From 8bfcab7c6b44038993fb1a7771ae0ce5d1bf6ea3 Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Mon, 9 Jun 2014 11:31:45 +0800 Subject: dt: wm8904: add device tree binding document Signed-off-by: Bo Shen Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/wm8904.txt | 33 ++++++++++++++++++++++ 1 file changed, 33 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wm8904.txt (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/wm8904.txt b/Documentation/devicetree/bindings/sound/wm8904.txt new file mode 100644 index 000000000000..e99f4097c83c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8904.txt @@ -0,0 +1,33 @@ +WM8904 audio CODEC + +This device supports I2C only. + +Required properties: + - compatible: "wlf,wm8904" + - reg: the I2C address of the device. + - clock-names: "mclk" + - clocks: reference to + + +Pins on the device (for linking into audio routes): + + * IN1L + * IN1R + * IN2L + * IN2R + * IN3L + * IN3R + * HPOUTL + * HPOUTR + * LINEOUTL + * LINEOUTR + * MICBIAS + +Examples: + +codec: wm8904@1a { + compatible = "wlf,wm8904"; + reg = <0x1a>; + clocks = <&pck0>; + clock-names = "mclk"; +}; -- cgit v1.2.3 From 2ad76541026ba5c6c93b5a5a6f1418aa89fa5f34 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 28 Mar 2014 19:05:04 +0100 Subject: ASoC: ak5386: Add regulators to documentation and fix sparse warning Document the newly added regulators to the DT binding document. Also, "static const char const *x" is not identical to "static const char * const x", which sparse now complains about. Fix it. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/ak5386.txt | 4 ++++ sound/soc/codecs/ak5386.c | 2 +- 2 files changed, 5 insertions(+), 1 deletion(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/ak5386.txt b/Documentation/devicetree/bindings/sound/ak5386.txt index dc3914fe6ce8..ec3df3abba0c 100644 --- a/Documentation/devicetree/bindings/sound/ak5386.txt +++ b/Documentation/devicetree/bindings/sound/ak5386.txt @@ -10,10 +10,14 @@ Optional properties: - reset-gpio : a GPIO spec for the reset/power down pin. If specified, it will be deasserted at probe time. + - va-supply : a regulator spec, providing 5.0V + - vd-supply : a regulator spec, providing 3.3V Example: spdif: ak5386@0 { compatible = "asahi-kasei,ak5386"; reset-gpio = <&gpio0 23>; + va-supply = <&vdd_5v0_reg>; + vd-supply = <&vdd_3v3_reg>; }; diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c index a30be5ce3fa4..8107a1cac876 100644 --- a/sound/soc/codecs/ak5386.c +++ b/sound/soc/codecs/ak5386.c @@ -19,7 +19,7 @@ #include #include -static const char const *supply_names[] = { +static const char * const supply_names[] = { "va", "vd" }; -- cgit v1.2.3 From 2b65df255c7e1f028f7b4f4d18fd41eecafad4bd Mon Sep 17 00:00:00 2001 From: Paul Handrigan Date: Mon, 23 Jun 2014 17:29:52 -0500 Subject: ASoC: cs4265: bindings: sound: Add binding for CS4265 CODEC. This patch adds binding documentation for the Cirrus Logic CS4265 I2C CODEC. Signed-off-by: Paul Handrigan Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/cs4265.txt | 29 ++++++++++++++++++++++ 1 file changed, 29 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs4265.txt (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/cs4265.txt b/Documentation/devicetree/bindings/sound/cs4265.txt new file mode 100644 index 000000000000..380fff8e4e83 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4265.txt @@ -0,0 +1,29 @@ +CS4265 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "cirrus,cs4265" + + - reg : the I2C address of the device for I2C. The I2C address depends on + the state of the AD0 pin. If AD0 is high, the i2c address is 0x4f. + If it is low, the i2c address is 0x4e. + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + +Examples: + +codec_ad0_high: cs4265@4f { /* AD0 Pin is high */ + compatible = "cirrus,cs4265"; + reg = <0x4f>; +}; + + +codec_ad0_low: cs4265@4e { /* AD0 Pin is low */ + compatible = "cirrus,cs4265"; + reg = <0x4e>; +}; -- cgit v1.2.3 From d9288d0ba12de1b5efb830b9128e4cc6877318fc Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 22 Jun 2014 17:56:23 -0700 Subject: ASoC: rsnd: SSI + DMA can select BUSIF Sound data needs to be sent to R-Car sound SSI when playback. But, there are 2 interfaces for it. 1st is SSITDR/SSIRDR which are mapped on SSI. 2nd is SSIn_BUSIF which are mapped on SSIU. 2nd SSIn_BUSIF is used when DMA transfer, and it is always used if sound data came from via SRC. But, we can use it when SSI+DMA case too. (Current driver is assuming 1st SSITDR/SSIRDR for it) 2nd SSIn_BUSIF can be used as FIFO. This is very helpful/useful for SSI+DMA. But DMA address / DMA ID are not same between 1st/2nd cases. This patch care about these settings. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/renesas,rsnd.txt | 1 + include/sound/rcar_snd.h | 1 + sound/soc/sh/rcar/core.c | 22 +++++--- sound/soc/sh/rcar/gen.c | 64 ++++++++++++++-------- sound/soc/sh/rcar/rsnd.h | 10 +++- sound/soc/sh/rcar/src.c | 37 +++++++++---- sound/soc/sh/rcar/ssi.c | 33 ++++++++++- 7 files changed, 123 insertions(+), 45 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 8346cab046cd..41a120c2389d 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -21,6 +21,7 @@ SSI subnode properties: - interrupts : Should contain SSI interrupt for PIO transfer - shared-pin : if shared clock pin - pio-transfer : use PIO transfer mode +- no-busif : BUSIF is not ussed when [mem -> SSI] via DMA case SRC subnode properties: no properties at this point diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index f4a706f82cb7..d76412b84b48 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -34,6 +34,7 @@ * B : SSI direction */ #define RSND_SSI_CLK_PIN_SHARE (1 << 31) +#define RSND_SSI_NO_BUSIF (1 << 30) /* SSI+DMA without BUSIF */ #define RSND_SSI(_dma_id, _pio_irq, _flags) \ { .dma_id = _dma_id, .pio_irq = _pio_irq, .flags = _flags } diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 7f68b33dcbbb..8c3707a68603 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -138,6 +138,17 @@ char *rsnd_mod_name(struct rsnd_mod *mod) return mod->ops->name; } +char *rsnd_mod_dma_name(struct rsnd_mod *mod) +{ + if (!mod || !mod->ops) + return "unknown"; + + if (!mod->ops->dma_name) + return mod->ops->name; + + return mod->ops->dma_name(mod); +} + void rsnd_mod_init(struct rsnd_priv *priv, struct rsnd_mod *mod, struct rsnd_mod_ops *ops, @@ -261,7 +272,7 @@ static int _rsnd_dma_of_name(char *dma_name, struct rsnd_mod *mod) { if (mod) return snprintf(dma_name, DMA_NAME_SIZE / 2, "%s%d", - rsnd_mod_name(mod), rsnd_mod_id(mod)); + rsnd_mod_dma_name(mod), rsnd_mod_id(mod)); else return snprintf(dma_name, DMA_NAME_SIZE / 2, "mem"); @@ -343,11 +354,8 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, dma_cap_zero(mask); dma_cap_set(DMA_SLAVE, mask); - if (dev->of_node) - rsnd_dma_of_name(dma, is_play, dma_name); - else - snprintf(dma_name, DMA_NAME_SIZE, - is_play ? "tx" : "rx"); + rsnd_dma_of_name(dma, is_play, dma_name); + rsnd_gen_dma_addr(priv, dma, &cfg, is_play, id); dev_dbg(dev, "dma name : %s\n", dma_name); @@ -359,8 +367,6 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, return -EIO; } - rsnd_gen_dma_addr(priv, dma, &cfg, is_play, id); - ret = dmaengine_slave_config(dma->chan, &cfg); if (ret < 0) goto rsnd_dma_init_err; diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 0280a11c0899..46677af6c748 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -165,15 +165,19 @@ static int rsnd_gen_regmap_init(struct rsnd_priv *priv, * * ex) R-Car H2 case * mod / DMAC in / DMAC out / DMAC PP in / DMAC pp out - * SSI : 0xec541000 / 0xec241008 / 0xec24100c / 0xec400000 / 0xec400000 + * SSI : 0xec541000 / 0xec241008 / 0xec24100c + * SSIU: 0xec541000 / 0xec100000 / 0xec100000 / 0xec400000 / 0xec400000 * SCU : 0xec500000 / 0xec000000 / 0xec004000 / 0xec300000 / 0xec304000 * CMD : 0xec500000 / 0xec008000 0xec308000 */ #define RDMA_SSI_I_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0x8) #define RDMA_SSI_O_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0xc) -#define RDMA_SSI_I_P(addr, i) (addr ##_reg - 0x00141000 + (0x1000 * i)) -#define RDMA_SSI_O_P(addr, i) (addr ##_reg - 0x00141000 + (0x1000 * i)) +#define RDMA_SSIU_I_N(addr, i) (addr ##_reg - 0x00441000 + (0x1000 * i)) +#define RDMA_SSIU_O_N(addr, i) (addr ##_reg - 0x00441000 + (0x1000 * i)) + +#define RDMA_SSIU_I_P(addr, i) (addr ##_reg - 0x00141000 + (0x1000 * i)) +#define RDMA_SSIU_O_P(addr, i) (addr ##_reg - 0x00141000 + (0x1000 * i)) #define RDMA_SRC_I_N(addr, i) (addr ##_reg - 0x00500000 + (0x400 * i)) #define RDMA_SRC_O_N(addr, i) (addr ##_reg - 0x004fc000 + (0x400 * i)) @@ -204,26 +208,36 @@ static void rsnd_gen2_dma_addr(struct rsnd_priv *priv, struct dma_addr { dma_addr_t src_addr; dma_addr_t dst_addr; - } dma_addrs[2][2][3] = { - { /* SRC */ - /* Capture */ - {{ 0, 0 }, - { RDMA_SRC_O_N(src, id), 0 }, - { RDMA_CMD_O_N(src, id), 0 }}, - /* Playback */ - {{ 0, 0, }, - { 0, RDMA_SRC_I_N(src, id) }, - { 0, RDMA_SRC_I_N(src, id) }} - }, { /* SSI */ - /* Capture */ - {{ RDMA_SSI_O_N(ssi, id), 0 }, - { RDMA_SSI_O_P(ssi, id), RDMA_SRC_I_P(src, id) }, - { RDMA_SSI_O_P(ssi, id), RDMA_SRC_I_P(src, id) }}, - /* Playback */ - {{ 0, RDMA_SSI_I_N(ssi, id) }, - { RDMA_SRC_O_P(src, id), RDMA_SSI_I_P(ssi, id) }, - { RDMA_CMD_O_P(src, id), RDMA_SSI_I_P(ssi, id) }} - } + } dma_addrs[3][2][3] = { + /* SRC */ + {{{ 0, 0 }, + /* Capture */ + { RDMA_SRC_O_N(src, id), 0 }, + { RDMA_CMD_O_N(src, id), 0 } }, + /* Playback */ + {{ 0, 0, }, + { 0, RDMA_SRC_I_N(src, id) }, + { 0, RDMA_SRC_I_N(src, id) } } + }, + /* SSI */ + /* Capture */ + {{{ RDMA_SSI_O_N(ssi, id), 0 }, + { 0, 0 }, + { 0, 0 } }, + /* Playback */ + {{ 0, RDMA_SSI_I_N(ssi, id) }, + { 0, 0 }, + { 0, 0 } } + }, + /* SSIU */ + /* Capture */ + {{{ RDMA_SSIU_O_N(ssi, id), 0 }, + { RDMA_SSIU_O_P(ssi, id), RDMA_SRC_I_P(src, id) }, + { RDMA_SSIU_O_P(ssi, id), RDMA_SRC_I_P(src, id) } }, + /* Playback */ + {{ 0, RDMA_SSIU_I_N(ssi, id) }, + { RDMA_SRC_O_P(src, id), RDMA_SSIU_I_P(ssi, id) }, + { RDMA_CMD_O_P(src, id), RDMA_SSIU_I_P(ssi, id) } } }, }; /* it shouldn't happen */ @@ -232,6 +246,10 @@ static void rsnd_gen2_dma_addr(struct rsnd_priv *priv, return; } + /* use SSIU or SSI ? */ + if (is_ssi && (0 == strcmp(rsnd_mod_dma_name(mod), "ssiu"))) + is_ssi++; + cfg->src_addr = dma_addrs[is_ssi][is_play][use_src + use_dvc].src_addr; cfg->dst_addr = dma_addrs[is_ssi][is_play][use_src + use_dvc].dst_addr; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 067a89e9f25c..a1466c1570bc 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -185,6 +185,7 @@ enum rsnd_mod_type { struct rsnd_mod_ops { char *name; + char* (*dma_name)(struct rsnd_mod *mod); int (*probe)(struct rsnd_mod *mod, struct rsnd_dai *rdai); int (*remove)(struct rsnd_mod *mod, @@ -224,6 +225,7 @@ void rsnd_mod_init(struct rsnd_priv *priv, enum rsnd_mod_type type, int id); char *rsnd_mod_name(struct rsnd_mod *mod); +char *rsnd_mod_dma_name(struct rsnd_mod *mod); /* * R-Car sound DAI @@ -391,8 +393,12 @@ struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id); unsigned int rsnd_src_get_ssi_rate(struct rsnd_priv *priv, struct rsnd_dai_stream *io, struct snd_pcm_runtime *runtime); -int rsnd_src_ssi_mode_init(struct rsnd_mod *ssi_mod, - struct rsnd_dai *rdai); +int rsnd_src_ssiu_start(struct rsnd_mod *ssi_mod, + struct rsnd_dai *rdai, + int use_busif); +int rsnd_src_ssiu_stop(struct rsnd_mod *ssi_mod, + struct rsnd_dai *rdai, + int use_busif); int rsnd_src_enable_ssi_irq(struct rsnd_mod *ssi_mod, struct rsnd_dai *rdai); diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 200eda019bc7..4d39505c21cf 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -106,18 +106,17 @@ struct rsnd_src { /* * Gen1/Gen2 common functions */ -int rsnd_src_ssi_mode_init(struct rsnd_mod *ssi_mod, - struct rsnd_dai *rdai) +int rsnd_src_ssiu_start(struct rsnd_mod *ssi_mod, + struct rsnd_dai *rdai, + int use_busif) { - struct rsnd_dai_stream *io = rsnd_mod_to_io(ssi_mod); - struct rsnd_mod *src_mod = rsnd_io_to_mod_src(io); int ssi_id = rsnd_mod_id(ssi_mod); /* * SSI_MODE0 */ rsnd_mod_bset(ssi_mod, SSI_MODE0, (1 << ssi_id), - src_mod ? 0 : (1 << ssi_id)); + !use_busif << ssi_id); /* * SSI_MODE1 @@ -143,6 +142,29 @@ int rsnd_src_ssi_mode_init(struct rsnd_mod *ssi_mod, 0x2 << shift : 0x1 << shift); } + /* + * DMA settings for SSIU + */ + if (use_busif) { + rsnd_mod_write(ssi_mod, SSI_BUSIF_ADINR, + rsnd_get_adinr(ssi_mod)); + rsnd_mod_write(ssi_mod, SSI_BUSIF_MODE, 1); + rsnd_mod_write(ssi_mod, SSI_CTRL, 0x1); + } + + return 0; +} + +int rsnd_src_ssiu_stop(struct rsnd_mod *ssi_mod, + struct rsnd_dai *rdai, + int use_busif) +{ + /* + * DMA settings for SSIU + */ + if (use_busif) + rsnd_mod_write(ssi_mod, SSI_CTRL, 0); + return 0; } @@ -467,9 +489,6 @@ static int rsnd_src_set_convert_rate_gen2(struct rsnd_mod *mod, if (ret < 0) return ret; - rsnd_mod_write(mod, SSI_BUSIF_ADINR, rsnd_get_adinr(mod)); - rsnd_mod_write(mod, SSI_BUSIF_MODE, 1); - rsnd_mod_write(mod, SRC_SRCCR, 0x00011110); rsnd_mod_write(mod, SRC_BSDSR, 0x01800000); @@ -554,7 +573,6 @@ static int rsnd_src_start_gen2(struct rsnd_mod *mod, rsnd_dma_start(rsnd_mod_to_dma(&src->mod)); - rsnd_mod_write(mod, SSI_CTRL, 0x1); rsnd_mod_write(mod, SRC_CTRL, val); return rsnd_src_start(mod, rdai); @@ -565,7 +583,6 @@ static int rsnd_src_stop_gen2(struct rsnd_mod *mod, { struct rsnd_src *src = rsnd_mod_to_src(mod); - rsnd_mod_write(mod, SSI_CTRL, 0); rsnd_mod_write(mod, SRC_CTRL, 0); rsnd_dma_stop(rsnd_mod_to_dma(&src->mod)); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 2df723df5d19..34e84009162b 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -90,6 +90,20 @@ struct rsnd_ssi { #define rsnd_ssi_mode_flags(p) ((p)->info->flags) #define rsnd_ssi_dai_id(ssi) ((ssi)->info->dai_id) +static int rsnd_ssi_use_busif(struct rsnd_mod *mod) +{ + struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + int use_busif = 0; + + if (!