps->rate_min; stream->rate_max = caps->rate_max; stream->formats = caps->formats; + stream->sig_bits = caps->sig_bits; } static void set_dai_flags(struct snd_soc_dai_driver *dai_drv, -- cgit v1.2.3 From 541070cec4f9be18ce9fcc74ac5e1036965ceb63 Mon Sep 17 00:00:00 2001 From: Shreyas NC Date: Tue, 23 Aug 2016 09:31:03 +0530 Subject: ASoC: Intel: Skylake: Parse manifest data Topology manifest has lib names and lib count info. So, define tokens to represent module private data and parse these tokens to fill up the manifest structure in the driver accordingly. Signed-off-by: Shreyas NC Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- include/uapi/sound/snd_sst_tokens.h | 8 +- sound/soc/intel/skylake/skl-topology.c | 194 ++++++++++++++++++++++++++++++++- 2 files changed, 200 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/snd_sst_tokens.h b/include/uapi/sound/snd_sst_tokens.h index f56a932736ca..1ee2e943d66a 100644 --- a/include/uapi/sound/snd_sst_tokens.h +++ b/include/uapi/sound/snd_sst_tokens.h @@ -153,6 +153,10 @@ * * %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 */ @@ -202,7 +206,9 @@ enum SKL_TKNS { SKL_TKN_U32_CAPS_PARAMS_ID, SKL_TKN_U32_CAPS_SIZE, SKL_TKN_U32_PROC_DOMAIN, - SKL_TKN_MAX = 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/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 000482e6e2f7..108ebb9ab329 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -2201,6 +2201,197 @@ static int skl_tplg_control_load(struct snd_soc_component *cmpnt, return 0; } +static int skl_tplg_fill_str_mfest_tkn(struct device *dev, + struct snd_soc_tplg_vendor_string_elem *str_elem, + struct skl_dfw_manifest *minfo) +{ + int tkn_count = 0; + static int ref_count; + + switch (str_elem->token) { + case SKL_TKN_STR_LIB_NAME: + if (ref_count > minfo->lib_count - 1) { + ref_count = 0; + return -EINVAL; + } + + strncpy(minfo->lib[ref_count].name, str_elem->string, + ARRAY_SIZE(minfo->lib[ref_count].name)); + ref_count++; + tkn_count++; + break; + + default: + dev_err(dev, "Not a string token %d", str_elem->token); + break; + } + + return tkn_count; +} + +static int skl_tplg_get_str_tkn(struct device *dev, + struct snd_soc_tplg_vendor_array *array, + struct skl_dfw_manifest *minfo) +{ + int tkn_count = 0, ret; + struct snd_soc_tplg_vendor_string_elem *str_elem; + + str_elem = (struct snd_soc_tplg_vendor_string_elem *)array->value; + while (tkn_count < array->num_elems) { + ret = skl_tplg_fill_str_mfest_tkn(dev, str_elem, minfo); + str_elem++; + + if (ret < 0) + return ret; + + tkn_count = tkn_count + ret; + } + + return tkn_count; +} + +static int skl_tplg_get_int_tkn(struct device *dev, + struct snd_soc_tplg_vendor_value_elem *tkn_elem, + struct skl_dfw_manifest *minfo) +{ + int tkn_count = 0; + + switch (tkn_elem->token) { + case SKL_TKN_U32_LIB_COUNT: + minfo->lib_count = tkn_elem->value; + tkn_count++; + break; + + default: + dev_err(dev, "Not a manifest token %d", tkn_elem->token); + return -EINVAL; + } + + return tkn_count; +} + +/* + * Fill the manifest structure by parsing the tokens based on the + * type. + */ +static int skl_tplg_get_manifest_tkn(struct device *dev, + char *pvt_data, struct skl_dfw_manifest *minfo, + int block_size) +{ + int tkn_count = 0, ret; + int off = 0, tuple_size = 0; + struct snd_soc_tplg_vendor_array *array; + struct snd_soc_tplg_vendor_value_elem *tkn_elem; + + if (block_size <= 0) + return -EINVAL; + + while (tuple_size < block_size) { + array = (struct snd_soc_tplg_vendor_array *)(pvt_data + off); + off += array->size; + switch (array->type) { + case SND_SOC_TPLG_TUPLE_TYPE_STRING: + ret = skl_tplg_get_str_tkn(dev, array, minfo); + + if (ret < 0) + return ret; + tkn_count += ret; + + tuple_size += tkn_count * + sizeof(struct snd_soc_tplg_vendor_string_elem); + continue; + + case SND_SOC_TPLG_TUPLE_TYPE_UUID: + dev_warn(dev, "no uuid tokens for skl tplf manifest"); + continue; + + default: + tkn_elem = array->value; + tkn_count = 0; + break; + } + + while (tkn_count <= array->num_elems - 1) { + ret = skl_tplg_get_int_tkn(dev, + tkn_elem, minfo); + if (ret < 0) + return ret; + + tkn_count = tkn_count + ret; + tkn_elem++; + tuple_size += tkn_count * + sizeof(struct snd_soc_tplg_vendor_value_elem); + break; + } + tkn_count = 0; + } + + return 0; +} + +/* + * Parse manifest private data for tokens. The private data block is + * preceded by descriptors for type and size of data block. + */ +static int skl_tplg_get_manifest_data(struct snd_soc_tplg_manifest *manifest, + struct device *dev, struct skl_dfw_manifest *minfo) +{ + struct snd_soc_tplg_vendor_array *array; + int num_blocks, block_size = 0, block_type, off = 0; + char *data; + int ret; + + /* Read the NUM_DATA_BLOCKS descriptor */ + array = (struct snd_soc_tplg_vendor_array *)manifest->priv.data; + ret = skl_tplg_get_desc_blocks(dev, array); + if (ret < 0) + return ret; + num_blocks = ret; + + off += array->size; + array = (struct snd_soc_tplg_vendor_array *) + (manifest->priv.data + off); + + /* Read the BLOCK_TYPE and BLOCK_SIZE descriptor */ + while (num_blocks > 0) { + ret = skl_tplg_get_desc_blocks(dev, array); + + if (ret < 0) + return ret; + block_type = ret; + off += array->size; + + array = (struct snd_soc_tplg_vendor_array *) + (manifest->priv.data + off); + + ret = skl_tplg_get_desc_blocks(dev, array); + + if (ret < 0) + return ret; + block_size = ret; + off += array->size; + + array = (struct snd_soc_tplg_vendor_array *) + (manifest->priv.data + off); + + data = (manifest->priv.data + off); + + if (block_type == SKL_TYPE_TUPLE) { + ret = skl_tplg_get_manifest_tkn(dev, data, minfo, + block_size); + + if (ret < 0) + return ret; + + --num_blocks; + } else { + return -EINVAL; + } + } + + return 0; +} + static int skl_manifest_load(struct snd_soc_component *cmpnt, struct snd_soc_tplg_manifest *manifest) { @@ -2211,7 +2402,8 @@ static int skl_manifest_load(struct snd_soc_component *cmpnt, int ret = 0; minfo = &skl->skl_sst->manifest; - memcpy(minfo, manifest->priv.data, sizeof(struct skl_dfw_manifest)); + + skl_tplg_get_manifest_data(manifest, bus->dev, minfo); if (minfo->lib_count > HDA_MAX_LIB) { dev_err(bus->dev, "Exceeding max Library count. Got:%d\n", -- cgit v1.2.3 From e5668caec5698f14f310fb06bb39595b21d2fe4a Mon Sep 17 00:00:00 2001 From: Nicolas Iooss Date: Tue, 23 Aug 2016 10:51:22 +0200 Subject: ASoC: simple-card-utils: add __printf attribute asoc_simple_card_set_dailink_name() uses devm_kvasprintf() to format some of its arguments. Adding a __printf attribute to this function makes it possible to detect at compile-time errors related to format strings. Signed-off-by: Nicolas Iooss Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 403ec92164fc..fd6412551145 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -27,6 +27,7 @@ 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, ...); -- cgit v1.2.3 From 398fa4db6c694e1f655cad78478a9e3fb030cc01 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Thu, 15 Sep 2016 07:25:19 +0900 Subject: ALSA: control: move layout of TLV payload to UAPI header In ALSA control interface, each element set can have threshold level information. This information is transferred between drivers/applications, in a shape of tlv packet. The layout of this packet is defined in 'uapi/sound/asound.h' (struct snd_ctl_tlv): struct snd_ctl_tlv { unsigned int numid; unsigned int length; unsigned int tlv[0]; }; Data in the payload (struct snd_ctl_tlv.tlv) is expected to be filled according to our own protocol. This protocol is described in 'include/sound/tlv.h'. A layout of the payload is expected as: struct snd_ctl_tlv.tlv[0]: one of SNDRV_CTL_TLVT_XXX struct snd_ctl_tlv.tlv[1]: Length of data struct snd_ctl_tlv.tlv[2...]: data Unfortunately, the macro is not exported to user land yet, thus applications cannot get to know the protocol. Additionally, ALSA control core has a feature called as 'user-defined' element set. This allows applications to add/remove arbitrary element sets with elements to control devices. Elements in the element set can be operated by the same way as the ones added by in-kernel implementation. For threshold level information of 'user-defined' element set, applications need to register the information to an element set. However, as described above, layout of the payload is closed in kernel land. This is quite inconvenient, too. This commit moves the protocol to UAPI header for TLV. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- include/sound/tlv.h | 61 ------------------------------------------------ include/uapi/sound/tlv.h | 60 +++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 60 insertions(+), 61 deletions(-) (limited to 'include') diff --git a/include/sound/tlv.h b/include/sound/tlv.h index df97d1966468..1f593b6747e9 100644 --- a/include/sound/tlv.h +++ b/include/sound/tlv.h @@ -22,67 +22,6 @@ * */ -/* - * 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 -#define TLV_ITEM(type, ...) \ - (type), TLV_LENGTH(__VA_ARGS__), __VA_ARGS__ -#define TLV_LENGTH(...) \ - ((unsigned int)sizeof((const unsigned int[]) { __VA_ARGS__ })) - -#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 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) } - -/* 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) } - -/* 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) } - -/* 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: */ -#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) \ - 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/tlv.h b/include/uapi/sound/tlv.h index ffc4f203146c..b741e0ef3643 100644 --- a/include/uapi/sound/tlv.h +++ b/include/uapi/sound/tlv.h @@ -28,4 +28,64 @@ #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 TLV_ITEM(type, ...) \ + (type), TLV_LENGTH(__VA_ARGS__), __VA_ARGS__ +#define TLV_LENGTH(...) \ + ((unsigned int)sizeof((const unsigned int[]) { __VA_ARGS__ })) + +#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 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) } + +/* 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) } + +/* 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) } + +/* 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: */ +#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) \ + SNDRV_CTL_TLVT_DB_RANGE, 6 * (num) * sizeof(unsigned int) + +#define TLV_DB_GAIN_MUTE -9999999 + #endif -- cgit v1.2.3 From 46e860f76804f86ac6062e9bb0d6dbea47f23899 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Thu, 15 Sep 2016 07:25:20 +0900 Subject: ALSA: rename TLV-related macros so that they're friendly to user applications In a previous commit, some macros newly appeared to UAPI header for TLV packet. These macros have short names and they easily bring name conflist to applications. The conflict can be avoided to rename them with a proper prefix. For this purpose, this commit renames these macros with prefix 'SNDRV_CTL_TLVD_'. