From 171a0138ab75fcbe1228c4af0442221efccfb197 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 10 Sep 2015 06:40:19 +0000 Subject: ASoC: ak4642: enable to use MCKO as fixed rate output pin on DT ak4642 chip can output clock via MCKO pin as audio reference clock. This patch supports it on DT Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/ak4642.txt | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/ak4642.txt b/Documentation/devicetree/bindings/sound/ak4642.txt index 623d4e70ae11..340784db6808 100644 --- a/Documentation/devicetree/bindings/sound/ak4642.txt +++ b/Documentation/devicetree/bindings/sound/ak4642.txt @@ -7,7 +7,14 @@ Required properties: - compatible : "asahi-kasei,ak4642" or "asahi-kasei,ak4643" or "asahi-kasei,ak4648" - reg : The chip select number on the I2C bus -Example: +Optional properties: + + - #clock-cells : common clock binding; shall be set to 0 + - clocks : common clock binding; MCKI clock + - clock-frequency : common clock binding; frequency of MCKO + - clock-output-names : common clock binding; MCKO clock name + +Example 1: &i2c { ak4648: ak4648@0x12 { @@ -15,3 +22,16 @@ Example: reg = <0x12>; }; }; + +Example 2: + +&i2c { + ak4643: codec@12 { + compatible = "asahi-kasei,ak4643"; + reg = <0x12>; + #clock-cells = <0>; + clocks = <&audio_clock>; + clock-frequency = <12288000>; + clock-output-names = "ak4643_mcko"; + }; +}; -- cgit v1.2.3 From e3d2cec8d49c01800373c25a5a06326f2a4304e6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 10 Sep 2015 06:49:54 +0000 Subject: ASoC: rsnd: add missing #sound-dai-cells explain on Document Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 2 ++ 1 file changed, 2 insertions(+) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 1173395b5e5c..776cf6aa8db9 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -30,6 +30,8 @@ Required properties: - rcar_sound,dai : DAI contents. The number of DAI subnode should be same as HW. see below for detail. +- #sound-dai-cells : it must be 0 if your system is using single DAI + it must be 1 if your system is using multi DAI SSI subnode properties: - interrupts : Should contain SSI interrupt for PIO transfer -- cgit v1.2.3 From ac37a45b0b6c8400719bb837f1c321079b72db53 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 10 Sep 2015 07:01:58 +0000 Subject: ASoC: rsnd: Add Gen3 initial support Renesas sound Gen3 is updated version of Gen2. We need to update driver for it, but basically it should works as Gen2 compatible. This is initial support for Gen3. Gen3 specific feature will be incrementally added in the future Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 2 ++ sound/soc/sh/rcar/core.c | 1 + 2 files changed, 3 insertions(+) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 776cf6aa8db9..bf6fd1af0a11 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -4,10 +4,12 @@ Required properties: - compatible : "renesas,rcar_sound-", fallbacks "renesas,rcar_sound-gen1" if generation1, and "renesas,rcar_sound-gen2" if generation2 + "renesas,rcar_sound-gen3" if generation3 Examples with soctypes are: - "renesas,rcar_sound-r8a7778" (R-Car M1A) - "renesas,rcar_sound-r8a7790" (R-Car H2) - "renesas,rcar_sound-r8a7791" (R-Car M2-W) + - "renesas,rcar_sound-r8a7795" (R-Car H3) - reg : Should contain the register physical address. required register is SRU/ADG/SSI if generation1 diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index f3feed5ce9b6..870f94415abc 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -110,6 +110,7 @@ static const struct rsnd_of_data rsnd_of_data_gen2 = { static const struct of_device_id rsnd_of_match[] = { { .compatible = "renesas,rcar_sound-gen1", .data = &rsnd_of_data_gen1 }, { .compatible = "renesas,rcar_sound-gen2", .data = &rsnd_of_data_gen2 }, + { .compatible = "renesas,rcar_sound-gen3", .data = &rsnd_of_data_gen2 }, /* gen2 compatible */ {}, }; MODULE_DEVICE_TABLE(of, rsnd_of_match); -- cgit v1.2.3 From 2a46db4a3787edb0dc07276f21f33bbaf01938f1 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 10 Sep 2015 07:04:45 +0000 Subject: ASoC: rsnd: add AUDIO_CLKOUT support Renesas sound has AUDIO_CLKOUT (in Gen1/Gen2) AUDIO_CLKOUT1/2/3 (in Gen3) This patch support these patches as clock provider. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/renesas,rsnd.txt | 3 + sound/soc/sh/rcar/adg.c | 98 +++++++++++++++++++++- 2 files changed, 97 insertions(+), 4 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index bf6fd1af0a11..c57cbd65736c 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -34,6 +34,9 @@ Required properties: see below for detail. - #sound-dai-cells : it must be 0 if your system is using single DAI it must be 1 if your system is using multi DAI +- #clock-cells : it must be 0 if your system has audio_clkout + it must be 1 if your system has audio_clkout0/1/2/3 +- clock-frequency : for all audio_clkout0/1/2/3 SSI subnode properties: - interrupts : Should contain SSI interrupt for PIO transfer diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index d4fb11a3ce64..3fecb87f45ba 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -7,6 +7,7 @@ * License. See the file "COPYING" in the main directory of this archive * for more details. */ +#include #include "rsnd.h" #define CLKA 0 @@ -15,6 +16,12 @@ #define CLKI 3 #define CLKMAX 4 +#define CLKOUT 0 +#define CLKOUT1 1 +#define CLKOUT2 2 +#define CLKOUT3 3 +#define CLKOUTMAX 4 + #define BRRx_MASK(x) (0x3FF & x) static struct rsnd_mod_ops adg_ops = { @@ -23,6 +30,8 @@ static struct rsnd_mod_ops adg_ops = { struct rsnd_adg { struct clk *clk[CLKMAX]; + struct clk *clkout[CLKOUTMAX]; + struct clk_onecell_data onecell; struct rsnd_mod mod; int rbga_rate_for_441khz; /* RBGA */ @@ -34,6 +43,11 @@ struct rsnd_adg { (i < CLKMAX) && \ ((pos) = adg->clk[i]); \ i++) +#define for_each_rsnd_clkout(pos, adg, i) \ + for (i = 0; \ + (i < CLKOUTMAX) && \ + ((pos) = adg->clkout[i]); \ + i++) #define rsnd_priv_to_adg(priv) ((struct rsnd_adg *)(priv)->adg) static u32 rsnd_adg_calculate_rbgx(unsigned long div) @@ -416,14 +430,25 @@ static void rsnd_adg_get_clkin(struct rsnd_priv *priv, dev_dbg(dev, "clk %d : %p : %ld\n", i, clk, clk_get_rate(clk)); } -static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg) +static void rsnd_adg_get_clkout(struct rsnd_priv *priv, + struct rsnd_adg *adg) { struct clk *clk; struct rsnd_mod *adg_mod = rsnd_mod_get(adg); struct device *dev = rsnd_priv_to_dev(priv); - unsigned long rate, div; + struct device_node *np = dev->of_node; u32 ckr, rbgx, rbga, rbgb; + u32 rate, req_rate, div; + uint32_t count = 0; + unsigned long req_48kHz_rate, req_441kHz_rate; int i; + const char *parent_clk_name = NULL; + static const char * const clkout_name[] = { + [CLKOUT] = "audio_clkout", + [CLKOUT1] = "audio_clkout1", + [CLKOUT2] = "audio_clkout2", + [CLKOUT3] = "audio_clkout3", + }; int brg_table[] = { [CLKA] = 0x0, [CLKB] = 0x1, @@ -431,6 +456,20 @@ static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg) [CLKI] = 0x2, }; + of_property_read_u32(np, "#clock-cells", &count); + + /* + * ADG supports BRRA/BRRB output only + * this means all clkout0/1/2/3 will be same rate + */ + of_property_read_u32(np, "clock-frequency", &req_rate); + req_48kHz_rate = 0; + req_441kHz_rate = 0; + if (0 == (req_rate % 44100)) + req_441kHz_rate = req_rate; + if (0 == (req_rate % 48000)) + req_48kHz_rate = req_rate; + /* * This driver is assuming that AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC * have 44.1kHz or 48kHz base clocks for now. @@ -454,22 +493,72 @@ static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg) /* RBGA */ if (!adg->rbga_rate_for_441khz && (0 == rate % 44100)) { div = 6; + if (req_441kHz_rate) + div = rate / req_441kHz_rate; rbgx = rsnd_adg_calculate_rbgx(div); if (BRRx_MASK(rbgx) == rbgx) { rbga = rbgx; adg->rbga_rate_for_441khz = rate / div; ckr |= brg_table[i] << 20; + if (req_441kHz_rate) + parent_clk_name = __clk_get_name(clk); } } /* RBGB */ if (!adg->rbgb_rate_for_48khz && (0 == rate % 48000)) { div = 6; + if (req_48kHz_rate) + div = rate / req_48kHz_rate; rbgx = rsnd_adg_calculate_rbgx(div); if (BRRx_MASK(rbgx) == rbgx) { rbgb = rbgx; adg->rbgb_rate_for_48khz = rate / div; ckr |= brg_table[i] << 16; + if (req_48kHz_rate) { + parent_clk_name = __clk_get_name(clk); + ckr |= 0x80000000; + } + } + } + } + + /* + * ADG supports BRRA/BRRB output only. + * this means all clkout0/1/2/3 will be * same rate + */ + + /* + * for clkout + */ + if (!count) { + clk = clk_register_fixed_rate(dev, clkout_name[i], + parent_clk_name, + (parent_clk_name) ? + 0 : CLK_IS_ROOT, req_rate); + if (!IS_ERR(clk)) { + adg->clkout[CLKOUT] = clk; + of_clk_add_provider(np, of_clk_src_simple_get, clk); + } + } + /* + * for clkout0/1/2/3 + */ + else { + for (i = 0; i < CLKOUTMAX; i++) { + clk = clk_register_fixed_rate(dev, clkout_name[i], + parent_clk_name, + (parent_clk_name) ? + 0 : CLK_IS_ROOT, + req_rate); + if (!IS_ERR(clk)) { + adg->onecell.clks = adg->clkout; + adg->onecell.clk_num = CLKOUTMAX; + + adg->clkout[i] = clk; + + of_clk_add_provider(np, of_clk_src_onecell_get, + &adg->onecell); } } } @@ -478,6 +567,8 @@ static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg) rsnd_mod_write(adg_mod, BRRA, rbga); rsnd_mod_write(adg_mod, BRRB, rbgb); + for_each_rsnd_clkout(clk, adg, i) + dev_dbg(dev, "clkout %d : %p : %ld\n", i, clk, clk_get_rate(clk)); dev_dbg(dev, "SSICKR = 0x%08x, BRRA/BRRB = 0x%x/0x%x\n", ckr, rbga, rbgb); } @@ -504,8 +595,7 @@ int rsnd_adg_probe(struct platform_device *pdev, adg->mod.priv = priv; rsnd_adg_get_clkin(priv, adg); - - rsnd_adg_ssi_clk_init(priv, adg); + rsnd_adg_get_clkout(priv, adg); priv->adg = adg; -- cgit v1.2.3 From 6131084a0bc966107021d8c89489f9cd1663b902 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Wed, 9 Sep 2015 21:27:43 +0300 Subject: ASoC: simple-card: Add tdm slot mask support to simple-card Adds DT binding for explicitly choosing a tdm mask for DAI and uses it in simple-card. The API for snd_soc_of_parse_tdm_slot() has also been changed. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/tdm-slot.txt | 11 +++++++++- include/sound/simple_card.h | 2 ++ include/sound/soc.h | 2 ++ sound/soc/generic/simple-card.c | 8 +++++-- sound/soc/soc-core.c | 25 ++++++++++++++++++++++ 5 files changed, 45 insertions(+), 3 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/tdm-slot.txt b/Documentation/devicetree/bindings/sound/tdm-slot.txt index 6a2c84247f91..34cf70e2cbc4 100644 --- a/Documentation/devicetree/bindings/sound/tdm-slot.txt +++ b/Documentation/devicetree/bindings/sound/tdm-slot.txt @@ -4,11 +4,15 @@ This specifies audio DAI's TDM slot. TDM slot properties: dai-tdm-slot-num : Number of slots in use. -dai-tdm-slot-width : Width in bits for each slot. +dai-tdm-slot-width : Width in bits for each slot. +dai-tdm-slot-tx-mask : Transmit direction slot mask, optional +dai-tdm-slot-rx-mask : Receive direction slot mask, optional For instance: dai-tdm-slot-num = <2>; dai-tdm-slot-width = <8>; + dai-tdm-slot-tx-mask = <0 1>; + dai-tdm-slot-rx-mask = <1 0>; And for each spcified driver, there could be one .of_xlate_tdm_slot_mask() to specify a explicit mapping of the channels and the slots. If it's absent @@ -18,3 +22,8 @@ tx and rx masks. For snd_soc_of_xlate_tdm_slot_mask(), the tx and rx masks will use a 1 bit for an active slot as default, and the default active bits are at the LSB of the masks. + +The explicit masks are given as array of integers, where the first +number presents bit-0 (LSB), second presents bit-1, etc. Any non zero +number is considered 1 and 0 is 0. snd_soc_of_xlate_tdm_slot_mask() +does not do anything, if either mask is set non zero value. diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h index b9b4f289fe6b..0399352f3a62 100644 --- a/include/sound/simple_card.h +++ b/include/sound/simple_card.h @@ -19,6 +19,8 @@ struct asoc_simple_dai { unsigned int sysclk; int slots; int slot_width; + unsigned int tx_slot_mask; + unsigned int rx_slot_mask; struct clk *clk; }; diff --git a/include/sound/soc.h b/include/sound/soc.h index 884e728b09d9..a76622d7bb2f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1601,6 +1601,8 @@ int snd_soc_of_parse_card_name(struct snd_soc_card *card, int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, const char *propname); int snd_soc_of_parse_tdm_slot(struct device_node *np, + unsigned int *tx_mask, + unsigned int *rx_mask, unsigned int *slots, unsigned int *slot_width); void snd_soc_of_parse_audio_prefix(struct snd_soc_card *card, diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 3ff76d419436..54c33204541f 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -151,7 +151,9 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, } if (set->slots) { - ret = snd_soc_dai_set_tdm_slot(dai, 0, 0, + ret = snd_soc_dai_set_tdm_slot(dai, + set->tx_slot_mask, + set->rx_slot_mask, set->slots, set->slot_width); if (ret && ret != -ENOTSUPP) { @@ -243,7 +245,9 @@ asoc_simple_card_sub_parse_of(struct device_node *np, return ret; /* Parse TDM slot */ - ret = snd_soc_of_parse_tdm_slot(np, &dai->slots, &dai->slot_width); + ret = snd_soc_of_parse_tdm_slot(np, &dai->tx_slot_mask, + &dai->rx_slot_mask, + &dai->slots, &dai->slot_width); if (ret) return ret; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6173d15236c3..c5e21ca0c015 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3291,13 +3291,38 @@ int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_simple_widgets); +static int snd_soc_of_get_slot_mask(struct device_node *np, + const char *prop_name, + unsigned int *mask) +{ + u32 val; + const u32 *of_slot_mask = of_get_property(np, prop_name, &val); + int i; + + if (!of_slot_mask) + return 0; + val /= sizeof(u32); + for (i = 0; i < val; i++) + if (be32_to_cpup(&of_slot_mask[i])) + *mask |= (1 << i); + + return val; +} + int snd_soc_of_parse_tdm_slot(struct device_node *np, + unsigned int *tx_mask, + unsigned int *rx_mask, unsigned int *slots, unsigned int *slot_width) { u32 val; int ret; + if (tx_mask) + snd_soc_of_get_slot_mask(np, "dai-tdm-slot-tx-mask", tx_mask); + if (rx_mask) + snd_soc_of_get_slot_mask(np, "dai-tdm-slot-rx-mask", rx_mask); + if (of_property_read_bool(np, "dai-tdm-slot-num")) { ret = of_property_read_u32(np, "dai-tdm-slot-num", &val); if (ret) -- cgit v1.2.3 From 4ec73d3c81be5a401546afc6628aaf8add253ff1 Mon Sep 17 00:00:00 2001 From: Maxime Ripard Date: Sat, 12 Sep 2015 15:26:23 +0200 Subject: ASoC: sunxi: Add the Allwinner A10 codec bindings The Allwinner SoCs have an in-SoC audio controller taking the role of a DAI and a codec. Add the binding documentation for that controller on the A10. Signed-off-by: Maxime Ripard Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/sun4i-codec.txt | 33 ++++++++++++++++++++++ 1 file changed, 33 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/sun4i-codec.txt (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/sun4i-codec.txt b/Documentation/devicetree/bindings/sound/sun4i-codec.txt new file mode 100644 index 000000000000..680144b74ae9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sun4i-codec.txt @@ -0,0 +1,33 @@ +* Allwinner A10 Codec + +Required properties: +- compatible: must be either "allwinner,sun4i-a10-codec" or + "allwinner,sun7i-a20-codec" +- reg: must contain the registers location and length +- interrupts: must contain the codec interrupt +- dmas: DMA channels 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: + - "apb": the parent APB clock for this controller + - "codec": the parent module clock +- 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. + + +Example: +codec: codec@01c22c00 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun7i-a20-codec"; + reg = <0x01c22c00 0x40>; + interrupts = <0 30 4>; + clocks = <&apb0_gates 0>, <&codec_clk>; + clock-names = "apb", "codec"; + dmas = <&dma 0 19>, <&dma 0 19>; + dma-names = "rx", "tx"; + routing = "Headphone Jack", "HP Right", + "Headphone Jack", "HP Left"; +}; -- cgit v1.2.3 From b07570628471777aabb5695284e1af4533e502da Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 15 Sep 2015 08:26:36 +0000 Subject: ASoC: add ak4613 support Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/ak4613.txt | 17 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4613.c | 469 +++++++++++++++++++++ 4 files changed, 493 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ak4613.txt create mode 100644 sound/soc/codecs/ak4613.c (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/ak4613.txt b/Documentation/devicetree/bindings/sound/ak4613.txt new file mode 100644 index 000000000000..15a919522b42 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4613.txt @@ -0,0 +1,17 @@ +AK4613 I2C transmitter + +This device supports I2C mode only. + +Required properties: + +- compatible : "asahi-kasei,ak4613" +- reg : The chip select number on the I2C bus + +Example: + +&i2c { + ak4613: ak4613@0x10 { + compatible = "asahi-kasei,ak4613"; + reg = <0x10>; + }; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0c9733ecd17f..a92e4d4b2eee 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -36,6 +36,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C select SND_SOC_AK4554 + select SND_SOC_AK4613 if I2C select SND_SOC_AK4641 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C @@ -319,6 +320,10 @@ config SND_SOC_AK4535 config SND_SOC_AK4554 tristate "AKM AK4554 CODEC" +config SND_SOC_AK4613 + tristate "AKM AK4613 CODEC" + depends on I2C + config SND_SOC_AK4641 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4a32077954ae..5b6c8af38a39 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -26,6 +26,7 @@ snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4554-objs := ak4554.o +snd-soc-ak4613-objs := ak4613.o snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o @@ -216,6 +217,7 @@ obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4554) += snd-soc-ak4554.o +obj-$(CONFIG_SND_SOC_AK4613) += snd-soc-ak4613.o obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c new file mode 100644 index 000000000000..fd96a8f9e2d3 --- /dev/null +++ b/sound/soc/codecs/ak4613.c @@ -0,0 +1,469 @@ +/* + * ak4613.c -- Asahi Kasei ALSA Soc Audio driver + * + * Copyright (C) 2015 Renesas Electronics Corporation + * Kuninori Morimoto + * + * Based on ak4642.c by Kuninori Morimoto + * Based on wm8731.c by Richard Purdie + * Based on ak4535.c by Richard Purdie + * Based on wm8753.c by Liam Girdwood + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#define PW_MGMT1 0x00 /* Power Management 1 */ +#define PW_MGMT2 0x01 /* Power Management 2 */ +#define PW_MGMT3 0x02 /* Power Management 3 */ +#define CTRL1 0x03 /* Control 1 */ +#define CTRL2 0x04 /* Control 2 */ +#define DEMP1 0x05 /* De-emphasis1 */ +#define DEMP2 0x06 /* De-emphasis2 */ +#define OFD 0x07 /* Overflow Detect */ +#define ZRD 0x08 /* Zero Detect */ +#define ICTRL 0x09 /* Input Control */ +#define OCTRL 0x0a /* Output Control */ +#define LOUT1 0x0b /* LOUT1 Volume Control */ +#define ROUT1 0x0c /* ROUT1 Volume Control */ +#define LOUT2 0x0d /* LOUT2 Volume Control */ +#define ROUT2 0x0e /* ROUT2 Volume Control */ +#define LOUT3 0x0f /* LOUT3 Volume Control */ +#define ROUT3 0x10 /* ROUT3 Volume Control */ +#define LOUT4 0x11 /* LOUT4 Volume Control */ +#define ROUT4 0x12 /* ROUT4 Volume Control */ +#define LOUT5 0x13 /* LOUT5 Volume Control */ +#define ROUT5 0x14 /* ROUT5 Volume Control */ +#define LOUT6 0x15 /* LOUT6 Volume Control */ +#define ROUT6 0x16 /* ROUT6 Volume Control */ + +/* PW_MGMT1 */ +#define RSTN BIT(0) +#define PMDAC BIT(1) +#define PMADC BIT(2) +#define PMVR BIT(3) + +/* PW_MGMT2 */ +#define PMAD_ALL 0x7 + +/* PW_MGMT3 */ +#define PMDA_ALL 0x3f + +/* CTRL1 */ +#define DIF0 BIT(3) +#define DIF1 BIT(4) +#define DIF2 BIT(5) +#define TDM0 BIT(6) +#define TDM1 BIT(7) +#define NO_FMT (0xff) +#define FMT_MASK (0xf8) + +/* CTRL2 */ +#define DFS_NORMAL_SPEED (0 << 2) +#define DFS_DOUBLE_SPEED (1 << 2) +#define DFS_QUAD_SPEED (2 << 2) + +struct ak4613_priv { + struct mutex lock; + + unsigned int fmt; + u8 fmt_ctrl; + int cnt; +}; + +struct ak4613_formats { + unsigned int width; + unsigned int fmt; +}; + +struct ak4613_interface { + struct ak4613_formats capture; + struct ak4613_formats playback; +}; + +static const struct reg_default ak4613_reg[] = { + { 0x0, 0x0f }, { 0x1, 0x07 }, { 0x2, 0x3f }, { 0x3, 0x20 }, + { 0x4, 0x20 }, { 0x5, 0x55 }, { 0x6, 0x05 }, { 0x7, 0x07 }, + { 0x8, 0x0f }, { 0x9, 0x07 }, { 0xa, 0x3f }, { 0xb, 0x00 }, + { 0xc, 0x00 }, { 0xd, 0x00 }, { 0xe, 0x00 }, { 0xf, 0x00 }, + { 0x10, 0x00 }, { 0x11, 0x00 }, { 0x12, 0x00 }, { 0x13, 0x00 }, + { 0x14, 0x00 }, { 0x15, 0x00 }, { 0x16, 0x00 }, +}; + +#define AUDIO_IFACE_IDX_TO_VAL(i) (i << 3) +#define AUDIO_IFACE(b, fmt) { b, SND_SOC_DAIFMT_##fmt } +static const struct ak4613_interface ak4613_iface[] = { + /* capture */ /* playback */ + [0] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(16, RIGHT_J) }, + [1] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(20, RIGHT_J) }, + [2] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, RIGHT_J) }, + [3] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, LEFT_J) }, + [4] = { AUDIO_IFACE(24, I2S), AUDIO_IFACE(24, I2S) }, +}; + +static const struct regmap_config ak4613_regmap_cfg = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 0x16, + .reg_defaults = ak4613_reg, + .num_reg_defaults = ARRAY_SIZE(ak4613_reg), +}; + +static const struct of_device_id ak4613_of_match[] = { + { .compatible = "asahi-kasei,ak4613", .data = &ak4613_regmap_cfg }, + {}, +}; +MODULE_DEVICE_TABLE(of, ak4613_of_match); + +static const struct i2c_device_id ak4613_i2c_id[] = { + { "ak4613", (kernel_ulong_t)&ak4613_regmap_cfg }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak4613_i2c_id); + +static const struct snd_soc_dapm_widget ak4613_dapm_widgets[] = { + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("LOUT1"), + SND_SOC_DAPM_OUTPUT("LOUT2"), + SND_SOC_DAPM_OUTPUT("LOUT3"), + SND_SOC_DAPM_OUTPUT("LOUT4"), + SND_SOC_DAPM_OUTPUT("LOUT5"), + SND_SOC_DAPM_OUTPUT("LOUT6"), + + SND_SOC_DAPM_OUTPUT("ROUT1"), + SND_SOC_DAPM_OUTPUT("ROUT2"), + SND_SOC_DAPM_OUTPUT("ROUT3"), + SND_SOC_DAPM_OUTPUT("ROUT4"), + SND_SOC_DAPM_OUTPUT("ROUT5"), + SND_SOC_DAPM_OUTPUT("ROUT6"), + + /* Inputs */ + SND_SOC_DAPM_INPUT("LIN1"), + SND_SOC_DAPM_INPUT("LIN2"), + + SND_SOC_DAPM_INPUT("RIN1"), + SND_SOC_DAPM_INPUT("RIN2"), + + /* DAC */ + SND_SOC_DAPM_DAC("DAC1", NULL, PW_MGMT3, 0, 0), + SND_SOC_DAPM_DAC("DAC2", NULL, PW_MGMT3, 1, 0), + SND_SOC_DAPM_DAC("DAC3", NULL, PW_MGMT3, 2, 0), + SND_SOC_DAPM_DAC("DAC4", NULL, PW_MGMT3, 3, 0), + SND_SOC_DAPM_DAC("DAC5", NULL, PW_MGMT3, 4, 0), + SND_SOC_DAPM_DAC("DAC6", NULL, PW_MGMT3, 5, 0), + + /* ADC */ + SND_SOC_DAPM_ADC("ADC1", NULL, PW_MGMT2, 0, 0), + SND_SOC_DAPM_ADC("ADC2", NULL, PW_MGMT2, 1, 0), +}; + +static const struct snd_soc_dapm_route ak4613_intercon[] = { + {"LOUT1", NULL, "DAC1"}, + {"LOUT2", NULL, "DAC2"}, + {"LOUT3", NULL, "DAC3"}, + {"LOUT4", NULL, "DAC4"}, + {"LOUT5", NULL, "DAC5"}, + {"LOUT6", NULL, "DAC6"}, + + {"ROUT1", NULL, "DAC1"}, + {"ROUT2", NULL, "DAC2"}, + {"ROUT3", NULL, "DAC3"}, + {"ROUT4", NULL, "DAC4"}, + {"ROUT5", NULL, "DAC5"}, + {"ROUT6", NULL, "DAC6"}, + + {"DAC1", NULL, "Playback"}, + {"DAC2", NULL, "Playback"}, + {"DAC3", NULL, "Playback"}, + {"DAC4", NULL, "Playback"}, + {"DAC5", NULL, "Playback"}, + {"DAC6", NULL, "Playback"}, + + {"Capture", NULL, "ADC1"}, + {"Capture", NULL, "ADC2"}, + + {"ADC1", NULL, "LIN1"}, + {"ADC2", NULL, "LIN2"}, + + {"ADC1", NULL, "RIN1"}, + {"ADC2", NULL, "RIN2"}, +}; + +static void ak4613_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + struct device *dev = codec->dev; + + mutex_lock(&priv->lock); + priv->cnt--; + if (priv->cnt < 0) { + dev_err(dev, "unexpected counter error\n"); + priv->cnt = 0; + } + if (!priv->cnt) + priv->fmt_ctrl = NO_FMT; + mutex_unlock(&priv->lock); +} + +static int ak4613_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + + fmt &= SND_SOC_DAIFMT_FORMAT_MASK; + + switch (fmt) { + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_I2S: + priv->fmt = fmt; + + break; + default: + return -EINVAL; + } + + return 0; +} + +static int ak4613_dai_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 ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + const struct ak4613_formats *fmts; + struct device *dev = codec->dev; + unsigned int width = params_width(params); + unsigned int fmt = priv->fmt; + unsigned int rate; + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int i, ret; + u8 fmt_ctrl, ctrl2; + + rate = params_rate(params); + switch (rate) { + case 32000: + case 44100: + case 48000: + ctrl2 = DFS_NORMAL_SPEED; + break; + case 88200: + case 96000: + ctrl2 = DFS_DOUBLE_SPEED; + break; + case 176400: + case 192000: + ctrl2 = DFS_QUAD_SPEED; + break; + default: + return -EINVAL; + } + + /* + * FIXME + * + * It doesn't support TDM at this point + */ + fmt_ctrl = NO_FMT; + for (i = 0; i < ARRAY_SIZE(ak4613_iface); i++) { + fmts = (is_play) ? &ak4613_iface[i].playback : + &ak4613_iface[i].capture; + + if (fmts->fmt != fmt) + continue; + + if (fmt == SND_SOC_DAIFMT_RIGHT_J) { + if (fmts->width != width) + continue; + } else { + if (fmts->width < width) + continue; + } + + fmt_ctrl = AUDIO_IFACE_IDX_TO_VAL(i); + break; + } + + ret = -EINVAL; + if (fmt_ctrl == NO_FMT) + goto hw_params_end; + + mutex_lock(&priv->lock); + if ((priv->fmt_ctrl == NO_FMT) || + (priv->fmt_ctrl == fmt_ctrl)) { + priv->fmt_ctrl = fmt_ctrl; + priv->cnt++; + ret = 0; + } + mutex_unlock(&priv->lock); + + if (ret < 0) + goto hw_params_end; + + snd_soc_update_bits(codec, CTRL1, FMT_MASK, fmt_ctrl); + snd_soc_write(codec, CTRL2, ctrl2); + +hw_params_end: + if (ret < 0) + dev_warn(dev, "unsupported data width/format combination\n"); + + return ret; +} + +static int ak4613_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u8 mgmt1 = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + mgmt1 |= RSTN; + /* fall through */ + case SND_SOC_BIAS_PREPARE: + mgmt1 |= PMADC | PMDAC; + /* fall through */ + case SND_SOC_BIAS_STANDBY: + mgmt1 |= PMVR; + /* fall through */ + case SND_SOC_BIAS_OFF: + default: + break; + } + + snd_soc_write(codec, PW_MGMT1, mgmt1); + + return 0; +} + +static const struct snd_soc_dai_ops ak4613_dai_ops = { + .shutdown = ak4613_dai_shutdown, + .set_fmt = ak4613_dai_set_fmt, + .hw_params = ak4613_dai_hw_params, +}; + +#define AK4613_PCM_RATE (SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_64000 |\ + SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_176400 |\ + SNDRV_PCM_RATE_192000) +#define AK4613_PCM_FMTBIT (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver ak4613_dai = { + .name = "ak4613-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = AK4613_PCM_RATE, + .formats = AK4613_PCM_FMTBIT, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = AK4613_PCM_RATE, + .formats = AK4613_PCM_FMTBIT, + }, + .ops = &ak4613_dai_ops, + .symmetric_rates = 1, +}; + +static int ak4613_resume(struct snd_soc_codec *codec) +{ + struct regmap *regmap = dev_get_regmap(codec->dev, NULL); + + regcache_mark_dirty(regmap); + return regcache_sync(regmap); +} + +static struct snd_soc_codec_driver soc_codec_dev_ak4613 = { + .resume = ak4613_resume, + .set_bias_level = ak4613_set_bias_level, + .dapm_widgets = ak4613_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4613_dapm_widgets), + .dapm_routes = ak4613_intercon, + .num_dapm_routes = ARRAY_SIZE(ak4613_intercon), +}; + +static int ak4613_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct device_node *np = dev->of_node; + const struct regmap_config *regmap_cfg; + struct regmap *regmap; + struct ak4613_priv *priv; + + regmap_cfg = NULL; + if (np) { + const struct of_device_id *of_id; + + of_id = of_match_device(ak4613_of_match, dev); + if (of_id) + regmap_cfg = of_id->data; + } else { + regmap_cfg = (const struct regmap_config *)id->driver_data; + } + + if (!regmap_cfg) + return -EINVAL; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->fmt_ctrl = NO_FMT; + priv->cnt = 0; + + mutex_init(&priv->lock); + + i2c_set_clientdata(i2c, priv); + + regmap = devm_regmap_init_i2c(i2c, regmap_cfg); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return snd_soc_register_codec(dev, &soc_codec_dev_ak4613, + &ak4613_dai, 1); +} + +static int ak4613_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static struct i2c_driver ak4613_i2c_driver = { + .driver = { + .name = "ak4613-codec", + .owner = THIS_MODULE, + .of_match_table = ak4613_of_match, + }, + .probe = ak4613_i2c_probe, + .remove = ak4613_i2c_remove, + .id_table = ak4613_i2c_id, +}; + +module_i2c_driver(ak4613_i2c_driver); + +MODULE_DESCRIPTION("Soc AK4613 driver"); +MODULE_AUTHOR("Kuninori Morimoto "); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From b29759ee29d5592b686bb42bb007179c215297c0 Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Tue, 29 Sep 2015 16:44:07 +0100 Subject: ASoC: da7219: Add bindings documentation for DA7219 audio codec Signed-off-by: Adam Thomson Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/da7219.txt | 106 +++++++++++++++++++++ 1 file changed, 106 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/da7219.txt (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/da7219.txt b/Documentation/devicetree/bindings/sound/da7219.txt new file mode 100644 index 000000000000..1b7030911a3b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/da7219.txt @@ -0,0 +1,106 @@ +Dialog Semiconductor DA7219 Audio Codec bindings + +DA7219 is an audio codec with advanced accessory detect features. + +====== + +Required properties: +- compatible : Should be "dlg,da7219" +- reg: Specifies the I2C slave address + +- interrupt-parent : Specifies the phandle of the interrupt controller to which + the IRQs from DA7219 are delivered to. +- interrupts : IRQ line info for DA7219. + (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt for + further information relating to interrupt properties) + +- VDD-supply: VDD power supply for the device +- VDDMIC-supply: VDDMIC power supply for the device +- VDDIO-supply: VDDIO power supply for the device + (See Documentation/devicetree/bindings/regulator/regulator.txt for further + information relating to regulators) + +Optional properties: +- interrupt-names : Name associated with interrupt line. Should be "wakeup" if + interrupt is to be used to wake system, otherwise "irq" should be used. +- wakeup-source: Flag to indicate this device can wake system (suspend/resume). + +- clocks : phandle and clock specifier for codec MCLK. +- clock-names : Clock name string for 'clocks' attribute, should be "mclk". + +- dlg,ldo-lvl : Required internal LDO voltage (mV) level for digital engine + [<1050>, <1100>, <1200>, <1400>] +- dlg,micbias-lvl : Voltage (mV) for Mic Bias + [<1800>, <2000>, <2200>, <2400>, <2600>] +- dlg,mic-amp-in-sel : Mic input source type + ["diff", "se_p", "se_n"] + +====== + +Child node - 'da7219_aad': + +Optional properties: +- dlg,micbias-pulse-lvl : Mic bias higher voltage pulse level (mV). + [<2800>, <2900>] +- dlg,micbias-pulse-time : Mic bias higher voltage pulse duration (ms) +- dlg,btn-cfg : Periodic button press measurements for 4-pole jack (ms) + [<2>, <5>, <10>, <50>, <100>, <200>, <500>] +- dlg,mic-det-thr : Impedance threshold for mic detection measurement (Ohms) + [<200>, <500>, <750>, <1000>] +- dlg,jack-ins-deb : Debounce time for jack insertion (ms) + [<5>, <10>, <20>, <50>, <100>, <200>, <500>, <1000>] +- dlg,jack-det-rate: Jack type detection latency (3/4 pole) + ["32ms_64ms", "64ms_128ms", "128ms_256ms", "256ms_512ms"] +- dlg,jack-rem-deb : Debounce time for jack removal (ms) + [<1>, <5>, <10>, <20>] +- dlg,a-d-btn-thr : Impedance threshold between buttons A and D + [0x0 - 0xFF] +- dlg,d-b-btn-thr : Impedance threshold between buttons D and B + [0x0 - 0xFF] +- dlg,b-c-btn-thr : Impedance threshold between buttons B and C + [0x0 - 0xFF] +- dlg,c-mic-btn-thr : Impedance threshold between button C and Mic + [0x0 - 0xFF] +- dlg,btn-avg : Number of 8-bit readings for averaged button measurement + [<1>, <2>, <4>, <8>] +- dlg,adc-1bit-rpt : Repeat count for 1-bit button measurement + [<1>, <2>, <4>, <8>] + +====== + +Example: + + codec: da7219@1a { + compatible = "dlg,da7219"; + reg = <0x1a>; + + interrupt-parent = <&gpio6>; + interrupts = <11 IRQ_TYPE_LEVEL_HIGH>; + + VDD-supply = <®_audio>; + VDDMIC-supply = <®_audio>; + VDDIO-supply = <®_audio>; + + clocks = <&clks 201>; + clock-names = "mclk"; + + dlg,ldo-lvl = <1200>; + dlg,micbias-lvl = <2600>; + dlg,mic-amp-in-sel = "diff"; + + da7219_aad { + dlg,btn-cfg = <50>; + dlg,mic-det-thr = <500>; + dlg,jack-ins-deb = <20>; + dlg,jack-det-rate = "32ms_64ms"; + dlg,jack-rem-deb = <1>; + + dlg,a-d-btn-thr = <0xa>; + dlg,d-b-btn-thr = <0x16>; + dlg,b-c-btn-thr = <0x21>; + dlg,c-mic-btn-thr = <0x3E>; + + dlg,btn-avg = <4>; + dlg,adc-1bit-rpt = <1>; + }; + }; -- cgit v1.2.3 From 34ca27f34f413b4a684fc7336911799da3ac84d5 Mon Sep 17 00:00:00 2001 From: Anatol Pomozov Date: Fri, 2 Oct 2015 09:49:14 -0700 Subject: ASoC: nau8825: Add driver for headset chip Nuvoton 8825 Sponsored-by: Google Chromium project Signed-off-by: Anatol Pomozov Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/nau8825.txt | 102 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8825.c | 1107 ++++++++++++++++++++ sound/soc/codecs/nau8825.h | 323 ++++++ 5 files changed, 1538 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8825.txt create mode 100644 sound/soc/codecs/nau8825.c create mode 100644 sound/soc/codecs/nau8825.h (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt new file mode 100644 index 000000000000..d3374231c871 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8825.txt @@ -0,0 +1,102 @@ +Nuvoton NAU8825 audio codec + +This device supports I2C only. + +Required properties: + - compatible : Must be "nuvoton,nau8825" + + - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1). + +Optional properties: + - nuvoton,jkdet-enable: Enable jack detection via JKDET pin. + - nuvoton,jkdet-pull-enable: Enable JKDET pin pull. If set - pin pull enabled, + otherwise pin in high impedance state. + - nuvoton,jkdet-pull-up: Pull-up JKDET pin. If set then JKDET pin is pull up, otherwise pull down. + - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low. + + - nuvoton,vref-impedance: VREF Impedance selection + 0 - Open + 1 - 25 kOhm + 2 - 125 kOhm + 3 - 2.5 kOhm + + - nuvoton,micbias-voltage: Micbias voltage level. + 0 - VDDA + 1 - VDDA + 2 - VDDA * 1.1 + 3 - VDDA * 1.2 + 4 - VDDA * 1.3 + 5 - VDDA * 1.4 + 6 - VDDA * 1.53 + 7 - VDDA * 1.53 + + - nuvoton,sar-threshold-num: Number of buttons supported + - nuvoton,sar-threshold: Impedance threshold for each button. Array that contains up to 8 buttons configuration. SAR value is calculated as + SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) + where MICBIAS is configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by 'nuvoton,sar-voltage', R - button impedance. + Refer datasheet section 10.2 for more information about threshold calculation. + + - nuvoton,sar-hysteresis: Button impedance measurement hysteresis. + + - nuvoton,sar-voltage: Reference voltage for button impedance measurement. + 0 - VDDA + 1 - VDDA + 2 - VDDA * 1.1 + 3 - VDDA * 1.2 + 4 - VDDA * 1.3 + 5 - VDDA * 1.4 + 6 - VDDA * 1.53 + 7 - VDDA * 1.53 + + - nuvoton,sar-compare-time: SAR compare time + 0 - 500 ns + 1 - 1 us + 2 - 2 us + 3 - 4 us + + - nuvoton,sar-sampling-time: SAR sampling time + 0 - 2 us + 1 - 4 us + 2 - 8 us + 3 - 16 us + + - nuvoton,short-key-debounce: Button short key press debounce time. + 0 - 30 ms + 1 - 50 ms + 2 - 100 ms + 3 - 30 ms + + - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms + - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms + + - clocks: list of phandle and clock specifier pairs according to common clock bindings for the + clocks described in clock-names + - clock-names: should include "mclk" for the MCLK master clock + +Example: + + headset: nau8825@1a { + compatible = "nuvoton,nau8825"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = ; + nuvoton,jkdet-enable; + nuvoton,jkdet-pull-enable; + nuvoton,jkdet-pull-up; + nuvoton,jkdet-polarity = ; + nuvoton,vref-impedance = <2>; + nuvoton,micbias-voltage = <6>; + // Setup 4 buttons impedance according to Android specification + nuvoton,sar-threshold-num = <4>; + nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; + nuvoton,sar-hysteresis = <1>; + nuvoton,sar-voltage = <0>; + nuvoton,sar-compare-time = <0>; + nuvoton,sar-sampling-time = <0>; + nuvoton,short-key-debounce = <2>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <7>; + + clock-names = "mclk"; + clocks = <&tegra_car TEGRA210_CLK_CLK_OUT_2>; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0142396bb42c..cc60ab92b21d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -79,6 +79,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C + select SND_SOC_NAU8825 if I2C select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER select SND_SOC_PCM3008 @@ -892,6 +893,9 @@ config SND_SOC_MC13783 config SND_SOC_ML26124 tristate +config SND_SOC_NAU8825 + tristate + config SND_SOC_TPA6130A2 tristate "Texas Instruments TPA6130A2 headphone amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7d7cc1b049c2..d7b0f41690b2 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -72,6 +72,7 @@ snd-soc-max98925-objs := max98925.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o +snd-soc-nau8825-objs := nau8825.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-pcm3008-objs := pcm3008.o @@ -263,6 +264,7 @@ obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c new file mode 100644 index 000000000000..f31a5008e879 --- /dev/null +++ b/sound/soc/codecs/nau8825.c @@ -0,0 +1,1107 @@ +/* + * Nuvoton NAU8825 audio codec driver + * + * Copyright 2015 Google Chromium project. + * Author: Anatol Pomozov + * Copyright 2015 Nuvoton Technology Corp. + * Co-author: Meng-Huang Kuo + * + * Licensed under the GPL-2. + */ + +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include +#include + + +#include "nau8825.h" + +static const struct reg_default nau8825_reg_defaults[] = { + { NAU8825_REG_ENA_CTRL, 0x00ff }, + { NAU8825_REG_CLK_DIVIDER, 0x0050 }, + { NAU8825_REG_FLL1, 0x0 }, + { NAU8825_REG_FLL2, 0x3126 }, + { NAU8825_REG_FLL3, 0x0008 }, + { NAU8825_REG_FLL4, 0x0010 }, + { NAU8825_REG_FLL5, 0x0 }, + { NAU8825_REG_FLL6, 0x6000 }, + { NAU8825_REG_FLL_VCO_RSV, 0xf13c }, + { NAU8825_REG_HSD_CTRL, 0x000c }, + { NAU8825_REG_JACK_DET_CTRL, 0x0 }, + { NAU8825_REG_INTERRUPT_MASK, 0x0 }, + { NAU8825_REG_INTERRUPT_DIS_CTRL, 0xffff }, + { NAU8825_REG_SAR_CTRL, 0x0015 }, + { NAU8825_REG_KEYDET_CTRL, 0x0110 }, + { NAU8825_REG_VDET_THRESHOLD_1, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_2, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_3, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_4, 0x0 }, + { NAU8825_REG_GPIO34_CTRL, 0x0 }, + { NAU8825_REG_GPIO12_CTRL, 0x0 }, + { NAU8825_REG_TDM_CTRL, 0x0 }, + { NAU8825_REG_I2S_PCM_CTRL1, 0x000b }, + { NAU8825_REG_I2S_PCM_CTRL2, 0x8010 }, + { NAU8825_REG_LEFT_TIME_SLOT, 0x0 }, + { NAU8825_REG_RIGHT_TIME_SLOT, 0x0 }, + { NAU8825_REG_BIQ_CTRL, 0x0 }, + { NAU8825_REG_BIQ_COF1, 0x0 }, + { NAU8825_REG_BIQ_COF2, 0x0 }, + { NAU8825_REG_BIQ_COF3, 0x0 }, + { NAU8825_REG_BIQ_COF4, 0x0 }, + { NAU8825_REG_BIQ_COF5, 0x0 }, + { NAU8825_REG_BIQ_COF6, 0x0 }, + { NAU8825_REG_BIQ_COF7, 0x0 }, + { NAU8825_REG_BIQ_COF8, 0x0 }, + { NAU8825_REG_BIQ_COF9, 0x0 }, + { NAU8825_REG_BIQ_COF10, 0x0 }, + { NAU8825_REG_ADC_RATE, 0x0010 }, + { NAU8825_REG_DAC_CTRL1, 0x0001 }, + { NAU8825_REG_DAC_CTRL2, 0x0 }, + { NAU8825_REG_DAC_DGAIN_CTRL, 0x0 }, + { NAU8825_REG_ADC_DGAIN_CTRL, 0x00cf }, + { NAU8825_REG_MUTE_CTRL, 0x0 }, + { NAU8825_REG_HSVOL_CTRL, 0x0 }, + { NAU8825_REG_DACL_CTRL, 0x02cf }, + { NAU8825_REG_DACR_CTRL, 0x00cf }, + { NAU8825_REG_ADC_DRC_KNEE_IP12, 0x1486 }, + { NAU8825_REG_ADC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8825_REG_ADC_DRC_SLOPES, 0x25ff }, + { NAU8825_REG_ADC_DRC_ATKDCY, 0x3457 }, + { NAU8825_REG_DAC_DRC_KNEE_IP12, 0x1486 }, + { NAU8825_REG_DAC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8825_REG_DAC_DRC_SLOPES, 0x25f9 }, + { NAU8825_REG_DAC_DRC_ATKDCY, 0x3457 }, + { NAU8825_REG_IMM_MODE_CTRL, 0x0 }, + { NAU8825_REG_CLASSG_CTRL, 0x0 }, + { NAU8825_REG_OPT_EFUSE_CTRL, 0x0 }, + { NAU8825_REG_MISC_CTRL, 0x0 }, + { NAU8825_REG_BIAS_ADJ, 0x0 }, + { NAU8825_REG_TRIM_SETTINGS, 0x0 }, + { NAU8825_REG_ANALOG_CONTROL_1, 0x0 }, + { NAU8825_REG_ANALOG_CONTROL_2, 0x0 }, + { NAU8825_REG_ANALOG_ADC_1, 0x0011 }, + { NAU8825_REG_ANALOG_ADC_2, 0x0020 }, + { NAU8825_REG_RDAC, 0x0008 }, + { NAU8825_REG_MIC_BIAS, 0x0006 }, + { NAU8825_REG_BOOST, 0x0 }, + { NAU8825_REG_FEPGA, 0x0 }, + { NAU8825_REG_POWER_UP_CONTROL, 0x0 }, + { NAU8825_REG_CHARGE_PUMP, 0x0 }, +}; + +static bool nau8825_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_ENA_CTRL: + case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: + case NAU8825_REG_INTERRUPT_MASK ... NAU8825_REG_KEYDET_CTRL: + case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL: + case NAU8825_REG_ADC_DRC_KNEE_IP12 ... NAU8825_REG_ADC_DRC_ATKDCY: + case NAU8825_REG_DAC_DRC_KNEE_IP12 ... NAU8825_REG_DAC_DRC_ATKDCY: + case NAU8825_REG_IMM_MODE_CTRL ... NAU8825_REG_IMM_RMS_R: + case NAU8825_REG_CLASSG_CTRL ... NAU8825_REG_OPT_EFUSE_CTRL: + case NAU8825_REG_MISC_CTRL: + case NAU8825_REG_I2C_DEVICE_ID ... NAU8825_REG_SARDOUT_RAM_STATUS: + case NAU8825_REG_BIAS_ADJ: + case NAU8825_REG_TRIM_SETTINGS ... NAU8825_REG_ANALOG_CONTROL_2: + case NAU8825_REG_ANALOG_ADC_1 ... NAU8825_REG_MIC_BIAS: + case NAU8825_REG_BOOST ... NAU8825_REG_FEPGA: + case NAU8825_REG_POWER_UP_CONTROL ... NAU8825_REG_GENERAL_STATUS: + return true; + default: + return false; + } + +} + +static bool nau8825_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_RESET ... NAU8825_REG_ENA_CTRL: + case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: + case NAU8825_REG_INTERRUPT_MASK: + case NAU8825_REG_INT_CLR_KEY_STATUS ... NAU8825_REG_KEYDET_CTRL: + case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL: + case NAU8825_REG_ADC_DRC_KNEE_IP12 ... NAU8825_REG_ADC_DRC_ATKDCY: + case NAU8825_REG_DAC_DRC_KNEE_IP12 ... NAU8825_REG_DAC_DRC_ATKDCY: + case NAU8825_REG_IMM_MODE_CTRL: + case NAU8825_REG_CLASSG_CTRL ... NAU8825_REG_OPT_EFUSE_CTRL: + case NAU8825_REG_MISC_CTRL: + case NAU8825_REG_BIAS_ADJ: + case NAU8825_REG_TRIM_SETTINGS ... NAU8825_REG_ANALOG_CONTROL_2: + case NAU8825_REG_ANALOG_ADC_1 ... NAU8825_REG_MIC_BIAS: + case NAU8825_REG_BOOST ... NAU8825_REG_FEPGA: + case NAU8825_REG_POWER_UP_CONTROL ... NAU8825_REG_CHARGE_PUMP: + return true; + default: + return false; + } +} + +static bool nau8825_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_RESET: + case NAU8825_REG_IRQ_STATUS: + case NAU8825_REG_INT_CLR_KEY_STATUS: + case NAU8825_REG_IMM_RMS_L: + case NAU8825_REG_IMM_RMS_R: + case NAU8825_REG_I2C_DEVICE_ID: + case NAU8825_REG_SARDOUT_RAM_STATUS: + case NAU8825_REG_CHARGE_PUMP_INPUT_READ: + case NAU8825_REG_GENERAL_STATUS: + return true; + default: + return false; + } +} + +static int nau8825_pump_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* Prevent startup click by letting charge pump to ramp up */ + msleep(10); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const char * const nau8825_adc_decimation[] = { + "32", "64", "128", "256" +}; + +static const struct soc_enum nau8825_adc_decimation_enum = + SOC_ENUM_SINGLE(NAU8825_REG_ADC_RATE, NAU8825_ADC_SYNC_DOWN_SFT, + ARRAY_SIZE(nau8825_adc_decimation), nau8825_adc_decimation); + +static const char * const nau8825_dac_oversampl[] = { + "64", "256", "128", "", "32" +}; + +static const struct soc_enum nau8825_dac_oversampl_enum = + SOC_ENUM_SINGLE(NAU8825_REG_DAC_CTRL1, NAU8825_DAC_OVERSAMPLE_SFT, + ARRAY_SIZE(nau8825_dac_oversampl), nau8825_dac_oversampl); + +static const DECLARE_TLV_DB_MINMAX_MUTE(adc_vol_tlv, -10300, 2400); +static const DECLARE_TLV_DB_MINMAX_MUTE(sidetone_vol_tlv, -4200, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -5400, 0); +static const DECLARE_TLV_DB_MINMAX(fepga_gain_tlv, -100, 3600); +static const DECLARE_TLV_DB_MINMAX_MUTE(crosstalk_vol_tlv, -9600, 2400); + +static const struct snd_kcontrol_new nau8825_controls[] = { + SOC_SINGLE_TLV("Mic Volume", NAU8825_REG_ADC_DGAIN_CTRL, + 0, 0xff, 0, adc_vol_tlv), + SOC_DOUBLE_TLV("Headphone Bypass Volume", NAU8825_REG_ADC_DGAIN_CTRL, + 12, 8, 0x0f, 0, sidetone_vol_tlv), + SOC_DOUBLE_TLV("Headphone Volume", NAU8825_REG_HSVOL_CTRL, + 6, 0, 0x3f, 1, dac_vol_tlv), + SOC_SINGLE_TLV("Frontend PGA Volume", NAU8825_REG_POWER_UP_CONTROL, + 8, 37, 0, fepga_gain_tlv), + SOC_DOUBLE_TLV("Headphone Crosstalk Volume", NAU8825_REG_DAC_DGAIN_CTRL, + 0, 8, 0xff, 0, crosstalk_vol_tlv), + + SOC_ENUM("ADC Decimation Rate", nau8825_adc_decimation_enum), + SOC_ENUM("DAC Oversampling Rate", nau8825_dac_oversampl_enum), +}; + +/* DAC Mux 0x33[9] and 0x34[9] */ +static const char * const nau8825_dac_src[] = { + "DACL", "DACR", +}; + +static SOC_ENUM_SINGLE_DECL( + nau8825_dacl_enum, NAU8825_REG_DACL_CTRL, + NAU8825_DACL_CH_SEL_SFT, nau8825_dac_src); + +static SOC_ENUM_SINGLE_DECL( + nau8825_dacr_enum, NAU8825_REG_DACR_CTRL, + NAU8825_DACR_CH_SEL_SFT, nau8825_dac_src); + +static const struct snd_kcontrol_new nau8825_dacl_mux = + SOC_DAPM_ENUM("DACL Source", nau8825_dacl_enum); + +static const struct snd_kcontrol_new nau8825_dacr_mux = + SOC_DAPM_ENUM("DACR Source", nau8825_dacr_enum); + + +static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { + SND_SOC_DAPM_AIF_OUT("AIFTX", "Capture", 0, NAU8825_REG_I2S_PCM_CTRL2, + 15, 1), + + SND_SOC_DAPM_INPUT("MIC"), + SND_SOC_DAPM_MICBIAS("MICBIAS", NAU8825_REG_MIC_BIAS, 8, 0), + + SND_SOC_DAPM_PGA("Frontend PGA", NAU8825_REG_POWER_UP_CONTROL, 14, 0, + NULL, 0), + + SND_SOC_DAPM_ADC("ADC", NULL, NAU8825_REG_ENA_CTRL, 8, 0), + SND_SOC_DAPM_SUPPLY("ADC Clock", NAU8825_REG_ENA_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Power", NAU8825_REG_ANALOG_ADC_2, 6, 0, NULL, + 0), + + /* ADC for button press detection */ + SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_ADC_EN_SFT, 0), + + SND_SOC_DAPM_DAC("ADACL", NULL, NAU8825_REG_RDAC, 12, 0), + SND_SOC_DAPM_DAC("ADACR", NULL, NAU8825_REG_RDAC, 13, 0), + SND_SOC_DAPM_SUPPLY("ADACL Clock", NAU8825_REG_RDAC, 8, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADACR Clock", NAU8825_REG_RDAC, 9, 0, NULL, 0), + + SND_SOC_DAPM_DAC("DDACR", NULL, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACR_SFT, 0), + SND_SOC_DAPM_DAC("DDACL", NULL, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACL_SFT, 0), + SND_SOC_DAPM_SUPPLY("DDAC Clock", NAU8825_REG_ENA_CTRL, 6, 0, NULL, 0), + + SND_SOC_DAPM_MUX("DACL Mux", SND_SOC_NOPM, 0, 0, &nau8825_dacl_mux), + SND_SOC_DAPM_MUX("DACR Mux", SND_SOC_NOPM, 0, 0, &nau8825_dacr_mux), + + SND_SOC_DAPM_PGA("HP amp L", NAU8825_REG_CLASSG_CTRL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP amp R", NAU8825_REG_CLASSG_CTRL, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP amp power", NAU8825_REG_CLASSG_CTRL, 0, 0, NULL, + 0), + + SND_SOC_DAPM_SUPPLY("Charge Pump", NAU8825_REG_CHARGE_PUMP, 5, 0, + nau8825_pump_event, SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_PGA("Output Driver R Stage 1", + NAU8825_REG_POWER_UP_CONTROL, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("Output Driver L Stage 1", + NAU8825_REG_POWER_UP_CONTROL, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Output Driver R Stage 2", + NAU8825_REG_POWER_UP_CONTROL, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("Output Driver L Stage 2", + NAU8825_REG_POWER_UP_CONTROL, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Output Driver R Stage 3", 1, + NAU8825_REG_POWER_UP_CONTROL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Output Driver L Stage 3", 1, + NAU8825_REG_POWER_UP_CONTROL, 0, 0, NULL, 0), + + SND_SOC_DAPM_PGA_S("Output DACL", 2, NAU8825_REG_CHARGE_PUMP, 8, 1, NULL, 0), + SND_SOC_DAPM_PGA_S("Output DACR", 2, NAU8825_REG_CHARGE_PUMP, 9, 1, NULL, 0), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route nau8825_dapm_routes[] = { + {"Frontend PGA", NULL, "MIC"}, + {"ADC", NULL, "Frontend PGA"}, + {"ADC", NULL, "ADC Clock"}, + {"ADC", NULL, "ADC Power"}, + {"AIFTX", NULL, "ADC"}, + + {"DDACL", NULL, "Playback"}, + {"DDACR", NULL, "Playback"}, + {"DDACL", NULL, "DDAC Clock"}, + {"DDACR", NULL, "DDAC Clock"}, + {"DACL Mux", "DACL", "DDACL"}, + {"DACL Mux", "DACR", "DDACR"}, + {"DACR Mux", "DACL", "DDACL"}, + {"DACR Mux", "DACR", "DDACR"}, + {"HP amp L", NULL, "DACL Mux"}, + {"HP amp R", NULL, "DACR Mux"}, + {"HP amp L", NULL, "HP amp power"}, + {"HP amp R", NULL, "HP amp power"}, + {"ADACL", NULL, "HP amp L"}, + {"ADACR", NULL, "HP amp R"}, + {"ADACL", NULL, "ADACL Clock"}, + {"ADACR", NULL, "ADACR Clock"}, + {"Output Driver L Stage 1", NULL, "ADACL"}, + {"Output Driver R Stage 1", NULL, "ADACR"}, + {"Output Driver L Stage 2", NULL, "Output Driver L Stage 1"}, + {"Output Driver R Stage 2", NULL, "Output Driver R Stage 1"}, + {"Output Driver L Stage 3", NULL, "Output Driver L Stage 2"}, + {"Output Driver R Stage 3", NULL, "Output Driver R Stage 2"}, + {"Output DACL", NULL, "Output Driver L Stage 3"}, + {"Output DACR", NULL, "Output Driver R Stage 3"}, + {"HPOL", NULL, "Output DACL"}, + {"HPOR", NULL, "Output DACR"}, + {"HPOL", NULL, "Charge Pump"}, + {"HPOR", NULL, "Charge Pump"}, +}; + +static int nau8825_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 nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0; + + switch (params_width(params)) { + case 16: + val_len |= NAU8825_I2S_DL_16; + break; + case 20: + val_len |= NAU8825_I2S_DL_20; + break; + case 24: + val_len |= NAU8825_I2S_DL_24; + break; + case 32: + val_len |= NAU8825_I2S_DL_32; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, + NAU8825_I2S_DL_MASK, val_len); + + return 0; +} + +static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + unsigned int ctrl1_val = 0, ctrl2_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= NAU8825_I2S_MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= NAU8825_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= NAU8825_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= NAU8825_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl1_val |= NAU8825_I2S_DF_RIGTH; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= NAU8825_I2S_DF_PCM_AB; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl1_val |= NAU8825_I2S_DF_PCM_AB; + ctrl1_val |= NAU8825_I2S_PCMB_EN; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, + NAU8825_I2S_DL_MASK | NAU8825_I2S_DF_MASK | + NAU8825_I2S_BP_MASK | NAU8825_I2S_PCMB_MASK, + ctrl1_val); + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, ctrl2_val); + + return 0; +} + +static const struct snd_soc_dai_ops nau8825_dai_ops = { + .hw_params = nau8825_hw_params, + .set_fmt = nau8825_set_dai_fmt, +}; + +#define NAU8825_RATES SNDRV_PCM_RATE_8000_192000 +#define NAU8825_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver nau8825_dai = { + .name = "nau8825-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .ops = &nau8825_dai_ops, +}; + +/** + * nau8825_enable_jack_detect - Specify a jack for event reporting + * + * @component: component to register the jack with + * @jack: jack to use to report headset and button events on + * + * After this function has been called the headset insert/remove and button + * events will be routed to the given jack. Jack can be null to stop + * reporting. + */ +int nau8825_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + struct regmap *regmap = nau8825->regmap; + + nau8825->jack = jack; + + /* Ground HP Outputs[1:0], needed for headset auto detection + * Enable Automatic Mic/Gnd switching reading on insert interrupt[6] + */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, + NAU8825_HSD_AUTO_MODE | NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L, + NAU8825_HSD_AUTO_MODE | NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L); + + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_EJECT_EN, 0); + + return 0; +} +EXPORT_SYMBOL_GPL(nau8825_enable_jack_detect); + + +static bool nau8825_is_jack_inserted(struct regmap *regmap) +{ + int status; + + regmap_read(regmap, NAU8825_REG_I2C_DEVICE_ID, &status); + return !