(rsnd_ssi_mode_flags(ssi) & RSND_SSI_NO_BUSIF)) + use_busif = 1; + if (rsnd_io_to_mod_src(io)) + use_busif = 1; + + return use_busif; +} + static void rsnd_ssi_status_check(struct rsnd_mod *mod, u32 bit) { @@ -289,8 +303,6 @@ static int rsnd_ssi_init(struct rsnd_mod *mod, ssi->cr_own = cr; ssi->err = -1; /* ignore 1st error */ - rsnd_src_ssi_mode_init(mod, rdai); - return 0; } @@ -389,6 +401,8 @@ static int rsnd_ssi_pio_start(struct rsnd_mod *mod, /* enable PIO IRQ */ ssi->cr_etc = UIEN | OIEN | DIEN; + rsnd_src_ssiu_start(mod, rdai, 0); + rsnd_src_enable_ssi_irq(mod, rdai); rsnd_ssi_hw_start(ssi, rdai, io); @@ -405,6 +419,8 @@ static int rsnd_ssi_pio_stop(struct rsnd_mod *mod, rsnd_ssi_hw_stop(ssi, rdai); + rsnd_src_ssiu_stop(mod, rdai, 0); + return 0; } @@ -457,6 +473,8 @@ static int rsnd_ssi_dma_start(struct rsnd_mod *mod, /* enable DMA transfer */ ssi->cr_etc = DMEN; + rsnd_src_ssiu_start(mod, rdai, rsnd_ssi_use_busif(mod)); + rsnd_dma_start(dma); rsnd_ssi_hw_start(ssi, ssi->rdai, io); @@ -482,11 +500,19 @@ static int rsnd_ssi_dma_stop(struct rsnd_mod *mod, rsnd_dma_stop(dma); + rsnd_src_ssiu_stop(mod, rdai, 1); + return 0; } +static char *rsnd_ssi_dma_name(struct rsnd_mod *mod) +{ + return rsnd_ssi_use_busif(mod) ? "ssiu" : SSI_NAME; +} + static struct rsnd_mod_ops rsnd_ssi_dma_ops = { .name = SSI_NAME, + .dma_name = rsnd_ssi_dma_name, .probe = rsnd_ssi_dma_probe, .remove = rsnd_ssi_dma_remove, .init = rsnd_ssi_init, @@ -595,6 +621,9 @@ static void rsnd_of_parse_ssi(struct platform_device *pdev, */ ssi_info->dma_id = of_get_property(np, "pio-transfer", NULL) ? 0 : 1; + + if (of_get_property(np, "no-busif", NULL)) + ssi_info->flags |= RSND_SSI_NO_BUSIF; } rsnd_of_parse_ssi_end: -- cgit v1.2.3 From 053e69d57cc6253b19ea661f929c8c1b6a907bff Mon Sep 17 00:00:00 2001 From: Wonjoon Lee Date: Fri, 20 Jun 2014 13:33:15 +0530 Subject: ASoC: max98090: Add max98091 compatible string The MAX98091 CODEC is the same as MAX98090 CODEC, but with an extra microphone. Existing driver for MAX98090 CODEC already has support for MAX98091 CODEC. Adding proper compatible string so that MAX98091 CODEC can be specified from device tree. Signed-off-by: Wonjoon Lee Signed-off-by: Doug Anderson Signed-off-by: Tushar Behera Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/max98090.txt | 2 +- sound/soc/codecs/max98090.c | 2 ++ 2 files changed, 3 insertions(+), 1 deletion(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt index a5e63fa47dc5..c454e67f54bb 100644 --- a/Documentation/devicetree/bindings/sound/max98090.txt +++ b/Documentation/devicetree/bindings/sound/max98090.txt @@ -4,7 +4,7 @@ This device supports I2C only. Required properties: -- compatible : "maxim,max98090". +- compatible : "maxim,max98090" or "maxim,max98091". - reg : The I2C address of the device. diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 3aec3ae78fe0..c00b36872dfe 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2447,12 +2447,14 @@ static const struct dev_pm_ops max98090_pm = { static const struct i2c_device_id max98090_i2c_id[] = { { "max98090", MAX98090 }, + { "max98091", MAX98091 }, { } }; MODULE_DEVICE_TABLE(i2c, max98090_i2c_id); static const struct of_device_id max98090_of_match[] = { { .compatible = "maxim,max98090", }, + { .compatible = "maxim,max98091", }, { } }; MODULE_DEVICE_TABLE(of, max98090_of_match); -- cgit v1.2.3 From 46aed597527384b30a6d49bff1806f6b1ed1fd77 Mon Sep 17 00:00:00 2001 From: Tushar Behera Date: Fri, 20 Jun 2014 13:33:16 +0530 Subject: ASoC: samsung: Extend snow driver to support MAX98091 Peach-pi board has MAX98091 CODEC. Extend snow machine driver to support this board. Signed-off-by: Tushar Behera Reviewed-by: Doug Anderson Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/snow.txt | 1 + sound/soc/samsung/snow.c | 1 + 2 files changed, 2 insertions(+) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/snow.txt b/Documentation/devicetree/bindings/sound/snow.txt index 678b191c37b8..e0b7a8207f5f 100644 --- a/Documentation/devicetree/bindings/sound/snow.txt +++ b/Documentation/devicetree/bindings/sound/snow.txt @@ -3,6 +3,7 @@ Audio Binding for Snow boards Required properties: - compatible : Can be one of the following, "google,snow-audio-max98090" or + "google,snow-audio-max98091" or "google,snow-audio-max98095" - samsung,i2s-controller: The phandle of the Samsung I2S controller - samsung,audio-codec: The phandle of the audio codec diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index 014c177840ba..8bbd348358dd 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -103,6 +103,7 @@ static int snow_probe(struct platform_device *pdev) static const struct of_device_id snow_of_match[] = { { .compatible = "google,snow-audio-max98090", }, + { .compatible = "google,snow-audio-max98091", }, { .compatible = "google,snow-audio-max98095", }, {}, }; -- cgit v1.2.3 From 34cb6123f90d264d63bdcd3ee0df0d2cb4b36aab Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 22 Jun 2014 17:59:28 -0700 Subject: ASoC: rsnd: add DT support to DVC Now, DVC can use DT Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/renesas,rsnd.txt | 8 +++++ sound/soc/sh/rcar/core.c | 13 ++++++-- sound/soc/sh/rcar/dvc.c | 38 ++++++++++++++++++++++ 3 files changed, 57 insertions(+), 2 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 41a120c2389d..aa697abf337e 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -13,6 +13,9 @@ Required properties: - rcar_sound,src : Should contain SRC feature. The number of SRC subnode should be same as HW. see below for detail. +- rcar_sound,dvc : Should contain DVC feature. + The number of DVC subnode should be same as HW. + see below for detail. - rcar_sound,dai : DAI contents. The number of DAI subnode should be same as HW. see below for detail. @@ -40,6 +43,11 @@ rcar_sound: rcar_sound@0xffd90000 { <0 0xec540000 0 0x1000>, /* SSIU */ <0 0xec541000 0 0x1280>; /* SSI */ + rcar_sound,dvc { + dvc0: dvc@0 { }; + dvc1: dvc@1 { }; + }; + rcar_sound,src { src0: src@0 { }; src1: src@1 { }; diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 49d9b31b185c..907d4802fd5c 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -286,7 +286,13 @@ static void rsnd_dma_of_path(struct rsnd_dma *dma, mod[i] = src; src = NULL; } else { - mod[i] = dvc; + if ((!is_play) && (this == src)) + this = dvc; + + mod[i] = (is_play) ? src : dvc; + i++; + mod[i] = (is_play) ? dvc : src; + src = NULL; dvc = NULL; } @@ -719,12 +725,13 @@ static void rsnd_of_parse_dai(struct platform_device *pdev, struct device_node *dai_node, *dai_np; struct device_node *ssi_node, *ssi_np; struct device_node *src_node, *src_np; + struct device_node *dvc_node, *dvc_np; struct device_node *playback, *capture; struct rsnd_dai_platform_info *dai_info; struct rcar_snd_info *info = rsnd_priv_to_info(priv); struct device *dev = &pdev->dev; int nr, i; - int dai_i, ssi_i, src_i; + int dai_i, ssi_i, src_i, dvc_i; if (!of_data) return; @@ -750,6 +757,7 @@ static void rsnd_of_parse_dai(struct platform_device *pdev, ssi_node = of_get_child_by_name(dev->of_node, "rcar_sound,ssi"); src_node = of_get_child_by_name(dev->of_node, "rcar_sound,src"); + dvc_node = of_get_child_by_name(dev->of_node, "rcar_sound,dvc"); #define mod_parse(name) \ if (name##_node) { \ @@ -785,6 +793,7 @@ if (name##_node) { \ mod_parse(ssi); mod_parse(src); + mod_parse(dvc); if (playback) of_node_put(playback); diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index deef310c75dc..9096fb03d001 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -235,6 +235,42 @@ struct rsnd_mod *rsnd_dvc_mod_get(struct rsnd_priv *priv, int id) return &((struct rsnd_dvc *)(priv->dvc) + id)->mod; } +static void rsnd_of_parse_dvc(struct platform_device *pdev, + const struct rsnd_of_data *of_data, + struct rsnd_priv *priv) +{ + struct device_node *node; + struct rsnd_dvc_platform_info *dvc_info; + struct rcar_snd_info *info = rsnd_priv_to_info(priv); + struct device *dev = &pdev->dev; + int nr; + + if (!of_data) + return; + + node = of_get_child_by_name(dev->of_node, "rcar_sound,dvc"); + if (!node) + return; + + nr = of_get_child_count(node); + if (!nr) + goto rsnd_of_parse_dvc_end; + + dvc_info = devm_kzalloc(dev, + sizeof(struct rsnd_dvc_platform_info) * nr, + GFP_KERNEL); + if (!dvc_info) { + dev_err(dev, "dvc info allocation error\n"); + goto rsnd_of_parse_dvc_end; + } + + info->dvc_info = dvc_info; + info->dvc_info_nr = nr; + +rsnd_of_parse_dvc_end: + of_node_put(node); +} + int rsnd_dvc_probe(struct platform_device *pdev, const struct rsnd_of_data *of_data, struct rsnd_priv *priv) @@ -246,6 +282,8 @@ int rsnd_dvc_probe(struct platform_device *pdev, char name[RSND_DVC_NAME_SIZE]; int i, nr; + rsnd_of_parse_dvc(pdev, of_data, priv); + nr = info->dvc_info_nr; if (!nr) return 0; -- cgit v1.2.3 From c46af3124b05ccb37d1311e442f191bb5c1d888e Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 3 Jul 2014 16:56:43 +0200 Subject: ASoC: tas5086: add regulator consumer support The TAS5086 has two power domains, DVDD and AVDD. Enable them both as long as the codec is in use. Also, switch on the power to identify the chip at device probe level, and switch it off again afterwards. The codec level will take care for power handling later. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/ti,tas5086.txt | 5 ++ sound/soc/codecs/tas5086.c | 67 +++++++++++++++++++--- 2 files changed, 63 insertions(+), 9 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/ti,tas5086.txt b/Documentation/devicetree/bindings/sound/ti,tas5086.txt index d2866a0d6a26..234dad296da7 100644 --- a/Documentation/devicetree/bindings/sound/ti,tas5086.txt +++ b/Documentation/devicetree/bindings/sound/ti,tas5086.txt @@ -31,6 +31,9 @@ Optional properties: Most systems should not set any of these properties. + - avdd-supply: Power supply for AVDD, providing 3.3V + - dvdd-supply: Power supply for DVDD, providing 3.3V + Examples: i2c_bus { @@ -39,5 +42,7 @@ Examples: reg = <0x1b>; reset-gpio = <&gpio 23 0>; ti,charge-period = <156000>; + avdd-supply = <&vdd_3v3_reg>; + dvdd-supply = <&vdd_3v3_reg>; }; }; diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index d48491a4a19d..be7194b43b7a 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -36,6 +36,7 @@ #include #include #include +#include #include #include #include @@ -240,6 +241,10 @@ static int tas5086_reg_read(void *context, unsigned int reg, return 0; } +static const char * const supply_names[] = { + "dvdd", "avdd" +}; + struct tas5086_private { struct regmap *regmap; unsigned int mclk, sclk; @@ -251,6 +256,7 @@ struct tas5086_private { int rate; /* GPIO driving Reset pin, if any */ int gpio_nreset; + struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; static int tas5086_deemph[] = { 0, 32000, 44100, 48000 }; @@ -773,6 +779,8 @@ static int tas5086_soc_suspend(struct snd_soc_codec *codec) if (ret < 0) return ret; + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + return 0; } @@ -781,6 +789,10 @@ static int tas5086_soc_resume(struct snd_soc_codec *codec) struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); int ret; + ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); + if (ret < 0) + return ret; + tas5086_reset(priv); regcache_mark_dirty(priv->regmap); @@ -812,6 +824,12 @@ static int tas5086_probe(struct snd_soc_codec *codec) struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); int i, ret; + ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable regulators: %d\n", ret); + return ret; + } + priv->pwm_start_mid_z = 0; priv->charge_period = 1300000; /* hardware default is 1300 ms */ @@ -832,16 +850,22 @@ static int tas5086_probe(struct snd_soc_codec *codec) } } + tas5086_reset(priv); ret = tas5086_init(codec->dev, priv); if (ret < 0) - return ret; + goto exit_disable_regulators; /* set master volume to 0 dB */ ret = regmap_write(priv->regmap, TAS5086_MASTER_VOL, 0x30); if (ret < 0) - return ret; + goto exit_disable_regulators; return 0; + +exit_disable_regulators: + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + + return ret; } static int tas5086_remove(struct snd_soc_codec *codec) @@ -852,6 +876,8 @@ static int tas5086_remove(struct snd_soc_codec *codec) /* Set codec to the reset state */ gpio_set_value(priv->gpio_nreset, 0); + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + return 0; }; @@ -900,6 +926,16 @@ static int tas5086_i2c_probe(struct i2c_client *i2c, if (!priv) return -ENOMEM; + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + priv->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret < 0) { + dev_err(dev, "Failed to get regulators: %d\n", ret); + return ret; + } + priv->regmap = devm_regmap_init(dev, NULL, i2c, &tas5086_regmap); if (IS_ERR(priv->regmap)) { ret = PTR_ERR(priv->regmap); @@ -919,21 +955,34 @@ static int tas5086_i2c_probe(struct i2c_client *i2c, gpio_nreset = -EINVAL; priv->gpio_nreset = gpio_nreset; + + ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); + if (ret < 0) { + dev_err(dev, "Failed to enable regulators: %d\n", ret); + return ret; + } + tas5086_reset(priv); /* The TAS5086 always returns 0x03 in its TAS5086_DEV_ID register */ ret = regmap_read(priv->regmap, TAS5086_DEV_ID, &i); - if (ret < 0) - return ret; - - if (i != 0x3) { + if (ret == 0 && i != 0x3) { dev_err(dev, "Failed to identify TAS5086 codec (got %02x)\n", i); - return -ENODEV; + ret = -ENODEV; } - return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tas5086, - &tas5086_dai, 1); + /* + * The chip has been identified, so we can turn off the power + * again until the dai link is set up. + */ + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + + if (ret == 0) + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tas5086, + &tas5086_dai, 1); + + return ret; } static int tas5086_i2c_remove(struct i2c_client *i2c) -- cgit v1.2.3 From 374a6679bef2e79fc2a27eea79b0400295e39df0 Mon Sep 17 00:00:00 2001 From: Rongjun Ying Date: Wed, 2 Jul 2014 10:45:42 +0800 Subject: ASoC: sirf: Add device tree binding for the USP audio device Signed-off-by: Rongjun Ying Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/sirf-usp.txt | 27 ++++++++++++++++++++++ 1 file changed, 27 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/sirf-usp.txt (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/sirf-usp.txt b/Documentation/devicetree/bindings/sound/sirf-usp.txt new file mode 100644 index 000000000000..02f85b32d359 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sirf-usp.txt @@ -0,0 +1,27 @@ +* SiRF SoC USP module + +Required properties: +- compatible: "sirf,prima2-usp-pcm" +- reg: Base address and size entries: +- dmas: List of DMA controller phandle and DMA request line ordered pairs. +- dma-names: Identifier string for each DMA request line in the dmas property. + These strings correspond 1:1 with the ordered pairs in dmas. + + One of the DMA channels will be responsible for transmission (should be + named "tx") and one for