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- include/sound/tlv.h | 26 ++++++++++++++++ include/uapi/sound/tlv.h | 80 ++++++++++++++++++++++++++++-------------------- 2 files changed, 72 insertions(+), 34 deletions(-) (limited to 'include') diff --git a/include/sound/tlv.h b/include/sound/tlv.h index 1f593b6747e9..6e2e7735d20a 100644 --- a/include/sound/tlv.h +++ b/include/sound/tlv.h @@ -24,4 +24,30 @@ #include +/* 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_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_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 + +#define TLV_DB_LINEAR_ITEM SNDRV_CTL_TLVD_DB_LINEAR_ITEM +#define DECLARE_TLV_DB_LINEAR SNDRV_CTL_TLVD_DECLARE_DB_LINEAR + +#define TLV_DB_RANGE_ITEM SNDRV_CTL_TLVD_DB_RANGE_ITEM +#define DECLARE_TLV_DB_RANGE SNDRV_CTL_TLVD_DECLARE_DB_RANGE +#define TLV_DB_RANGE_HEAD SNDRV_CTL_TLVD_DB_RANGE_HEAD + +#define TLV_DB_GAIN_MUTE SNDRV_CTL_TLVD_DB_GAIN_MUTE + #endif /* __SOUND_TLV_H */ diff --git a/include/uapi/sound/tlv.h b/include/uapi/sound/tlv.h index b741e0ef3643..f3c198f8bee7 100644 --- a/include/uapi/sound/tlv.h +++ b/include/uapi/sound/tlv.h @@ -36,41 +36,51 @@ * block_length = (length + (sizeof(unsigned int) - 1)) & * ~(sizeof(unsigned int) - 1)) .... */ -#define TLV_ITEM(type, ...) \ - (type), TLV_LENGTH(__VA_ARGS__), __VA_ARGS__ -#define TLV_LENGTH(...) \ +#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 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 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 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 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 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 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 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 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 @@ -78,14 +88,16 @@ * dB values (which is also required for all other mixer controls). */ /* Each item is: */ -#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__) } +#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__) \ + } /* The below assumes that each item TLV is 4 words like DB_SCALE or LINEAR */ -#define TLV_DB_RANGE_HEAD(num) \ +#define SNDRV_CTL_TLVD_DB_RANGE_HEAD(num) \ SNDRV_CTL_TLVT_DB_RANGE, 6 * (num) * sizeof(unsigned int) -#define TLV_DB_GAIN_MUTE -9999999 +#define SNDRV_CTL_TLVD_DB_GAIN_MUTE -9999999 #endif -- cgit v1.2.3 From a16039cbf1a1ee7e76d7d100f9e613998919ab91 Mon Sep 17 00:00:00 2001 From: Andrej Krutak Date: Sun, 18 Sep 2016 20:59:32 +0200 Subject: ALSA: line6: Add hwdep interface to access the POD control messages We must do it this way, because e.g. POD X3 won't play any sound unless the host listens on the bulk EP, so we cannot export it only via libusb. The driver currently doesn't use the bulk EP messages in other way, in future it could e.g. sense/modify volume(s). Signed-off-by: Andrej Krutak Signed-off-by: Takashi Iwai --- include/uapi/sound/asound.h | 3 +- sound/usb/line6/driver.c | 152 ++++++++++++++++++++++++++++++++++++++++++-- sound/usb/line6/driver.h | 23 ++++++- 3 files changed, 171 insertions(+), 7 deletions(-) (limited to 'include') 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/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index 8a71d45ce945..f9224ec141c8 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -17,6 +17,7 @@ #include #include +#include #include "capture.h" #include "driver.h" @@ -303,7 +304,7 @@ static void line6_data_received(struct urb *urb) for (;;) { done = line6_midibuf_read(mb, line6->buffer_message, - LINE6_MESSAGE_MAXLEN); + LINE6_MIDI_MESSAGE_MAXLEN); if (done == 0) break; @@ -315,8 +316,11 @@ static void line6_data_received(struct urb *urb) line6->process_message(line6); } } else { + line6->buffer_message = urb->transfer_buffer; + line6->message_length = urb->actual_length; if (line6->process_message) line6->process_message(line6); + line6->buffer_message = NULL; } line6_start_listen(line6); @@ -473,14 +477,17 @@ static void line6_destruct(struct snd_card *card) struct usb_line6 *line6 = card->private_data; struct usb_device *usbdev = line6->usbdev; - /* free buffer memory first: */ - if (line6->properties->capabilities & LINE6_CAP_CONTROL_MIDI) + /* Free buffer memory first. We cannot depend on the existence of private + * data from the (podhd) module, it may be gone already during this call + */ + if (line6->buffer_message) kfree(line6->buffer_message); kfree(line6->buffer_listen); /* then free URBs: */ usb_free_urb(line6->urb_listen); + line6->urb_listen = NULL; /* decrement reference counters: */ usb_put_dev(usbdev); @@ -522,6 +529,138 @@ static void line6_get_interval(struct usb_line6 *line6) } } + +/* Enable buffering of incoming messages, flush the buffer */ +static int line6_hwdep_open(struct snd_hwdep *hw, struct file *file) +{ + struct usb_line6 *line6 = hw->private_data; + + /* NOTE: hwdep layer provides atomicity here */ + + line6->messages.active = 1; + + return 0; +} + +/* Stop buffering */ +static int line6_hwdep_release(struct snd_hwdep *hw, struct file *file) +{ + struct usb_line6 *line6 = hw->private_data; + + line6->messages.active = 0; + + return 0; +} + +/* Read from circular buffer, return to user */ +static long +line6_hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, + loff_t *offset) +{ + struct usb_line6 *line6 = hwdep->private_data; + long rv = 0; + unsigned int out_count; + + if (mutex_lock_interruptible(&line6->messages.read_lock)) + return -ERESTARTSYS; + + while (kfifo_len(&line6->messages.fifo) == 0) { + mutex_unlock(&line6->messages.read_lock); + + rv = wait_event_interruptible( + line6->messages.wait_queue, + kfifo_len(&line6->messages.fifo) != 0); + if (rv < 0) + return rv; + + if (mutex_lock_interruptible(&line6->messages.read_lock)) + return -ERESTARTSYS; + } + + if (kfifo_peek_len(&line6->messages.fifo) > count) { + /* Buffer too small; allow re-read of the current item... */ + rv = -EINVAL; + } else { + rv = kfifo_to_user(&line6->messages.fifo, buf, count, &out_count); + if (rv == 0) + rv = out_count; + } + + mutex_unlock(&line6->messages.read_lock); + return rv; +} + +/* Write directly (no buffering) to device by user*/ +static long +line6_hwdep_write(struct snd_hwdep *hwdep, const char __user *data, long count, + loff_t *offset) +{ + struct usb_line6 *line6 = hwdep->private_data; + int rv; + char *data_copy; + + if (count > line6->max_packet_size * LINE6_RAW_MESSAGES_MAXCOUNT) { + /* This is an arbitrary limit - still better than nothing... */ + return -EINVAL; + } + + data_copy = memdup_user(data, count); + if (IS_ERR(ERR_PTR)) + return -ENOMEM; + + rv = line6_send_raw_message(line6, data_copy, count); + + kfree(data_copy); + return rv; +} + +static const struct snd_hwdep_ops hwdep_ops = { + .open = line6_hwdep_open, + .release = line6_hwdep_release, + .read = line6_hwdep_read, + .write = line6_hwdep_write, +}; + +/* Insert into circular buffer */ +static void line6_hwdep_push_message(struct usb_line6 *line6) +{ + if (!line6->messages.active) + return; + + if (kfifo_avail(&line6->messages.fifo) >= line6->message_length) { + /* No race condition here, there's only one writer */ + kfifo_in(&line6->messages.fifo, + line6->buffer_message, line6->message_length); + } /* else TODO: signal overflow */ + + wake_up_interruptible(&line6->messages.wait_queue); +} + +static int line6_hwdep_init(struct usb_line6 *line6) +{ + int err; + struct snd_hwdep *hwdep; + + /* TODO: usb_driver_claim_interface(); */ + line6->process_message = line6_hwdep_push_message; + line6->messages.active = 0; + init_waitqueue_head(&line6->messages.wait_queue); + mutex_init(&line6->messages.read_lock); + INIT_KFIFO(line6->messages.fifo); + + err = snd_hwdep_new(line6->card, "config", 0, &hwdep); + if (err < 0) + goto end; + strcpy(hwdep->name, "config"); + hwdep->iface = SNDRV_HWDEP_IFACE_LINE6; + hwdep->ops = hwdep_ops; + hwdep->private_data = line6; + hwdep->exclusive = true; + +end: + return err; +} + static int line6_init_cap_control(struct usb_line6 *line6) { int ret; @@ -536,9 +675,13 @@ static int line6_init_cap_control(struct usb_line6 *line6) return -ENOMEM; if (line6->properties->capabilities & LINE6_CAP_CONTROL_MIDI) { - line6->buffer_message = kmalloc(LINE6_MESSAGE_MAXLEN, GFP_KERNEL); + line6->buffer_message = kmalloc(LINE6_MIDI_MESSAGE_MAXLEN, GFP_KERNEL); if (!line6->buffer_message) return -ENOMEM; + } else { + ret = line6_hwdep_init(line6); + if (ret < 0) + return ret; } ret = line6_start_listen(line6); @@ -716,3 +859,4 @@ EXPORT_SYMBOL_GPL(line6_resume); MODULE_AUTHOR(DRIVER_AUTHOR); MODULE_DESCRIPTION(DRIVER_DESC); MODULE_LICENSE("GPL"); + diff --git a/sound/usb/line6/driver.h b/sound/usb/line6/driver.h index 88cf1e750060..7e3a3aada222 100644 --- a/sound/usb/line6/driver.h +++ b/sound/usb/line6/driver.h @@ -12,8 +12,9 @@ #ifndef DRIVER_H #define DRIVER_H -#include #include +#include +#include #include #include "midi.h" @@ -32,7 +33,16 @@ #define LINE6_TIMEOUT 1 #define LINE6_BUFSIZE_LISTEN 64 -#define LINE6_MESSAGE_MAXLEN 256 +#define LINE6_MIDI_MESSAGE_MAXLEN 256 + +#define LINE6_RAW_MESSAGES_MAXCOUNT_ORDER 7 +/* 4k packets are common, BUFSIZE * MAXCOUNT should be bigger... */ +#define LINE6_RAW_MESSAGES_MAXCOUNT (1 << LINE6_RAW_MESSAGES_MAXCOUNT_ORDER) + + +#if LINE6_BUFSIZE_LISTEN > 65535 +#error "Use dynamic fifo instead" +#endif /* Line 6 MIDI control commands @@ -156,6 +166,15 @@ struct usb_line6 { /* Length of message to be processed, generated from MIDI layer */ int message_length; + /* Circular buffer for non-MIDI control messages */ + struct { + struct mutex read_lock; + wait_queue_head_t wait_queue; + unsigned int active:1; + STRUCT_KFIFO_REC_2(LINE6_BUFSIZE_LISTEN * LINE6_RAW_MESSAGES_MAXCOUNT) + fifo; + } messages; + /* If MIDI is supported, buffer_message contains the pre-processed data; * otherwise the data is only in urb_listen (buffer_incoming). */ -- cgit v1.2.3 From 2b26dd4c1fc5f83bc088f4a053120ca03817045e Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Mon, 19 Sep 2016 19:26:08 +0800 Subject: ASoC: rt5660: add rt5660 codec driver This is the initial codec driver for rt5660 Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/rt5660.txt | 47 + include/sound/rt5660.h | 31 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5660.c | 1353 ++++++++++++++++++++ sound/soc/codecs/rt5660.h | 847 ++++++++++++ 6 files changed, 2286 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt5660.txt create mode 100644 include/sound/rt5660.h create mode 100644 sound/soc/codecs/rt5660.c create mode 100644 sound/soc/codecs/rt5660.h (limited to 'include') diff --git a/Documentation/devicetree/bindings/sound/rt5660.txt b/Documentation/devicetree/bindings/sound/rt5660.txt new file mode 100644 index 000000000000..30be5f921930 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5660.txt @@ -0,0 +1,47 @@ +RT5660 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5660". + +- reg : The I2C address of the device. + +Optional properties: + +- clocks: The phandle of the master clock to the CODEC +- clock-names: Should be "mclk" + +- realtek,in1-differential +- realtek,in3-differential + Boolean. Indicate MIC1/3 input are differential, rather than single-ended. + +- realtek,poweroff-in-suspend + Boolean. If the codec will be powered off in suspend, the resume should be + added delay time for waiting codec power ready. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using GPIO2 pin as dmic1 data pin + 2: using IN1P pin as dmic1 data pin + +Pins on the device (for linking into audio routes) for RT5660: + + * DMIC L1 + * DMIC R1 + * IN1P + * IN1N + * IN2P + * IN3P + * IN3N + * SPO + * LOUTL + * LOUTR + +Example: + +rt5660 { + compatible = "realtek,rt5660"; + reg = <0x1c>; +}; 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 + * + * 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/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1cd6ab344d67..890affe5eb65 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -112,6 +112,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5645 if I2C select SND_SOC_RT5651 if I2C select SND_SOC_RT5659 if I2C + select SND_SOC_RT5660 if I2C select SND_SOC_RT5670 if I2C select SND_SOC_RT5677 if I2C && SPI_MASTER select SND_SOC_SGTL5000 if I2C @@ -645,6 +646,7 @@ config SND_SOC_RL6231 default y if SND_SOC_RT5645=y default y if SND_SOC_RT5651=y default y if SND_SOC_RT5659=y + default y if SND_SOC_RT5660=y default y if SND_SOC_RT5670=y default y if SND_SOC_RT5677=y default m if SND_SOC_RT5514=m @@ -653,6 +655,7 @@ config SND_SOC_RL6231 default m if SND_SOC_RT5645=m default m if SND_SOC_RT5651=m default m if SND_SOC_RT5659=m + default m if SND_SOC_RT5660=m default m if SND_SOC_RT5670=m default m if SND_SOC_RT5677=m @@ -697,6 +700,9 @@ config SND_SOC_RT5651 config SND_SOC_RT5659 tristate +config SND_SOC_RT5660 + tristate + config SND_SOC_RT5670 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 58036af2c7d9..68fed20ccfaf 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -112,6 +112,7 @@ snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o snd-soc-rt5651-objs := rt5651.o snd-soc-rt5659-objs := rt5659.o +snd-soc-rt5660-objs := rt5660.o snd-soc-rt5670-objs := rt5670.o snd-soc-rt5677-objs := rt5677.o snd-soc-rt5677-spi-objs := rt5677-spi.o @@ -333,6 +334,7 @@ obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o obj-$(CONFIG_SND_SOC_RT5651) += snd-soc-rt5651.o obj-$(CONFIG_SND_SOC_RT5659) += snd-soc-rt5659.o +obj-$(CONFIG_SND_SOC_RT5660) += snd-soc-rt5660.o obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o diff --git a/sound/soc/codecs/rt5660.c b/sound/soc/codecs/rt5660.c new file mode 100644 index 000000000000..9f0933ced804 --- /dev/null +++ b/sound/soc/codecs/rt5660.c @@ -0,0 +1,1353 @@ +/* + * rt5660.c -- RT5660 ALSA SoC audio codec driver + * + * Copyright 2016 Realtek Semiconductor Corp. + * Author: Oder Chiou + * + * 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 +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "rl6231.h" +#include "rt5660.h" + +#define RT5660_DEVICE_ID 0x6338 + +#define RT5660_PR_RANGE_BASE (0xff + 1) +#define RT5660_PR_SPACING 0x100 + +#define RT5660_PR_BASE (RT5660_PR_RANGE_BASE + (0 * RT5660_PR_SPACING)) + +static const struct regmap_range_cfg rt5660_ranges[] = { + { .name = "PR", .range_min = RT5660_PR_BASE, + .range_max = RT5660_PR_BASE + 0xf3, + .selector_reg = RT5660_PRIV_INDEX, + .selector_mask = 0xff, + .selector_shift = 0x0, + .window_start = RT5660_PRIV_DATA, + .window_len = 0x1, }, +}; + +static const struct reg_sequence rt5660_patch[] = { + { RT5660_ALC_PGA_CTRL2, 0x44c3 }, + { RT5660_PR_BASE + 0x3d, 0x2600 }, +}; + +static const struct reg_default rt5660_reg[] = { + { 0x00, 0x0000 }, + { 0x01, 0xc800 }, + { 0x02, 0xc8c8 }, + { 0x0d, 0x1010 }, + { 0x0e, 0x1010 }, + { 0x19, 0xafaf }, + { 0x1c, 0x2f2f }, + { 0x1e, 0x0000 }, + { 0x27, 0x6060 }, + { 0x29, 0x8080 }, + { 0x2a, 0x4242 }, + { 0x2f, 0x0000 }, + { 0x3b, 0x0000 }, + { 0x3c, 0x007f }, + { 0x3d, 0x0000 }, + { 0x3e, 0x007f }, + { 0x45, 0xe000 }, + { 0x46, 0x003e }, + { 0x48, 0xf800 }, + { 0x4a, 0x0004 }, + { 0x4d, 0x0000 }, + { 0x4e, 0x0000 }, + { 0x4f, 0x01ff }, + { 0x50, 0x0000 }, + { 0x51, 0x0000 }, + { 0x52, 0x01ff }, + { 0x61, 0x0000 }, + { 0x62, 0x0000 }, + { 0x63, 0x00c0 }, + { 0x64, 0x0000 }, + { 0x65, 0x0000 }, + { 0x66, 0x0000 }, + { 0x70, 0x8000 }, + { 0x73, 0x7000 }, + { 0x74, 0x3c00 }, + { 0x75, 0x2800 }, + { 0x80, 0x0000 }, + { 0x81, 0x0000 }, + { 0x82, 0x0000 }, + { 0x8c, 0x0228 }, + { 0x8d, 0xa000 }, + { 0x8e, 0x0000 }, + { 0x92, 0x0000 }, + { 0x93, 0x3000 }, + { 0xa1, 0x0059 }, + { 0xa2, 0x0001 }, + { 0xa3, 0x5c80 }, + { 0xa4, 0x0146 }, + { 0xa5, 0x1f1f }, + { 0xa6, 0x78c6 }, + { 0xa7, 0xe5ec }, + { 0xa8, 0xba61 }, + { 0xa9, 0x3c78 }, + { 0xaa, 0x8ae2 }, + { 0xab, 0xe5ec }, + { 0xac, 0xc600 }, + { 0xad, 0xba61 }, + { 0xae, 0x17ed }, + { 0xb0, 0x2080 }, + { 0xb1, 0x0000 }, + { 0xb3, 0x001f }, + { 0xb4, 0x020c }, + { 0xb5, 0x1f00 }, + { 0xb6, 0x0000 }, + { 0xb7, 0x4000 }, + { 0xbb, 0x0000 }, + { 0xbd, 0x0000 }, + { 0xbe, 0x0000 }, + { 0xbf, 0x0100 }, + { 0xc0, 0x0000 }, + { 0xc2, 0x0000 }, + { 0xd3, 0xa220 }, + { 0xd9, 0x0809 }, + { 0xda, 0x0000 }, + { 0xe0, 0x8000 }, + { 0xe1, 0x0200 }, + { 0xe2, 0x8000 }, + { 0xe3, 0x0200 }, + { 0xe4, 0x0f20 }, + { 0xe5, 0x001f }, + { 0xe6, 0x020c }, + { 0xe7, 0x1f00 }, + { 0xe8, 0x0000 }, + { 0xe9, 0x4000 }, + { 0xea, 0x00a6 }, + { 0xeb, 0x04c3 }, + { 0xec, 0x27c8 }, + { 0xed, 0x7418 }, + { 0xee, 0xbf50 }, + { 0xef, 0x0045 }, + { 0xf0, 0x0007 }, + { 0xfa, 0x0000 }, + { 0xfd, 0x0000 }, + { 0xfe, 0x10ec }, + { 0xff, 0x6338 }, +}; + +static bool rt5660_volatile_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5660_ranges); i++) + if ((reg >= rt5660_ranges[i].window_start && + reg <= rt5660_ranges[i].window_start + + rt5660_ranges[i].window_len) || + (reg >= rt5660_ranges[i].range_min && + reg <= rt5660_ranges[i].range_max)) + return true; + + switch (reg) { + case RT5660_RESET: + case RT5660_PRIV_DATA: + case RT5660_EQ_CTRL1: + case RT5660_IRQ_CTRL2: + case RT5660_INT_IRQ_ST: + case RT5660_VENDOR_ID: + case RT5660_VENDOR_ID1: + case RT5660_VENDOR_ID2: + return true; + default: + return false; + } +} + +static bool rt5660_readable_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5660_ranges); i++) + if ((reg >= rt5660_ranges[i].window_start && + reg <= rt5660_ranges[i].window_start + + rt5660_ranges[i].window_len) || + (reg >= rt5660_ranges[i].range_min && + reg <= rt5660_ranges[i].range_max)) + return true; + + switch (reg) { + case RT5660_RESET: + case RT5660_SPK_VOL: + case RT5660_LOUT_VOL: + case RT5660_IN1_IN2: + case RT5660_IN3_IN4: + case RT5660_DAC1_DIG_VOL: + case RT5660_STO1_ADC_DIG_VOL: + case RT5660_ADC_BST_VOL1: + case RT5660_STO1_ADC_MIXER: + case RT5660_AD_DA_MIXER: + case RT5660_STO_DAC_MIXER: + case RT5660_DIG_INF1_DATA: + case RT5660_REC_L1_MIXER: + case RT5660_REC_L2_MIXER: + case RT5660_REC_R1_MIXER: + case RT5660_REC_R2_MIXER: + case RT5660_LOUT_MIXER: + case RT5660_SPK_MIXER: + case RT5660_SPO_MIXER: + case RT5660_SPO_CLSD_RATIO: + case RT5660_OUT_L_GAIN1: + case RT5660_OUT_L_GAIN2: + case RT5660_OUT_L1_MIXER: + case RT5660_OUT_R_GAIN1: + case RT5660_OUT_R_GAIN2: + case RT5660_OUT_R1_MIXER: + case RT5660_PWR_DIG1: + case RT5660_PWR_DIG2: + case RT5660_PWR_ANLG1: + case RT5660_PWR_ANLG2: + case RT5660_PWR_MIXER: + case RT5660_PWR_VOL: + case RT5660_PRIV_INDEX: + case RT5660_PRIV_DATA: + case RT5660_I2S1_SDP: + case RT5660_ADDA_CLK1: + case RT5660_ADDA_CLK2: + case RT5660_DMIC_CTRL1: + case RT5660_GLB_CLK: + case RT5660_PLL_CTRL1: + case RT5660_PLL_CTRL2: + case RT5660_CLSD_AMP_OC_CTRL: + case RT5660_CLSD_AMP_CTRL: + case RT5660_LOUT_AMP_CTRL: + case RT5660_SPK_AMP_SPKVDD: + case RT5660_MICBIAS: + case RT5660_CLSD_OUT_CTRL1: + case RT5660_CLSD_OUT_CTRL2: + case RT5660_DIPOLE_MIC_CTRL1: + case RT5660_DIPOLE_MIC_CTRL2: + case RT5660_DIPOLE_MIC_CTRL3: + case RT5660_DIPOLE_MIC_CTRL4: + case RT5660_DIPOLE_MIC_CTRL5: + case RT5660_DIPOLE_MIC_CTRL6: + case RT5660_DIPOLE_MIC_CTRL7: + case RT5660_DIPOLE_MIC_CTRL8: + case RT5660_DIPOLE_MIC_CTRL9: + case RT5660_DIPOLE_MIC_CTRL10: + case RT5660_DIPOLE_MIC_CTRL11: + case RT5660_DIPOLE_MIC_CTRL12: + case RT5660_EQ_CTRL1: + case RT5660_EQ_CTRL2: + case RT5660_DRC_AGC_CTRL1: + case RT5660_DRC_AGC_CTRL2: + case RT5660_DRC_AGC_CTRL3: + case RT5660_DRC_AGC_CTRL4: + case RT5660_DRC_AGC_CTRL5: + case RT5660_JD_CTRL: + case RT5660_IRQ_CTRL1: + case RT5660_IRQ_CTRL2: + case RT5660_INT_IRQ_ST: + case RT5660_GPIO_CTRL1: + case RT5660_GPIO_CTRL2: + case RT5660_WIND_FILTER_CTRL1: + case RT5660_SV_ZCD1: + case RT5660_SV_ZCD2: + case RT5660_DRC1_LM_CTRL1: + case RT5660_DRC1_LM_CTRL2: + case RT5660_DRC2_LM_CTRL1: + case RT5660_DRC2_LM_CTRL2: + case RT5660_MULTI_DRC_CTRL: + case RT5660_DRC2_CTRL1: + case RT5660_DRC2_CTRL2: + case RT5660_DRC2_CTRL3: + case RT5660_DRC2_CTRL4: + case RT5660_DRC2_CTRL5: + case RT5660_ALC_PGA_CTRL1: + case RT5660_ALC_PGA_CTRL2: + case RT5660_ALC_PGA_CTRL3: + case RT5660_ALC_PGA_CTRL4: + case RT5660_ALC_PGA_CTRL5: + case RT5660_ALC_PGA_CTRL6: + case RT5660_ALC_PGA_CTRL7: + case RT5660_GEN_CTRL1: + case RT5660_GEN_CTRL2: + case RT5660_GEN_CTRL3: + case RT5660_VENDOR_ID: + case RT5660_VENDOR_ID1: + case RT5660_VENDOR_ID2: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(rt5660_out_vol_tlv, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(rt5660_dac_vol_tlv, -6525, 75, 0); +static const DECLARE_TLV_DB_SCALE(rt5660_adc_vol_tlv, -1725, 75, 0); +static const DECLARE_TLV_DB_SCALE(rt5660_adc_bst_tlv, 0, 1200, 0); +static const DECLARE_TLV_DB_SCALE(rt5660_bst_tlv, -1200, 75, 0); + +static const struct snd_kcontrol_new rt5660_snd_controls[] = { + /* Speaker Output Volume */ + SOC_SINGLE("Speaker Playback Switch", RT5660_SPK_VOL, RT5660_L_MUTE_SFT, + 1, 1), + SOC_SINGLE_TLV("Speaker Playback Volume", RT5660_SPK_VOL, + RT5660_L_VOL_SFT, 39, 1, rt5660_out_vol_tlv), + + /* OUTPUT Control */ + SOC_DOUBLE("OUT Playback Switch", RT5660_LOUT_VOL, RT5660_L_MUTE_SFT, + RT5660_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("OUT Playback Volume", RT5660_LOUT_VOL, RT5660_L_VOL_SFT, + RT5660_R_VOL_SFT, 39, 1, rt5660_out_vol_tlv), + + /* DAC Digital Volume */ + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5660_DAC1_DIG_VOL, + RT5660_DAC_L1_VOL_SFT, RT5660_DAC_R1_VOL_SFT, 87, 0, + rt5660_dac_vol_tlv), + + /* IN1/IN2/IN3 Control */ + SOC_SINGLE_TLV("IN1 Boost Volume", RT5660_IN1_IN2, RT5660_BST_SFT1, 69, + 0, rt5660_bst_tlv), + SOC_SINGLE_TLV("IN2 Boost Volume", RT5660_IN1_IN2, RT5660_BST_SFT2, 69, + 0, rt5660_bst_tlv), + SOC_SINGLE_TLV("IN3 Boost Volume", RT5660_IN3_IN4, RT5660_BST_SFT3, 69, + 0, rt5660_bst_tlv), + + /* ADC Digital Volume Control */ + SOC_DOUBLE("ADC Capture Switch", RT5660_STO1_ADC_DIG_VOL, + RT5660_L_MUTE_SFT, RT5660_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("ADC Capture Volume", RT5660_STO1_ADC_DIG_VOL, + RT5660_ADC_L_VOL_SFT, RT5660_ADC_R_VOL_SFT, 63, 0, + rt5660_adc_vol_tlv), + + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("STO1 ADC Boost Gain Volume", RT5660_ADC_BST_VOL1, + RT5660_STO1_ADC_L_BST_SFT, RT5660_STO1_ADC_R_BST_SFT, 3, 0, + rt5660_adc_bst_tlv), +}; + +/** + * rt5660_set_dmic_clk - Set parameter of dmic. + * + * @w: DAPM widget. + * @kcontrol: The kcontrol of this widget. + * @event: Event id. + * + */ +static int rt5660_set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + int idx, rate; + + rate = rt5660->sysclk / rl6231_get_pre_div(rt5660->regmap, + RT5660_ADDA_CLK1, RT5660_I2S_PD1_SFT); + idx = rl6231_calc_dmic_clk(rate); + if (idx < 0) + dev_err(codec->dev, "Failed to set DMIC clock\n"); + else + snd_soc_update_bits(codec, RT5660_DMIC_CTRL1, + RT5660_DMIC_CLK_MASK, idx << RT5660_DMIC_CLK_SFT); + + return idx; +} + +static int rt5660_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + unsigned int val; + + val = snd_soc_read(codec, RT5660_GLB_CLK); + val &= RT5660_SCLK_SRC_MASK; + if (val == RT5660_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5660_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5660_STO1_ADC_MIXER, + RT5660_M_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5660_STO1_ADC_MIXER, + RT5660_M_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5660_STO1_ADC_MIXER, + RT5660_M_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5660_STO1_ADC_MIXER, + RT5660_M_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5660_AD_DA_MIXER, + RT5660_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5660_AD_DA_MIXER, + RT5660_M_DAC1_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5660_AD_DA_MIXER, + RT5660_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5660_AD_DA_MIXER, + RT5660_M_DAC1_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_sto_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5660_STO_DAC_MIXER, + RT5660_M_DAC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5660_STO_DAC_MIXER, + RT5660_M_DAC_R1_STO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_sto_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5660_STO_DAC_MIXER, + RT5660_M_DAC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5660_STO_DAC_MIXER, + RT5660_M_DAC_L1_STO_R_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5660_rec_l_mix[] = { + SOC_DAPM_SINGLE("BST3 Switch", RT5660_REC_L2_MIXER, + RT5660_M_BST3_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5660_REC_L2_MIXER, + RT5660_M_BST2_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5660_REC_L2_MIXER, + RT5660_M_BST1_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXL Switch", RT5660_REC_L2_MIXER, + RT5660_M_OM_L_RM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_rec_r_mix[] = { + SOC_DAPM_SINGLE("BST3 Switch", RT5660_REC_R2_MIXER, + RT5660_M_BST3_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5660_REC_R2_MIXER, + RT5660_M_BST2_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5660_REC_R2_MIXER, + RT5660_M_BST1_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXR Switch", RT5660_REC_R2_MIXER, + RT5660_M_OM_R_RM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_spk_mix[] = { + SOC_DAPM_SINGLE("BST3 Switch", RT5660_SPK_MIXER, + RT5660_M_BST3_SM_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5660_SPK_MIXER, + RT5660_M_BST1_SM_SFT, 1, 1), + SOC_DAPM_SINGLE("DACL Switch", RT5660_SPK_MIXER, + RT5660_M_DACL_SM_SFT, 1, 1), + SOC_DAPM_SINGLE("DACR Switch", RT5660_SPK_MIXER, + RT5660_M_DACR_SM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTMIXL Switch", RT5660_SPK_MIXER, + RT5660_M_OM_L_SM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_out_l_mix[] = { + SOC_DAPM_SINGLE("BST3 