(status & NAU8825_GPIO2JD1); +} + +static void nau8825_restart_jack_detection(struct regmap *regmap) +{ + /* this will restart the entire jack detection process including MIC/GND + * switching and create interrupts. We have to go from 0 to 1 and back + * to 0 to restart. + */ + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_RESTART, NAU8825_JACK_DET_RESTART); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_RESTART, 0); +} + +static void nau8825_eject_jack(struct nau8825 *nau8825) +{ + struct snd_soc_dapm_context *dapm = nau8825->dapm; + struct regmap *regmap = nau8825->regmap; + + snd_soc_dapm_disable_pin(dapm, "SAR"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + /* Detach 2kOhm Resistors from MICBIAS to MICGND1/2 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, 0); + /* ground HPL/HPR, MICGRND1/2 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0xf, 0xf); + + snd_soc_dapm_sync(dapm); +} + +static int nau8825_button_decode(int value) +{ + int buttons = 0; + + /* The chip supports up to 8 buttons, but ALSA defines only 6 buttons */ + if (value & BIT(0)) + buttons |= SND_JACK_BTN_0; + if (value & BIT(1)) + buttons |= SND_JACK_BTN_1; + if (value & BIT(2)) + buttons |= SND_JACK_BTN_2; + if (value & BIT(3)) + buttons |= SND_JACK_BTN_3; + if (value & BIT(4)) + buttons |= SND_JACK_BTN_4; + if (value & BIT(5)) + buttons |= SND_JACK_BTN_5; + + return buttons; +} + +static int nau8825_jack_insert(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + struct snd_soc_dapm_context *dapm = nau8825->dapm; + int jack_status_reg, mic_detected; + int type = 0; + + regmap_read(regmap, NAU8825_REG_GENERAL_STATUS, &jack_status_reg); + mic_detected = (jack_status_reg >> 10) & 3; + + switch (mic_detected) { + case 0: + /* no mic */ + type = SND_JACK_HEADPHONE; + break; + case 1: + dev_dbg(nau8825->dev, "OMTP (micgnd1) mic connected\n"); + type = SND_JACK_HEADSET; + + /* Unground MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 3 << 2, + 1 << 2); + /* Attach 2kOhm Resistor from MICBIAS to MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, + NAU8825_MICBIAS_JKR2); + /* Attach SARADC to MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_INPUT_MASK, + NAU8825_SAR_INPUT_JKR2); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); + snd_soc_dapm_sync(dapm); + break; + case 2: + case 3: + dev_dbg(nau8825->dev, "CTIA (micgnd2) mic connected\n"); + type = SND_JACK_HEADSET; + + /* Unground MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 3 << 2, + 2 << 2); + /* Attach 2kOhm Resistor from MICBIAS to MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, + NAU8825_MICBIAS_JKSLV); + /* Attach SARADC to MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_INPUT_MASK, + NAU8825_SAR_INPUT_JKSLV); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); + snd_soc_dapm_sync(dapm); + break; + } + + if (type & SND_JACK_HEADPHONE) { + /* Unground HPL/R */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0x3, 0); + } + + return type; +} + +#define NAU8825_BUTTONS (SND_JACK_BTN_0 | SND_JACK_BTN_1 | \ + SND_JACK_BTN_2 | SND_JACK_BTN_3) + +static irqreturn_t nau8825_interrupt(int irq, void *data) +{ + struct nau8825 *nau8825 = (struct nau8825 *)data; + struct regmap *regmap = nau8825->regmap; + int active_irq, clear_irq = 0, event = 0, event_mask = 0; + + regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq); + + if ((active_irq & NAU8825_JACK_EJECTION_IRQ_MASK) == + NAU8825_JACK_EJECTION_DETECTED) { + + nau8825_eject_jack(nau8825); + event_mask |= SND_JACK_HEADSET; + clear_irq = NAU8825_JACK_EJECTION_IRQ_MASK; + } else if (active_irq & NAU8825_KEY_SHORT_PRESS_IRQ) { + int key_status; + + regmap_read(regmap, NAU8825_REG_INT_CLR_KEY_STATUS, + &key_status); + + /* upper 8 bits of the register are for short pressed keys, + * lower 8 bits - for long pressed buttons + */ + nau8825->button_pressed = nau8825_button_decode( + key_status >> 8); + + event |= nau8825->button_pressed; + event_mask |= NAU8825_BUTTONS; + clear_irq = NAU8825_KEY_SHORT_PRESS_IRQ; + } else if (active_irq & NAU8825_KEY_RELEASE_IRQ) { + event_mask = NAU8825_BUTTONS; + clear_irq = NAU8825_KEY_RELEASE_IRQ; + } else if (active_irq & NAU8825_HEADSET_COMPLETION_IRQ) { + if (nau8825_is_jack_inserted(regmap)) { + event |= nau8825_jack_insert(nau8825); + } else { + dev_warn(nau8825->dev, "Headset completion IRQ fired but no headset connected\n"); + nau8825_eject_jack(nau8825); + } + + event_mask |= SND_JACK_HEADSET; + clear_irq = NAU8825_HEADSET_COMPLETION_IRQ; + } + + if (!clear_irq) + clear_irq = active_irq; + /* clears the rightmost interruption */ + regmap_write(regmap, NAU8825_REG_INT_CLR_KEY_STATUS, clear_irq); + + if (event_mask) + snd_soc_jack_report(nau8825->jack, event, event_mask); + + return IRQ_HANDLED; +} + +static void nau8825_setup_buttons(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_TRACKING_GAIN_MASK, + nau8825->sar_voltage << NAU8825_SAR_TRACKING_GAIN_SFT); + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_COMPARE_TIME_MASK, + nau8825->sar_compare_time << NAU8825_SAR_COMPARE_TIME_SFT); + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_SAMPLING_TIME_MASK, + nau8825->sar_sampling_time << NAU8825_SAR_SAMPLING_TIME_SFT); + + regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL, + NAU8825_KEYDET_LEVELS_NR_MASK, + (nau8825->sar_threshold_num - 1) << NAU8825_KEYDET_LEVELS_NR_SFT); + regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL, + NAU8825_KEYDET_HYSTERESIS_MASK, + nau8825->sar_hysteresis << NAU8825_KEYDET_HYSTERESIS_SFT); + regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL, + NAU8825_KEYDET_SHORTKEY_DEBOUNCE_MASK, + nau8825->key_debounce << NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT); + + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_1, + (nau8825->sar_threshold[0] << 8) | nau8825->sar_threshold[1]); + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_2, + (nau8825->sar_threshold[2] << 8) | nau8825->sar_threshold[3]); + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_3, + (nau8825->sar_threshold[4] << 8) | nau8825->sar_threshold[5]); + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_4, + (nau8825->sar_threshold[6] << 8) | nau8825->sar_threshold[7]); + + /* Enable short press and release interruptions */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_KEY_SHORT_PRESS_EN | NAU8825_IRQ_KEY_RELEASE_EN, + 0); +} + +static void nau8825_init_regs(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + /* Enable Bias/Vmid */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_VMID, NAU8825_BIAS_VMID); + regmap_update_bits(nau8825->regmap, NAU8825_REG_BOOST, + NAU8825_GLOBAL_BIAS_EN, NAU8825_GLOBAL_BIAS_EN); + + /* VMID Tieoff */ + regmap_update_bits(regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_VMID_SEL_MASK, + nau8825->vref_impedance << NAU8825_BIAS_VMID_SEL_SFT); + /* Disable Boost Driver, Automatic Short circuit protection enable */ + regmap_update_bits(regmap, NAU8825_REG_BOOST, + NAU8825_PRECHARGE_DIS | NAU8825_HP_BOOST_G_DIS | + NAU8825_SHORT_SHUTDOWN_EN, + NAU8825_PRECHARGE_DIS | NAU8825_HP_BOOST_G_DIS | + NAU8825_SHORT_SHUTDOWN_EN); + + regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL, + NAU8825_JKDET_OUTPUT_EN, + nau8825->jkdet_enable ? 0 : NAU8825_JKDET_OUTPUT_EN); + regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL, + NAU8825_JKDET_PULL_EN, + nau8825->jkdet_pull_enable ? 0 : NAU8825_JKDET_PULL_EN); + regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL, + NAU8825_JKDET_PULL_UP, + nau8825->jkdet_pull_up ? NAU8825_JKDET_PULL_UP : 0); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_POLARITY, + /* jkdet_polarity - 1 is for active-low */ + nau8825->jkdet_polarity ? 0 : NAU8825_JACK_POLARITY); + + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_INSERT_DEBOUNCE_MASK, + nau8825->jack_insert_debounce << NAU8825_JACK_INSERT_DEBOUNCE_SFT); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_EJECT_DEBOUNCE_MASK, + nau8825->jack_eject_debounce << NAU8825_JACK_EJECT_DEBOUNCE_SFT); + + /* Mask unneeded IRQs: 1 - disable, 0 - enable */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, 0x7ff, 0x7ff); + + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_VOLTAGE_MASK, nau8825->micbias_voltage); + + if (nau8825->sar_threshold_num) + nau8825_setup_buttons(nau8825); + + /* Default oversampling/decimations settings are unusable + * (audible hiss). Set it to something better. + */ + regmap_update_bits(regmap, NAU8825_REG_ADC_RATE, + NAU8825_ADC_SYNC_DOWN_MASK, NAU8825_ADC_SYNC_DOWN_128); + regmap_update_bits(regmap, NAU8825_REG_DAC_CTRL1, + NAU8825_DAC_OVERSAMPLE_MASK, NAU8825_DAC_OVERSAMPLE_128); +} + +static const struct regmap_config nau8825_regmap_config = { + .val_bits = 16, + .reg_bits = 16, + + .max_register = NAU8825_REG_MAX, + .readable_reg = nau8825_readable_reg, + .writeable_reg = nau8825_writeable_reg, + .volatile_reg = nau8825_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8825_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8825_reg_defaults), +}; + +static int nau8825_codec_probe(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + + nau8825->dapm = dapm; + + /* The interrupt clock is gated by x1[10:8], + * one of them needs to be enabled all the time for + * interrupts to happen. + */ + snd_soc_dapm_force_enable_pin(dapm, "DDACR"); + snd_soc_dapm_sync(dapm); + + /* Unmask interruptions. Handler uses dapm object so we can enable + * interruptions only after dapm is fully initialized. + */ + regmap_write(nau8825->regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0); + nau8825_restart_jack_detection(nau8825->regmap); + + return 0; +} + +static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, + unsigned int freq) +{ + struct regmap *regmap = nau8825->regmap; + int ret; + + switch (clk_id) { + case NAU8825_CLK_MCLK: + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); + regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, 0); + + /* We selected MCLK source but the clock itself managed externally */ + if (!nau8825->mclk) + break; + + if (!nau8825->mclk_freq) { + ret = clk_prepare_enable(nau8825->mclk); + if (ret) { + dev_err(nau8825->dev, "Unable to prepare codec mclk\n"); + return ret; + } + } + + if (nau8825->mclk_freq != freq) { + nau8825->mclk_freq = freq; + + freq = clk_round_rate(nau8825->mclk, freq); + ret = clk_set_rate(nau8825->mclk, freq); + if (ret) { + dev_err(nau8825->dev, "Unable to set mclk rate\n"); + return ret; + } + } + + break; + case NAU8825_CLK_INTERNAL: + regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, + NAU8825_DCO_EN); + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); + + if (nau8825->mclk_freq) { + clk_disable_unprepare(nau8825->mclk); + nau8825->mclk_freq = 0; + } + + break; + default: + dev_err(nau8825->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + + dev_dbg(nau8825->dev, "Sysclk is %dHz and clock id is %d\n", freq, + clk_id); + return 0; +} + +static int nau8825_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + return nau8825_configure_sysclk(nau8825, clk_id, freq); +} + +static int nau8825_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + if (nau8825->mclk_freq) { + ret = clk_prepare_enable(nau8825->mclk); + if (ret) { + dev_err(nau8825->dev, "Unable to prepare codec mclk\n"); + return ret; + } + } + + ret = regcache_sync(nau8825->regmap); + if (ret) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + + break; + + case SND_SOC_BIAS_OFF: + if (nau8825->mclk_freq) + clk_disable_unprepare(nau8825->mclk); + + regcache_mark_dirty(nau8825->regmap); + break; + } + return 0; +} + +static struct snd_soc_codec_driver nau8825_codec_driver = { + .probe = nau8825_codec_probe, + .set_sysclk = nau8825_set_sysclk, + .set_bias_level = nau8825_set_bias_level, + .suspend_bias_off = true, + + .controls = nau8825_controls, + .num_controls = ARRAY_SIZE(nau8825_controls), + .dapm_widgets = nau8825_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8825_dapm_widgets), + .dapm_routes = nau8825_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8825_dapm_routes), +}; + +static