reception (should be named "rx"). + +- clocks: USP controller clock source +- pinctrl-names: Must contain a "default" entry. +- pinctrl-NNN: One property must exist for each entry in pinctrl-names. + +Example: +usp0: usp@b0080000 { + compatible = "sirf,prima2-usp-pcm"; + reg = <0xb0080000 0x10000>; + clocks = <&clks 28>; + dmas = <&dmac1 1>, <&dmac1 2>; + dma-names = "rx", "tx"; + pinctrl-names = "default"; + pinctrl-0 = <&usp0_only_utfs_pins_a>; +}; + -- cgit v1.2.3 From 00ad93e26375d974801886070738d0c7cf187fdf Mon Sep 17 00:00:00 2001 From: Tushar Behera Date: Fri, 4 Jul 2014 14:22:59 +0530 Subject: ASoC: samsung: Make card name for Snow configurable Snow sound-card driver supports multiple boards with different audio codecs. Updating the sound card name per board basis would provide some more information to the end-user. Signed-off-by: Tushar Behera Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/snow.txt | 4 ++++ sound/soc/samsung/snow.c | 3 +++ 2 files changed, 7 insertions(+) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/snow.txt b/Documentation/devicetree/bindings/sound/snow.txt index e0b7a8207f5f..6df74f15687f 100644 --- a/Documentation/devicetree/bindings/sound/snow.txt +++ b/Documentation/devicetree/bindings/sound/snow.txt @@ -8,11 +8,15 @@ Required properties: - samsung,i2s-controller: The phandle of the Samsung I2S controller - samsung,audio-codec: The phandle of the audio codec +Optional: +- samsung,model: The name of the sound-card + Example: sound { compatible = "google,snow-audio-max98095"; + samsung,model = "Snow-I2S-MAX98095"; samsung,i2s-controller = <&i2s0>; samsung,audio-codec = <&max98095>; }; diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index 8bbd348358dd..0acf5d0eed53 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -92,6 +92,9 @@ static int snow_probe(struct platform_device *pdev) card->dev = &pdev->dev; + /* Update card-name if provided through DT, else use default name */ + snd_soc_of_parse_card_name(card, "samsung,model"); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); -- cgit v1.2.3 From d4796e35fb67df4ec36fa5914689ced8bc51a614 Mon Sep 17 00:00:00 2001 From: Jianqun Xu Date: Sat, 5 Jul 2014 19:10:38 +0800 Subject: ASoC: dt-bindings: add rockchip i2s bindings Add devicetree bindings for i2s controller found on rk3066, rk3188 and rk3288 processors from rockchip. Signed-off-by: Jianqun Xu Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/rockchip-i2s.txt | 37 ++++++++++++++++++++++ 1 file changed, 37 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-i2s.txt (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt new file mode 100644 index 000000000000..6c55fcfe5e1d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt @@ -0,0 +1,37 @@ +* Rockchip I2S controller + +The I2S bus (Inter-IC sound bus) is a serial link for digital +audio data transfer between devices in the system. + +Required properties: + +- compatible: should be one of the followings + - "rockchip,rk3066-i2s": for rk3066 + - "rockchip,rk3188-i2s", "rockchip,rk3066-i2s": for rk3188 + - "rockchip,rk3288-i2s", "rockchip,rk3066-i2s": for rk3288 +- reg: physical base address of the controller and length of memory mapped + region. +- interrupts: should contain the I2S interrupt. +- #address-cells: should be 1. +- #size-cells: should be 0. +- dmas: DMA specifiers for tx and rx dma. See the DMA client binding, + Documentation/devicetree/bindings/dma/dma.txt +- dma-names: should include "tx" and "rx". +- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. +- clock-names: should contain followings: + - "i2s_hclk": clock for I2S BUS + - "i2s_clk" : clock for I2S controller + +Example for rk3288 I2S controller: + +i2s@ff890000 { + compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s"; + reg = <0xff890000 0x10000>; + interrupts = ; + #address-cells = <1>; + #size-cells = <0>; + dmas = <&pdma1 0>, <&pdma1 1>; + dma-names = "rx", "tx"; + clock-names = "i2s_hclk", "i2s_clk"; + clocks = <&cru HCLK_I2S0>, <&cru SCLK_I2S0>; +}; -- cgit v1.2.3 From 83e3fbd6f3949d21c178a76af72cdfb5be0274ee Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 4 Jul 2014 15:13:44 +0200 Subject: ASoC: samsung: Document Odroid X2/U3 audio subsystem bindings Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- .../bindings/sound/samsung,odroidx2-max98090.txt | 35 ++++++++++++++++++++++ 1 file changed, 35 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt b/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt new file mode 100644 index 000000000000..9148f72319e1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt @@ -0,0 +1,35 @@ +Samsung Exynos Odroid X2/U3 audio complex with MAX98090 codec + +Required properties: + - compatible : "samsung,odroidx2-audio" - for Odroid X2 board, + "samsung,odroidu3-audio" - for Odroid U3 board + - samsung,model : the user-visible name of this sound complex + - samsung,i2s-controller : the phandle of the I2S controller + - samsung,audio-codec : the phandle of the MAX98090 audio codec + - samsung,audio-routing : a list of the connections between audio + components; each entry is a pair of strings, the first being the + connection's sink, the second being the connection's source; + valid names for sources and sinks are the MAX98090's pins (as + documented in its binding), and the jacks on the board + For Odroid X2: + * Headphone Jack + * Mic Jack + * DMIC + + For Odroid U3: + * Headphone Jack + * Speakers + +Example: + +sound { + compatible = "samsung,odroidu3-audio"; + samsung,i2s-controller = <&i2s0>; + samsung,audio-codec = <&max98090>; + samsung,model = "Odroid-X2"; + samsung,audio-routing = + "Headphone Jack", "HPL", + "Headphone Jack", "HPR", + "IN1", "Mic Jack", + "Mic Jack", "MICBIAS"; +}; -- cgit v1.2.3 From 892028acf020e5f9f8e8894b404400b360f5fe4f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 15 Jul 2014 16:58:59 +0200 Subject: ALSA: hda - Update model list documentation for IDT codecs Just add missing recent entries. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'Documentation') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index d1ab5e17eb13..a5e754714344 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -284,6 +284,11 @@ STAC92HD83* hp-zephyr HP Zephyr hp-led HP with broken BIOS for mute LED hp-inv-led HP with broken BIOS for inverted mute LED + hp-mic-led HP with mic-mute LED + headset-jack Dell Latitude with a 4-pin headset jack + hp-envy-bass Pin fixup for HP Envy bass speaker (NID 0x0f) + hp-envy-ts-bass Pin fixup for HP Envy TS bass speaker (NID 0x10) + hp-bnb13-eq Hardware equalizer setup for HP laptops auto BIOS setup (default) STAC92HD95 -- cgit v1.2.3 From 5df7f71d5cdfbcbfd7e1b68df9994609d33f7e58 Mon Sep 17 00:00:00 2001 From: Dan Murphy Date: Mon, 14 Jul 2014 15:10:45 -0500 Subject: ASoC: tas2552: Support TI TAS2552 Amplifier Support the TI TAS2552 Class D amplifier. The TAS2552 is a high efficiency Class-D audio power amplifier with advanced battery current management and an integrated Class-G boost The device constantly measures the current and voltage across the load and provides a digital stream of this information. Signed-off-by: Dan Murphy Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/tas2552.txt | 26 + include/sound/tas2552-plat.h | 25 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2552.c | 540 +++++++++++++++++++++ sound/soc/codecs/tas2552.h | 129 +++++ 6 files changed, 727 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2552.txt create mode 100644 include/sound/tas2552-plat.h create mode 100644 sound/soc/codecs/tas2552.c create mode 100644 sound/soc/codecs/tas2552.h (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/tas2552.txt b/Documentation/devicetree/bindings/sound/tas2552.txt new file mode 100644 index 000000000000..55e2a0af5645 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2552.txt @@ -0,0 +1,26 @@ +Texas Instruments - tas2552 Codec module + +The tas2552 serial control bus communicates through I2C protocols + +Required properties: + - compatible - One of: + "ti,tas2552" - TAS2552 + - reg - I2C slave address + - supply-*: Required supply regulators are: + "vbat" battery voltage + "iovdd" I/O Voltage + "avdd" Analog DAC Voltage + +Optional properties: + - enable-gpio - gpio pin to enable/disable the device + +Example: + +tas2552: tas2552@41 { + compatible = "ti,tas2552"; + reg = <0x41>; + enable-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; +}; + +For more product information please see the link below: +http://www.ti.com/product/TAS2552 diff --git a/include/sound/tas2552-plat.h b/include/sound/tas2552-plat.h new file mode 100644 index 000000000000..65e7627ba38e --- /dev/null +++ b/include/sound/tas2552-plat.h @@ -0,0 +1,25 @@ +/* + * TAS2552 driver platform header + * + * Copyright (C) 2014 Texas Instruments Inc. + * + * Author: Dan Murphy + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#ifndef TAS2552_PLAT_H +#define TAS2552_PLAT_H + +struct tas2552_platform_data { + int enable_gpio; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cbfa1e18f651..480881747ec5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -91,6 +91,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STA350 if I2C select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS + select SND_SOC_TAS2552 if I2C select SND_SOC_TAS5086 if I2C select SND_SOC_TLV320AIC23_I2C if I2C select SND_SOC_TLV320AIC23_SPI if SPI_MASTER @@ -513,6 +514,10 @@ config SND_SOC_STA529 config SND_SOC_STAC9766 tristate +config SND_SOC_TAS2552 + tristate "Texas Instruments TAS2552 Mono Audio amplifier" + depends on I2C + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index be3377b8d73f..d79de052ef8e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -160,6 +160,7 @@ snd-soc-wm-hubs-objs := wm_hubs.o # Amp snd-soc-max9877-objs := max9877.o snd-soc-tpa6130a2-objs := tpa6130a2.o +snd-soc-tas2552-objs := tas2552.o obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o obj-$(CONFIG_SND_SOC_AB8500_CODEC) += snd-soc-ab8500-codec.o @@ -251,6 +252,7 @@ obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STA350) += snd-soc-sta350.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o +obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c new file mode 100644 index 000000000000..f0760af5a21e --- /dev/null +++ b/sound/soc/codecs/tas2552.c @@ -0,0 +1,540 @@ +/* + * tas2552.c - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier + * + * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com + * + * Author: Dan Murphy + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include + +#include +#include +#include +#include +#include +#include + +#include "tas2552.h" + +static struct reg_default tas2552_reg_defs[] = { + {TAS2552_CFG_1, 0x22}, + {TAS2552_CFG_3, 0x80}, + {TAS2552_DOUT, 0x00}, + {TAS2552_OUTPUT_DATA, 0xc0}, + {TAS2552_PDM_CFG, 0x01}, + {TAS2552_PGA_GAIN, 0x00}, + {TAS2552_BOOST_PT_CTRL, 0x0f}, + {TAS2552_RESERVED_0D, 0x00}, + {TAS2552_LIMIT_RATE_HYS, 0x08}, + {TAS2552_CFG_2, 0xef}, + {TAS2552_SER_CTRL_1, 0x00}, + {TAS2552_SER_CTRL_2, 0x00}, + {TAS2552_PLL_CTRL_1, 0x10}, + {TAS2552_PLL_CTRL_2, 0x00}, + {TAS2552_PLL_CTRL_3, 0x00}, + {TAS2552_BTIP, 0x8f}, + {TAS2552_BTS_CTRL, 0x80}, + {TAS2552_LIMIT_RELEASE, 0x04}, + {TAS2552_LIMIT_INT_COUNT, 0x00}, + {TAS2552_EDGE_RATE_CTRL, 0x40}, + {TAS2552_VBAT_DATA, 0x00}, +}; + +#define TAS2552_NUM_SUPPLIES 3 +static const char *tas2552_supply_names[TAS2552_NUM_SUPPLIES] = { + "vbat", /* vbat voltage */ + "iovdd", /* I/O Voltage */ + "avdd", /* Analog DAC Voltage */ +}; + +struct tas2552_data { + struct snd_soc_codec *codec; + struct regmap *regmap; + struct i2c_client *tas2552_client; + struct regulator_bulk_data supplies[TAS2552_NUM_SUPPLIES]; + struct gpio_desc *enable_gpio; + unsigned char regs[TAS2552_VBAT_DATA]; + unsigned int mclk; +}; + +static void tas2552_sw_shutdown(struct tas2552_data *tas_data, int sw_shutdown) +{ + u8 cfg1_reg; + + if (sw_shutdown) + cfg1_reg = 0; + else + cfg1_reg = TAS2552_SWS_MASK; + + snd_soc_update_bits(tas_data->codec, TAS2552_CFG_1, + TAS2552_SWS_MASK, cfg1_reg); +} + +static int tas2552_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 tas2552_data *tas2552 = dev_get_drvdata(codec->dev); + int sample_rate, pll_clk; + int d; + u8 p, j; + + /* Turn on Class D amplifier */ + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN_MASK, + TAS2552_CLASSD_EN); + + if (!tas2552->mclk) + return -EINVAL; + + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, 0); + + if (tas2552->mclk == TAS2552_245MHZ_CLK || + tas2552->mclk == TAS2552_225MHZ_CLK) { + /* By pass the PLL configuration */ + snd_soc_update_bits(codec, TAS2552_PLL_CTRL_2, + TAS2552_PLL_BYPASS_MASK, + TAS2552_PLL_BYPASS); + } else { + /* Fill in the PLL control registers for J & D + * PLL_CLK = (.5 * freq * J.D) / 2^p + * Need to fill in J and D here based on incoming freq + */ + p = snd_soc_read(codec, TAS2552_PLL_CTRL_1); + p = (p >> 7); + sample_rate = params_rate(params); + + if (sample_rate == 48000) + pll_clk = TAS2552_245MHZ_CLK; + else if (sample_rate == 44100) + pll_clk = TAS2552_225MHZ_CLK; + else { + dev_vdbg(codec->dev, "Substream sample rate is not found %i\n", + params_rate(params)); + return -EINVAL; + } + + j = (pll_clk * 2 * (1 << p)) / tas2552->mclk; + d = (pll_clk * 2 * (1 << p)) % tas2552->mclk; + + snd_soc_update_bits(codec, TAS2552_PLL_CTRL_1, + TAS2552_PLL_J_MASK, j); + snd_soc_write(codec, TAS2552_PLL_CTRL_2, + (d >> 7) & TAS2552_PLL_D_UPPER_MASK); + snd_soc_write(codec, TAS2552_PLL_CTRL_3, + d & TAS2552_PLL_D_LOWER_MASK); + + } + + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, + TAS2552_PLL_ENABLE); + + return 0; +} + +static int tas2552_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + u8 serial_format; + u8 serial_control_mask; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + serial_format = 0x00; + break; + case SND_SOC_DAIFMT_CBS_CFM: + serial_format = TAS2552_WORD_CLK_MASK; + break; + case SND_SOC_DAIFMT_CBM_CFS: + serial_format = TAS2552_BIT_CLK_MASK; + break; + case SND_SOC_DAIFMT_CBM_CFM: + serial_format = (TAS2552_BIT_CLK_MASK | TAS2552_WORD_CLK_MASK); + break; + default: + dev_vdbg(codec->dev, "DAI Format master is not found\n"); + return -EINVAL; + } + + serial_control_mask = TAS2552_BIT_CLK_MASK | TAS2552_WORD_CLK_MASK; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + serial_format &= TAS2552_DAIFMT_I2S_MASK; + break; + case