Switch", RT5660_OUT_L1_MIXER, + RT5660_M_BST3_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5660_OUT_L1_MIXER, + RT5660_M_BST2_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5660_OUT_L1_MIXER, + RT5660_M_BST1_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIXL Switch", RT5660_OUT_L1_MIXER, + RT5660_M_RM_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DACR Switch", RT5660_OUT_L1_MIXER, + RT5660_M_DAC_R_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DACL Switch", RT5660_OUT_L1_MIXER, + RT5660_M_DAC_L_OM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_out_r_mix[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5660_OUT_R1_MIXER, + RT5660_M_BST2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5660_OUT_R1_MIXER, + RT5660_M_BST1_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIXR Switch", RT5660_OUT_R1_MIXER, + RT5660_M_RM_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DACR Switch", RT5660_OUT_R1_MIXER, + RT5660_M_DAC_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DACL Switch", RT5660_OUT_R1_MIXER, + RT5660_M_DAC_L_OM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_spo_mix[] = { + SOC_DAPM_SINGLE("DACR Switch", RT5660_SPO_MIXER, + RT5660_M_DAC_R_SPM_SFT, 1, 1), + SOC_DAPM_SINGLE("DACL Switch", RT5660_SPO_MIXER, + RT5660_M_DAC_L_SPM_SFT, 1, 1), + SOC_DAPM_SINGLE("SPKVOL Switch", RT5660_SPO_MIXER, + RT5660_M_SV_SPM_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5660_SPO_MIXER, + RT5660_M_BST1_SPM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_lout_mix[] = { + SOC_DAPM_SINGLE("DAC Switch", RT5660_LOUT_MIXER, + RT5660_M_DAC1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTMIX Switch", RT5660_LOUT_MIXER, + RT5660_M_LOVOL_LM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new spk_vol_control = + SOC_DAPM_SINGLE("Switch", RT5660_SPK_VOL, + RT5660_VOL_L_SFT, 1, 1); + +static const struct snd_kcontrol_new lout_l_vol_control = + SOC_DAPM_SINGLE("Switch", RT5660_LOUT_VOL, + RT5660_VOL_L_SFT, 1, 1); + +static const struct snd_kcontrol_new lout_r_vol_control = + SOC_DAPM_SINGLE("Switch", RT5660_LOUT_VOL, + RT5660_VOL_R_SFT, 1, 1); + +/* Interface data select */ +static const char * const rt5660_data_select[] = { + "L/R", "R/L", "L/L", "R/R" +}; + +static const SOC_ENUM_SINGLE_DECL(rt5660_if1_dac_enum, + RT5660_DIG_INF1_DATA, RT5660_IF1_DAC_IN_SFT, rt5660_data_select); + +static const SOC_ENUM_SINGLE_DECL(rt5660_if1_adc_enum, + RT5660_DIG_INF1_DATA, RT5660_IF1_ADC_IN_SFT, rt5660_data_select); + +static const struct snd_kcontrol_new rt5660_if1_dac_swap_mux = + SOC_DAPM_ENUM("IF1 DAC Swap Source", rt5660_if1_dac_enum); + +static const struct snd_kcontrol_new rt5660_if1_adc_swap_mux = + SOC_DAPM_ENUM("IF1 ADC Swap Source", rt5660_if1_adc_enum); + +static int rt5660_lout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5660_LOUT_AMP_CTRL, + RT5660_LOUT_CO_MASK | RT5660_LOUT_CB_MASK, + RT5660_LOUT_CO_EN | RT5660_LOUT_CB_PU); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5660_LOUT_AMP_CTRL, + RT5660_LOUT_CO_MASK | RT5660_LOUT_CB_MASK, + RT5660_LOUT_CO_DIS | RT5660_LOUT_CB_PD); + break; + + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt5660_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("LDO2", RT5660_PWR_ANLG1, + RT5660_PWR_LDO2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL1", RT5660_PWR_ANLG2, + RT5660_PWR_PLL_BIT, 0, NULL, 0), + + /* MICBIAS */ + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5660_PWR_ANLG2, + RT5660_PWR_MB1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5660_PWR_ANLG2, + RT5660_PWR_MB2_BIT, 0, NULL, 0), + + /* Input Side */ + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC L1"), + SND_SOC_DAPM_INPUT("DMIC R1"), + + SND_SOC_DAPM_INPUT("IN1P"), + SND_SOC_DAPM_INPUT("IN1N"), + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN3P"), + SND_SOC_DAPM_INPUT("IN3N"), + + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + rt5660_set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC Power", RT5660_DMIC_CTRL1, + RT5660_DMIC_1_EN_SFT, 0, NULL, 0), + + /* Boost */ + SND_SOC_DAPM_PGA("BST1", RT5660_PWR_ANLG2, RT5660_PWR_BST1_BIT, 0, + NULL, 0), + SND_SOC_DAPM_PGA("BST2", RT5660_PWR_ANLG2, RT5660_PWR_BST2_BIT, 0, + NULL, 0), + SND_SOC_DAPM_PGA("BST3", RT5660_PWR_ANLG2, RT5660_PWR_BST3_BIT, 0, + NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIXL", RT5660_PWR_MIXER, RT5660_PWR_RM_L_BIT, + 0, rt5660_rec_l_mix, ARRAY_SIZE(rt5660_rec_l_mix)), + SND_SOC_DAPM_MIXER("RECMIXR", RT5660_PWR_MIXER, RT5660_PWR_RM_R_BIT, + 0, rt5660_rec_r_mix, ARRAY_SIZE(rt5660_rec_r_mix)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC R", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_SUPPLY("ADC L power", RT5660_PWR_DIG1, + RT5660_PWR_ADC_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC R power", RT5660_PWR_DIG1, + RT5660_PWR_ADC_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC clock", RT5660_PR_BASE + RT5660_CHOP_DAC_ADC, + 12, 0, NULL, 0), + + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("adc stereo1 filter", RT5660_PWR_DIG2, + RT5660_PWR_ADC_S1F_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Sto1 ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5660_sto1_adc_l_mix, ARRAY_SIZE(rt5660_sto1_adc_l_mix)), + SND_SOC_DAPM_MIXER("Sto1 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5660_sto1_adc_r_mix, ARRAY_SIZE(rt5660_sto1_adc_r_mix)), + + /* ADC */ + SND_SOC_DAPM_ADC("Stereo1 ADC MIXL", NULL, RT5660_STO1_ADC_DIG_VOL, + RT5660_L_MUTE_SFT, 1), + SND_SOC_DAPM_ADC("Stereo1 ADC MIXR", NULL, RT5660_STO1_ADC_DIG_VOL, + RT5660_R_MUTE_SFT, 1), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5660_PWR_DIG1, RT5660_PWR_I2S1_BIT, 0, + NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MUX("IF1 DAC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5660_if1_dac_swap_mux), + SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MUX("IF1 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5660_if1_adc_swap_mux), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0, rt5660_dac_l_mix, + ARRAY_SIZE(rt5660_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0, rt5660_dac_r_mix, + ARRAY_SIZE(rt5660_dac_r_mix)), + + /* DAC Mixer */ + SND_SOC_DAPM_SUPPLY("dac stereo1 filter", RT5660_PWR_DIG2, + RT5660_PWR_DAC_S1F_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5660_sto_dac_l_mix, ARRAY_SIZE(rt5660_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5660_sto_dac_r_mix, ARRAY_SIZE(rt5660_sto_dac_r_mix)), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC L1", NULL, RT5660_PWR_DIG1, + RT5660_PWR_DAC_L1_BIT, 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, RT5660_PWR_DIG1, + RT5660_PWR_DAC_R1_BIT, 0), + + /* OUT Mixer */ + SND_SOC_DAPM_MIXER("SPK MIX", RT5660_PWR_MIXER, RT5660_PWR_SM_BIT, + 0, rt5660_spk_mix, ARRAY_SIZE(rt5660_spk_mix)), + SND_SOC_DAPM_MIXER("OUT MIXL", RT5660_PWR_MIXER, RT5660_PWR_OM_L_BIT, + 0, rt5660_out_l_mix, ARRAY_SIZE(rt5660_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5660_PWR_MIXER, RT5660_PWR_OM_R_BIT, + 0, rt5660_out_r_mix, ARRAY_SIZE(rt5660_out_r_mix)), + + /* Output Volume */ + SND_SOC_DAPM_SWITCH("SPKVOL", RT5660_PWR_VOL, + RT5660_PWR_SV_BIT, 0, &spk_vol_control), + SND_SOC_DAPM_PGA("DAC 1", SND_SOC_NOPM, + 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("LOUTVOL", SND_SOC_NOPM, + 0, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("LOUTVOL L", SND_SOC_NOPM, + RT5660_PWR_LV_L_BIT, 0, &lout_l_vol_control), + SND_SOC_DAPM_SWITCH("LOUTVOL R", SND_SOC_NOPM, + RT5660_PWR_LV_R_BIT, 0, &lout_r_vol_control), + + /* HPO/LOUT/Mono Mixer */ + SND_SOC_DAPM_MIXER("SPO MIX", SND_SOC_NOPM, 0, + 0, rt5660_spo_mix, ARRAY_SIZE(rt5660_spo_mix)), + SND_SOC_DAPM_MIXER("LOUT MIX", SND_SOC_NOPM, 0, 0, + rt5660_lout_mix, ARRAY_SIZE(rt5660_lout_mix)), + SND_SOC_DAPM_SUPPLY("VREF HP", RT5660_GEN_CTRL1, + RT5660_PWR_VREF_HP_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("LOUT amp", 1, RT5660_PWR_ANLG1, + RT5660_PWR_HA_BIT, 0, rt5660_lout_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("SPK amp", 1, RT5660_PWR_DIG1, + RT5660_PWR_CLS_D_BIT, 0, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("LOUTL"), + SND_SOC_DAPM_OUTPUT("LOUTR"), + SND_SOC_DAPM_OUTPUT("SPO"), +}; + +static const struct snd_soc_dapm_route rt5660_dapm_routes[] = { + { "MICBIAS1", NULL, "LDO2" }, + { "MICBIAS2", NULL, "LDO2" }, + + { "BST1", NULL, "IN1P" }, + { "BST1", NULL, "IN1N" }, + { "BST2", NULL, "IN2P" }, + { "BST3", NULL, "IN3P" }, + { "BST3", NULL, "IN3N" }, + + { "RECMIXL", "BST3 Switch", "BST3" }, + { "RECMIXL", "BST2 Switch", "BST2" }, + { "RECMIXL", "BST1 Switch", "BST1" }, + { "RECMIXL", "OUT MIXL Switch", "OUT MIXL" }, + + { "RECMIXR", "BST3 Switch", "BST3" }, + { "RECMIXR", "BST2 Switch", "BST2" }, + { "RECMIXR", "BST1 Switch", "BST1" }, + { "RECMIXR", "OUT MIXR Switch", "OUT MIXR" }, + + { "ADC L", NULL, "RECMIXL" }, + { "ADC L", NULL, "ADC L power" }, + { "ADC L", NULL, "ADC clock" }, + { "ADC R", NULL, "RECMIXR" }, + { "ADC R", NULL, "ADC R power" }, + { "ADC R", NULL, "ADC clock" }, + + {"DMIC L1", NULL, "DMIC CLK"}, + {"DMIC L1", NULL, "DMIC Power"}, + {"DMIC R1", NULL, "DMIC CLK"}, + {"DMIC R1", NULL, "DMIC Power"}, + + { "Sto1 ADC MIXL", "ADC1 Switch", "ADC L" }, + { "Sto1 ADC MIXL", "ADC2 Switch", "DMIC L1" }, + { "Sto1 ADC MIXR", "ADC1 Switch", "ADC R" }, + { "Sto1 ADC MIXR", "ADC2 Switch", "DMIC R1" }, + + { "Stereo1 ADC MIXL", NULL, "Sto1 ADC MIXL" }, + { "Stereo1 ADC MIXL", NULL, "adc stereo1 filter" }, + { "adc stereo1 filter", NULL, "PLL1", rt5660_is_sys_clk_from_pll }, + + { "Stereo1 ADC MIXR", NULL, "Sto1 ADC MIXR" }, + { "Stereo1 ADC MIXR", NULL, "adc stereo1 filter" }, + { "adc stereo1 filter", NULL, "PLL1", rt5660_is_sys_clk_from_pll }, + + { "IF1 ADC", NULL, "Stereo1 ADC MIXL" }, + { "IF1 ADC", NULL, "Stereo1 ADC MIXR" }, + { "IF1 ADC", NULL, "I2S1" }, + + { "IF1 ADC Swap Mux", "L/R", "IF1 ADC" }, + { "IF1 ADC Swap Mux", "R/L", "IF1 ADC" }, + { "IF1 ADC Swap Mux", "L/L", "IF1 ADC" }, + { "IF1 ADC Swap Mux", "R/R", "IF1 ADC" }, + { "AIF1TX", NULL, "IF1 ADC Swap Mux" }, + + { "IF1 DAC", NULL, "AIF1RX" }, + { "IF1 DAC", NULL, "I2S1" }, + + { "IF1 DAC Swap Mux", "L/R", "IF1 DAC" }, + { "IF1 DAC Swap Mux", "R/L", "IF1 DAC" }, + { "IF1 DAC Swap Mux", "L/L", "IF1 DAC" }, + { "IF1 DAC Swap Mux", "R/R", "IF1 DAC" }, + + { "IF1 DAC L", NULL, "IF1 DAC Swap Mux" }, + { "IF1 DAC R", NULL, "IF1 DAC Swap Mux" }, + + { "DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL" }, + { "DAC1 MIXL", "DAC1 Switch", "IF1 DAC L" }, + { "DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR" }, + { "DAC1 MIXR", "DAC1 Switch", "IF1 DAC R" }, + + { "Stereo DAC MIXL", "DAC L1 Switch", "DAC1 MIXL" }, + { "Stereo DAC MIXL", "DAC R1 Switch", "DAC1 MIXR" }, + { "Stereo DAC MIXL", NULL, "dac stereo1 filter" }, + { "dac stereo1 filter", NULL, "PLL1", rt5660_is_sys_clk_from_pll }, + { "Stereo DAC MIXR", "DAC R1 Switch", "DAC1 MIXR" }, + { "Stereo DAC MIXR", "DAC L1 Switch", "DAC1 MIXL" }, + { "Stereo DAC MIXR", NULL, "dac stereo1 filter" }, + { "dac stereo1 filter", NULL, "PLL1", rt5660_is_sys_clk_from_pll }, + + { "DAC L1", NULL, "Stereo DAC MIXL" }, + { "DAC R1", NULL, "Stereo DAC MIXR" }, + + { "SPK MIX", "BST3 Switch", "BST3" }, + { "SPK MIX", "BST1 Switch", "BST1" }, + { "SPK MIX", "DACL Switch", "DAC L1" }, + { "SPK MIX", "DACR Switch", "DAC R1" }, + { "SPK MIX", "OUTMIXL Switch", "OUT MIXL" }, + + { "OUT MIXL", "BST3 Switch", "BST3" }, + { "OUT MIXL", "BST2 Switch", "BST2" }, + { "OUT MIXL", "BST1 Switch", "BST1" }, + { "OUT MIXL", "RECMIXL Switch", "RECMIXL" }, + { "OUT MIXL", "DACR Switch", "DAC R1" }, + { "OUT MIXL", "DACL Switch", "DAC L1" }, + + { "OUT MIXR", "BST2 Switch", "BST2" }, + { "OUT MIXR", "BST1 Switch", "BST1" }, + { "OUT MIXR", "RECMIXR Switch", "RECMIXR" }, + { "OUT MIXR", "DACR Switch", "DAC R1" }, + { "OUT MIXR", "DACL Switch", "DAC L1" }, + + { "SPO MIX", "DACR Switch", "DAC R1" }, + { "SPO MIX", "DACL Switch", "DAC L1" }, + { "SPO MIX", "SPKVOL Switch", "SPKVOL" }, + { "SPO MIX", "BST1 Switch", "BST1" }, + + { "SPKVOL", "Switch", "SPK MIX" }, + { "LOUTVOL L", "Switch", "OUT MIXL" }, + { "LOUTVOL R", "Switch", "OUT MIXR" }, + + { "LOUTVOL", NULL, "LOUTVOL