void nau8825_reset_chip(struct regmap *regmap) +{ + regmap_write(regmap, NAU8825_REG_RESET, 0x00); + regmap_write(regmap, NAU8825_REG_RESET, 0x00); +} + +static int nau8825_read_device_properties(struct device *dev, + struct nau8825 *nau8825) { + + nau8825->jkdet_enable = device_property_read_bool(dev, + "nuvoton,jkdet-enable"); + nau8825->jkdet_pull_enable = device_property_read_bool(dev, + "nuvoton,jkdet-pull-enable"); + nau8825->jkdet_pull_up = device_property_read_bool(dev, + "nuvoton,jkdet-pull-up"); + device_property_read_u32(dev, "nuvoton,jkdet-polarity", + &nau8825->jkdet_polarity); + device_property_read_u32(dev, "nuvoton,micbias-voltage", + &nau8825->micbias_voltage); + device_property_read_u32(dev, "nuvoton,vref-impedance", + &nau8825->vref_impedance); + device_property_read_u32(dev, "nuvoton,sar-threshold-num", + &nau8825->sar_threshold_num); + device_property_read_u32_array(dev, "nuvoton,sar-threshold", + nau8825->sar_threshold, nau8825->sar_threshold_num); + device_property_read_u32(dev, "nuvoton,sar-hysteresis", + &nau8825->sar_hysteresis); + device_property_read_u32(dev, "nuvoton,sar-voltage", + &nau8825->sar_voltage); + device_property_read_u32(dev, "nuvoton,sar-compare-time", + &nau8825->sar_compare_time); + device_property_read_u32(dev, "nuvoton,sar-sampling-time", + &nau8825->sar_sampling_time); + device_property_read_u32(dev, "nuvoton,short-key-debounce", + &nau8825->key_debounce); + device_property_read_u32(dev, "nuvoton,jack-insert-debounce", + &nau8825->jack_insert_debounce); + device_property_read_u32(dev, "nuvoton,jack-eject-debounce", + &nau8825->jack_eject_debounce); + + nau8825->mclk = devm_clk_get(dev, "mclk"); + if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) { + return -EPROBE_DEFER; + } else if (PTR_ERR(nau8825->mclk) == -ENOENT) { + /* The MCLK is managed externally or not used at all */ + nau8825->mclk = NULL; + dev_info(dev, "No 'mclk' clock found, assume MCLK is managed externally"); + } else if (IS_ERR(nau8825->mclk)) { + return -EINVAL; + } + + return 0; +} + +static int nau8825_setup_irq(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + int ret; + + /* IRQ Output Enable */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_OUTPUT_EN, NAU8825_IRQ_OUTPUT_EN); + + /* Enable internal VCO needed for interruptions */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); + + /* Enable DDACR needed for interrupts + * It is the same as force_enable_pin("DDACR") we do later + */ + regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACR, NAU8825_ENABLE_DACR); + + /* Chip needs one FSCLK cycle in order to generate interrupts, + * as we cannot guarantee one will be provided by the system. Turning + * master mode on then off enables us to generate that FSCLK cycle + * with a minimum of contention on the clock bus. + */ + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); + + ret = devm_request_threaded_irq(nau8825->dev, nau8825->irq, NULL, + nau8825_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "nau8825", nau8825); + + if (ret) { + dev_err(nau8825->dev, "Cannot request irq %d (%d)\n", + nau8825->irq, ret); + return ret; + } + + return 0; +} + +static int nau8825_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8825 *nau8825 = dev_get_platdata(&i2c->dev); + int ret, value; + + if (!nau8825) { + nau8825 = devm_kzalloc(dev, sizeof(*nau8825), GFP_KERNEL); + if (!nau8825) + return -ENOMEM; + ret = nau8825_read_device_properties(dev, nau8825); + if (ret) + return ret; + } + + i2c_set_clientdata(i2c, nau8825); + + nau8825->regmap = devm_regmap_init_i2c(i2c, &nau8825_regmap_config); + if (IS_ERR(nau8825->regmap)) + return PTR_ERR(nau8825->regmap); + nau8825->dev = dev; + nau8825->irq = i2c->irq; + + nau8825_reset_chip(nau8825->regmap); + ret = regmap_read(nau8825->regmap, NAU8825_REG_I2C_DEVICE_ID, &value); + if (ret < 0) { + dev_err(dev, "Failed to read device id from the NAU8825: %d\n", + ret); + return ret; + } + if ((value & NAU8825_SOFTWARE_ID_MASK) != + NAU8825_SOFTWARE_ID_NAU8825) { + dev_err(dev, "Not a NAU8825 chip\n"); + return -ENODEV; + } + + nau8825_init_regs(nau8825); + + if (i2c->irq) + nau8825_setup_irq(nau8825); + + return snd_soc_register_codec(&i2c->dev, &nau8825_codec_driver, + &nau8825_dai, 1); +} + +static int nau8825_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id nau8825_i2c_ids[] = { + { "nau8825", 0 }, + { } +}; + +#ifdef CONFIG_OF +static const struct of_device_id nau8825_of_ids[] = { + { .compatible = "nuvoton,nau8825", }, + {} +}; +MODULE_DEVICE_TABLE(of, nau8825_of_ids); +#endif + +static struct i2c_driver nau8825_driver = { + .driver = { + .name = "nau8825", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(nau8825_of_ids), + }, + .probe = nau8825_i2c_probe, + .remove = nau8825_i2c_remove, + .id_table = nau8825_i2c_ids, +}; +module_i2c_driver(nau8825_driver); + +MODULE_DESCRIPTION("ASoC nau8825 driver"); +MODULE_AUTHOR("Anatol Pomozov "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h new file mode 100644 index 000000000000..8774923502b4 --- /dev/null +++ b/sound/soc/codecs/nau8825.h @@ -0,0 +1,323 @@ +/* + * NAU8825 ALSA SoC audio driver + * + * Copyright 2015 Google Inc. + * Author: Anatol Pomozov + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __NAU8825_H__ +#define __NAU8825_H__ + +#define NAU8825_REG_RESET 0x00 +#define NAU8825_REG_ENA_CTRL 0x01 +#define NAU8825_REG_CLK_DIVIDER 0x03 +#define NAU8825_REG_FLL1 0x04 +#define NAU8825_REG_FLL2 0x05 +#define NAU8825_REG_FLL3 0x06 +#define NAU8825_REG_FLL4 0x07 +#define NAU8825_REG_FLL5 0x08 +#define NAU8825_REG_FLL6 0x09 +#define NAU8825_REG_FLL_VCO_RSV 0x0a +#define NAU8825_REG_HSD_CTRL 0x0c +#define NAU8825_REG_JACK_DET_CTRL 0x0d +#define NAU8825_REG_INTERRUPT_MASK 0x0f +#define NAU8825_REG_IRQ_STATUS 0x10 +#define NAU8825_REG_INT_CLR_KEY_STATUS 0x11 +#define NAU8825_REG_INTERRUPT_DIS_CTRL 0x12 +#define NAU8825_REG_SAR_CTRL 0x13 +#define NAU8825_REG_KEYDET_CTRL 0x14 +#define NAU8825_REG_VDET_THRESHOLD_1 0x15 +#define NAU8825_REG_VDET_THRESHOLD_2 0x16 +#define NAU8825_REG_VDET_THRESHOLD_3 0x17 +#define NAU8825_REG_VDET_THRESHOLD_4 0x18 +#define NAU8825_REG_GPIO34_CTRL 0x19 +#define NAU8825_REG_GPIO12_CTRL 0x1a +#define NAU8825_REG_TDM_CTRL 0x1b +#define NAU8825_REG_I2S_PCM_CTRL1 0x1c +#define NAU8825_REG_I2S_PCM_CTRL2 0x1d +#define NAU8825_REG_LEFT_TIME_SLOT 0x1e +#define NAU8825_REG_RIGHT_TIME_SLOT 0x1f +#define NAU8825_REG_BIQ_CTRL 0x20 +#define NAU8825_REG_BIQ_COF1 0x21 +#define NAU8825_REG_BIQ_COF2 0x22 +#define NAU8825_REG_BIQ_COF3 0x23 +#define NAU8825_REG_BIQ_COF4 0x24 +#define NAU8825_REG_BIQ_COF5 0x25 +#define NAU8825_REG_BIQ_COF6 0x26 +#define NAU8825_REG_BIQ_COF7 0x27 +#define NAU8825_REG_BIQ_COF8 0x28 +#define NAU8825_REG_BIQ_COF9 0x29 +#define NAU8825_REG_BIQ_COF10 0x2a +#define NAU8825_REG_ADC_RATE 0x2b +#define NAU8825_REG_DAC_CTRL1 0x2c +#define NAU8825_REG_DAC_CTRL2 0x2d +#define NAU8825_REG_DAC_DGAIN_CTRL 0x2f +#define NAU8825_REG_ADC_DGAIN_CTRL 0x30 +#define NAU8825_REG_MUTE_CTRL 0x31 +#define NAU8825_REG_HSVOL_CTRL 0x32 +#define NAU8825_REG_DACL_CTRL 0x33 +#define NAU8825_REG_DACR_CTRL 0x34 +#define NAU8825_REG_ADC_DRC_KNEE_IP12 0x38 +#define NAU8825_REG_ADC_DRC_KNEE_IP34 0x39 +#define NAU8825_REG_ADC_DRC_SLOPES 0x3a +#define NAU8825_REG_ADC_DRC_ATKDCY 0x3b +#define NAU8825_REG_DAC_DRC_KNEE_IP12 0x45 +#define NAU8825_REG_DAC_DRC_KNEE_IP34 0x46 +#define NAU8825_REG_DAC_DRC_SLOPES 0x47 +#define NAU8825_REG_DAC_DRC_ATKDCY 0x48 +#define NAU8825_REG_IMM_MODE_CTRL 0x4c +#define NAU8825_REG_IMM_RMS_L 0x4d +#define NAU8825_REG_IMM_RMS_R 0x4e +#define NAU8825_REG_CLASSG_CTRL 0x50 +#define NAU8825_REG_OPT_EFUSE_CTRL 0x51 +#define NAU8825_REG_MISC_CTRL 0x55 +#define NAU8825_REG_I2C_DEVICE_ID 0x58 +#define NAU8825_REG_SARDOUT_RAM_STATUS 0x59 +#define NAU8825_REG_BIAS_ADJ 0x66 +#define NAU8825_REG_TRIM_SETTINGS 0x68 +#define NAU8825_REG_ANALOG_CONTROL_1 0x69 +#define NAU8825_REG_ANALOG_CONTROL_2 0x6a +#define NAU8825_REG_ANALOG_ADC_1 0x71 +#define NAU8825_REG_ANALOG_ADC_2 0x72 +#define NAU8825_REG_RDAC 0x73 +#define NAU8825_REG_MIC_BIAS 0x74 +#define NAU8825_REG_BOOST 0x76 +#define NAU8825_REG_FEPGA 0x77 +#define NAU8825_REG_POWER_UP_CONTROL 0x7f +#define NAU8825_REG_CHARGE_PUMP 0x80 +#define NAU8825_REG_CHARGE_PUMP_INPUT_READ 0x81 +#define NAU8825_REG_GENERAL_STATUS 0x82 +#define NAU8825_REG_MAX NAU8825_REG_GENERAL_STATUS + +/* ENA_CTRL (0x1) */ +#define NAU8825_ENABLE_DACR_SFT 10 +#define NAU8825_ENABLE_DACR (1 << NAU8825_ENABLE_DACR_SFT) +#define NAU8825_ENABLE_DACL_SFT 9 +#define NAU8825_ENABLE_ADC_SFT 8 +#define NAU8825_ENABLE_SAR_SFT 1 + +/* CLK_DIVIDER (0x3) */ +#define NAU8825_CLK_SRC_SFT 15 +#define NAU8825_CLK_SRC_MASK (1 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_VCO (1 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_MCLK (0 << NAU8825_CLK_SRC_SFT) + +/* FLL6 (0x9) */ +#define NAU8825_DCO_EN (1 << 15) + +/* HSD_CTRL (0xc) */ +#define NAU8825_HSD_AUTO_MODE (1 << 6) +/* 0 - short to GND, 1 - open */ +#define NAU8825_SPKR_DWN1R (1 << 1) +#define NAU8825_SPKR_DWN1L (1 << 0) + +/* JACK_DET_CTRL (0xd) */ +#define NAU8825_JACK_DET_RESTART (1 << 9) +#define NAU8825_JACK_INSERT_DEBOUNCE_SFT 5 +#define NAU8825_JACK_INSERT_DEBOUNCE_MASK (0x7 << NAU8825_JACK_INSERT_DEBOUNCE_SFT) +#define NAU8825_JACK_EJECT_DEBOUNCE_SFT 2 +#define NAU8825_JACK_EJECT_DEBOUNCE_MASK (0x7 << NAU8825_JACK_EJECT_DEBOUNCE_SFT) +#define NAU8825_JACK_POLARITY (1 << 1) /* 0 - active low, 1 - active high */ + +/* INTERRUPT_MASK (0xf) */ +#define NAU8825_IRQ_OUTPUT_EN (1 << 11) +#define NAU8825_IRQ_HEADSET_COMPLETE_EN (1 << 10) +#define NAU8825_IRQ_KEY_RELEASE_EN (1 << 7) +#define NAU8825_IRQ_KEY_SHORT_PRESS_EN (1 << 5) +#define NAU8825_IRQ_EJECT_EN (1 << 2) + +/* IRQ_STATUS (0x10) */ +#define NAU8825_HEADSET_COMPLETION_IRQ (1 << 10) +#define NAU8825_SHORT_CIRCUIT_IRQ (1 << 9) +#define NAU8825_IMPEDANCE_MEAS_IRQ (1 << 8) +#define NAU8825_KEY_IRQ_MASK (0x7 << 5) +#define NAU8825_KEY_RELEASE_IRQ (1 << 7) +#define NAU8825_KEY_LONG_PRESS_IRQ (1 << 6) +#define NAU8825_KEY_SHORT_PRESS_IRQ (1 << 5) +#define NAU8825_MIC_DETECTION_IRQ (1 << 4) +#define NAU8825_JACK_EJECTION_IRQ_MASK (3 << 2) +#define NAU8825_JACK_EJECTION_DETECTED (1 << 2) +#define NAU8825_JACK_INSERTION_IRQ_MASK (3 << 0) +#define NAU8825_JACK_INSERTION_DETECTED (1 << 0) + +/* INTERRUPT_DIS_CTRL (0x12) */ +#define NAU8825_IRQ_HEADSET_COMPLETE_DIS (1 << 10) +#define NAU8825_IRQ_KEY_RELEASE_DIS (1 << 7) +#define NAU8825_IRQ_KEY_SHORT_PRESS_DIS (1 << 5) +#define NAU8825_IRQ_EJECT_DIS (1 << 2) + +/* SAR_CTRL (0x13) */ +#define NAU8825_SAR_ADC_EN_SFT 12 +#define NAU8825_SAR_ADC_EN (1 << NAU8825_SAR_ADC_EN_SFT) +#define NAU8825_SAR_INPUT_MASK (1 << 11) +#define