SND_SOC_DAIFMT_DSP_A: + serial_format |= TAS2552_DAIFMT_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + serial_format |= TAS2552_DAIFMT_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + serial_format |= TAS2552_DAIFMT_LEFT_J; + break; + default: + dev_vdbg(codec->dev, "DAI Format is not found\n"); + return -EINVAL; + } + + if (fmt & SND_SOC_DAIFMT_FORMAT_MASK) + serial_control_mask |= TAS2552_DATA_FORMAT_MASK; + + snd_soc_update_bits(codec, TAS2552_SER_CTRL_1, serial_control_mask, + serial_format); + + return 0; +} + +static int tas2552_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct tas2552_data *tas2552 = dev_get_drvdata(codec->dev); + + tas2552->mclk = freq; + + return 0; +} + +static int tas2552_mute(struct snd_soc_dai *dai, int mute) +{ + u8 cfg1_reg; + struct snd_soc_codec *codec = dai->codec; + + if (mute) + cfg1_reg = TAS2552_MUTE_MASK; + else + cfg1_reg = ~TAS2552_MUTE_MASK; + + snd_soc_update_bits(codec, TAS2552_CFG_1, TAS2552_MUTE_MASK, cfg1_reg); + + return 0; +} + +#ifdef CONFIG_PM_RUNTIME +static int tas2552_runtime_suspend(struct device *dev) +{ + struct tas2552_data *tas2552 = dev_get_drvdata(dev); + + tas2552_sw_shutdown(tas2552, 0); + + if (tas2552->enable_gpio) + gpiod_set_value(tas2552->enable_gpio, 0); + + regcache_cache_only(tas2552->regmap, true); + regcache_mark_dirty(tas2552->regmap); + + return 0; +} + +static int tas2552_runtime_resume(struct device *dev) +{ + struct tas2552_data *tas2552 = dev_get_drvdata(dev); + + if (tas2552->enable_gpio) + gpiod_set_value(tas2552->enable_gpio, 1); + + tas2552_sw_shutdown(tas2552, 1); + + regcache_cache_only(tas2552->regmap, false); + regcache_sync(tas2552->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops tas2552_pm = { + SET_RUNTIME_PM_OPS(tas2552_runtime_suspend, tas2552_runtime_resume, + NULL) +}; + +static void tas2552_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, 0); +} + +static struct snd_soc_dai_ops tas2552_speaker_dai_ops = { + .hw_params = tas2552_hw_params, + .set_sysclk = tas2552_set_dai_sysclk, + .set_fmt = tas2552_set_dai_fmt, + .shutdown = tas2552_shutdown, + .digital_mute = tas2552_mute, +}; + +/* Formats supported by TAS2552 driver. */ +#define TAS2552_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +/* TAS2552 dai structure. */ +static struct snd_soc_dai_driver tas2552_dai[] = { + { + .name = "tas2552-amplifier", + .playback = { + .stream_name = "Speaker", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = TAS2552_FORMATS, + }, + .ops = &tas2552_speaker_dai_ops, + }, +}; + +/* + * DAC digital volumes. From -7 to 24 dB in 1 dB steps + */ +static DECLARE_TLV_DB_SCALE(dac_tlv, -7, 100, 24); + +static const struct snd_kcontrol_new tas2552_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Driver Playback Volume", + TAS2552_PGA_GAIN, 0, 0x1f, 1, dac_tlv), +}; + +static const struct reg_default tas2552_init_regs[] = { + { TAS2552_RESERVED_0D, 0xc0 }, +}; + +static int tas2552_codec_probe(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + int ret; + + tas2552->codec = codec; + + ret = regulator_bulk_enable(ARRAY_SIZE(tas2552->supplies), + tas2552->supplies); + + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", + ret); + return ret; + } + + if (tas2552->enable_gpio) + gpiod_set_value(tas2552->enable_gpio, 1); + + ret = pm_runtime_get_sync(codec->dev); + if (ret < 0) { + dev_err(codec->dev, "Enabling device failed: %d\n", + ret); + goto probe_fail; + } + + snd_soc_write(codec, TAS2552_CFG_1, TAS2552_MUTE_MASK | + TAS2552_PLL_SRC_BCLK); + snd_soc_write(codec, TAS2552_CFG_3, TAS2552_I2S_OUT_SEL | + TAS2552_DIN_SRC_SEL_AVG_L_R | TAS2552_88_96KHZ); + snd_soc_write(codec, TAS2552_DOUT, TAS2552_PDM_DATA_I); + snd_soc_write(codec, TAS2552_OUTPUT_DATA, TAS2552_PDM_DATA_V_I | 0x8); + snd_soc_write(codec, TAS2552_PDM_CFG, TAS2552_PDM_BCLK_SEL); + snd_soc_write(codec, TAS2552_BOOST_PT_CTRL, TAS2552_APT_DELAY_200 | + TAS2552_APT_THRESH_2_1_7); + + ret = regmap_register_patch(tas2552->regmap, tas2552_init_regs, + ARRAY_SIZE(tas2552_init_regs)); + if (ret != 0) { + dev_err(codec->dev, "Failed to write init registers: %d\n", + ret); + goto patch_fail; + } + + snd_soc_write(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN | + TAS2552_BOOST_EN | TAS2552_APT_EN | + TAS2552_LIM_EN); + return 0; + +patch_fail: + pm_runtime_put(codec->dev); +probe_fail: + if (tas2552->enable_gpio) + gpiod_set_value(tas2552->enable_gpio, 0); + + regulator_bulk_disable(ARRAY_SIZE(tas2552->supplies), + tas2552->supplies); + return -EIO; +} + +static int tas2552_codec_remove(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + + if (tas2552->enable_gpio) + gpiod_set_value(tas2552->enable_gpio, 0); + + return 0; +}; + +#ifdef CONFIG_PM +static int tas2552_suspend(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = regulator_bulk_disable(ARRAY_SIZE(tas2552->supplies), + tas2552->supplies); + + if (ret != 0) + dev_err(codec->dev, "Failed to disable supplies: %d\n", + ret); + return 0; +} + +static int tas2552_resume(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(tas2552->supplies), + tas2552->supplies); + + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", + ret); + } + + return 0; +} +#else +#define tas2552_suspend NULL +#define tas2552_resume NULL +#endif + +static struct snd_soc_codec_driver soc_codec_dev_tas2552 = { + .probe = tas2552_codec_probe, + .remove = tas2552_codec_remove, + .suspend = tas2552_suspend, + .resume = tas2552_resume, + .controls = tas2552_snd_controls, + .num_controls = ARRAY_SIZE(tas2552_snd_controls), +}; + +static const struct regmap_config tas2552_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = TAS2552_MAX_REG, + .reg_defaults = tas2552_reg_defs, + .num_reg_defaults = ARRAY_SIZE(tas2552_reg_defs), + .cache_type = REGCACHE_RBTREE, +}; + +static int tas2552_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device *dev; + struct tas2552_data *data; + int ret; + int i; + + dev = &client->dev; + data = devm_kzalloc(&client->dev, sizeof(*data), GFP_KERNEL); + if (data == NULL) + return -ENOMEM; + + data->enable_gpio = devm_gpiod_get(dev, "enable"); + if (IS_ERR(data->enable_gpio)) { + ret = PTR_ERR(data->enable_gpio); + if (ret != -ENOENT && ret != -ENOSYS) + return ret; + + data->enable_gpio = NULL; + } else { + gpiod_direction_output(data->enable_gpio, 0); + } + + data->tas2552_client = client; + data->regmap = devm_regmap_init_i2c(client, &tas2552_regmap_config); + if (IS_ERR(data->regmap)) { + ret = PTR_ERR(data->regmap); + dev_err(&client->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(data->supplies); i++) + data->supplies[i].supply = tas2552_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(data->supplies), + data->supplies); + if (ret != 0) + dev_err(dev, "Failed to request supplies: %d\n", ret); + + pm_runtime_set_active(&client->dev); + pm_runtime_set_autosuspend_delay(&client->dev, 1000); + pm_runtime_use_autosuspend(&client->dev); + pm_runtime_enable(&client->dev); + pm_runtime_mark_last_busy(&client->dev); + pm_runtime_put_sync_autosuspend(&client->dev); + + dev_set_drvdata(&client->dev, data); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_dev_tas2552, + tas2552_dai, ARRAY_SIZE(tas2552_dai)); + if (ret < 0) + dev_err(&client->dev, "Failed to register codec: %d\n", ret); + + return 0; +} + +static int tas2552_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id tas2552_id[] = { + { "tas2552", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas2552_id); + +#if IS_ENABLED(CONFIG_OF) +static const struct of_device_id tas2552_of_match[] = { + { .compatible = "ti,tas2552", }, + {}, +}; +MODULE_DEVICE_TABLE(of, tas2552_of_match); +#endif + +static struct i2c_driver tas2552_i2c_driver = { + .driver = { + .name = "tas2552", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tas2552_of_match), + .pm = &tas2552_pm, + }, + .probe = tas2552_probe, + .remove = tas2552_i2c_remove, + .id_table = tas2552_id, +}; + +module_i2c_driver(tas2552_i2c_driver); + +MODULE_AUTHOR("Dan Muprhy "); +MODULE_DESCRIPTION("TAS2552 Audio amplifier driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2552.h b/sound/soc/codecs/tas2552.h new file mode 100644 index 000000000000..6cea8f31bf88 --- /dev/null +++ b/sound/soc/codecs/tas2552.h @@ -0,0 +1,129 @@ +/* + * tas2552.h - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier + * + * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com + * + * Author: Dan Murphy + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#ifndef __TAS2552_H__ +#define __TAS2552_H__ + +/* Register Address Map */ +#define TAS2552_DEVICE_STATUS 0x00 +#define TAS2552_CFG_1 0x01 +#define TAS2552_CFG_2 0x02 +#define TAS2552_CFG_3 0x03 +#define TAS2552_DOUT 0x04 +#define TAS2552_SER_CTRL_1 0x05 +#define TAS2552_SER_CTRL_2 0x06 +#define TAS2552_OUTPUT_DATA 0x07 +#define TAS2552_PLL_CTRL_1 0x08 +#define TAS2552_PLL_CTRL_2 0x09 +#define TAS2552_PLL_CTRL_3 0x0a +#define TAS2552_BTIP 0x0b +#define TAS2552_BTS_CTRL 0x0c +#define TAS2552_RESERVED_0D 0x0d +#define TAS2552_LIMIT_RATE_HYS 0x0e +#define TAS2552_LIMIT_RELEASE 0x0f +#define TAS2552_LIMIT_INT_COUNT 0x10 +#define TAS2552_PDM_CFG 0x11 +#define TAS2552_PGA_GAIN 0x12 +#define TAS2552_EDGE_RATE_CTRL 0x13 +#define TAS2552_BOOST_PT_CTRL 0x14 +#define TAS2552_VER_NUM 0x16 +#define TAS2552_VBAT_DATA 0x19 +#define TAS2552_MAX_REG 0x20 + +/* CFG1 Register Masks */ +#define TAS2552_MUTE_MASK (1 << 2) +#define TAS2552_SWS_MASK (1 << 1) +#define TAS2552_WCLK_MASK 0x07 +#define TAS2552_CLASSD_EN_MASK (1 << 7) + +/* CFG2 Register Masks */ +#define TAS2552_CLASSD_EN (1 << 7) +#define TAS2552_BOOST_EN (1 << 6) +#define TAS2552_APT_EN (1 << 5) +#define TAS2552_PLL_ENABLE (1 << 3) +#define TAS2552_LIM_EN (1 << 2) +#define TAS2552_IVSENSE_EN (1 << 1) + +/* CFG3 Register Masks */ +#define TAS2552_WORD_CLK_MASK (1 << 7) +#define TAS2552_BIT_CLK_MASK (1 << 6) +#define TAS2552_DATA_FORMAT_MASK (0x11 << 2) + +#define TAS2552_DAIFMT_I2S_MASK 0xf3 +#define TAS2552_DAIFMT_DSP (1 << 3) +#define TAS2552_DAIFMT_RIGHT_J (1 << 4) +#define TAS2552_DAIFMT_LEFT_J (0x11 << 3) + +#define TAS2552_PLL_SRC_MCLK 0x00 +#define TAS2552_PLL_SRC_BCLK (1 << 3) +#define TAS2552_PLL_SRC_IVCLKIN (1 << 4) +#define TAS2552_PLL_SRC_1_8_FIXED (0x11 << 3) + +#define TAS2552_DIN_SRC_SEL_MUTED 0x00 +#define TAS2552_DIN_SRC_SEL_LEFT (1 << 4) +#define TAS2552_DIN_SRC_SEL_RIGHT (1 << 5) +#define TAS2552_DIN_SRC_SEL_AVG_L_R (0x11 << 4) + +#define TAS2552_PDM_IN_SEL (1 << 5) +#define TAS2552_I2S_OUT_SEL (1 << 6) +#define TAS2552_ANALOG_IN_SEL (1 << 7) + +/* CFG3 WCLK Dividers */ +#define TAS2552_8KHZ 0x00 +#define TAS2552_11_12KHZ (1 << 1) +#define TAS2552_16KHZ (1 << 2) +#define TAS2552_22_24KHZ (1 << 3) +#define TAS2552_32KHZ (1 << 4) +#define TAS2552_44_48KHZ (1 << 5) +#define TAS2552_88_96KHZ (1 << 6) +#define TAS2552_176_192KHZ (1 << 7) + +/* OUTPUT_DATA register */ +#define TAS2552_PDM_DATA_I 0x00 +#define TAS2552_PDM_DATA_V (1 << 6) +#define TAS2552_PDM_DATA_I_V (1 << 7) +#define TAS2552_PDM_DATA_V_I (0x11 << 6) + +/* PDM CFG Register */ +#define TAS2552_PDM_DATA_ES_RISE 0x4 + +#define TAS2552_PDM_PLL_CLK_SEL 0x00 +#define TAS2552_PDM_IV_CLK_SEL (1 << 1) +#define TAS2552_PDM_BCLK_SEL (1 << 2) +#define TAS2552_PDM_MCLK_SEL (1 << 3) + +/* Boost pass-through register */ +#define TAS2552_APT_DELAY_50 0x00 +#define TAS2552_APT_DELAY_75 (1 << 1) +#define TAS2552_APT_DELAY_125 (1 << 2) +#define TAS2552_APT_DELAY_200 (1 << 3) + +#define TAS2552_APT_THRESH_2_5 0x00 +#define TAS2552_APT_THRESH_1_7 (1 << 3) +#define TAS2552_APT_THRESH_1_4_1_1 (1 << 4) +#define TAS2552_APT_THRESH_2_1_7 (0x11 << 2) + +/* PLL Control Register */ +#define TAS2552_245MHZ_CLK 24576000 +#define TAS2552_225MHZ_CLK 22579200 +#define TAS2552_PLL_J_MASK 0x7f +#define TAS2552_PLL_D_UPPER_MASK 0x3f +#define TAS2552_PLL_D_LOWER_MASK 0xff +#define TAS2552_PLL_BYPASS_MASK 0x80 +#define TAS2552_PLL_BYPASS 0x80 + +#endif -- cgit v1.2.3 From 3117bb3109dc223e186302f5dc8ce9ed04adca90 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 29 Jul 2014 18:08:53 +0800 Subject: ASoC: fsl_asrc: Add ASRC ASoC CPU DAI and platform drivers The Asynchronous Sample Rate Converter (ASRC) converts the sampling rate of a signal associated with an input clock into a signal associated with a different output clock. The driver currently works as a Front End of DPCM with other Back Ends DAI links such as ESAI<->CS42888 and SSI<->WM8962 and SAI. It converts the original sample rate to a common rate supported by Back Ends for playback while converts the common rate of Back Ends to a desired rate for capture. It has 3 pairs to support three different substreams within totally 10 channels. Signed-off-by: Nicolin Chen Reviewed-by: Varka Bhadram Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/fsl,asrc.txt | 60 ++ sound/soc/fsl/Kconfig | 9 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_asrc.c | 992 +++++++++++++++++++++ sound/soc/fsl/fsl_asrc.h | 461 ++++++++++ sound/soc/fsl/fsl_asrc_dma.c | 386 ++++++++ 6 files changed, 1910 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,asrc.txt create mode 100644 sound/soc/fsl/fsl_asrc.c create mode 100644 sound/soc/fsl/fsl_asrc.h create mode 100644 sound/soc/fsl/fsl_asrc_dma.c (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/fsl,asrc.txt b/Documentation/devicetree/bindings/sound/fsl,asrc.txt new file mode 100644 index 000000000000..b93362a570be --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,asrc.txt @@ -0,0 +1,60 @@ +Freescale Asynchronous Sample Rate Converter (ASRC) Controller + +The Asynchronous Sample Rate Converter (ASRC) converts the sampling rate of a +signal associated with an input clock into a signal associated with a different +output clock. The driver currently works as a Front End of DPCM with other Back +Ends Audio controller such as ESAI, SSI and SAI. It has three pairs to support +three substreams within totally 10 channels. + +Required properties: + + - compatible : Contains "fsl,imx35-asrc" or "fsl,imx53-asrc". + + - reg : Offset and length of the register set for the device. + + - interrupts : Contains the spdif interrupt. + + - dmas : Generic dma devicetree binding as described in + Documentation/devicetree/bindings/dma/dma.txt. + + - dma-names : Contains "rxa", "rxb", "rxc", "txa", "txb" and "txc". + + - clocks : Contains an entry for each entry in clock-names. + + - clock-names : Contains the following entries + "mem" Peripheral