L" }, + { "LOUTVOL", NULL, "LOUTVOL R" }, + + { "DAC 1", NULL, "DAC L1" }, + { "DAC 1", NULL, "DAC R1" }, + + { "LOUT MIX", "DAC Switch", "DAC 1" }, + { "LOUT MIX", "OUTMIX Switch", "LOUTVOL" }, + + { "LOUT amp", NULL, "LOUT MIX" }, + { "LOUT amp", NULL, "VREF HP" }, + { "LOUTL", NULL, "LOUT amp" }, + { "LOUTR", NULL, "LOUT amp" }, + + { "SPK amp", NULL, "SPO MIX" }, + { "SPO", NULL, "SPK amp" }, +}; + +static int rt5660_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0, val_clk, mask_clk; + int pre_div, bclk_ms, frame_size; + + rt5660->lrck[dai->id] = params_rate(params); + pre_div = rl6231_get_clk_info(rt5660->sysclk, rt5660->lrck[dai->id]); + if (pre_div < 0) { + dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n", + rt5660->lrck[dai->id], dai->id); + return -EINVAL; + } + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); + return frame_size; + } + + if (frame_size > 32) + bclk_ms = 1; + else + bclk_ms = 0; + + rt5660->bclk[dai->id] = rt5660->lrck[dai->id] * (32 << bclk_ms); + + dev_dbg(dai->dev, "bclk is %dHz and lrck is %dHz\n", + rt5660->bclk[dai->id], rt5660->lrck[dai->id]); + dev_dbg(dai->dev, "bclk_ms is %d and pre_div is %d for iis %d\n", + bclk_ms, pre_div, dai->id); + + switch (params_width(params)) { + case 16: + break; + case 20: + val_len |= RT5660_I2S_DL_20; + break; + case 24: + val_len |= RT5660_I2S_DL_24; + break; + case 8: + val_len |= RT5660_I2S_DL_8; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5660_AIF1: + mask_clk = RT5660_I2S_BCLK_MS1_MASK | RT5660_I2S_PD1_MASK; + val_clk = bclk_ms << RT5660_I2S_BCLK_MS1_SFT | + pre_div << RT5660_I2S_PD1_SFT; + snd_soc_update_bits(codec, RT5660_I2S1_SDP, RT5660_I2S_DL_MASK, + val_len); + snd_soc_update_bits(codec, RT5660_ADDA_CLK1, mask_clk, val_clk); + break; + + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + + return 0; +} + +static int rt5660_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5660->master[dai->id] = 1; + break; + + case SND_SOC_DAIFMT_CBS_CFS: + reg_val |= RT5660_I2S_MS_S; + rt5660->master[dai->id] = 0; + break; + + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + + case SND_SOC_DAIFMT_IB_NF: + reg_val |= RT5660_I2S_BP_INV; + break; + + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT5660_I2S_DF_LEFT; + break; + + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5660_I2S_DF_PCM_A; + break; + + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5660_I2S_DF_PCM_B; + break; + + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5660_AIF1: + snd_soc_update_bits(codec, RT5660_I2S1_SDP, + RT5660_I2S_MS_MASK | RT5660_I2S_BP_MASK | + RT5660_I2S_DF_MASK, reg_val); + break; + + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + + return 0; +} + +static int rt5660_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + if (freq == rt5660->sysclk && clk_id == rt5660->sysclk_src) + return 0; + + switch (clk_id) { + case RT5660_SCLK_S_MCLK: + reg_val |= RT5660_SCLK_SRC_MCLK; + break; + + case RT5660_SCLK_S_PLL1: + reg_val |= RT5660_SCLK_SRC_PLL1; + break; + + case RT5660_SCLK_S_RCCLK: + reg_val |= RT5660_SCLK_SRC_RCCLK; + break; + + default: + dev_err(codec->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + + snd_soc_update_bits(codec, RT5660_GLB_CLK, RT5660_SCLK_SRC_MASK, + reg_val); + + rt5660->sysclk = freq; + rt5660->sysclk_src = clk_id; + + dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id); + + return 0; +} + +static int rt5660_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + struct rl6231_pll_code pll_code; + int ret; + + if (source == rt5660->pll_src && freq_in == rt5660->pll_in && + freq_out == rt5660->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(codec->dev, "PLL disabled\n"); + + rt5660->pll_in = 0; + rt5660->pll_out = 0; + snd_soc_update_bits(codec, RT5660_GLB_CLK, + RT5660_SCLK_SRC_MASK, RT5660_SCLK_SRC_MCLK); + return 0; + } + + switch (source) { + case RT5660_PLL1_S_MCLK: + snd_soc_update_bits(codec, RT5660_GLB_CLK, + RT5660_PLL1_SRC_MASK, RT5660_PLL1_SRC_MCLK); + break; + + case RT5660_PLL1_S_BCLK: + snd_soc_update_bits(codec, RT5660_GLB_CLK, + RT5660_PLL1_SRC_MASK, RT5660_PLL1_SRC_BCLK1); + break; + + default: + dev_err(codec->dev, "Unknown PLL source %d\n", source); + return -EINVAL; + } + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(codec->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(codec->dev, "bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); + + snd_soc_write(codec, RT5660_PLL_CTRL1, + pll_code.n_code << RT5660_PLL_N_SFT | pll_code.k_code); + snd_soc_write(codec, RT5660_PLL_CTRL2, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5660_PLL_M_SFT | + pll_code.m_bp << RT5660_PLL_M_BP_SFT); + + rt5660->pll_in = freq_in; + rt5660->pll_out = freq_out; + rt5660->pll_src = source; + + return 0; +} + +static int rt5660_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + snd_soc_update_bits(codec, RT5660_GEN_CTRL1, + RT5660_DIG_GATE_CTRL, RT5660_DIG_GATE_CTRL); + + if (IS_ERR(rt5660->mclk)) + break; + + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_ON) { + clk_disable_unprepare(rt5660->mclk); + } else { + ret = clk_prepare_enable(rt5660->mclk); + if (ret) + return ret; + } + break; + + case SND_SOC_BIAS_STANDBY: + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + snd_soc_update_bits(codec, RT5660_PWR_ANLG1, + RT5660_PWR_VREF1 | RT5660_PWR_MB | + RT5660_PWR_BG | RT5660_PWR_VREF2, + RT5660_PWR_VREF1 | RT5660_PWR_MB | + RT5660_PWR_BG | RT5660_PWR_VREF2); + usleep_range(10000, 15000); + snd_soc_update_bits(codec, RT5660_PWR_ANLG1, + RT5660_PWR_FV1 | RT5660_PWR_FV2, + RT5660_PWR_FV1 | RT5660_PWR_FV2); + } + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, RT5660_GEN_CTRL1, + RT5660_DIG_GATE_CTRL, 0); + break; + + default: + break; + } + + return 0; +} + +static int rt5660_probe(struct snd_soc_codec *codec) +{ + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + + rt5660->codec = codec; + + return 0; +} + +static int rt5660_remove(struct snd_soc_codec *codec) +{ + return snd_soc_write(codec, RT5660_RESET, 0); +} + +#ifdef CONFIG_PM +static int rt5660_suspend(struct snd_soc_codec *codec) +{ + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5660->regmap, true); + regcache_mark_dirty(rt5660->regmap); + + return 0; +} + +static int rt5660_resume(struct snd_soc_codec *codec) +{ + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + + if (rt5660->pdata.poweroff_codec_in_suspend) + usleep_range(350000, 400000); + + regcache_cache_only(rt5660->regmap, false); + regcache_sync(rt5660->regmap); + + return 0; +} +#else +#define rt5660_suspend NULL +#define rt5660_resume NULL +#endif + +#define RT5660_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT5660_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static const struct snd_soc_dai_ops rt5660_aif_dai_ops = { + .hw_params = rt5660_hw_params, + .set_fmt = rt5660_set_dai_fmt, + .set_sysclk = rt5660_set_dai_sysclk, + .set_pll = rt5660_set_dai_pll, +}; + +static struct snd_soc_dai_driver rt5660_dai[] = { + { + .name = "rt5660-aif1", + .id = RT5660_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5660_STEREO_RATES, + .formats = RT5660_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5660_STEREO_RATES, + .formats = RT5660_FORMATS, + }, + .ops = &rt5660_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt5660 = { + .probe = rt5660_probe, + .remove = rt5660_remove, + .suspend = rt5660_suspend, + .resume = rt5660_resume, + .set_bias_level = rt5660_set_bias_level, + .idle_bias_off = true, + .component_driver = { + .controls = rt5660_snd_controls, + .num_controls = ARRAY_SIZE(rt5660_snd_controls), + .dapm_widgets = rt5660_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5660_dapm_widgets), + .dapm_routes = rt5660_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5660_dapm_routes), + }, +}; + +static const struct regmap_config rt5660_regmap = { + .reg_bits = 8, + .val_bits = 16, + .use_single_rw = true, + + .max_register = RT5660_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5660_ranges) * + RT5660_PR_SPACING), + .volatile_reg = rt5660_volatile_register, + .readable_reg = rt5660_readable_register, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5660_reg, + .num_reg_defaults = ARRAY_SIZE(rt5660_reg), + .ranges = rt5660_ranges, + .num_ranges = ARRAY_SIZE(rt5660_ranges), +}; + +static const struct i2c_device_id rt5660_i2c_id[] = { + { "rt5660", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5660_i2c_id); + +static const struct of_device_id rt5660_of_match[] = { + { .compatible = "realtek,rt5660", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5660_of_match); + +static const struct acpi_device_id rt5660_acpi_match[] = { + { "10EC5660", 0 }, + { }, +}; +MODULE_DEVICE_TABLE(acpi, rt5660_acpi_match); + +static int rt5660_parse_dt(struct rt5660_priv *rt5660, struct device *dev) +{ + rt5660->pdata.in1_diff = device_property_read_bool(dev, + "realtek,in1-differential"); + rt5660->pdata.in3_diff = device_property_read_bool(dev, + "realtek,in3-differential"); + rt5660->pdata.poweroff_codec_in_suspend = device_property_read_bool(dev, + "realtek,poweroff-in-suspend"); + device_property_read_u32(dev, "realtek,dmic1-data-pin", + &rt5660->pdata.dmic1_data_pin); + + return 0; +} + +static int rt5660_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5660_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5660_priv *rt5660; + int ret; + unsigned int val; + + rt5660 = devm_kzalloc(&i2c->dev, sizeof(struct rt5660_priv), + GFP_KERNEL); + + if (rt5660 == NULL) + return -ENOMEM; + + /* Check if MCLK provided */ + rt5660->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (PTR_ERR(rt5660->mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + + i2c_set_clientdata(i2c, rt5660); + + if (pdata) + rt5660->pdata = *pdata; + else if (i2c->dev.of_node) + rt5660_parse_dt(rt5660, &i2c->dev); + + rt5660->regmap = devm_regmap_init_i2c(i2c, &rt5660_regmap); + if (IS_ERR(rt5660->regmap)) { + ret = PTR_ERR(rt5660->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt5660->regmap, RT5660_VENDOR_ID2, &val); + if (val != RT5660_DEVICE_ID) { + dev_err(&i2c->dev, + "Device with ID register %#x is not rt5660\n", val); + return -ENODEV; + } + + regmap_write(rt5660->regmap, RT5660_RESET, 0); + + ret = regmap_register_patch(rt5660->regmap, rt5660_patch, + ARRAY_SIZE(rt5660_patch)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + + if (rt5660->pdata.dmic1_data_pin) { + regmap_update_bits(rt5660->regmap, RT5660_GPIO_CTRL1, + RT5660_GP1_PIN_MASK, RT5660_GP1_PIN_DMIC1_SCL); + + if (rt5660->pdata.dmic1_data_pin == RT5660_DMIC1_DATA_GPIO2) + regmap_update_bits(rt5660->regmap, RT5660_DMIC_CTRL1, + RT5660_SEL_DMIC_DATA_MASK, + RT5660_SEL_DMIC_DATA_GPIO2); + else if (rt5660->pdata.dmic1_data_pin == RT5660_DMIC1_DATA_IN1P) + regmap_update_bits(rt5660->regmap, RT5660_DMIC_CTRL1, + RT5660_SEL_DMIC_DATA_MASK, + RT5660_SEL_DMIC_DATA_IN1P); + } + + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5660, + rt5660_dai, ARRAY_SIZE(rt5660_dai)); +} + +static int rt5660_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +static struct i2c_driver rt5660_i2c_driver = { + .driver = { + .name = "rt5660", + .acpi_match_table = ACPI_PTR(rt5660_acpi_match), + .of_match_table = of_match_ptr(rt5660_of_match), + }, + .probe = rt5660_i2c_probe, + .remove = rt5660_i2c_remove, + .id_table = rt5660_i2c_id, +}; +module_i2c_driver(rt5660_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5660 driver"); +MODULE_AUTHOR("Oder Chiou "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5660.h b/sound/soc/codecs/rt5660.h new file mode 100644 index 000000000000..6cdb9269ec9e --- /dev/null +++ b/sound/soc/codecs/rt5660.h @@ -0,0 +1,847 @@ +/* + * rt5660.h -- RT5660 ALSA SoC audio driver + * + * Copyright 2016 Realtek Semiconductor Corp. + * Author: Oder Chiou + * + * 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 _RT5660_H +#define _RT5660_H + +#include +#include + +/* Info */ +#define RT5660_RESET 0x00 +#define RT5660_VENDOR_ID 0xfd +#define RT5660_VENDOR_ID1 0xfe +#define RT5660_VENDOR_ID2 0xff +/* I/O - Output */ +#define RT5660_SPK_VOL 0x01 +#define RT5660_LOUT_VOL 0x02 +/* I/O - Input */ +#define RT5660_IN1_IN2 0x0d +#define RT5660_IN3_IN4 0x0e +/* I/O - ADC/DAC/DMIC */ +#define RT5660_DAC1_DIG_VOL 0x19 +#define RT5660_STO1_ADC_DIG_VOL 0x1c +#define RT5660_ADC_BST_VOL1 0x1e +/* Mixer - D-D */ +#define RT5660_STO1_ADC_MIXER 0x27 +#define RT5660_AD_DA_MIXER 0x29 +#define RT5660_STO_DAC_MIXER 0x2a +#define RT5660_DIG_INF1_DATA 0x2f +/* Mixer - ADC */ +#define RT5660_REC_L1_MIXER 0x3b +#define RT5660_REC_L2_MIXER 0x3c +#define RT5660_REC_R1_MIXER 0x3d +#define RT5660_REC_R2_MIXER 0x3e +/* Mixer - DAC */ +#define RT5660_LOUT_MIXER 0x45 +#define RT5660_SPK_MIXER 0x46 +#define RT5660_SPO_MIXER 0x48 +#define RT5660_SPO_CLSD_RATIO 0x4a +#define RT5660_OUT_L_GAIN1 0x4d +#define RT5660_OUT_L_GAIN2 0x4e +#define RT5660_OUT_L1_MIXER 0x4f +#define RT5660_OUT_R_GAIN1 0x50 +#define RT5660_OUT_R_GAIN2 0x51 +#define RT5660_OUT_R1_MIXER 0x52 +/* Power */ +#define RT5660_PWR_DIG1 0x61 +#define RT5660_PWR_DIG2 0x62 +#define RT5660_PWR_ANLG1 0x63 +#define RT5660_PWR_ANLG2 0x64 +#define RT5660_PWR_MIXER 0x65 +#define RT5660_PWR_VOL 0x66 +/* Private Register Control */ +#define RT5660_PRIV_INDEX 0x6a +#define RT5660_PRIV_DATA 0x6c +/* Format - ADC/DAC */ +#define RT5660_I2S1_SDP 0x70 +#define RT5660_ADDA_CLK1 0x73 +#define RT5660_ADDA_CLK2 0x74 +#define RT5660_DMIC_CTRL1 0x75 +/* Function - Analog */ +#define RT5660_GLB_CLK 0x80 +#define RT5660_PLL_CTRL1 0x81 +#define RT5660_PLL_CTRL2 0x82 +#define RT5660_CLSD_AMP_OC_CTRL 0x8c +#define RT5660_CLSD_AMP_CTRL 0x8d +#define RT5660_LOUT_AMP_CTRL 0x8e +#define RT5660_SPK_AMP_SPKVDD 0x92 +#define RT5660_MICBIAS 0x93 +#define RT5660_CLSD_OUT_CTRL1 0xa1 +#define RT5660_CLSD_OUT_CTRL2 0xa2 +#define RT5660_DIPOLE_MIC_CTRL1 0xa3 +#define RT5660_DIPOLE_MIC_CTRL2 0xa4 +#define RT5660_DIPOLE_MIC_CTRL3 0xa5 +#define RT5660_DIPOLE_MIC_CTRL4 0xa6 +#define RT5660_DIPOLE_MIC_CTRL5 0xa7 +#define RT5660_DIPOLE_MIC_CTRL6 0xa8 +#define RT5660_DIPOLE_MIC_CTRL7 0xa9 +#define RT5660_DIPOLE_MIC_CTRL8 0xaa +#define RT5660_DIPOLE_MIC_CTRL9 0xab +#define RT5660_DIPOLE_MIC_CTRL10 0xac +#define RT5660_DIPOLE_MIC_CTRL11 0xad +#define RT5660_DIPOLE_MIC_CTRL12 0xae +/* Function - Digital */ +#define RT5660_EQ_CTRL1 0xb0 +#define RT5660_EQ_CTRL2 0xb1 +#define RT5660_DRC_AGC_CTRL1 0xb3 +#define RT5660_DRC_AGC_CTRL2 0xb4 +#define RT5660_DRC_AGC_CTRL3 0xb5 +#define RT5660_DRC_AGC_CTRL4 0xb6 +#define RT5660_DRC_AGC_CTRL5 0xb7 +#define RT5660_JD_CTRL 0xbb +#define RT5660_IRQ_CTRL1 0xbd +#define RT5660_IRQ_CTRL2 0xbe +#define RT5660_INT_IRQ_ST 0xbf +#define RT5660_GPIO_CTRL1 0xc0 +#define RT5660_GPIO_CTRL2 0xc2 +#define RT5660_WIND_FILTER_CTRL1 0xd3 +#define RT5660_SV_ZCD1 0xd9 +#define RT5660_SV_ZCD2 0xda +#define RT5660_DRC1_LM_CTRL1 0xe0 +#define RT5660_DRC1_LM_CTRL2 0xe1 +#define RT5660_DRC2_LM_CTRL1 0xe2 +#define RT5660_DRC2_LM_CTRL2 0xe3 +#define RT5660_MULTI_DRC_CTRL 0xe4 +#define RT5660_DRC2_CTRL1 0xe5 +#define RT5660_DRC2_CTRL2 0xe6 +#define RT5660_DRC2_CTRL3 0xe7 +#define RT5660_DRC2_CTRL4 0xe8 +#define RT5660_DRC2_CTRL5 0xe9 +#define RT5660_ALC_PGA_CTRL1 0xea +#define RT5660_ALC_PGA_CTRL2 0xeb +#define RT5660_ALC_PGA_CTRL3 0xec +#define RT5660_ALC_PGA_CTRL4 0xed +#define RT5660_ALC_PGA_CTRL5 0xee +#define RT5660_ALC_PGA_CTRL6 0xef +#define RT5660_ALC_PGA_CTRL7 0xf0 + +/* General Control */ +#define RT5660_GEN_CTRL1 0xfa +#define RT5660_GEN_CTRL2 0xfb +#define RT5660_GEN_CTRL3 0xfc + +/* Index of Codec Private Register definition */ +#define RT5660_CHOP_DAC_ADC 0x3d + +/* Global Definition */ +#define RT5660_L_MUTE (0x1 << 15) +#define RT5660_L_MUTE_SFT 15 +#define RT5660_VOL_L_MUTE (0x1 << 14) +#define RT5660_VOL_L_SFT 14 +#define RT5660_R_MUTE (0x1 << 7) +#define RT5660_R_MUTE_SFT 7 +#define RT5660_VOL_R_MUTE (0x1 << 6) +#define RT5660_VOL_R_SFT 6 +#define RT5660_L_VOL_MASK (0x3f << 8) +#define RT5660_L_VOL_SFT 8 +#define RT5660_R_VOL_MASK (0x3f) +#define RT5660_R_VOL_SFT 0 + +/* IN1 and IN2 Control (0x0d) */ +#define RT5660_IN_DF1 (0x1 << 15) +#define RT5660_IN_SFT1 15 +#define RT5660_BST_MASK1 (0x7f << 8) +#define RT5660_BST_SFT1 8 +#define RT5660_IN_DF2 (0x1 << 7) +#define RT5660_IN_SFT2 7 +#define RT5660_BST_MASK2 (0x7f << 0) +#define RT5660_BST_SFT2 0 + +/* IN3 and IN4 Control (0x0e) */ +#define RT5660_IN_DF3 (0x1 << 15) +#define RT5660_IN_SFT3 15 +#define RT5660_BST_MASK3 (0x7f << 8) +#define RT5660_BST_SFT3 8 +#define RT5660_IN_DF4 (0x1 << 7) +#define RT5660_IN_SFT4 7 +#define RT5660_BST_MASK4 (0x7f << 0) +#define RT5660_BST_SFT4 0 + +/* DAC1 Digital Volume (0x19) */ +#define RT5660_DAC_L1_VOL_MASK (0x7f << 9) +#define RT5660_DAC_L1_VOL_SFT 9 +#define RT5660_DAC_R1_VOL_MASK (0x7f << 1) +#define RT5660_DAC_R1_VOL_SFT 1 + +/* ADC Digital Volume Control (0x1c) */ +#define RT5660_ADC_L_VOL_MASK (0x3f << 9) +#define RT5660_ADC_L_VOL_SFT 9 +#define RT5660_ADC_R_VOL_MASK (0x3f << 1) +#define RT5660_ADC_R_VOL_SFT 1 + +/* ADC Boost Volume Control (0x1e) */ +#define RT5660_STO1_ADC_L_BST_MASK (0x3 << 14) +#define RT5660_STO1_ADC_L_BST_SFT 14 +#define RT5660_STO1_ADC_R_BST_MASK (0x3 << 12) +#define RT5660_STO1_ADC_R_BST_SFT 12 + +/* Stereo ADC Mixer Control (0x27) */ +#define RT5660_M_ADC_L1 (0x1 << 14) +#define RT5660_M_ADC_L1_SFT 14 +#define RT5660_M_ADC_L2 (0x1 << 13) +#define RT5660_M_ADC_L2_SFT 13 +#define RT5660_M_ADC_R1 (0x1 << 6) +#define RT5660_M_ADC_R1_SFT 6 +#define RT5660_M_ADC_R2 (0x1 << 5) +#define RT5660_M_ADC_R2_SFT 5 + +/* ADC Mixer to DAC Mixer Control (0x29) */ +#define RT5660_M_ADCMIX_L (0x1 << 15) +#define RT5660_M_ADCMIX_L_SFT 15 +#define RT5660_M_DAC1_L (0x1 << 14) +#define RT5660_M_DAC1_L_SFT 14 +#define RT5660_M_ADCMIX_R (0x1 << 7) +#define RT5660_M_ADCMIX_R_SFT 7 +#define RT5660_M_DAC1_R (0x1 << 6) +#define RT5660_M_DAC1_R_SFT 6 + +/* Stereo DAC Mixer Control (0x2a) */ +#define RT5660_M_DAC_L1 (0x1 << 14) +#define RT5660_M_DAC_L1_SFT 14 +#define RT5660_DAC_L1_STO_L_VOL_MASK (0x1 << 13) +#define RT5660_DAC_L1_STO_L_VOL_SFT 13 +#define RT5660_M_DAC_R1_STO_L (0x1 << 9) +#define RT5660_M_DAC_R1_STO_L_SFT 9 +#define RT5660_DAC_R1_STO_L_VOL_MASK (0x1 << 8) +#define RT5660_DAC_R1_STO_L_VOL_SFT 8 +#define RT5660_M_DAC_R1 (0x1 << 6) +#define RT5660_M_DAC_R1_SFT 6 +#define RT5660_DAC_R1_STO_R_VOL_MASK (0x1 << 5) +#define RT5660_DAC_R1_STO_R_VOL_SFT 5 +#define RT5660_M_DAC_L1_STO_R (0x1 << 1) +#define RT5660_M_DAC_L1_STO_R_SFT 1 +#define RT5660_DAC_L1_STO_R_VOL_MASK (0x1) +#define RT5660_DAC_L1_STO_R_VOL_SFT 0 + +/* Digital Interface Data Control (0x2f) */ +#define RT5660_IF1_DAC_IN_SEL (0x3 << 14) +#define RT5660_IF1_DAC_IN_SFT 14 +#define RT5660_IF1_ADC_IN_SEL (0x3 << 12) +#define RT5660_IF1_ADC_IN_SFT 12 + +/* REC Left Mixer Control 1 (0x3b) */ +#define RT5660_G_BST3_RM_L_MASK (0x7 << 4) +#define RT5660_G_BST3_RM_L_SFT 4 +#define RT5660_G_BST2_RM_L_MASK (0x7 << 1) +#define RT5660_G_BST2_RM_L_SFT 1 + +/* REC Left Mixer Control 2 (0x3c) */ +#define RT5660_G_BST1_RM_L_MASK (0x7 << 13) +#define RT5660_G_BST1_RM_L_SFT 13 +#define RT5660_G_OM_L_RM_L_MASK (0x7 << 10) +#define RT5660_G_OM_L_RM_L_SFT 10 +#define RT5660_M_BST3_RM_L (0x1 << 3) +#define RT5660_M_BST3_RM_L_SFT 3 +#define RT5660_M_BST2_RM_L (0x1 << 2) +#define RT5660_M_BST2_RM_L_SFT 2 +#define RT5660_M_BST1_RM_L (0x1 << 1) +#define RT5660_M_BST1_RM_L_SFT 1 +#define RT5660_M_OM_L_RM_L (0x1) +#define RT5660_M_OM_L_RM_L_SFT 0 + +/* REC Right Mixer Control 1 (0x3d) */ +#define RT5660_G_BST3_RM_R_MASK (0x7 << 4) +#define RT5660_G_BST3_RM_R_SFT 4 +#define RT5660_G_BST2_RM_R_MASK (0x7 << 1) +#define RT5660_G_BST2_RM_R_SFT 1 + +/* REC Right Mixer Control 2 (0x3e) */ +#define RT5660_G_BST1_RM_R_MASK (0x7 << 13) +#define RT5660_G_BST1_RM_R_SFT 13 +#define RT5660_G_OM_R_RM_R_MASK (0x7 << 10) +#define RT5660_G_OM_R_RM_R_SFT 10 +#define RT5660_M_BST3_RM_R (0x1 << 3) +#define RT5660_M_BST3_RM_R_SFT 3 +#define RT5660_M_BST2_RM_R (0x1 << 2) +#define RT5660_M_BST2_RM_R_SFT 2 +#define RT5660_M_BST1_RM_R (0x1 << 1) +#define RT5660_M_BST1_RM_R_SFT 1 +#define RT5660_M_OM_R_RM_R (0x1) +#define RT5660_M_OM_R_RM_R_SFT 0 + +/* LOUTMIX Control (0x45) */ +#define RT5660_M_DAC1_LM (0x1 << 14) +#define RT5660_M_DAC1_LM_SFT 14 +#define RT5660_M_LOVOL_M (0x1 << 13) +#define RT5660_M_LOVOL_LM_SFT 13 + +/* SPK Mixer Control (0x46) */ +#define RT5660_G_BST3_SM_MASK (0x3 << 14) +#define RT5660_G_BST3_SM_SFT 14 +#define RT5660_G_BST1_SM_MASK (0x3 << 12) +#define RT5660_G_BST1_SM_SFT 12 +#define RT5660_G_DACl_SM_MASK (0x3 << 10) +#define RT5660_G_DACl_SM_SFT 10 +#define RT5660_G_DACR_SM_MASK (0x3 << 8) +#define RT5660_G_DACR_SM_SFT 8 +#define RT5660_G_OM_L_SM_MASK (0x3 << 6) +#define RT5660_G_OM_L_SM_SFT 6 +#define RT5660_M_DACR_SM (0x1 << 5) +#define RT5660_M_DACR_SM_SFT 5 +#define RT5660_M_BST1_SM (0x1 << 4) +#define RT5660_M_BST1_SM_SFT 4 +#define RT5660_M_BST3_SM (0x1 << 3) +#define RT5660_M_BST3_SM_SFT 3 +#define RT5660_M_DACL_SM (0x1 << 2) +#define RT5660_M_DACL_SM_SFT 2 +#define RT5660_M_OM_L_SM (0x1 << 1) +#define RT5660_M_OM_L_SM_SFT 1 + +/* SPOMIX Control (0x48) */ +#define RT5660_M_DAC_R_SPM (0x1 << 14) +#define RT5660_M_DAC_R_SPM_SFT 14 +#define RT5660_M_DAC_L_SPM (0x1 << 13) +#define RT5660_M_DAC_L_SPM_SFT 13 +#define RT5660_M_SV_SPM (0x1 << 12) +#define RT5660_M_SV_SPM_SFT 12 +#define RT5660_M_BST1_SPM (0x1 << 11) +#define RT5660_M_BST1_SPM_SFT 11 + +/* Output Left Mixer Control 1 (0x4d) */ +#define RT5660_G_BST3_OM_L_MASK (0x7 << 13) +#define RT5660_G_BST3_OM_L_SFT 13 +#define RT5660_G_BST2_OM_L_MASK (0x7 << 10) +#define RT5660_G_BST2_OM_L_SFT 10 +#define RT5660_G_BST1_OM_L_MASK (0x7 << 7) +#define RT5660_G_BST1_OM_L_SFT 7 +#define RT5660_G_RM_L_OM_L_MASK (0x7 << 1) +#define RT5660_G_RM_L_OM_L_SFT 1 + +/* Output Left Mixer Control 2 (0x4e) */ +#define RT5660_G_DAC_R1_OM_L_MASK (0x7 << 10) +#define RT5660_G_DAC_R1_OM_L_SFT 10 +#define RT5660_G_DAC_L1_OM_L_MASK (0x7 << 7) +#define RT5660_G_DAC_L1_OM_L_SFT 7 + +/* Output Left Mixer Control 3 (0x4f) */ +#define RT5660_M_BST3_OM_L (0x1 << 5) +#define RT5660_M_BST3_OM_L_SFT 5 +#define RT5660_M_BST2_OM_L (0x1 << 4) +#define RT5660_M_BST2_OM_L_SFT 4 +#define RT5660_M_BST1_OM_L (0x1 << 3) +#define RT5660_M_BST1_OM_L_SFT 3 +#define RT5660_M_RM_L_OM_L (0x1 << 2) +#define RT5660_M_RM_L_OM_L_SFT 2 +#define RT5660_M_DAC_R_OM_L (0x1 << 1) +#define RT5660_M_DAC_R_OM_L_SFT 1 +#define RT5660_M_DAC_L_OM_L (0x1) +#define RT5660_M_DAC_L_OM_L_SFT 0 + +/* Output Right Mixer Control 1 (0x50) */ +#define RT5660_G_BST2_OM_R_MASK (0x7 << 10) +#define RT5660_G_BST2_OM_R_SFT 10 +#define RT5660_G_BST1_OM_R_MASK (0x7 << 7) +#define RT5660_G_BST1_OM_R_SFT 7 +#define RT5660_G_RM_R_OM_R_MASK (0x7 << 1) +#define RT5660_G_RM_R_OM_R_SFT 1 + +/* Output Right Mixer Control 2 (0x51) */ +#define RT5660_G_DAC_L_OM_R_MASK (0x7 << 10) +#define RT5660_G_DAC_L_OM_R_SFT 10 +#define RT5660_G_DAC_R_OM_R_MASK (0x7 << 7) +#define RT5660_G_DAC_R_OM_R_SFT 7 + +/* Output Right Mixer Control 3 (0x52) */ +#define RT5660_M_BST2_OM_R (0x1 << 4) +#define RT5660_M_BST2_OM_R_SFT 4 +#define RT5660_M_BST1_OM_R (0x1 << 3) +#define RT5660_M_BST1_OM_R_SFT 3 +#define RT5660_M_RM_R_OM_R (0x1 << 2) +#define RT5660_M_RM_R_OM_R_SFT 2 +#define RT5660_M_DAC_L_OM_R (0x1 << 1) +#define RT5660_M_DAC_L_OM_R_SFT 1 +#define RT5660_M_DAC_R_OM_R (0x1) +#define RT5660_M_DAC_R_OM_R_SFT 0 + +/* Power Management for Digital 1 (0x61) */ +#define RT5660_PWR_I2S1 (0x1 << 15) +#define RT5660_PWR_I2S1_BIT 15 +#define RT5660_PWR_DAC_L1 (0x1 << 12) +#define RT5660_PWR_DAC_L1_BIT 12 +#define RT5660_PWR_DAC_R1 (0x1 << 11) +#define RT5660_PWR_DAC_R1_BIT 11 +#define RT5660_PWR_ADC_L (0x1 << 2) +#define RT5660_PWR_ADC_L_BIT 2 +#define RT5660_PWR_ADC_R (0x1 << 1) +#define RT5660_PWR_ADC_R_BIT 1 +#define RT5660_PWR_CLS_D (0x1) +#define RT5660_PWR_CLS_D_BIT 0 + +/* Power Management for Digital 2 (0x62) */ +#define RT5660_PWR_ADC_S1F (0x1 << 15) +#define RT5660_PWR_ADC_S1F_BIT 15 +#define RT5660_PWR_DAC_S1F (0x1 << 11) +#define RT5660_PWR_DAC_S1F_BIT 11 + +/* Power Management for Analog 1 (0x63) */ +#define RT5660_PWR_VREF1 (0x1 << 15) +#define RT5660_PWR_VREF1_BIT 15 +#define RT5660_PWR_FV1 (0x1 << 14) +#define RT5660_PWR_FV1_BIT 14 +#define RT5660_PWR_MB (0x1 << 13) +#define RT5660_PWR_MB_BIT 13 +#define RT5660_PWR_BG (0x1 << 11) +#define RT5660_PWR_BG_BIT 11 +#define RT5660_PWR_HP_L (0x1 << 7) +#define RT5660_PWR_HP_L_BIT 7 +#define RT5660_PWR_HP_R (0x1 << 6) +#define RT5660_PWR_HP_R_BIT 6 +#define RT5660_PWR_HA (0x1 << 5) +#define RT5660_PWR_HA_BIT 5 +#define RT5660_PWR_VREF2 (0x1 << 4) +#define RT5660_PWR_VREF2_BIT 4 +#define RT5660_PWR_FV2 (0x1 << 3) +#define RT5660_PWR_FV2_BIT 3 +#define RT5660_PWR_LDO2 (0x1 << 2) +#define RT5660_PWR_LDO2_BIT 2 + +/* Power Management for Analog 2 (0x64) */ +#define RT5660_PWR_BST1 (0x1 << 15) +#define RT5660_PWR_BST1_BIT 15 +#define RT5660_PWR_BST2 (0x1 << 14) +#define RT5660_PWR_BST2_BIT 14 +#define RT5660_PWR_BST3 (0x1 << 13) +#define RT5660_PWR_BST3_BIT 13 +#define RT5660_PWR_MB1 (0x1 << 11) +#define RT5660_PWR_MB1_BIT 11 +#define RT5660_PWR_MB2 (0x1 << 10) +#define RT5660_PWR_MB2_BIT 10 +#define RT5660_PWR_PLL (0x1 << 9) +#define RT5660_PWR_PLL_BIT 9 + +/* Power Management for Mixer (0x65) */ +#define RT5660_PWR_OM_L (0x1 << 15) +#define RT5660_PWR_OM_L_BIT 15 +#define RT5660_PWR_OM_R (0x1 << 14) +#define RT5660_PWR_OM_R_BIT 14 +#define RT5660_PWR_SM (0x1 << 13) +#define RT5660_PWR_SM_BIT 13 +#define RT5660_PWR_RM_L (0x1 << 11) +#define RT5660_PWR_RM_L_BIT 11 +#define RT5660_PWR_RM_R (0x1 << 10) +#define RT5660_PWR_RM_R_BIT 10 + +/* Power Management for Volume (0x66) */ +#define RT5660_PWR_SV (0x1 << 15) +#define RT5660_PWR_SV_BIT 15 +#define RT5660_PWR_LV_L (0x1 << 11) +#define RT5660_PWR_LV_L_BIT 11 +#define RT5660_PWR_LV_R (0x1 << 10) +#define RT5660_PWR_LV_R_BIT 10 + +/* I2S1 Audio Serial Data Port Control (0x70) */ +#define RT5660_I2S_MS_MASK (0x1 << 15) +#define RT5660_I2S_MS_SFT 15 +#define RT5660_I2S_MS_M (0x0 << 15) +#define RT5660_I2S_MS_S (0x1 << 15) +#define RT5660_I2S_O_CP_MASK (0x3 << 10) +#define RT5660_I2S_O_CP_SFT 10 +#define RT5660_I2S_O_CP_OFF (0x0 << 10) +#define RT5660_I2S_O_CP_U_LAW (0x1 << 10) +#define RT5660_I2S_O_CP_A_LAW (0x2 << 10) +#define RT5660_I2S_I_CP_MASK (0x3 << 8) +#define RT5660_I2S_I_CP_SFT 8 +#define RT5660_I2S_I_CP_OFF (0x0 << 8) +#define RT5660_I2S_I_CP_U_LAW (0x1 << 8) +#define RT5660_I2S_I_CP_A_LAW (0x2 << 8) +#define RT5660_I2S_BP_MASK (0x1 << 7) +#define RT5660_I2S_BP_SFT 7 +#define RT5660_I2S_BP_NOR (0x0 << 7) +#define RT5660_I2S_BP_INV (0x1 << 7) +#define RT5660_I2S_DL_MASK (0x3 << 2) +#define RT5660_I2S_DL_SFT 2 +#define RT5660_I2S_DL_16 (0x0 << 2) +#define RT5660_I2S_DL_20 (0x1 << 2) +#define RT5660_I2S_DL_24 (0x2 << 2) +#define RT5660_I2S_DL_8 (0x3 << 2) +#define RT5660_I2S_DF_MASK (0x3) +#define RT5660_I2S_DF_SFT 0 +#define RT5660_I2S_DF_I2S (0x0) +#define RT5660_I2S_DF_LEFT (0x1) +#define RT5660_I2S_DF_PCM_A (0x2) +#define RT5660_I2S_DF_PCM_B (0x3) + +/* ADC/DAC Clock Control 1 (0x73) */ +#define RT5660_I2S_BCLK_MS1_MASK (0x1 << 15) +#define RT5660_I2S_BCLK_MS1_SFT 15 +#define RT5660_I2S_BCLK_MS1_32 (0x0 << 15) +#define RT5660_I2S_BCLK_MS1_64 (0x1 << 15) +#define RT5660_I2S_PD1_MASK (0x7 << 12) +#define RT5660_I2S_PD1_SFT 12 +#define RT5660_I2S_PD1_1 (0x0 << 12) +#define RT5660_I2S_PD1_2 (0x1 << 12) +#define RT5660_I2S_PD1_3 (0x2 << 12) +#define RT5660_I2S_PD1_4 (0x3 << 12) +#define RT5660_I2S_PD1_6 (0x4 << 12) +#define RT5660_I2S_PD1_8 (0x5 << 12) +#define RT5660_I2S_PD1_12 (0x6 << 12) +#define RT5660_I2S_PD1_16 (0x7 << 12) +#define RT5660_DAC_OSR_MASK (0x3 << 2) +#define RT5660_DAC_OSR_SFT 2 +#define RT5660_DAC_OSR_128 (0x0 << 2) +#define RT5660_DAC_OSR_64 (0x1 << 2) +#define RT5660_DAC_OSR_32 (0x2 << 2) +#define RT5660_DAC_OSR_16 (0x3 << 2) +#define RT5660_ADC_OSR_MASK (0x3) +#define RT5660_ADC_OSR_SFT 0 +#define RT5660_ADC_OSR_128 (0x0) +#define RT5660_ADC_OSR_64 (0x1) +#define RT5660_ADC_OSR_32 (0x2) +#define RT5660_ADC_OSR_16 (0x3) + +/* ADC/DAC Clock Control 2 (0x74) */ +#define RT5660_RESET_ADF (0x1 << 13) +#define RT5660_RESET_ADF_SFT 13 +#define RT5660_RESET_DAF (0x1 << 12) +#define RT5660_RESET_DAF_SFT 12 +#define RT5660_DAHPF_EN (0x1 << 11) +#define RT5660_DAHPF_EN_SFT 11 +#define RT5660_ADHPF_EN (0x1 << 10) +#define RT5660_ADHPF_EN_SFT 10 + +/* Digital Microphone Control (0x75) */ +#define RT5660_DMIC_1_EN_MASK (0x1 << 15) +#define RT5660_DMIC_1_EN_SFT 15 +#define RT5660_DMIC_1_DIS (0x0 << 15) +#define RT5660_DMIC_1_EN (0x1 << 15) +#define RT5660_DMIC_1L_LH_MASK (0x1 << 13) +#define RT5660_DMIC_1L_LH_SFT 13 +#define RT5660_DMIC_1L_LH_RISING (0x0 << 13) +#define RT5660_DMIC_1L_LH_FALLING (0x1 << 13) +#define RT5660_DMIC_1R_LH_MASK (0x1 << 12) +#define RT5660_DMIC_1R_LH_SFT 12 +#define RT5660_DMIC_1R_LH_RISING (0x0 << 12) +#define RT5660_DMIC_1R_LH_FALLING (0x1 << 12) +#define RT5660_SEL_DMIC_DATA_MASK (0x1 << 11) +#define RT5660_SEL_DMIC_DATA_SFT 11 +#define RT5660_SEL_DMIC_DATA_GPIO2 (0x0 << 11) +#define RT5660_SEL_DMIC_DATA_IN1P (0x1 << 11) +#define RT5660_DMIC_CLK_MASK (0x7 << 5) +#define RT5660_DMIC_CLK_SFT 5 + +/* Global Clock Control (0x80) */ +#define RT5660_SCLK_SRC_MASK (0x3 << 14) +#define RT5660_SCLK_SRC_SFT 14 +#define RT5660_SCLK_SRC_MCLK (0x0 << 14) +#define RT5660_SCLK_SRC_PLL1 (0x1 << 14) +#define RT5660_SCLK_SRC_RCCLK (0x2 << 14) +#define RT5660_PLL1_SRC_MASK (0x3 << 12) +#define RT5660_PLL1_SRC_SFT 12 +#define RT5660_PLL1_SRC_MCLK (0x0 << 12) +#define RT5660_PLL1_SRC_BCLK1 (0x1 << 12) +#define RT5660_PLL1_SRC_RCCLK (0x2 << 12) +#define RT5660_PLL1_PD_MASK (0x1 << 3) +#define RT5660_PLL1_PD_SFT 3 +#define RT5660_PLL1_PD_1 (0x0 << 3) +#define RT5660_PLL1_PD_2 (0x1 << 3) + +#define RT5660_PLL_INP_MAX 40000000 +#define RT5660_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x81) */ +#define RT5660_PLL_N_MAX 0x1ff +#define RT5660_PLL_N_MASK (RT5660_PLL_N_MAX << 7) +#define RT5660_PLL_N_SFT 7 +#define RT5660_PLL_K_MAX 0x1f +#define RT5660_PLL_K_MASK (RT5660_PLL_K_MAX) +#define RT5660_PLL_K_SFT 0 + +/* PLL M/N/K Code Control 2 (0x82) */ +#define RT5660_PLL_M_MAX 0xf +#define RT5660_PLL_M_MASK (RT5660_PLL_M_MAX << 12) +#define RT5660_PLL_M_SFT 12 +#define RT5660_PLL_M_BP (0x1 << 11) +#define RT5660_PLL_M_BP_SFT 11 + +/* Class D Over Current Control (0x8c) */ +#define RT5660_CLSD_OC_MASK (0x1 << 9) +#define RT5660_CLSD_OC_SFT 9 +#define RT5660_CLSD_OC_PU (0x0 << 9) +#define RT5660_CLSD_OC_PD (0x1 << 9) +#define RT5660_AUTO_PD_MASK (0x1 << 8) +#define RT5660_AUTO_PD_SFT 8 +#define RT5660_AUTO_PD_DIS (0x0 << 8) +#define RT5660_AUTO_PD_EN (0x1 << 8) +#define RT5660_CLSD_OC_TH_MASK (0x3f) +#define RT5660_CLSD_OC_TH_SFT 0 + +/* Class D Output Control (0x8d) */ +#define RT5660_CLSD_RATIO_MASK (0xf << 12) +#define RT5660_CLSD_RATIO_SFT 12 + +/* Lout Amp Control 1 (0x8e) */ +#define RT5660_LOUT_CO_MASK (0x1 << 4) +#define RT5660_LOUT_CO_SFT 4 +#define RT5660_LOUT_CO_DIS (0x0 << 4) +#define RT5660_LOUT_CO_EN (0x1 << 4) +#define RT5660_LOUT_CB_MASK (0x1) +#define RT5660_LOUT_CB_SFT 0 +#define RT5660_LOUT_CB_PD (0x0) +#define RT5660_LOUT_CB_PU (0x1) + +/* SPKVDD detection control (0x92) */ +#define RT5660_SPKVDD_DET_MASK (0x1 << 15) +#define RT5660_SPKVDD_DET_SFT 15 +#define RT5660_SPKVDD_DET_DIS (0x0 << 15) +#define RT5660_SPKVDD_DET_EN (0x1 << 15) +#define RT5660_SPK_AG_MASK (0x1 << 14) +#define RT5660_SPK_AG_SFT 14 +#define RT5660_SPK_AG_DIS (0x0 << 14) +#define RT5660_SPK_AG_EN (0x1 << 14) + +/* Micbias Control (0x93) */ +#define RT5660_MIC1_BS_MASK (0x1 << 15) +#define RT5660_MIC1_BS_SFT 15 +#define RT5660_MIC1_BS_9AV (0x0 << 15) +#define RT5660_MIC1_BS_75AV (0x1 << 15) +#define RT5660_MIC2_BS_MASK (0x1 << 14) +#define RT5660_MIC2_BS_SFT 14 +#define RT5660_MIC2_BS_9AV (0x0 << 14) +#define RT5660_MIC2_BS_75AV (0x1 << 14) +#define RT5660_MIC1_OVCD_MASK (0x1 << 11) +#define RT5660_MIC1_OVCD_SFT 11 +#define RT5660_MIC1_OVCD_DIS (0x0 << 11) +#define RT5660_MIC1_OVCD_EN (0x1 << 11) +#define RT5660_MIC1_OVTH_MASK (0x3 << 9) +#define RT5660_MIC1_OVTH_SFT 9 +#define RT5660_MIC1_OVTH_600UA (0x0 << 9) +#define RT5660_MIC1_OVTH_1500UA (0x1 << 9) +#define RT5660_MIC1_OVTH_2000UA (0x2 << 9) +#define RT5660_MIC2_OVCD_MASK (0x1 << 8) +#define RT5660_MIC2_OVCD_SFT 8 +#define RT5660_MIC2_OVCD_DIS (0x0 << 8) +#define RT5660_MIC2_OVCD_EN (0x1 << 8) +#define RT5660_MIC2_OVTH_MASK (0x3 << 6) +#define RT5660_MIC2_OVTH_SFT 6 +#define RT5660_MIC2_OVTH_600UA (0x0 << 6) +#define RT5660_MIC2_OVTH_1500UA (0x1 << 6) +#define RT5660_MIC2_OVTH_2000UA (0x2 << 6) +#define RT5660_PWR_CLK25M_MASK (0x1 << 4) +#define RT5660_PWR_CLK25M_SFT 4 +#define RT5660_PWR_CLK25M_PD (0x0 << 4) +#define RT5660_PWR_CLK25M_PU (0x1 << 4) + +/* EQ Control 1 (0xb0) */ +#define RT5660_EQ_SRC_MASK (0x1 << 15) +#define RT5660_EQ_SRC_SFT 15 +#define RT5660_EQ_SRC_DAC (0x0 << 15) +#define RT5660_EQ_SRC_ADC (0x1 << 15) +#define RT5660_EQ_UPD (0x1 << 14) +#define RT5660_EQ_UPD_BIT 14 + +/* Jack Detect Control (0xbb) */ +#define RT5660_JD_MASK (0x3 << 14) +#define RT5660_JD_SFT 14 +#define RT5660_JD_DIS (0x0 << 14) +#define RT5660_JD_GPIO1 (0x1 << 14) +#define RT5660_JD_GPIO2 (0x2 << 14) +#define RT5660_JD_LOUT_MASK (0x1 << 11) +#define RT5660_JD_LOUT_SFT 11 +#define RT5660_JD_LOUT_DIS (0x0 << 11) +#define RT5660_JD_LOUT_EN (0x1 << 11) +#define RT5660_JD_LOUT_TRG_MASK (0x1 << 10) +#define RT5660_JD_LOUT_TRG_SFT 10 +#define RT5660_JD_LOUT_TRG_LO (0x0 << 10) +#define RT5660_JD_LOUT_TRG_HI (0x1 << 10) +#define RT5660_JD_SPO_MASK (0x1 << 9) +#define RT5660_JD_SPO_SFT 9 +#define RT5660_JD_SPO_DIS (0x0 << 9) +#define RT5660_JD_SPO_EN (0x1 << 9) +#define RT5660_JD_SPO_TRG_MASK (0x1 << 8) +#define RT5660_JD_SPO_TRG_SFT 8 +#define RT5660_JD_SPO_TRG_LO (0x0 << 8) +#define RT5660_JD_SPO_TRG_HI (0x1 << 8) + +/* IRQ Control 1 (0xbd) */ +#define RT5660_IRQ_JD_MASK (0x1 << 15) +#define RT5660_IRQ_JD_SFT 15 +#define RT5660_IRQ_JD_BP (0x0 << 15) +#define RT5660_IRQ_JD_NOR (0x1 << 15) +#define RT5660_IRQ_OT_MASK (0x1 << 14) +#define RT5660_IRQ_OT_SFT 14 +#define RT5660_IRQ_OT_BP (0x0 << 14) +#define RT5660_IRQ_OT_NOR (0x1 << 14) +#define RT5660_JD_STKY_MASK (0x1 << 13) +#define RT5660_JD_STKY_SFT 13 +#define RT5660_JD_STKY_DIS (0x0 << 13) +#define RT5660_JD_STKY_EN (0x1 << 13) +#define RT5660_OT_STKY_MASK (0x1 << 12) +#define RT5660_OT_STKY_SFT 12 +#define RT5660_OT_STKY_DIS (0x0 << 12) +#define RT5660_OT_STKY_EN (0x1 << 12) +#define RT5660_JD_P_MASK (0x1 << 11) +#define RT5660_JD_P_SFT 11 +#define RT5660_JD_P_NOR (0x0 << 11) +#define RT5660_JD_P_INV (0x1 << 11) +#define RT5660_OT_P_MASK (0x1 << 10) +#define RT5660_OT_P_SFT 10 +#define RT5660_OT_P_NOR (0x0 << 10) +#define RT5660_OT_P_INV (0x1 << 10) + +/* IRQ Control 2 (0xbe) */ +#define RT5660_IRQ_MB1_OC_MASK (0x1 << 15) +#define RT5660_IRQ_MB1_OC_SFT 15 +#define RT5660_IRQ_MB1_OC_BP (0x0 << 15) +#define RT5660_IRQ_MB1_OC_NOR (0x1 << 15) +#define RT5660_IRQ_MB2_OC_MASK (0x1 << 14) +#define RT5660_IRQ_MB2_OC_SFT 14 +#define RT5660_IRQ_MB2_OC_BP (0x0 << 14) +#define RT5660_IRQ_MB2_OC_NOR (0x1 << 14) +#define RT5660_MB1_OC_STKY_MASK (0x1 << 11) +#define RT5660_MB1_OC_STKY_SFT 11 +#define RT5660_MB1_OC_STKY_DIS (0x0 << 11) +#define RT5660_MB1_OC_STKY_EN (0x1 << 11) +#define RT5660_MB2_OC_STKY_MASK (0x1 << 10) +#define RT5660_MB2_OC_STKY_SFT 10 +#define RT5660_MB2_OC_STKY_DIS (0x0 << 10) +#define RT5660_MB2_OC_STKY_EN (0x1 << 10) +#define RT5660_MB1_OC_P_MASK (0x1 << 7) +#define RT5660_MB1_OC_P_SFT 7 +#define RT5660_MB1_OC_P_NOR (0x0 << 7) +#define RT5660_MB1_OC_P_INV (0x1 << 7) +#define RT5660_MB2_OC_P_MASK (0x1 << 6) +#define RT5660_MB2_OC_P_SFT 6 +#define RT5660_MB2_OC_P_NOR (0x0 << 6) +#define RT5660_MB2_OC_P_INV (0x1 << 6) +#define RT5660_MB1_OC_CLR (0x1 << 3) +#define RT5660_MB1_OC_CLR_SFT 3 +#define RT5660_MB2_OC_CLR (0x1 << 2) +#define RT5660_MB2_OC_CLR_SFT 2 + +/* GPIO Control 1 (0xc0) */ +#define RT5660_GP2_PIN_MASK (0x1 << 14) +#define RT5660_GP2_PIN_SFT 14 +#define RT5660_GP2_PIN_GPIO2 (0x0 << 14) +#define RT5660_GP2_PIN_DMIC1_SDA (0x1 << 14) +#define RT5660_GP1_PIN_MASK (0x3 << 12) +#define RT5660_GP1_PIN_SFT 12 +#define RT5660_GP1_PIN_GPIO1 (0x0 << 12) +#define RT5660_GP1_PIN_DMIC1_SCL (0x1 << 12) +#define RT5660_GP1_PIN_IRQ (0x2 << 12) +#define RT5660_GPIO_M_MASK (0x1 << 9) +#define RT5660_GPIO_M_SFT 9 +#define RT5660_GPIO_M_FLT (0x0 << 9) +#define RT5660_GPIO_M_PH (0x1 << 9) + +/* GPIO Control 3 (0xc2) */ +#define RT5660_GP2_PF_MASK (0x1 << 5) +#define RT5660_GP2_PF_SFT 5 +#define RT5660_GP2_PF_IN (0x0 << 5) +#define RT5660_GP2_PF_OUT (0x1 << 5) +#define RT5660_GP2_OUT_MASK (0x1 << 4) +#define RT5660_GP2_OUT_SFT 4 +#define RT5660_GP2_OUT_LO (0x0 << 