access clock to access registers. + "ipg" Peripheral clock to driver module. + "asrck_<0-f>" Clock sources for input and output clock. + + - big-endian : If this property is absent, the little endian mode + will be in use as default. Otherwise, the big endian + mode will be in use for all the device registers. + + - fsl,asrc-rate : Defines a mutual sample rate used by DPCM Back Ends. + + - fsl,asrc-width : Defines a mutual sample width used by DPCM Back Ends. + +Example: + +asrc: asrc@02034000 { + compatible = "fsl,imx53-asrc"; + reg = <0x02034000 0x4000>; + interrupts = <0 50 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clks 107>, <&clks 107>, <&clks 0>, + <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>, + <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>, + <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>, + <&clks 107>, <&clks 0>, <&clks 0>; + clock-names = "mem", "ipg", "asrck0", + "asrck_1", "asrck_2", "asrck_3", "asrck_4", + "asrck_5", "asrck_6", "asrck_7", "asrck_8", + "asrck_9", "asrck_a", "asrck_b", "asrck_c", + "asrck_d", "asrck_e", "asrck_f"; + dmas = <&sdma 17 23 1>, <&sdma 18 23 1>, <&sdma 19 23 1>, + <&sdma 20 23 1>, <&sdma 21 23 1>, <&sdma 22 23 1>; + dma-names = "rxa", "rxb", "rxc", + "txa", "txb", "txc"; + fsl,asrc-rate = <48000>; + fsl,asrc-width = <16>; + status = "okay"; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 37933629cbed..56fd32b3a881 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -2,6 +2,15 @@ menu "SoC Audio for Freescale CPUs" comment "Common SoC Audio options for Freescale CPUs:" +config SND_SOC_FSL_ASRC + tristate "Asynchronous Sample Rate Converter (ASRC) module support" + select REGMAP_MMIO + help + Say Y if you want to add Asynchronous Sample Rate Converter (ASRC) + support for the Freescale CPUs. + This option is only useful for out-of-tree drivers since + in-tree drivers select it automatically. + config SND_SOC_FSL_SAI tristate "Synchronous Audio Interface (SAI) module support" select REGMAP_MMIO diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index db254e358c18..9ff59267eac9 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -11,6 +11,7 @@ snd-soc-p1022-rdk-objs := p1022_rdk.o obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o # Freescale SSI/DMA/SAI/SPDIF Support +snd-soc-fsl-asrc-objs := fsl_asrc.o fsl_asrc_dma.o snd-soc-fsl-sai-objs := fsl_sai.o snd-soc-fsl-ssi-y := fsl_ssi.o snd-soc-fsl-ssi-$(CONFIG_DEBUG_FS) += fsl_ssi_dbg.o @@ -18,6 +19,7 @@ snd-soc-fsl-spdif-objs := fsl_spdif.o snd-soc-fsl-esai-objs := fsl_esai.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o +obj-$(CONFIG_SND_SOC_FSL_ASRC) += snd-soc-fsl-asrc.o obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c new file mode 100644 index 000000000000..27a4a7084343 --- /dev/null +++ b/sound/soc/fsl/fsl_asrc.c @@ -0,0 +1,992 @@ +/* + * Freescale ASRC ALSA SoC Digital Audio Interface (DAI) driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "fsl_asrc.h" + +#define IDEAL_RATIO_DECIMAL_DEPTH 26 + +#define pair_err(fmt, ...) \ + dev_err(&asrc_priv->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__) + +#define pair_dbg(fmt, ...) \ + dev_dbg(&asrc_priv->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__) + +/* Sample rates are aligned with that defined in pcm.h file */ +static const u8 process_option[][8][2] = { + /* 32kHz 44.1kHz 48kHz 64kHz 88.2kHz 96kHz 176kHz 192kHz */ + {{0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 5512Hz */ + {{0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 8kHz */ + {{0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 11025Hz */ + {{0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 16kHz */ + {{0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 22050Hz */ + {{0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0},}, /* 32kHz */ + {{0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0},}, /* 44.1kHz */ + {{0, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0},}, /* 48kHz */ + {{1, 2}, {0, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0},}, /* 64kHz */ + {{1, 2}, {1, 2}, {1, 2}, {1, 1}, {1, 1}, {1, 1}, {1, 1}, {1, 1},}, /* 88.2kHz */ + {{1, 2}, {1, 2}, {1, 2}, {1, 1}, {1, 1}, {1, 1}, {1, 1}, {1, 1},}, /* 96kHz */ + {{2, 2}, {2, 2}, {2, 2}, {2, 1}, {2, 1}, {2, 1}, {2, 1}, {2, 1},}, /* 176kHz */ + {{2, 2}, {2, 2}, {2, 2}, {2, 1}, {2, 1}, {2, 1}, {2, 1}, {2, 1},}, /* 192kHz */ +}; + +/* Corresponding to process_option */ +static int supported_input_rate[] = { + 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, 88200, + 96000, 176400, 192000, +}; + +static int supported_asrc_rate[] = { + 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000, +}; + +/** + * The following tables map the relationship between asrc_inclk/asrc_outclk in + * fsl_asrc.h and the registers of ASRCSR + */ +static unsigned char input_clk_map_imx35[] = { + 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0xa, 0xb, 0xc, 0xd, 0xe, 0xf, +}; + +static unsigned char output_clk_map_imx35[] = { + 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0xa, 0xb, 0xc, 0xd, 0xe, 0xf, +}; + +/* i.MX53 uses the same map for input and output */ +static unsigned char input_clk_map_imx53[] = { +/* 0x0 0x1 0x2 0x3 0x4 0x5 0x6 0x7 0x8 0x9 0xa 0xb 0xc 0xd 0xe 0xf */ + 0x0, 0x1, 0x2, 0x7, 0x4, 0x5, 0x6, 0x3, 0x8, 0x9, 0xa, 0xb, 0xc, 0xf, 0xe, 0xd, +}; + +static unsigned char output_clk_map_imx53[] = { +/* 0x0 0x1 0x2 0x3 0x4 0x5 0x6 0x7 0x8 0x9 0xa 0xb 0xc 0xd 0xe 0xf */ + 0x8, 0x9, 0xa, 0x7, 0xc, 0x5, 0x6, 0xb, 0x0, 0x1, 0x2, 0x3, 0x4, 0xf, 0xe, 0xd, +}; + +static unsigned char *clk_map[2]; + +/** + * Request ASRC pair + * + * It assigns pair by the order of A->C->B because allocation of pair B, + * within range [ANCA, ANCA+ANCB-1], depends on the channels of pair A + * while pair A and pair C are comparatively independent. + */ +static int fsl_asrc_request_pair(int channels, struct fsl_asrc_pair *pair) +{ + enum asrc_pair_index index = ASRC_INVALID_PAIR; + struct fsl_asrc *asrc_priv = pair->asrc_priv; + struct device *dev = &asrc_priv->pdev->dev; + unsigned long lock_flags; + int i, ret = 0; + + spin_lock_irqsave(&asrc_priv->lock, lock_flags); + + for (i = ASRC_PAIR_A; i < ASRC_PAIR_MAX_NUM; i++) { + if (asrc_priv->pair[i] != NULL) + continue; + + index = i; + + if (i != ASRC_PAIR_B) + break; + } + + if (index == ASRC_INVALID_PAIR) { + dev_err(dev, "all pairs are busy now\n"); + ret = -EBUSY; + } else if (asrc_priv->channel_avail < channels) { + dev_err(dev, "can't afford required channels: %d\n", channels); + ret = -EINVAL; + } else { + asrc_priv->channel_avail -= channels; + asrc_priv->pair[index] = pair; + pair->channels = channels; + pair->index = index; + } + + spin_unlock_irqrestore(&asrc_priv->lock, lock_flags); + + return ret; +} + +/** + * Release ASRC pair + * + * It clears the resource from asrc_priv and releases the occupied channels. + */ +static void fsl_asrc_release_pair(struct fsl_asrc_pair *pair) +{ + struct fsl_asrc *asrc_priv = pair->asrc_priv; + enum asrc_pair_index index = pair->index; + unsigned long lock_flags; + + /* Make sure the pair is disabled */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_ASRCEi_MASK(index), 0); + + spin_lock_irqsave(&asrc_priv->lock, lock_flags); + + asrc_priv->channel_avail += pair->channels; + asrc_priv->pair[index] = NULL; + pair->error = 0; + + spin_unlock_irqrestore(&asrc_priv->lock, lock_flags); +} + +/** + * Configure input and output thresholds + */ +static void fsl_asrc_set_watermarks(struct fsl_asrc_pair *pair, u32 in, u32 out) +{ + struct fsl_asrc *asrc_priv = pair->asrc_priv; + enum asrc_pair_index index = pair->index; + + regmap_update_bits(asrc_priv->regmap, REG_ASRMCR(index), + ASRMCRi_EXTTHRSHi_MASK | + ASRMCRi_INFIFO_THRESHOLD_MASK | + ASRMCRi_OUTFIFO_THRESHOLD_MASK, + ASRMCRi_EXTTHRSHi | + ASRMCRi_INFIFO_THRESHOLD(in) | + ASRMCRi_OUTFIFO_THRESHOLD(out)); +} + +/** + * Calculate the total divisor between asrck clock rate and sample rate + * + * It follows the formula clk_rate = samplerate * (2 ^ prescaler) * divider + */ +static u32 fsl_asrc_cal_asrck_divisor(struct fsl_asrc_pair *pair, u32 div) +{ + u32 ps; + + /* Calculate the divisors: prescaler [2^0, 2^7], divder [1, 8] */ + for (ps = 0; div > 8; ps++) + div >>= 1; + + return ((div - 1) << ASRCDRi_AxCPi_WIDTH) | ps; +} + +/** + * Calculate and set the ratio for Ideal Ratio mode only + * + * The ratio is a 32-bit fixed point value with 26 fractional bits. + */ +static int fsl_asrc_set_ideal_ratio(struct fsl_asrc_pair *pair, + int inrate, int outrate) +{ + struct fsl_asrc *asrc_priv = pair->asrc_priv; + enum asrc_pair_index index = pair->index; + unsigned long ratio; + int i; + + if (!outrate) { + pair_err("output rate should not be zero\n"); + return -EINVAL; + } + + /* Calculate the intergal part of the ratio */ + ratio = (inrate / outrate) << IDEAL_RATIO_DECIMAL_DEPTH; + + /* ... and then the 26 depth decimal part */ + inrate %= outrate; + + for (i = 1; i <= IDEAL_RATIO_DECIMAL_DEPTH; i++) { + inrate <<= 1; + + if (inrate < outrate) + continue; + + ratio |= 1 << (IDEAL_RATIO_DECIMAL_DEPTH - i); + inrate -= outrate; + + if (!inrate) + break; + } + + regmap_write(asrc_priv->regmap, REG_ASRIDRL(index), ratio); + regmap_write(asrc_priv->regmap, REG_ASRIDRH(index), ratio >> 24); + + return 0; +} + +/** + * Configure the assigned ASRC pair + * + * It configures those ASRC registers according to a configuration instance + * of struct asrc_config which includes in/output sample rate, width, channel + * and clock settings. + */ +static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) +{ + struct asrc_config *config = pair->config; + struct fsl_asrc *asrc_priv = pair->asrc_priv; + enum asrc_pair_index index = pair->index; + u32 inrate = config->input_sample_rate, indiv; + u32 outrate = config->output_sample_rate, outdiv; + bool ideal = config->inclk == INCLK_NONE; + u32 clk_index[2], div[2]; + int in, out, channels; + struct clk *clk; + + if (!config) { + pair_err("invalid pair config\n"); + return -EINVAL; + } + + /* Validate channels */ + if (config->channel_num < 1 || config->channel_num > 10) { + pair_err("does not support %d channels\n", config->channel_num); + return -EINVAL; + } + + /* Validate output width */ + if (config->output_word_width == ASRC_WIDTH_8_BIT) { + pair_err("does not support 8bit width output\n"); + return -EINVAL; + } + + /* Validate input and output sample rates */ + for (in = 0; in < ARRAY_SIZE(supported_input_rate); in++) + if (inrate == supported_input_rate[in]) + break; + + if (in == ARRAY_SIZE(supported_input_rate)) { + pair_err("unsupported input sample rate: %dHz\n", inrate); + return -EINVAL; + } + + for (out = 0; out < ARRAY_SIZE(supported_asrc_rate); out++) + if (outrate == supported_asrc_rate[out]) + break; + + if (out == ARRAY_SIZE(supported_asrc_rate)) { + pair_err("unsupported output sample rate: %dHz\n", outrate); + return -EINVAL; + } + + /* Validate input and output clock sources */ + clk_index[IN] = clk_map[IN][config->inclk]; + clk_index[OUT] = clk_map[OUT][config->outclk]; + + /* We only have output clock for ideal ratio mode */ + clk = asrc_priv->asrck_clk[clk_index[ideal ? OUT : IN]]; + + div[IN] = clk_get_rate(clk) / inrate; + if (div[IN] == 0) { + pair_err("failed to support input sample rate %dHz by asrck_%x\n", + inrate, clk_index[ideal ? OUT : IN]); + return -EINVAL; + } + + clk = asrc_priv->asrck_clk[clk_index[OUT]]; + + /* Use fixed output rate for Ideal Ratio mode (INCLK_NONE) */ + if (ideal) + div[OUT] = clk_get_rate(clk) / IDEAL_RATIO_RATE; + else + div[OUT] = clk_get_rate(clk) / outrate; + + if (div[OUT] == 0) { + pair_err("failed to support output sample rate %dHz by asrck_%x\n", + outrate, clk_index[OUT]); + return -EINVAL; + } + + /* Set the channel number */ + channels = config->channel_num; + + if (asrc_priv->channel_bits < 4) + channels /= 2; + + /* Update channels for current pair */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCNCR, + ASRCNCR_ANCi_MASK(index, asrc_priv->channel_bits), + ASRCNCR_ANCi(index, channels, asrc_priv->channel_bits)); + + /* Default setting: Automatic selection for processing mode */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_ATSi_MASK(index), ASRCTR_ATS(index)); + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_USRi_MASK(index), 0); + + /* Set the input and output clock sources */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCSR, + ASRCSR_AICSi_MASK(index) | ASRCSR_AOCSi_MASK(index), + ASRCSR_AICS(index, clk_index[IN]) | + ASRCSR_AOCS(index, clk_index[OUT])); + + /* Calculate the input clock divisors */ + indiv = fsl_asrc_cal_asrck_divisor(pair, div[IN]); + outdiv = fsl_asrc_cal_asrck_divisor(pair, div[OUT]); + + /* Suppose indiv and outdiv includes prescaler, so add its MASK too */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCDR(index), + ASRCDRi_AOCPi_MASK(index) | ASRCDRi_AICPi_MASK(index) | + ASRCDRi_AOCDi_MASK(index) | ASRCDRi_AICDi_MASK(index), + ASRCDRi_AOCP(index, outdiv) | ASRCDRi_AICP(index, indiv)); + + /* Implement word_width configurations */ + regmap_update_bits(asrc_priv->regmap, REG_ASRMCR1(index), + ASRMCR1i_OW16_MASK | ASRMCR1i_IWD_MASK, + ASRMCR1i_OW16(config->output_word_width) | + ASRMCR1i_IWD(config->input_word_width)); + + /* Enable BUFFER STALL */ + regmap_update_bits(asrc_priv->regmap, REG_ASRMCR(index), + ASRMCRi_BUFSTALLi_MASK, ASRMCRi_BUFSTALLi); + + /* Set default thresholds for input and output FIFO */ + fsl_asrc_set_watermarks(pair, ASRC_INPUTFIFO_THRESHOLD, + ASRC_INPUTFIFO_THRESHOLD); + + /* Configure the followings only for Ideal Ratio mode */ + if (!ideal) + return 0; + + /* Clear ASTSx bit to use Ideal Ratio mode */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_ATSi_MASK(index), 0); + + /* Enable Ideal Ratio mode */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_IDRi_MASK(index) | ASRCTR_USRi_MASK(index), + ASRCTR_IDR(index) | ASRCTR_USR(index)); + + /* Apply configurations for pre- and post-processing */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCFG, + ASRCFG_PREMODi_MASK(index) | ASRCFG_POSTMODi_MASK(index), + ASRCFG_PREMOD(index, process_option[in][out][0]) | + ASRCFG_POSTMOD(index, process_option[in][out][1])); + + return fsl_asrc_set_ideal_ratio(pair, inrate, outrate); +} + +/** + * Start the assigned ASRC pair + * + * It enables the assigned pair and makes it stopped at the stall level. + */ +static void fsl_asrc_start_pair(struct fsl_asrc_pair *pair) +{ + struct fsl_asrc *asrc_priv = pair->asrc_priv; + enum asrc_pair_index index = pair->index; + int reg, retry = 10, i; + + /* Enable the current pair */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_ASRCEi_MASK(index), ASRCTR_ASRCE(index)); + + /* Wait for status of initialization */ + do { + udelay(5); + regmap_read(asrc_priv->regmap, REG_ASRCFG, ®); + reg &= ASRCFG_INIRQi_MASK(index); + } while (!reg && --retry); + + /* Make the input fifo to ASRC STALL level */ + regmap_read(asrc_priv->regmap, REG_ASRCNCR, ®); + for (i = 0; i < pair->channels * 4; i++) + regmap_write(asrc_priv->regmap, REG_ASRDI(index), 0); + + /* Enable overload interrupt */ + regmap_write(asrc_priv->regmap, REG_ASRIER, ASRIER_AOLIE); +} + +/** + * Stop the assigned ASRC pair + */ +static void fsl_asrc_stop_pair(struct fsl_asrc_pair *pair) +{ + struct fsl_asrc *asrc_priv = pair->asrc_priv; + enum asrc_pair_index index = pair->index; + + /* Stop the current pair */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_ASRCEi_MASK(index), 0); +} + +/** + * Get DMA channel according to the pair and direction. + */ +struct dma_chan *fsl_asrc_get_dma_channel(struct fsl_asrc_pair *pair, bool dir) +{ + struct fsl_asrc *asrc_priv = pair->asrc_priv; + enum asrc_pair_index index = pair->index; + char name[4]; + + sprintf(name, "%cx%c", dir == IN ? 