4) +#define RT5660_GP2_OUT_HI (0x1 << 4) +#define RT5660_GP2_P_MASK (0x1 << 3) +#define RT5660_GP2_P_SFT 3 +#define RT5660_GP2_P_NOR (0x0 << 3) +#define RT5660_GP2_P_INV (0x1 << 3) +#define RT5660_GP1_PF_MASK (0x1 << 2) +#define RT5660_GP1_PF_SFT 2 +#define RT5660_GP1_PF_IN (0x0 << 2) +#define RT5660_GP1_PF_OUT (0x1 << 2) +#define RT5660_GP1_OUT_MASK (0x1 << 1) +#define RT5660_GP1_OUT_SFT 1 +#define RT5660_GP1_OUT_LO (0x0 << 1) +#define RT5660_GP1_OUT_HI (0x1 << 1) +#define RT5660_GP1_P_MASK (0x1) +#define RT5660_GP1_P_SFT 0 +#define RT5660_GP1_P_NOR (0x0) +#define RT5660_GP1_P_INV (0x1) + +/* Soft volume and zero cross control 1 (0xd9) */ +#define RT5660_SV_MASK (0x1 << 15) +#define RT5660_SV_SFT 15 +#define RT5660_SV_DIS (0x0 << 15) +#define RT5660_SV_EN (0x1 << 15) +#define RT5660_SPO_SV_MASK (0x1 << 14) +#define RT5660_SPO_SV_SFT 14 +#define RT5660_SPO_SV_DIS (0x0 << 14) +#define RT5660_SPO_SV_EN (0x1 << 14) +#define RT5660_OUT_SV_MASK (0x1 << 12) +#define RT5660_OUT_SV_SFT 12 +#define RT5660_OUT_SV_DIS (0x0 << 12) +#define RT5660_OUT_SV_EN (0x1 << 12) +#define RT5660_ZCD_DIG_MASK (0x1 << 11) +#define RT5660_ZCD_DIG_SFT 11 +#define RT5660_ZCD_DIG_DIS (0x0 << 11) +#define RT5660_ZCD_DIG_EN (0x1 << 11) +#define RT5660_ZCD_MASK (0x1 << 10) +#define RT5660_ZCD_SFT 10 +#define RT5660_ZCD_PD (0x0 << 10) +#define RT5660_ZCD_PU (0x1 << 10) +#define RT5660_SV_DLY_MASK (0xf) +#define RT5660_SV_DLY_SFT 0 + +/* Soft volume and zero cross control 2 (0xda) */ +#define RT5660_ZCD_SPO_MASK (0x1 << 15) +#define RT5660_ZCD_SPO_SFT 15 +#define RT5660_ZCD_SPO_DIS (0x0 << 15) +#define RT5660_ZCD_SPO_EN (0x1 << 15) +#define RT5660_ZCD_OMR_MASK (0x1 << 8) +#define RT5660_ZCD_OMR_SFT 8 +#define RT5660_ZCD_OMR_DIS (0x0 << 8) +#define RT5660_ZCD_OMR_EN (0x1 << 8) +#define RT5660_ZCD_OML_MASK (0x1 << 7) +#define RT5660_ZCD_OML_SFT 7 +#define RT5660_ZCD_OML_DIS (0x0 << 7) +#define RT5660_ZCD_OML_EN (0x1 << 7) +#define RT5660_ZCD_SPM_MASK (0x1 << 6) +#define RT5660_ZCD_SPM_SFT 6 +#define RT5660_ZCD_SPM_DIS (0x0 << 6) +#define RT5660_ZCD_SPM_EN (0x1 << 6) +#define RT5660_ZCD_RMR_MASK (0x1 << 5) +#define RT5660_ZCD_RMR_SFT 5 +#define RT5660_ZCD_RMR_DIS (0x0 << 5) +#define RT5660_ZCD_RMR_EN (0x1 << 5) +#define RT5660_ZCD_RML_MASK (0x1 << 4) +#define RT5660_ZCD_RML_SFT 4 +#define RT5660_ZCD_RML_DIS (0x0 << 4) +#define RT5660_ZCD_RML_EN (0x1 << 4) + +/* General Control 1 (0xfa) */ +#define RT5660_PWR_VREF_HP (0x1 << 11) +#define RT5660_PWR_VREF_HP_SFT 11 +#define RT5660_DIG_GATE_CTRL (0x1) +#define RT5660_DIG_GATE_CTRL_SFT 0 + +/* System Clock Source */ +#define RT5660_SCLK_S_MCLK 0 +#define RT5660_SCLK_S_PLL1 1 +#define RT5660_SCLK_S_RCCLK 2 + +/* PLL1 Source */ +#define RT5660_PLL1_S_MCLK 0 +#define RT5660_PLL1_S_BCLK 1 + +enum { + RT5660_AIF1, + RT5660_AIFS, +}; + +struct rt5660_priv { + struct snd_soc_codec *codec; + struct rt5660_platform_data pdata; + struct regmap *regmap; + struct clk *mclk; + + int sysclk; + int sysclk_src; + int lrck[RT5660_AIFS]; + int bclk[RT5660_AIFS]; + int master[RT5660_AIFS]; + + int pll_src; + int pll_in; + int pll_out; +}; + +#endif -- cgit v1.2.3 From 318824d3d0795309b448563faa698d3c02035db4 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sat, 24 Sep 2016 19:28:23 +0900 Subject: ALSA: control: cage TLV_DB_RANGE_HEAD in kernel land because it was obsoleted In commit bf1d1c9b6179 ("ALSA: tlv: add DECLARE_TLV_DB_RANGE()"), the new macro was added so that "dB range information can be specified without having to count the items manually for TLV_DB_RANGE_HEAD()". In short, TLV_DB_RANGE_HEAD macro was obsoleted. In commit 46e860f76804 ("ALSA: rename TLV-related macros so that they're friendly to user applications"), TLV-related macros are exposed for applications in user land to get content of data structured by Type/Length/Value shape. The commit managed to expose TLV-related macros as many as possible, while obsoleted TLV_DB_RANGE_HEAD() was included to the list of exposed macros. This situation brings some confusions to application developers because they might think all exposed macros have their own purpose and useful for applications. For the reason, this commit moves TLV_DB_RANGE_HEAD macro from UAPI header to a header for kernel land, again. The above commit is done within the same development period for kernel 4.9, thus not published yet. This commit might certainly brings no confusions to user land. Reference: commit bf1d1c9b6179 ("ALSA: tlv: add DECLARE_TLV_DB_RANGE()") Reference: commit 46e860f76804 ("ALSA: rename TLV-related macros so that they're friendly to user applications") Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- include/sound/tlv.h | 9 ++++++++- include/uapi/sound/tlv.h | 3 --- 2 files changed, 8 insertions(+), 4 deletions(-) (limited to 'include') diff --git a/include/sound/tlv.h b/include/sound/tlv.h index 6e2e7735d20a..3677ebb928d5 100644 --- a/include/sound/tlv.h +++ b/include/sound/tlv.h @@ -46,8 +46,15 @@ #define TLV_DB_RANGE_ITEM SNDRV_CTL_TLVD_DB_RANGE_ITEM #define DECLARE_TLV_DB_RANGE SNDRV_CTL_TLVD_DECLARE_DB_RANGE -#define TLV_DB_RANGE_HEAD SNDRV_CTL_TLVD_DB_RANGE_HEAD #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()"). + */ +#define TLV_DB_RANGE_HEAD(num) \ + SNDRV_CTL_TLVT_DB_RANGE, 6 * (num) * sizeof(unsigned int) + #endif /* __SOUND_TLV_H */ diff --git a/include/uapi/sound/tlv.h b/include/uapi/sound/tlv.h index f3c198f8bee7..b4df440c015b 100644 --- a/include/uapi/sound/tlv.h +++ b/include/uapi/sound/tlv.h @@ -94,9 +94,6 @@ unsigned int name[] = { \ SNDRV_CTL_TLVD_DB_RANGE_ITEM(__VA_ARGS__) \ } -/* The below assumes that each item TLV is 4 words like DB_SCALE or LINEAR */ -#define SNDRV_CTL_TLVD_DB_RANGE_HEAD(num) \ - SNDRV_CTL_TLVT_DB_RANGE, 6 * (num) * sizeof(unsigned int) #define SNDRV_CTL_TLVD_DB_GAIN_MUTE -9999999 -- cgit v1.2.3 From bb0c35fcaf8f2ad3383dd43ca8abf5203cd06cc3 Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Mon, 26 Sep 2016 14:29:21 +0100 Subject: ASoC: da7219: Disable AAD if codec is not a wake-up source Currently if AAD is enabled in the device, during system suspend the feature remains, regardless of whether the codec is a wake-up source or not. This means some additional power is being used which is unnecessary, and can causes issues with some platforms' IRQ handlers where state changes during system suspend aren't captured. This patch updates the driver to disable AAD during suspend, if we're not a wake-up source, and then re-enables this on resume. Signed-off-by: Adam Thomson Signed-off-by: Mark Brown --- include/sound/da7219.h | 2 ++ sound/soc/codecs/da7219-aad.c | 56 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/da7219-aad.h | 5 ++++ sound/soc/codecs/da7219.c | 30 +++++++++++++---------- sound/soc/codecs/da7219.h | 1 + 5 files changed, 81 insertions(+), 13 deletions(-) (limited to 'include') 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/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index fc27dab3d6ba..2b8914dd5990 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -796,6 +796,62 @@ static void da7219_aad_handle_pdata(struct snd_soc_codec *codec) } +/* + * Suspend/Resume + */ + +void da7219_aad_suspend(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct da7219_aad_priv *da7219_aad = da7219->aad; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + u8 micbias_ctrl; + + if (da7219_aad->jack) { + /* Disable jack detection during suspend */ + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1, + DA7219_ACCDET_EN_MASK, 0); + + /* + * If we have a 4-pole jack inserted, then micbias will be + * enabled. We can disable micbias here, and keep a note to + * re-enable it on resume. If jack removal occurred during + * suspend then this will be dealt with through the IRQ handler. + */ + if (da7219_aad->jack_inserted) { + micbias_ctrl = snd_soc_read(codec, DA7219_MICBIAS_CTRL); + if (micbias_ctrl & DA7219_MICBIAS1_EN_MASK) { + snd_soc_dapm_disable_pin(dapm, "Mic Bias"); + snd_soc_dapm_sync(dapm); + da7219_aad->micbias_resume_enable = true; + } + } + } +} + +void da7219_aad_resume(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct da7219_aad_priv *da7219_aad = da7219->aad; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + + if (da7219_aad->jack) { + /* Re-enable micbias if previously enabled for 4-pole jack */ + if (da7219_aad->jack_inserted && + da7219_aad->micbias_resume_enable) { + snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); + snd_soc_dapm_sync(dapm); + da7219_aad->micbias_resume_enable = false; + } + + /* Re-enable jack detection */ + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1, + DA7219_ACCDET_EN_MASK, + DA7219_ACCDET_EN_MASK); + } +} + + /* * Init/Exit */ diff --git a/sound/soc/codecs/da7219-aad.h b/sound/soc/codecs/da7219-aad.h index a34be4828f97..117a3d7ccd31 100644 --- a/sound/soc/codecs/da7219-aad.h +++ b/sound/soc/codecs/da7219-aad.h @@ -201,12 +201,17 @@ struct da7219_aad_priv { struct work_struct hptest_work; struct snd_soc_jack *jack; + bool micbias_resume_enable; bool jack_inserted; }; /* AAD control */ void da7219_aad_jack_det(struct snd_soc_codec *codec, struct snd_soc_jack *jack); +/* Suspend/Resume */ +void da7219_aad_suspend(struct snd_soc_codec *codec); +void da7219_aad_resume(struct snd_soc_codec *codec); + /* Init/Exit */ int da7219_aad_init(struct snd_soc_codec *codec); void da7219_aad_exit(struct snd_soc_codec *codec); diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index eecb6d6c29cf..65f7e9807659 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1482,6 +1482,8 @@ static struct da7219_pdata *da7219_fw_to_pdata(struct snd_soc_codec *codec) if (!pdata) return NULL; + pdata->wakeup_source = device_property_read_bool(dev, "wakeup-source"); + if (device_property_read_u32(dev, "dlg,micbias-lvl", &of_val32) >= 0) pdata->micbias_lvl = da7219_fw_micbias_lvl(dev, of_val32); else @@ -1524,20 +1526,21 @@ static int da7219_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: - if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) /* Master bias */ snd_soc_update_bits(codec, DA7219_REFERENCES, DA7219_BIAS_EN_MASK, DA7219_BIAS_EN_MASK); - } else { + + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_PREPARE) { /* Remove MCLK */ if (da7219->mclk) clk_disable_unprepare(da7219->mclk); } break; case SND_SOC_BIAS_OFF: - /* Only disable master bias if jack detection not active */ - if (!da7219->aad->jack) + /* Only disable master bias if we're not a wake-up source */ + if (!da7219->wakeup_source) snd_soc_update_bits(codec, DA7219_REFERENCES, DA7219_BIAS_EN_MASK, 0); @@ -1603,6 +1606,8 @@ static void da7219_handle_pdata(struct snd_soc_codec *codec) if (pdata) { u8 micbias_lvl = 0; + da7219->wakeup_source = pdata->wakeup_source; + /* Mic Bias voltages */ switch (pdata->micbias_lvl) { case DA7219_MICBIAS_1_6V: @@ -1737,11 +1742,11 @@ static int da7219_suspend(struct snd_soc_codec *codec) { struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); - snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); + /* Suspend AAD if we're not a wake-up source */ + if (!da7219->wakeup_source) + da7219_aad_suspend(codec); - /* Put device into standby mode if jack detection disabled */ - if (!da7219->aad->jack) - snd_soc_write(codec, DA7219_SYSTEM_ACTIVE, 0); + snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -1750,13 +1755,12 @@ static int da7219_resume(struct snd_soc_codec *codec) { struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); - /* Put device into active mode if previously pushed to standby */ - if (!da7219->aad->jack) - snd_soc_write(codec, DA7219_SYSTEM_ACTIVE, - DA7219_SYSTEM_ACTIVE_MASK); - snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Resume AAD if previously suspended */ + if (!da7219->wakeup_source) + da7219_aad_resume(codec); + return 0; } #else diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h index f1b3ad835270..66d3bad86739 100644 --- a/sound/soc/codecs/da7219.h +++ b/sound/soc/codecs/da7219.h @@ -803,6 +803,7 @@ struct da7219_priv { struct da7219_aad_priv *aad; struct da7219_pdata *pdata; + bool wakeup_source; struct regulator_bulk_data supplies[DA7219_NUM_SUPPLIES]; struct regmap *regmap; struct mutex lock; -- cgit v1.2.3