'r' : 't', index + 'a'); + + return dma_request_slave_channel(&asrc_priv->pdev->dev, name); +} +EXPORT_SYMBOL_GPL(fsl_asrc_get_dma_channel); + +static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct fsl_asrc *asrc_priv = snd_soc_dai_get_drvdata(dai); + int width = snd_pcm_format_width(params_format(params)); + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + unsigned int channels = params_channels(params); + unsigned int rate = params_rate(params); + struct asrc_config config; + int word_width, ret; + + ret = fsl_asrc_request_pair(channels, pair); + if (ret) { + dev_err(dai->dev, "fail to request asrc pair\n"); + return ret; + } + + pair->config = &config; + + if (width == 16) + width = ASRC_WIDTH_16_BIT; + else + width = ASRC_WIDTH_24_BIT; + + if (asrc_priv->asrc_width == 16) + word_width = ASRC_WIDTH_16_BIT; + else + word_width = ASRC_WIDTH_24_BIT; + + config.pair = pair->index; + config.channel_num = channels; + config.inclk = INCLK_NONE; + config.outclk = OUTCLK_ASRCK1_CLK; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + config.input_word_width = width; + config.output_word_width = word_width; + config.input_sample_rate = rate; + config.output_sample_rate = asrc_priv->asrc_rate; + } else { + config.input_word_width = word_width; + config.output_word_width = width; + config.input_sample_rate = asrc_priv->asrc_rate; + config.output_sample_rate = rate; + } + + ret = fsl_asrc_config_pair(pair); + if (ret) { + dev_err(dai->dev, "fail to config asrc pair\n"); + return ret; + } + + return 0; +} + +static int fsl_asrc_dai_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + + if (pair) + fsl_asrc_release_pair(pair); + + return 0; +} + +static int fsl_asrc_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + fsl_asrc_start_pair(pair); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + fsl_asrc_stop_pair(pair); + break; + default: + return -EINVAL; + } + + return 0; +} + +static struct snd_soc_dai_ops fsl_asrc_dai_ops = { + .hw_params = fsl_asrc_dai_hw_params, + .hw_free = fsl_asrc_dai_hw_free, + .trigger = fsl_asrc_dai_trigger, +}; + +static int fsl_asrc_dai_probe(struct snd_soc_dai *dai) +{ + struct fsl_asrc *asrc_priv = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, &asrc_priv->dma_params_tx, + &asrc_priv->dma_params_rx); + + return 0; +} + +#define FSL_ASRC_RATES SNDRV_PCM_RATE_8000_192000 +#define FSL_ASRC_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FORMAT_S20_3LE) + +static struct snd_soc_dai_driver fsl_asrc_dai = { + .probe = fsl_asrc_dai_probe, + .playback = { + .stream_name = "ASRC-Playback", + .channels_min = 1, + .channels_max = 10, + .rates = FSL_ASRC_RATES, + .formats = FSL_ASRC_FORMATS, + }, + .capture = { + .stream_name = "ASRC-Capture", + .channels_min = 1, + .channels_max = 10, + .rates = FSL_ASRC_RATES, + .formats = FSL_ASRC_FORMATS, + }, + .ops = &fsl_asrc_dai_ops, +}; + +static const struct snd_soc_component_driver fsl_asrc_component = { + .name = "fsl-asrc-dai", +}; + +static bool fsl_asrc_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case REG_ASRCTR: + case REG_ASRIER: + case REG_ASRCNCR: + case REG_ASRCFG: + case REG_ASRCSR: + case REG_ASRCDR1: + case REG_ASRCDR2: + case REG_ASRSTR: + case REG_ASRPM1: + case REG_ASRPM2: + case REG_ASRPM3: + case REG_ASRPM4: + case REG_ASRPM5: + case REG_ASRTFR1: + case REG_ASRCCR: + case REG_ASRDOA: + case REG_ASRDOB: + case REG_ASRDOC: + case REG_ASRIDRHA: + case REG_ASRIDRLA: + case REG_ASRIDRHB: + case REG_ASRIDRLB: + case REG_ASRIDRHC: + case REG_ASRIDRLC: + case REG_ASR76K: + case REG_ASR56K: + case REG_ASRMCRA: + case REG_ASRFSTA: + case REG_ASRMCRB: + case REG_ASRFSTB: + case REG_ASRMCRC: + case REG_ASRFSTC: + case REG_ASRMCR1A: + case REG_ASRMCR1B: + case REG_ASRMCR1C: + return true; + default: + return false; + } +} + +static bool fsl_asrc_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case REG_ASRSTR: + case REG_ASRDIA: + case REG_ASRDIB: + case REG_ASRDIC: + case REG_ASRDOA: + case REG_ASRDOB: + case REG_ASRDOC: + case REG_ASRFSTA: + case REG_ASRFSTB: + case REG_ASRFSTC: + case REG_ASRCFG: + return true; + default: + return false; + } +} + +static bool fsl_asrc_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case REG_ASRCTR: + case REG_ASRIER: + case REG_ASRCNCR: + case REG_ASRCFG: + case REG_ASRCSR: + case REG_ASRCDR1: + case REG_ASRCDR2: + case REG_ASRSTR: + case REG_ASRPM1: + case REG_ASRPM2: + case REG_ASRPM3: + case REG_ASRPM4: + case REG_ASRPM5: + case REG_ASRTFR1: + case REG_ASRCCR: + case REG_ASRDIA: + case REG_ASRDIB: + case REG_ASRDIC: + case REG_ASRIDRHA: + case REG_ASRIDRLA: + case REG_ASRIDRHB: + case REG_ASRIDRLB: + case REG_ASRIDRHC: + case REG_ASRIDRLC: + case REG_ASR76K: + case REG_ASR56K: + case REG_ASRMCRA: + case REG_ASRMCRB: + case REG_ASRMCRC: + case REG_ASRMCR1A: + case REG_ASRMCR1B: + case REG_ASRMCR1C: + return true; + default: + return false; + } +} + +static struct regmap_config fsl_asrc_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + + .max_register = REG_ASRMCR1C, + .readable_reg = fsl_asrc_readable_reg, + .volatile_reg = fsl_asrc_volatile_reg, + .writeable_reg = fsl_asrc_writeable_reg, + .cache_type = REGCACHE_RBTREE, +}; + +/** + * Initialize ASRC registers with a default configurations + */ +static int fsl_asrc_init(struct fsl_asrc *asrc_priv) +{ + /* Halt ASRC internal FP when input FIFO needs data for pair A, B, C */ + regmap_write(asrc_priv->regmap, REG_ASRCTR, ASRCTR_ASRCEN); + + /* Disable interrupt by default */ + regmap_write(asrc_priv->regmap, REG_ASRIER, 0x0); + + /* Apply recommended settings for parameters from Reference Manual */ + regmap_write(asrc_priv->regmap, REG_ASRPM1, 0x7fffff); + regmap_write(asrc_priv->regmap, REG_ASRPM2, 0x255555); + regmap_write(asrc_priv->regmap, REG_ASRPM3, 0xff7280); + regmap_write(asrc_priv->regmap, REG_ASRPM4, 0xff7280); + regmap_write(asrc_priv->regmap, REG_ASRPM5, 0xff7280); + + /* Base address for task queue FIFO. Set to 0x7C */ + regmap_update_bits(asrc_priv->regmap, REG_ASRTFR1, + ASRTFR1_TF_BASE_MASK, ASRTFR1_TF_BASE(0xfc)); + + /* Set the processing clock for 76KHz to 133M */ + regmap_write(asrc_priv->regmap, REG_ASR76K, 0x06D6); + + /* Set the processing clock for 56KHz to 133M */ + return regmap_write(asrc_priv->regmap, REG_ASR56K, 0x0947); +} + +/** + * Interrupt handler for ASRC + */ +static irqreturn_t fsl_asrc_isr(int irq, void *dev_id) +{ + struct fsl_asrc *asrc_priv = (struct fsl_asrc *)dev_id; + struct device *dev = &asrc_priv->pdev->dev; + enum asrc_pair_index index; + u32 status; + + regmap_read(asrc_priv->regmap, REG_ASRSTR, &status); + + /* Clean overload error */ + regmap_write(asrc_priv->regmap, REG_ASRSTR, ASRSTR_AOLE); + + /* + * We here use dev_dbg() for all exceptions because ASRC itself does + * not care if FIFO overflowed or underrun while a warning in the + * interrupt would result a ridged conversion. + */ + for (index = ASRC_PAIR_A; index < ASRC_PAIR_MAX_NUM; index++) { + if (!asrc_priv->pair[index]) + continue; + + if (status & ASRSTR_ATQOL) { + asrc_priv->pair[index]->error |= ASRC_TASK_Q_OVERLOAD; + dev_dbg(dev, "ASRC Task Queue FIFO overload\n"); + } + + if (status & ASRSTR_AOOL(index)) { + asrc_priv->pair[index]->error |= ASRC_OUTPUT_TASK_OVERLOAD; + pair_dbg("Output Task Overload\n"); + } + + if (status & ASRSTR_AIOL(index)) { + asrc_priv->pair[index]->error |= ASRC_INPUT_TASK_OVERLOAD; + pair_dbg("Input Task Overload\n"); + } + + if (status & ASRSTR_AODO(index)) { + asrc_priv->pair[index]->error |= ASRC_OUTPUT_BUFFER_OVERFLOW; + pair_dbg("Output Data Buffer has overflowed\n"); + } + + if (status & ASRSTR_AIDU(index)) { + asrc_priv->pair[index]->error |= ASRC_INPUT_BUFFER_UNDERRUN; + pair_dbg("Input Data Buffer has underflowed\n"); + } + } + + return IRQ_HANDLED; +} + +static int fsl_asrc_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct fsl_asrc *asrc_priv; + struct resource *res; + void __iomem *regs; + int irq, ret, i; + char tmp[16]; + + asrc_priv = devm_kzalloc(&pdev->dev, sizeof(*asrc_priv), GFP_KERNEL); + if (!asrc_priv) + return -ENOMEM; + + asrc_priv->pdev = pdev; + strcpy(asrc_priv->name, np->name); + + /* Get the addresses and IRQ */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + asrc_priv->paddr = res->start; + + /* Register regmap and let it prepare core clock */ + if (of_property_read_bool(np, "big-endian")) + fsl_asrc_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; + + asrc_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "mem", regs, + &fsl_asrc_regmap_config); + if (IS_ERR(asrc_priv->regmap)) { + dev_err(&pdev->dev, "failed to init regmap\n"); + return PTR_ERR(asrc_priv->regmap); + } + + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); + return irq; + } + + ret = devm_request_irq(&pdev->dev, irq, fsl_asrc_isr, 0, + asrc_priv->name, asrc_priv); + if (ret) { + dev_err(&pdev->dev, "failed to claim irq %u: %d\n", irq, ret); + return ret; + } + + asrc_priv->mem_clk = devm_clk_get(&pdev->dev, "mem"); + if (IS_ERR(asrc_priv->mem_clk)) { + dev_err(&pdev->dev, "failed to get mem clock\n"); + return PTR_ERR(asrc_priv->ipg_clk); + } + + asrc_priv->ipg_clk = devm_clk_get(&pdev->dev, "ipg"); + if (IS_ERR(asrc_priv->ipg_clk)) { + dev_err(&pdev->dev, "failed to get ipg clock\n"); + return PTR_ERR(asrc_priv->ipg_clk); + } + + for (i = 0; i < ASRC_CLK_MAX_NUM; i++) { + sprintf(tmp, "asrck_%x", i); + asrc_priv->asrck_clk[i] = devm_clk_get(&pdev->dev, tmp); + if (IS_ERR(asrc_priv->asrck_clk[i])) { + dev_err(&pdev->dev, "failed to get %s clock\n", tmp); + return PTR_ERR(asrc_priv->asrck_clk[i]); + } + } + + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx35-asrc")) { + asrc_priv->channel_bits = 3; + clk_map[IN] = input_clk_map_imx35; + clk_map[OUT] = output_clk_map_imx35; + } else { + asrc_priv->channel_bits = 4; + clk_map[IN] = input_clk_map_imx53; + clk_map[OUT] = output_clk_map_imx53; + } + + ret = fsl_asrc_init(asrc_priv); + if (ret) { + dev_err(&pdev->dev, "failed to init asrc %d\n", ret); + return -EINVAL; + } + + asrc_priv->channel_avail = 10; + + ret = of_property_read_u32(np, "fsl,asrc-rate", + &asrc_priv->asrc_rate); + if (ret) { + dev_err(&pdev->dev, "failed to get output rate\n"); + return -EINVAL; + } + + ret = of_property_read_u32(np, "fsl,asrc-width", + &asrc_priv->asrc_width); + if (ret) { + dev_err(&pdev->dev, "failed to get output width\n"); + return -EINVAL; + } + + if (asrc_priv->asrc_width != 16 && asrc_priv->asrc_width != 24) { + dev_warn(&pdev->dev, "unsupported width, switching to 24bit\n"); + asrc_priv->asrc_width = 24; + } + + platform_set_drvdata(pdev, asrc_priv); + pm_runtime_enable(&pdev->dev); + spin_lock_init(&asrc_priv->lock); + + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_asrc_component, + &fsl_asrc_dai, 1); + if (ret) { + dev_err(&pdev->dev, "failed to register ASoC DAI\n"); + return ret; + } + + ret = devm_snd_soc_register_platform(&pdev->dev, &fsl_asrc_platform); + if (ret) { + dev_err(&pdev->dev, "failed to register ASoC platform\n"); + return ret; + } + + dev_info(&pdev->dev, "driver registered\n"); + + return 0; +} + +#if CONFIG_PM_RUNTIME +static int fsl_asrc_runtime_resume(struct device *dev) +{ + struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); + int i; + + clk_prepare_enable(asrc_priv->mem_clk); + clk_prepare_enable(asrc_priv->ipg_clk); + for (i = 0; i < ASRC_CLK_MAX_NUM; i++) + clk_prepare_enable(asrc_priv->asrck_clk[i]); + + return 0; +} + +static int fsl_asrc_runtime_suspend(struct device *dev) +{ + struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); + int i; + + for (i = 0; i < ASRC_CLK_MAX_NUM; i++) + clk_disable_unprepare(asrc_priv->asrck_clk[i]); + clk_disable_unprepare(asrc_priv->ipg_clk); + clk_disable_unprepare(asrc_priv->mem_clk); + + return 0; +} +#endif /* CONFIG_PM_RUNTIME */ + +#if CONFIG_PM_SLEEP +static int fsl_asrc_suspend(struct device *dev) +{ + struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); + + regcache_cache_only(asrc_priv->regmap, true); + regcache_mark_dirty(asrc_priv->regmap); + + return 0; +} + +static int fsl_asrc_resume(struct device *dev) +{ + struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); + u32 asrctr; + + /* Stop all pairs provisionally */ + regmap_read(asrc_priv->regmap, REG_ASRCTR, &asrctr); + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_ASRCEi_ALL_MASK, 0); + + /* Restore all registers */ + regcache_cache_only(asrc_priv->regmap, false); + regcache_sync(asrc_priv->regmap); + + /* Restart enabled pairs */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_ASRCEi_ALL_MASK, asrctr); + + return 0; +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops fsl_asrc_pm = { + SET_RUNTIME_PM_OPS(fsl_asrc_runtime_suspend, fsl_asrc_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(fsl_asrc_suspend, fsl_asrc_resume) +}; + +static const struct of_device_id fsl_asrc_ids[] = { + { .compatible = "fsl,imx35-asrc", }, + { .compatible = "fsl,imx53-asrc", }, + {} +}; +MODULE_DEVICE_TABLE(of, fsl_asrc_ids); + +static struct platform_driver fsl_asrc_driver = { + .probe = fsl_asrc_probe, + .driver = { + .name = "fsl-asrc", + .of_match_table = fsl_asrc_ids, + .pm = &fsl_asrc_pm, + }, +}; +module_platform_driver(fsl_asrc_driver); + +MODULE_DESCRIPTION("Freescale ASRC ASoC driver"); +MODULE_AUTHOR("Nicolin Chen "); +MODULE_ALIAS("platform:fsl-asrc"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h new file mode 100644 index 000000000000..a3f211f53c23 --- /dev/null +++ b/sound/soc/fsl/fsl_asrc.h @@ -0,0 +1,461 @@ +/* + * fsl_asrc.h - Freescale ASRC ALSA SoC header file + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#ifndef _FSL_ASRC_H +#define _FSL_ASRC_H + +#define IN 0 +#define OUT 1 + +#define ASRC_DMA_BUFFER_NUM 2 +#define ASRC_INPUTFIFO_THRESHOLD 32 +#define ASRC_OUTPUTFIFO_THRESHOLD 32 +#define ASRC_FIFO_THRESHOLD_MIN 0 +#define ASRC_FIFO_THRESHOLD_MAX 63 +#define ASRC_DMA_BUFFER_SIZE (1024 * 48 * 4) +#define ASRC_MAX_BUFFER_SIZE (1024 * 48) +#define ASRC_OUTPUT_LAST_SAMPLE 8 + +#define IDEAL_RATIO_RATE 1000000 + +#define REG_ASRCTR 0x00 +#define REG_ASRIER 0x04 +#define REG_ASRCNCR 0x0C +#define REG_ASRCFG 0x10 +#define REG_ASRCSR 0x14 + +#define REG_ASRCDR1 0x18 +#define REG_ASRCDR2 0x1C +#define REG_ASRCDR(i) ((i < 2) ? REG_ASRCDR1 : REG_ASRCDR2) + +#define REG_ASRSTR 0x20 +#define REG_ASRRA 0x24 +#define REG_ASRRB 0x28 +#define REG_ASRRC 0x2C +#define REG_ASRPM1 0x40 +#define REG_ASRPM2 0x44 +#define REG_ASRPM3 0x48 +#define REG_ASRPM4 0x4C +#define REG_ASRPM5 0x50 +#define REG_ASRTFR1 0x54 +#define REG_ASRCCR 0x5C + +#define REG_ASRDIA 0x60 +#define REG_ASRDOA 0x64 +#define REG_ASRDIB 0x68 +#define REG_ASRDOB 0x6C +#define REG_ASRDIC 0x70 +#define REG_ASRDOC 0x74 +#define REG_ASRDI(i) (REG_ASRDIA + (i << 3)) +#define REG_ASRDO(i) (REG_ASRDOA + (i << 3)) +#define REG_ASRDx(x, i) (x == IN ? REG_ASRDI(i) : REG_ASRDO(i)) + +#define REG_ASRIDRHA 0x80 +#define REG_ASRIDRLA 0x84 +#define REG_ASRIDRHB 0x88 +#define REG_ASRIDRLB 0x8C +#define REG_ASRIDRHC 0x90 +#define REG_ASRIDRLC 0x94 +#define REG_ASRIDRH(i) (REG_ASRIDRHA + (i << 3)) +#define REG_ASRIDRL(i) (REG_ASRIDRLA + (i << 3)) + +#define REG_ASR76K 0x98 +#define REG_ASR56K 0x9C + +#define REG_ASRMCRA 0xA0 +#define REG_ASRFSTA 0xA4 +#define REG_ASRMCRB 0xA8 +#define REG_ASRFSTB 0xAC +#define REG_ASRMCRC 0xB0 +#define REG_ASRFSTC 0xB4 +#define REG_ASRMCR(i) (REG_ASRMCRA + (i << 3)) +#define REG_ASRFST(i) (REG_ASRFSTA + (i << 3)) + +#define REG_ASRMCR1A 0xC0 +#define REG_ASRMCR1B 0xC4 +#define REG_ASRMCR1C 0xC8 +#define REG_ASRMCR1(i) (REG_ASRMCR1A + (i << 2)) + + +/* REG0 0x00 REG_ASRCTR */ +#define ASRCTR_ATSi_SHIFT(i) (20 + i) +#define ASRCTR_ATSi_MASK(i) (1 << ASRCTR_ATSi_SHIFT(i)) +#define ASRCTR_ATS(i) (1 << ASRCTR_ATSi_SHIFT(i)) +#define ASRCTR_USRi_SHIFT(i) (14 + (i << 1)) +#define ASRCTR_USRi_MASK(i) (1 << ASRCTR_USRi_SHIFT(i)) +#define ASRCTR_USR(i) (1 << ASRCTR_USRi_SHIFT(i)) +#define ASRCTR_IDRi_SHIFT(i) (13 + (i << 1)) +#define ASRCTR_IDRi_MASK(i) (1 << ASRCTR_IDRi_SHIFT(i)) +#define ASRCTR_IDR(i) (1 << ASRCTR_IDRi_SHIFT(i)) +#define ASRCTR_SRST_SHIFT 4 +#define ASRCTR_SRST_MASK (1 << ASRCTR_SRST_SHIFT) +#define ASRCTR_SRST (1 << ASRCTR_SRST_SHIFT) +#define ASRCTR_ASRCEi_SHIFT(i) (1 + i) +#define ASRCTR_ASRCEi_MASK(i) (1 << ASRCTR_ASRCEi_SHIFT(i)) +#define ASRCTR_ASRCE(i) (1 << ASRCTR_ASRCEi_SHIFT(i)) +#define ASRCTR_ASRCEi_ALL_MASK (0x7 << ASRCTR_ASRCEi_SHIFT(0)) +#define ASRCTR_ASRCEN_SHIFT 0 +#define ASRCTR_ASRCEN_MASK (1 << ASRCTR_ASRCEN_SHIFT) +#define ASRCTR_ASRCEN (1 << ASRCTR_ASRCEN_SHIFT) + +/* REG1 0x04 REG_ASRIER */ +#define ASRIER_AFPWE_SHIFT 7 +#define ASRIER_AFPWE_MASK (1 << ASRIER_AFPWE_SHIFT) +#define ASRIER_AFPWE (1 << ASRIER_AFPWE_SHIFT) +#define ASRIER_AOLIE_SHIFT 6 +#define ASRIER_AOLIE_MASK (1 << ASRIER_AOLIE_SHIFT) +#define ASRIER_AOLIE (1 << ASRIER_AOLIE_SHIFT) +#define ASRIER_ADOEi_SHIFT(i) (3 + i) +#define ASRIER_ADOEi_MASK(i) (1 << ASRIER_ADOEi_SHIFT(i)) +#define ASRIER_ADOE(i) (1 << ASRIER_ADOEi_SHIFT(i)) +#define ASRIER_ADIEi_SHIFT(i) (0 + i) +#define ASRIER_ADIEi_MASK(i) (1 << ASRIER_ADIEi_SHIFT(i)) +#define ASRIER_ADIE(i) (1 << ASRIER_ADIEi_SHIFT(i)) + +/* REG2 0x0C REG_ASRCNCR */ +#define ASRCNCR_ANCi_SHIFT(i, b) (b * i) +#define ASRCNCR_ANCi_MASK(i, b) (((1 << b) - 1) << ASRCNCR_ANCi_SHIFT(i, b)) +#define ASRCNCR_ANCi(i, v, b) ((v << ASRCNCR_ANCi_SHIFT(i, b)) & ASRCNCR_ANCi_MASK(i, b)) + +/* REG3 0x10 REG_ASRCFG */ +#define ASRCFG_INIRQi_SHIFT(i) (21 + i) +#define ASRCFG_INIRQi_MASK(i) (1 << ASRCFG_INIRQi_SHIFT(i)) +#define ASRCFG_INIRQi (1 << ASRCFG_INIRQi_SHIFT(i)) +#define ASRCFG_NDPRi_SHIFT(i) (18 + i) +#define ASRCFG_NDPRi_MASK(i) (1 << ASRCFG_NDPRi_SHIFT(i)) +#define ASRCFG_NDPRi (1 << ASRCFG_NDPRi_SHIFT(i)) +#define ASRCFG_POSTMODi_SHIFT(i) (8 + (i << 2)) +#define ASRCFG_POSTMODi_WIDTH 2 +#define ASRCFG_POSTMODi_MASK(i) (((1 << ASRCFG_POSTMODi_WIDTH) - 1) << ASRCFG_POSTMODi_SHIFT(i)) +#define ASRCFG_POSTMOD(i, v) ((v) << ASRCFG_POSTMODi_SHIFT(i)) +#define ASRCFG_POSTMODi_UP(i) (0 << ASRCFG_POSTMODi_SHIFT(i)) +#define ASRCFG_POSTMODi_DCON(i) (1 << ASRCFG_POSTMODi_SHIFT(i)) +#define ASRCFG_POSTMODi_DOWN(i) (2 << ASRCFG_POSTMODi_SHIFT(i)) +#define ASRCFG_PREMODi_SHIFT(i) (6 + (i << 2)) +#define ASRCFG_PREMODi_WIDTH 2 +#define ASRCFG_PREMODi_MASK(i) (((1 << ASRCFG_PREMODi_WIDTH) - 1) << ASRCFG_PREMODi_SHIFT(i)) +#define ASRCFG_PREMOD(i, v) ((v) << ASRCFG_PREMODi_SHIFT(i)) +#define ASRCFG_PREMODi_UP(i) (0 << ASRCFG_PREMODi_SHIFT(i)) +#define ASRCFG_PREMODi_DCON(i) (1 << ASRCFG_PREMODi_SHIFT(i)) +#define ASRCFG_PREMODi_DOWN(i) (2 << ASRCFG_PREMODi_SHIFT(i)) +#define ASRCFG_PREMODi_BYPASS(i) (3 << ASRCFG_PREMODi_SHIFT(i)) + +/* REG4 0x14 REG_ASRCSR */ +#define ASRCSR_AxCSi_WIDTH 4 +#define ASRCSR_AxCSi_MASK ((1 << ASRCSR_AxCSi_WIDTH) - 1) +#define ASRCSR_AOCSi_SHIFT(i) (12 + (i << 2)) +#define ASRCSR_AOCSi_MASK(i) (((1 << ASRCSR_AxCSi_WIDTH) - 1) << ASRCSR_AOCSi_SHIFT(i)) +#define ASRCSR_AOCS(i, v) ((v) << ASRCSR_AOCSi_SHIFT(i)) +#define ASRCSR_AICSi_SHIFT(i) (i << 2) +#define ASRCSR_AICSi_MASK(i) (((1 << ASRCSR_AxCSi_WIDTH) - 1) << ASRCSR_AICSi_SHIFT(i)) +#define ASRCSR_AICS(i, v) ((v) << ASRCSR_AICSi_SHIFT(i)) + +/* REG5&6 0x18 & 0x1C REG_ASRCDR1 & ASRCDR2 */ +#define ASRCDRi_AxCPi_WIDTH 3 +#define ASRCDRi_AICPi_SHIFT(i) (0 + (i % 2) * 6) +#define ASRCDRi_AICPi_MASK(i) (((1 << ASRCDRi_AxCPi_WIDTH) - 1) << ASRCDRi_AICPi_SHIFT(i)) +#define ASRCDRi_AICP(i, v) ((v) << ASRCDRi_AICPi_SHIFT(i)) +#define ASRCDRi_AICDi_SHIFT(i) (3 + (i % 2) * 6) +#define ASRCDRi_AICDi_MASK(i) (((1 << ASRCDRi_AxCPi_WIDTH) - 1) << ASRCDRi_AICDi_SHIFT(i)) +#define ASRCDRi_AICD(i, v) ((v) << ASRCDRi_AICDi_SHIFT(i)) +#define ASRCDRi_AOCPi_SHIFT(i) ((i < 2) ? 12 + i * 6 : 6) +#define ASRCDRi_AOCPi_MASK(i) (((1 << ASRCDRi_AxCPi_WIDTH) - 1) << ASRCDRi_AOCPi_SHIFT(i)) +#define ASRCDRi_AOCP(i, v) ((v) << ASRCDRi_AOCPi_SHIFT(i)) +#define ASRCDRi_AOCDi_SHIFT(i) ((i < 2) ? 15 + i * 6 : 9) +#define ASRCDRi_AOCDi_MASK(i) (((1 << ASRCDRi_AxCPi_WIDTH) - 1) << ASRCDRi_AOCDi_SHIFT(i)) +#define ASRCDRi_AOCD(i, v) ((v) << ASRCDRi_AOCDi_SHIFT(i)) + +/* REG7 0x20 REG_ASRSTR */ +#define ASRSTR_DSLCNT_SHIFT 21 +#define ASRSTR_DSLCNT_MASK (1 << ASRSTR_DSLCNT_SHIFT) +#define ASRSTR_DSLCNT (1 << ASRSTR_DSLCNT_SHIFT) +#define ASRSTR_ATQOL_SHIFT 20 +#define ASRSTR_ATQOL_MASK (1 << ASRSTR_ATQOL_SHIFT) +#define ASRSTR_ATQOL (1 << ASRSTR_ATQOL_SHIFT) +#define ASRSTR_AOOLi_SHIFT(i) (17 + i) +#define ASRSTR_AOOLi_MASK(i) (1 << ASRSTR_AOOLi_SHIFT(i)) +#define ASRSTR_AOOL(i) (1 << ASRSTR_AOOLi_SHIFT(i)) +#define ASRSTR_AIOLi_SHIFT(i) (14 + i) +#define ASRSTR_AIOLi_MASK(i) (1 << ASRSTR_AIOLi_SHIFT(i)) +#define ASRSTR_AIOL(i) (1 << ASRSTR_AIOLi_SHIFT(i)) +#define ASRSTR_AODOi_SHIFT(i) (11 + i) +#define ASRSTR_AODOi_MASK(i) (1 << ASRSTR_AODOi_SHIFT(i)) +#define ASRSTR_AODO(i) (1 << ASRSTR_AODOi_SHIFT(i)) +#define ASRSTR_AIDUi_SHIFT(i) (8 + i) +#define ASRSTR_AIDUi_MASK(i) (1 << ASRSTR_AIDUi_SHIFT(i)) +#define ASRSTR_AIDU(i) (1 << ASRSTR_AIDUi_SHIFT(i)) +#define ASRSTR_FPWT_SHIFT 7 +#define ASRSTR_FPWT_MASK (1 << ASRSTR_FPWT_SHIFT) +#define ASRSTR_FPWT (1 << ASRSTR_FPWT_SHIFT) +#define ASRSTR_AOLE_SHIFT 6 +#define ASRSTR_AOLE_MASK (1 << ASRSTR_AOLE_SHIFT) +#define ASRSTR_AOLE (1 << ASRSTR_AOLE_SHIFT) +#define ASRSTR_AODEi_SHIFT(i) (3 + i) +#define ASRSTR_AODFi_MASK(i) (1 << ASRSTR_AODEi_SHIFT(i)) +#define ASRSTR_AODF(i) (1 << ASRSTR_AODEi_SHIFT(i)) +#define ASRSTR_AIDEi_SHIFT(i) (0 + i) +#define ASRSTR_AIDEi_MASK(i) (1 << ASRSTR_AIDEi_SHIFT(i)) +#define ASRSTR_AIDE(i) (1 << ASRSTR_AIDEi_SHIFT(i)) + +/* REG10 0x54 REG_ASRTFR1 */ +#define ASRTFR1_TF_BASE_WIDTH 7 +#define ASRTFR1_TF_BASE_SHIFT 6 +#define ASRTFR1_TF_BASE_MASK (((1 << ASRTFR1_TF_BASE_WIDTH) - 1) << ASRTFR1_TF_BASE_SHIFT) +#define ASRTFR1_TF_BASE(i) ((i) << ASRTFR1_TF_BASE_SHIFT) + +/* + * REG22 0xA0 REG_ASRMCRA + * REG24 0xA8 REG_ASRMCRB + * REG26 0xB0 REG_ASRMCRC + */ +#define ASRMCRi_ZEROBUFi_SHIFT 23 +#define ASRMCRi_ZEROBUFi_MASK (1 << ASRMCRi_ZEROBUFi_SHIFT) +#define ASRMCRi_ZEROBUFi (1 << ASRMCRi_ZEROBUFi_SHIFT) +#define ASRMCRi_EXTTHRSHi_SHIFT 22 +#define ASRMCRi_EXTTHRSHi_MASK (1 << ASRMCRi_EXTTHRSHi_SHIFT) +#define ASRMCRi_EXTTHRSHi (1 << ASRMCRi_EXTTHRSHi_SHIFT) +#define ASRMCRi_BUFSTALLi_SHIFT 21 +#define ASRMCRi_BUFSTALLi_MASK (1 << ASRMCRi_BUFSTALLi_SHIFT) +#define ASRMCRi_BUFSTALLi (1 << ASRMCRi_BUFSTALLi_SHIFT) +#define ASRMCRi_BYPASSPOLYi_SHIFT 20 +#define ASRMCRi_BYPASSPOLYi_MASK (1 << ASRMCRi_BYPASSPOLYi_SHIFT) +#define ASRMCRi_BYPASSPOLYi (1 << ASRMCRi_BYPASSPOLYi_SHIFT) +#define ASRMCRi_OUTFIFO_THRESHOLD_WIDTH 6 +#define ASRMCRi_OUTFIFO_THRESHOLD_SHIFT 12 +#define ASRMCRi_OUTFIFO_THRESHOLD_MASK (((1 << ASRMCRi_OUTFIFO_THRESHOLD_WIDTH) - 1) << ASRMCRi_OUTFIFO_THRESHOLD_SHIFT) +#define ASRMCRi_OUTFIFO_THRESHOLD(v) (((v) << ASRMCRi_OUTFIFO_THRESHOLD_SHIFT) & ASRMCRi_OUTFIFO_THRESHOLD_MASK) +#define ASRMCRi_RSYNIFi_SHIFT 11 +#define ASRMCRi_RSYNIFi_MASK (1 << ASRMCRi_RSYNIFi_SHIFT) +#define ASRMCRi_RSYNIFi (1 << ASRMCRi_RSYNIFi_SHIFT) +#define ASRMCRi_RSYNOFi_SHIFT 10 +#define ASRMCRi_RSYNOFi_MASK (1 << ASRMCRi_RSYNOFi_SHIFT) +#define ASRMCRi_RSYNOFi (1 << ASRMCRi_RSYNOFi_SHIFT) +#define ASRMCRi_INFIFO_THRESHOLD_WIDTH 6 +#define ASRMCRi_INFIFO_THRESHOLD_SHIFT 0 +#define ASRMCRi_INFIFO_THRESHOLD_MASK (((1 << ASRMCRi_INFIFO_THRESHOLD_WIDTH) - 1) << ASRMCRi_INFIFO_THRESHOLD_SHIFT) +#define ASRMCRi_INFIFO_THRESHOLD(v) (((v) << ASRMCRi_INFIFO_THRESHOLD_SHIFT) & ASRMCRi_INFIFO_THRESHOLD_MASK) + +/* + * REG23 0xA4 REG_ASRFSTA + * REG25 0xAC REG_ASRFSTB + * REG27 0xB4 REG_ASRFSTC + */ +#define ASRFSTi_OAFi_SHIFT 23 +#define ASRFSTi_OAFi_MASK (1 << ASRFSTi_OAFi_SHIFT) +#define ASRFSTi_OAFi (1 << ASRFSTi_OAFi_SHIFT) +#define ASRFSTi_OUTPUT_FIFO_WIDTH 7 +#define ASRFSTi_OUTPUT_FIFO_SHIFT 12 +#define ASRFSTi_OUTPUT_FIFO_MASK (((1 << ASRFSTi_OUTPUT_FIFO_WIDTH) - 1) << ASRFSTi_OUTPUT_FIFO_SHIFT) +#define ASRFSTi_IAEi_SHIFT 11 +#define ASRFSTi_IAEi_MASK (1 << ASRFSTi_OAFi_SHIFT) +#define ASRFSTi_IAEi (1 << ASRFSTi_OAFi_SHIFT) +#define ASRFSTi_INPUT_FIFO_WIDTH 7 +#define ASRFSTi_INPUT_FIFO_SHIFT 0 +#define ASRFSTi_INPUT_FIFO_MASK ((1 << ASRFSTi_INPUT_FIFO_WIDTH) - 1) + +/* REG28 0xC0 & 0xC4 & 0xC8 REG_ASRMCR1i */ +#define ASRMCR1i_IWD_WIDTH 3 +#define ASRMCR1i_IWD_SHIFT 9 +#define ASRMCR1i_IWD_MASK (((1 << ASRMCR1i_IWD_WIDTH) - 1) << ASRMCR1i_IWD_SHIFT) +#define ASRMCR1i_IWD(v) ((v) << ASRMCR1i_IWD_SHIFT) +#define ASRMCR1i_IMSB_SHIFT 8 +#define ASRMCR1i_IMSB_MASK (1 << ASRMCR1i_IMSB_SHIFT) +#define ASRMCR1i_IMSB_MSB (1 << ASRMCR1i_IMSB_SHIFT) +#define ASRMCR1i_IMSB_LSB (0 << ASRMCR1i_IMSB_SHIFT) +#define ASRMCR1i_OMSB_SHIFT 2 +#define ASRMCR1i_OMSB_MASK (1 << ASRMCR1i_OMSB_SHIFT) +#define ASRMCR1i_OMSB_MSB (1 << ASRMCR1i_OMSB_SHIFT) +#define ASRMCR1i_OMSB_LSB (0 << ASRMCR1i_OMSB_SHIFT) +#define ASRMCR1i_OSGN_SHIFT 1 +#define ASRMCR1i_OSGN_MASK (1 << ASRMCR1i_OSGN_SHIFT) +#define ASRMCR1i_OSGN (1 << ASRMCR1i_OSGN_SHIFT) +#define ASRMCR1i_OW16_SHIFT 0 +#define ASRMCR1i_OW16_MASK (1 << ASRMCR1i_OW16_SHIFT) +#define ASRMCR1i_OW16(v) ((v) << ASRMCR1i_OW16_SHIFT) + + +enum asrc_pair_index { + ASRC_INVALID_PAIR = -1, + ASRC_PAIR_A = 0, + ASRC_PAIR_B = 1, + ASRC_PAIR_C = 2, +}; + +#define ASRC_PAIR_MAX_NUM (ASRC_PAIR_C + 1) + +enum asrc_inclk { + INCLK_NONE = 0x03, + INCLK_ESAI_RX = 0x00, + INCLK_SSI1_RX = 0x01, + INCLK_SSI2_RX = 0x02, + INCLK_SSI3_RX = 0x07, + INCLK_SPDIF_RX = 0x04, + INCLK_MLB_CLK = 0x05, + INCLK_PAD = 0x06, + INCLK_ESAI_TX = 0x08, + INCLK_SSI1_TX = 0x09, + INCLK_SSI2_TX = 0x0a, + INCLK_SSI3_TX = 0x0b, + INCLK_SPDIF_TX = 0x0c, + INCLK_ASRCK1_CLK = 0x0f, +}; + +enum asrc_outclk { + OUTCLK_NONE = 0x03, + OUTCLK_ESAI_TX = 0x00, + OUTCLK_SSI1_TX = 0x01, + OUTCLK_SSI2_TX = 0x02, + OUTCLK_SSI3_TX = 0x07, + OUTCLK_SPDIF_TX = 0x04, + OUTCLK_MLB_CLK = 0x05, + OUTCLK_PAD = 0x06, + OUTCLK_ESAI_RX = 0x08, + OUTCLK_SSI1_RX = 0x09, + OUTCLK_SSI2_RX = 0x0a, + OUTCLK_SSI3_RX = 0x0b, + OUTCLK_SPDIF_RX = 0x0c, + OUTCLK_ASRCK1_CLK = 0x0f, +}; + +#define ASRC_CLK_MAX_NUM 16 + +enum asrc_word_width { + ASRC_WIDTH_24_BIT = 0, + ASRC_WIDTH_16_BIT = 1, + ASRC_WIDTH_8_BIT = 2, +}; + +struct asrc_config { + enum asrc_pair_index pair; + unsigned int channel_num; + unsigned int buffer_num; + unsigned int dma_buffer_size; + unsigned int input_sample_rate; + unsigned int output_sample_rate; + enum asrc_word_width input_word_width; + enum asrc_word_width output_word_width; + enum asrc_inclk inclk; + enum asrc_outclk outclk; +}; + +struct asrc_req { + unsigned int chn_num; + enum asrc_pair_index index; +}; + +struct asrc_querybuf { + unsigned int buffer_index; + unsigned int input_length; + unsigned int output_length; + unsigned long input_offset; + unsigned long output_offset; +}; + +struct asrc_convert_buffer { + void *input_buffer_vaddr; + void *output_buffer_vaddr; + unsigned int input_buffer_length; + unsigned int output_buffer_length; +}; + +struct asrc_status_flags { + enum asrc_pair_index index; + unsigned int overload_error; +}; + +enum asrc_error_status { + ASRC_TASK_Q_OVERLOAD = 0x01, + ASRC_OUTPUT_TASK_OVERLOAD = 0x02, + ASRC_INPUT_TASK_OVERLOAD = 0x04, + ASRC_OUTPUT_BUFFER_OVERFLOW = 0x08, + ASRC_INPUT_BUFFER_UNDERRUN = 0x10, +}; + +struct dma_block { + dma_addr_t dma_paddr; + void *dma_vaddr; + unsigned int length; +}; + +/** + * fsl_asrc_pair: ASRC Pair private data + * + * @asrc_priv: pointer to its parent module + * @config: configuration profile + * @error: error record + * @index: pair index (ASRC_PAIR_A, ASRC_PAIR_B, ASRC_PAIR_C) + * @channels: occupied channel number + * @desc: input and output dma descriptors + * @dma_chan: inputer and output DMA channels + * @dma_data: private dma data + * @pos: hardware pointer position + * @private: pair private area + */ +struct fsl_asrc_pair { + struct fsl_asrc *asrc_priv; + struct asrc_config *config; + unsigned int error; + + enum asrc_pair_index index; + unsigned int channels; + + struct dma_async_tx_descriptor *desc[2]; + struct dma_chan *dma_chan[2]; + struct imx_dma_data dma_data; + unsigned int pos; + + void *private; +}; + +/** + * fsl_asrc_pair: ASRC private data + * + * @dma_params_rx: DMA parameters for receive channel + * @dma_params_tx: DMA parameters for transmit channel + * @pdev: platform device pointer + * @regmap: regmap handler + * @paddr: physical address to the base address of registers + * @mem_clk: clock source to access register + * @ipg_clk: clock source to drive peripheral + * @asrck_clk: clock sources to driver ASRC internal logic + * @lock: spin lock for resource protection + * @pair: pair pointers + * @channel_bits: width of ASRCNCR register for each pair + * @channel_avail: non-occupied channel numbers + * @asrc_rate: default sample rate for ASoC Back-Ends + * @asrc_width: default sample width for ASoC Back-Ends + * @name: driver name + */ +struct fsl_asrc { + struct snd_dmaengine_dai_dma_data dma_params_rx; + struct snd_dmaengine_dai_dma_data dma_params_tx; + struct platform_device *pdev; + struct regmap *regmap; + unsigned long paddr; + struct clk *mem_clk; + struct clk *ipg_clk; + struct clk *asrck_clk[ASRC_CLK_MAX_NUM]; + spinlock_t lock; + + struct fsl_asrc_pair *pair[ASRC_PAIR_MAX_NUM]; + unsigned int channel_bits; + unsigned int channel_avail; + + int asrc_rate; + int asrc_width; + + char name[32]; +}; + +extern struct snd_soc_platform_driver fsl_asrc_platform; +struct dma_chan *fsl_asrc_get_dma_channel(struct fsl_asrc_pair *pair, bool dir); +#endif /* _FSL_ASRC_H */ diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c new file mode 100644 index 000000000000..5b1e73e97817 --- /dev/null +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -0,0 +1,386 @@ +/* + * Freescale ASRC ALSA SoC Platform (DMA) driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include +#include +#include +#include +#include + +#include "fsl_asrc.h" + +#define FSL_ASRC_DMABUF_SIZE (256 * 1024) + +static struct snd_pcm_hardware snd_imx_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .buffer_bytes_max = FSL_ASRC_DMABUF_SIZE, + .period_bytes_min = 128, + .period_bytes_max = 65535, /* Limited by SDMA engine */ + .periods_min = 2, + .periods_max = 255, + .fifo_size = 0, +}; + +static bool filter(struct dma_chan *chan, void *param) +{ + if (!imx_dma_is_general_purpose(chan)) + return false; + + chan->private = param; + + return true; +} + +static void fsl_asrc_dma_complete(void *arg) +{ + struct snd_pcm_substream *substream = arg; + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + + pair->pos += snd_pcm_lib_period_bytes(substream); + if (pair->pos >= snd_pcm_lib_buffer_bytes(substream)) + pair->pos = 0; + + snd_pcm_period_elapsed(substream); +} + +static int fsl_asrc_dma_prepare_and_submit(struct snd_pcm_substream *substream) +{ + u8 dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? OUT : IN; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + struct device *dev = rtd->platform->dev; + unsigned long flags = DMA_CTRL_ACK; + + /* Prepare and submit Front-End DMA channel */ + if (!substream->runtime->no_period_wakeup) + flags |= DMA_PREP_INTERRUPT; + + pair->pos = 0; + pair->desc[!dir] = dmaengine_prep_dma_cyclic( + pair->dma_chan[!dir], runtime->dma_addr, + snd_pcm_lib_buffer_bytes(substream), + snd_pcm_lib_period_bytes(substream), + dir == OUT ? DMA_TO_DEVICE : DMA_FROM_DEVICE, flags); + if (!pair->desc[!dir]) { + dev_err(dev, "failed to prepare slave DMA for Front-End\n"); + return -ENOMEM; + } + + pair->desc[!dir]->callback = fsl_asrc_dma_complete; + pair->desc[!dir]->callback_param = substream; + + dmaengine_submit(pair->desc[!dir]); + + /* Prepare and submit Back-End DMA channel */ + pair->desc[dir] = dmaengine_prep_dma_cyclic( + pair->dma_chan[dir], 0xffff, 64, 64, DMA_DEV_TO_DEV, 0); + if (!pair->desc[dir]) { + dev_err(dev, "failed to prepare slave DMA for Back-End\n"); + return -ENOMEM; + } + + dmaengine_submit(pair->desc[dir]); + + return 0; +} + +static int fsl_asrc_dma_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = fsl_asrc_dma_prepare_and_submit(substream); + if (ret) + return ret; + dma_async_issue_pending(pair->dma_chan[IN]); + dma_async_issue_pending(pair->dma_chan[OUT]); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + dmaengine_terminate_all(pair->dma_chan[OUT]); + dmaengine_terminate_all(pair->dma_chan[IN]); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + enum dma_slave_buswidth buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_dmaengine_dai_dma_data *dma_params_fe = NULL; + struct snd_dmaengine_dai_dma_data *dma_params_be = NULL; + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + struct fsl_asrc *asrc_priv = pair->asrc_priv; + struct dma_slave_config config_fe, config_be; + enum asrc_pair_index index = pair->index; + struct device *dev = rtd->platform->dev; + int stream = substream->stream; + struct imx_dma_data *tmp_data; + struct snd_soc_dpcm *dpcm; + struct dma_chan *tmp_chan; + struct device *dev_be; + u8 dir = tx ? OUT : IN; + dma_cap_mask_t mask; + int ret; + + /* Fetch the Back-End dma_data from DPCM */ + list_for_each_entry(dpcm, &rtd->dpcm[stream].be_clients, list_be) { + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *substream_be; + struct snd_soc_dai *dai = be->cpu_dai; + + if (dpcm->fe != rtd) + continue; + + substream_be = snd_soc_dpcm_get_substream(be, stream); + dma_params_be = snd_soc_dai_get_dma_data(dai, substream_be); + dev_be = dai->dev; + break; + } + + if (!dma_params_be) { + dev_err(dev, "failed to get the substream of Back-End\n"); + return -EINVAL; + } + + /* Override dma_data of the Front-End and config its dmaengine */ + dma_params_fe = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_params_fe->addr = asrc_priv->paddr + REG_ASRDx(!dir, index); + dma_params_fe->maxburst = dma_params_be->maxburst; + + pair->dma_chan[!dir] = fsl_asrc_get_dma_channel(pair, !dir); + if (!pair->dma_chan[!dir]) { + dev_err(dev, "failed to request DMA channel\n"); + return -EINVAL; + } + + memset(&config_fe, 0, sizeof(config_fe)); + ret = snd_dmaengine_pcm_prepare_slave_config(substream, params, &config_fe); + if (ret) { + dev_err(dev, "failed to prepare DMA config for Front-End\n"); + return ret; + } + + ret = dmaengine_slave_config(pair->dma_chan[!dir], &config_fe); + if (ret) { + dev_err(dev, "failed to config DMA channel for Front-End\n"); + return ret; + } + + /* Request and config DMA channel for Back-End */ + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + dma_cap_set(DMA_CYCLIC, mask); + + /* Get DMA request of Back-End */ + tmp_chan = dma_request_slave_channel(dev_be, tx ? "tx" : "rx"); + tmp_data = tmp_chan->private; + pair->dma_data.dma_request = tmp_data->dma_request; + dma_release_channel(tmp_chan); + + /* Get DMA request of Front-End */ + tmp_chan = fsl_asrc_get_dma_channel(pair, dir); + tmp_data = tmp_chan->private; + pair->dma_data.dma_request2 = tmp_data->dma_request; + pair->dma_data.peripheral_type = tmp_data->peripheral_type; + pair->dma_data.priority = tmp_data->priority; + dma_release_channel(tmp_chan); + + pair->dma_chan[dir] = dma_request_channel(mask, filter, &pair->dma_data); + if (!pair->dma_chan[dir]) { + dev_err(dev, "failed to request DMA channel for Back-End\n"); + return -EINVAL; + } + + if (asrc_priv->asrc_width == 16) + buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; + else + buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; + + config_be.direction = DMA_DEV_TO_DEV; + config_be.src_addr_width = buswidth; + config_be.src_maxburst = dma_params_be->maxburst; + config_be.dst_addr_width = buswidth; + config_be.dst_maxburst = dma_params_be->maxburst; + + if (tx) { + config_be.src_addr = asrc_priv->paddr + REG_ASRDO(index); + config_be.dst_addr = dma_params_be->addr; + } else { + config_be.dst_addr = asrc_priv->paddr + REG_ASRDI(index); + config_be.src_addr = dma_params_be->addr; + } + + ret = dmaengine_slave_config(pair->dma_chan[dir], &config_be); + if (ret) { + dev_err(dev, "failed to config DMA channel for Back-End\n"); + return ret; + } + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + return 0; +} + +static int fsl_asrc_dma_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + + snd_pcm_set_runtime_buffer(substream, NULL); + + if (pair->dma_chan[IN]) + dma_release_channel(pair->dma_chan[IN]); + + if (pair->dma_chan[OUT]) + dma_release_channel(pair->dma_chan[OUT]); + + pair->dma_chan[IN] = NULL; + pair->dma_chan[OUT] = NULL; + + return 0; +} + +static int fsl_asrc_dma_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct device *dev = rtd->platform->dev; + struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); + struct fsl_asrc_pair *pair; + + pair = kzalloc(sizeof(struct fsl_asrc_pair), GFP_KERNEL); + if (!pair) { + dev_err(dev, "failed to allocate pair\n"); + return -ENOMEM; + } + + pair->asrc_priv = asrc_priv; + + runtime->private_data = pair; + + snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); + + return 0; +} + +static int fsl_asrc_dma_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + struct fsl_asrc *asrc_priv = pair->asrc_priv; + + if (pair && asrc_priv->pair[pair->index] == pair) + asrc_priv->pair[pair->index] = NULL; + + kfree(pair); + + return 0; +} + +static snd_pcm_uframes_t fsl_asrc_dma_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + + return bytes_to_frames(substream->runtime, pair->pos); +} + +static struct snd_pcm_ops fsl_asrc_dma_pcm_ops = { + .ioctl = snd_pcm_lib_ioctl, + .hw_params = fsl_asrc_dma_hw_params, + .hw_free = fsl_asrc_dma_hw_free, + .trigger = fsl_asrc_dma_trigger, + .open = fsl_asrc_dma_startup, + .close = fsl_asrc_dma_shutdown, + .pointer = fsl_asrc_dma_pcm_pointer, +}; + +static int fsl_asrc_dma_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm_substream *substream; + struct snd_pcm *pcm = rtd->pcm; + int ret, i; + + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); + if (ret) { + dev_err(card->dev, "failed to set DMA mask\n"); + return ret; + } + + for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) { + substream = pcm->streams[i].substream; + if (!substream) + continue; + + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev, + FSL_ASRC_DMABUF_SIZE, &substream->dma_buffer); + if (ret) { + dev_err(card->dev, "failed to allocate DMA buffer\n"); + goto err; + } + } + + return 0; + +err: + if (--i == 0 && pcm->streams[i].substream) + snd_dma_free_pages(&pcm->streams[i].substream->dma_buffer); + + return ret; +} + +static void fsl_asrc_dma_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + int i; + + for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) { + substream = pcm->streams[i].substream; + if (!substream) + continue; + + snd_dma_free_pages(&substream->dma_buffer); + substream->dma_buffer.area = NULL; + substream->dma_buffer.addr = 0; + } +} + +struct snd_soc_platform_driver fsl_asrc_platform = { + .ops = &fsl_asrc_dma_pcm_ops, + .pcm_new = fsl_asrc_dma_pcm_new, + .pcm_free = fsl_asrc_dma_pcm_free, +}; +EXPORT_SYMBOL_GPL(fsl_asrc_platform); -- cgit v1.2.3 From f42bb22243d2ae264d721b055f836059fe35321f Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 4 Aug 2014 15:17:55 +0200 Subject: ALSA: virtuoso: add Xonar Essence STX II support Just add the PCI ID for the STX II. It appears to work the same as the STX, except for the addition of the not-yet-supported daughterboard. Tested-by: Mario Tested-by: corubba Cc: Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 4 ++-- sound/pci/Kconfig | 4 ++-- sound/pci/oxygen/virtuoso.c | 1 + sound/pci/oxygen/xonar_pcm179x.c | 12 ++++++++++-- 4 files changed, 15 insertions(+), 6 deletions(-) (limited to 'Documentation') diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 7ccf933bfbe0..48148d6d9307 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -2026,8 +2026,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. ------------------- Module for sound cards based on the Asus AV66/AV100/AV200 chips, - i.e., Xonar D1, DX, D2, D2X, DS, Essence ST (Deluxe), Essence STX, - HDAV1.3 (Deluxe), and HDAV1.3 Slim. + i.e., Xonar D1, DX, D2, D2X, DS, DSX, Essence ST (Deluxe), + Essence STX (II), HDAV1.3 (Deluxe), and HDAV1.3 Slim. This module supports autoprobe and multiple cards. diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 3a3a3a71088b..50dd0086cfb1 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -858,8 +858,8 @@ config SND_VIRTUOSO select SND_JACK if INPUT=y || INPUT=SND help Say Y here to include support for sound cards based on the - Asus AV66/AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X, DS, - Essence ST (Deluxe), and Essence STX. + Asus AV66/AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X, DS, DSX, + Essence ST (Deluxe), and Essence STX (II). Support for the HDAV1.3 (Deluxe) and HDAV1.3 Slim is experimental; for the Xense, missing. diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 64b9fda5f04a..dbbbacfd535e 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -53,6 +53,7 @@ static DEFINE_PCI_DEVICE_TABLE(xonar_ids) = { { OXYGEN_PCI_SUBID(0x1043, 0x835e) }, { OXYGEN_PCI_SUBID(0x1043, 0x838e) }, { OXYGEN_PCI_SUBID(0x1043, 0x8522) }, + { OXYGEN_PCI_SUBID(0x1043, 0x85f4) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } }; diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index c8c7f2c9b355..e02605931669 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -100,8 +100,8 @@ */ /* - * Xonar Essence ST (Deluxe)/STX - * ----------------------------- + * Xonar Essence ST (Deluxe)/STX (II) + * ---------------------------------- * * CMI8788: * @@ -1138,6 +1138,14 @@ int get_xonar_pcm179x_model(struct oxygen *chip, chip->model.resume = xonar_stx_resume; chip->model.set_dac_params = set_pcm1796_params; break; + case 0x85f4: + chip->model = model_xonar_st; + /* TODO: daughterboard support */ + chip->model.shortname = "Xonar STX II"; + chip->model.init = xonar_stx_init; + chip->model.resume = xonar_stx_resume; + chip->model.set_dac_params = set_pcm1796_params; + break; default: return -EINVAL; } -- cgit v1.2.3