NAU8825_SAR_INPUT_JKSLV (1 << 11) +#define NAU8825_SAR_INPUT_JKR2 (0 << 11) +#define NAU8825_SAR_TRACKING_GAIN_SFT 8 +#define NAU8825_SAR_TRACKING_GAIN_MASK (0x7 << NAU8825_SAR_TRACKING_GAIN_SFT) +#define NAU8825_SAR_COMPARE_TIME_SFT 2 +#define NAU8825_SAR_COMPARE_TIME_MASK (3 << 2) +#define NAU8825_SAR_SAMPLING_TIME_SFT 0 +#define NAU8825_SAR_SAMPLING_TIME_MASK (3 << 0) + +/* KEYDET_CTRL (0x14) */ +#define NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT 12 +#define NAU8825_KEYDET_SHORTKEY_DEBOUNCE_MASK (0x3 << NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT) +#define NAU8825_KEYDET_LEVELS_NR_SFT 8 +#define NAU8825_KEYDET_LEVELS_NR_MASK (0x7 << 8) +#define NAU8825_KEYDET_HYSTERESIS_SFT 0 +#define NAU8825_KEYDET_HYSTERESIS_MASK 0xf + +/* GPIO12_CTRL (0x1a) */ +#define NAU8825_JKDET_PULL_UP (1 << 11) /* 0 - pull down, 1 - pull up */ +#define NAU8825_JKDET_PULL_EN (1 << 9) /* 0 - enable pull, 1 - disable */ +#define NAU8825_JKDET_OUTPUT_EN (1 << 8) /* 0 - enable input, 1 - enable output */ + +/* I2S_PCM_CTRL1 (0x1c) */ +#define NAU8825_I2S_BP_SFT 7 +#define NAU8825_I2S_BP_MASK (1 << NAU8825_I2S_BP_SFT) +#define NAU8825_I2S_BP_INV (1 << NAU8825_I2S_BP_SFT) +#define NAU8825_I2S_PCMB_SFT 6 +#define NAU8825_I2S_PCMB_MASK (1 << NAU8825_I2S_PCMB_SFT) +#define NAU8825_I2S_PCMB_EN (1 << NAU8825_I2S_PCMB_SFT) +#define NAU8825_I2S_DL_SFT 2 +#define NAU8825_I2S_DL_MASK (0x3 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_16 (0 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_20 (1 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_24 (2 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_32 (3 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DF_SFT 0 +#define NAU8825_I2S_DF_MASK (0x3 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_RIGTH (0 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_LEFT (1 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_I2S (2 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_PCM_AB (3 << NAU8825_I2S_DF_SFT) + +/* I2S_PCM_CTRL2 (0x1d) */ +#define NAU8825_I2S_TRISTATE (1 << 15) /* 0 - normal mode, 1 - Hi-Z output */ +#define NAU8825_I2S_MS_SFT 3 +#define NAU8825_I2S_MS_MASK (1 << NAU8825_I2S_MS_SFT) +#define NAU8825_I2S_MS_MASTER (1 << NAU8825_I2S_MS_SFT) +#define NAU8825_I2S_MS_SLAVE (0 << NAU8825_I2S_MS_SFT) + +/* ADC_RATE (0x2b) */ +#define NAU8825_ADC_SYNC_DOWN_SFT 0 +#define NAU8825_ADC_SYNC_DOWN_MASK 0x3 +#define NAU8825_ADC_SYNC_DOWN_32 0 +#define NAU8825_ADC_SYNC_DOWN_64 1 +#define NAU8825_ADC_SYNC_DOWN_128 2 +#define NAU8825_ADC_SYNC_DOWN_256 3 + +/* DAC_CTRL1 (0x2c) */ +#define NAU8825_DAC_CLIP_OFF (1 << 7) +#define NAU8825_DAC_OVERSAMPLE_SFT 0 +#define NAU8825_DAC_OVERSAMPLE_MASK 0x7 +#define NAU8825_DAC_OVERSAMPLE_64 0 +#define NAU8825_DAC_OVERSAMPLE_256 1 +#define NAU8825_DAC_OVERSAMPLE_128 2 +#define NAU8825_DAC_OVERSAMPLE_32 4 + +/* MUTE_CTRL (0x31) */ +#define NAU8825_DAC_ZERO_CROSSING_EN (1 << 9) +#define NAU8825_DAC_SOFT_MUTE (1 << 9) + +/* HSVOL_CTRL (0x32) */ +#define NAU8825_HP_MUTE (1 << 15) + +/* DACL_CTRL (0x33) */ +#define NAU8825_DACL_CH_SEL_SFT 9 + +/* DACR_CTRL (0x34) */ +#define NAU8825_DACR_CH_SEL_SFT 9 + +/* I2C_DEVICE_ID (0x58) */ +#define NAU8825_GPIO2JD1 (1 << 7) +#define NAU8825_SOFTWARE_ID_MASK 0x3 +#define NAU8825_SOFTWARE_ID_NAU8825 0x0 + +/* BIAS_ADJ (0x66) */ +#define NAU8825_BIAS_VMID (1 << 6) +#define NAU8825_BIAS_VMID_SEL_SFT 4 +#define NAU8825_BIAS_VMID_SEL_MASK (3 << NAU8825_BIAS_VMID_SEL_SFT) + +/* ANALOG_CONTROL_2 (0x6a) */ +#define NAU8825_HP_NON_CLASSG_CURRENT_2xADJ (1 << 12) +#define NAU8825_DAC_CAPACITOR_MSB (1 << 1) +#define NAU8825_DAC_CAPACITOR_LSB (1 << 0) + +/* ANALOG_ADC_2 (0x72) */ +#define NAU8825_ADC_VREFSEL_MASK (0x3 << 8) +#define NAU8825_ADC_VREFSEL_ANALOG (0 << 8) +#define NAU8825_ADC_VREFSEL_VMID (1 << 8) +#define NAU8825_ADC_VREFSEL_VMID_PLUS_0_5DB (2 << 8) +#define NAU8825_ADC_VREFSEL_VMID_PLUS_1DB (3 << 8) +#define NAU8825_POWERUP_ADCL (1 << 6) + +/* MIC_BIAS (0x74) */ +#define NAU8825_MICBIAS_JKSLV (1 << 14) +#define NAU8825_MICBIAS_JKR2 (1 << 12) +#define NAU8825_MICBIAS_POWERUP_SFT 8 +#define NAU8825_MICBIAS_VOLTAGE_SFT 0 +#define NAU8825_MICBIAS_VOLTAGE_MASK 0x7 + +/* BOOST (0x76) */ +#define NAU8825_PRECHARGE_DIS (1 << 13) +#define NAU8825_GLOBAL_BIAS_EN (1 << 12) +#define NAU8825_HP_BOOST_G_DIS (1 << 8) +#define NAU8825_SHORT_SHUTDOWN_EN (1 << 6) + +/* POWER_UP_CONTROL (0x7f) */ +#define NAU8825_POWERUP_INTEGR_R (1 << 5) +#define NAU8825_POWERUP_INTEGR_L (1 << 4) +#define NAU8825_POWERUP_DRV_IN_R (1 << 3) +#define NAU8825_POWERUP_DRV_IN_L (1 << 2) +#define NAU8825_POWERUP_HP_DRV_R (1 << 1) +#define NAU8825_POWERUP_HP_DRV_L (1 << 0) + +/* CHARGE_PUMP (0x80) */ +#define NAU8825_JAMNODCLOW (1 << 10) +#define NAU8825_POWER_DOWN_DACR (1 << 9) +#define NAU8825_POWER_DOWN_DACL (1 << 8) +#define NAU8825_CHANRGE_PUMP_EN (1 << 5) + + +/* System Clock Source */ +enum { + NAU8825_CLK_MCLK = 0, + NAU8825_CLK_INTERNAL, +}; + +struct nau8825 { + struct device *dev; + struct regmap *regmap; + struct snd_soc_dapm_context *dapm; + struct snd_soc_jack *jack; + struct clk *mclk; + int irq; + int mclk_freq; /* 0 - mclk is disabled */ + int button_pressed; + int micbias_voltage; + int vref_impedance; + bool jkdet_enable; + bool jkdet_pull_enable; + bool jkdet_pull_up; + int jkdet_polarity; + int sar_threshold_num; + int sar_threshold[8]; + int sar_hysteresis; + int sar_voltage; + int sar_compare_time; + int sar_sampling_time; + int key_debounce; + int jack_insert_debounce; + int jack_eject_debounce; +}; + +int nau8825_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); + + +#endif /* __NAU8825_H__ */ -- cgit v1.2.3 From c570b82c5e6ac78be35b4e72594c0f1b1888fce1 Mon Sep 17 00:00:00 2001 From: Maxime Ripard Date: Wed, 7 Oct 2015 11:59:58 +0100 Subject: ASoC: sun4i-codec: Remove the routing property Most of the boards have their headphone jack directly connected to the matching pins of the SoCs. Since most of the time we will have the same routing path, it makes no sense to put that in the DTS, since it will only be some useless duplication there. It also fixes the following warning messages that were seen so far, on boards where we were using the bindings in the documentation example. sun4i-codec 1c22c00.codec: ASoC: no sink widget found for Headphone Jack sun4i-codec 1c22c00.codec: ASoC: Failed to add route HP Left -> direct -> Headphone Jack sun4i-codec 1c22c00.codec: ASoC: no sink widget found for Headphone Jack sun4i-codec 1c22c00.codec: ASoC: Failed to add route HP Right -> direct -> Headphone Jack Reported-by: Priit Laes Signed-off-by: Maxime Ripard Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/sun4i-codec.txt | 6 ------ sound/soc/sunxi/sun4i-codec.c | 7 ------- 2 files changed, 13 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/sun4i-codec.txt b/Documentation/devicetree/bindings/sound/sun4i-codec.txt index 680144b74ae9..c92966bd5488 100644 --- a/Documentation/devicetree/bindings/sound/sun4i-codec.txt +++ b/Documentation/devicetree/bindings/sound/sun4i-codec.txt @@ -13,10 +13,6 @@ Required properties: - clock-names: should contain followings: - "apb": the parent APB clock for this controller - "codec": the parent module clock -- 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. - Example: codec: codec@01c22c00 { @@ -28,6 +24,4 @@ codec: codec@01c22c00 { clock-names = "apb", "codec"; dmas = <&dma 0 19>, <&dma 0 19>; dma-names = "rx", "tx"; - routing = "Headphone Jack", "HP Right", - "Headphone Jack", "HP Left"; }; diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 47780552dcd0..bcbf4da168b6 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -571,7 +571,6 @@ static struct snd_soc_dai_link *sun4i_codec_create_link(struct device *dev, static struct snd_soc_card *sun4i_codec_create_card(struct device *dev) { struct snd_soc_card *card; - int ret; card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); if (!card) @@ -584,12 +583,6 @@ static struct snd_soc_card *sun4i_codec_create_card(struct device *dev) card->dev = dev; card->name = "sun4i-codec"; - ret = snd_soc_of_parse_audio_routing(card, "routing"); - if (ret) { - dev_err(dev, "Failed to create our audio routing\n"); - return NULL; - } - return card; }; -- cgit v1.2.3 From a97d4e93a9bc556d0b3c2efb7695eb4c79938de7 Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Wed, 7 Oct 2015 14:27:13 +0100 Subject: ASoC: da7213: Add bindings documentation for codec driver Signed-off-by: Adam Thomson Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/da7213.txt | 41 ++++++++++++++++++++++ 1 file changed, 41 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/da7213.txt (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/da7213.txt b/Documentation/devicetree/bindings/sound/da7213.txt new file mode 100644 index 000000000000..7280e828ab91 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/da7213.txt @@ -0,0 +1,41 @@ +Dialog Semiconductor DA7213 Audio Codec bindings + +====== + +Required properties: +- compatible : Should be "dlg,da7213" +- reg: Specifies the I2C slave address + +Optional properties: +- clocks : phandle and clock specifier for codec MCLK. +- clock-names : Clock name string for 'clocks' attribute, should be "mclk". + +- dlg,micbias1-lvl : Voltage (mV) for Mic Bias 1 + [<1600>, <2200>, <2500>, <3000>] +- dlg,micbias2-lvl : Voltage (mV) for Mic Bias 2 + [<1600>, <2200>, <2500>, <3000>] +- dlg,dmic-data-sel : DMIC channel select based on clock edge. + ["lrise_rfall", "lfall_rrise"] +- dlg,dmic-samplephase : When to sample audio from DMIC. + ["on_clkedge", "between_clkedge"] +- dlg,dmic-clkrate : DMIC clock frequency (MHz). + [<1500000>, <3000000>] + +====== + +Example: + + codec_i2c: da7213@1a { + compatible = "dlg,da7213"; + reg = <0x1a>; + + clocks = <&clks 201>; + clock-names = "mclk"; + + dlg,micbias1-lvl = <2500>; + dlg,micbias2-lvl = <2500>; + + dlg,dmic-data-sel = "lrise_rfall"; + dlg,dmic-samplephase = "between_clkedge"; + dlg,dmic-clkrate = <3000000>; + }; -- cgit v1.2.3 From 955da48532f6d6254cca38f6f247ddeee1ebd722 Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Thu, 8 Oct 2015 11:21:54 +0100 Subject: ASoC: da7213: Correct units description of dmic-clkrate Description previously stated MHz whereas the resolution is in Hz. This has now has been updated to align with reality. Signed-off-by: Adam Thomson Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/da7213.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/da7213.txt b/Documentation/devicetree/bindings/sound/da7213.txt index 7280e828ab91..58902802d56c 100644 --- a/Documentation/devicetree/bindings/sound/da7213.txt +++ b/Documentation/devicetree/bindings/sound/da7213.txt @@ -18,7 +18,7 @@ Optional properties: ["lrise_rfall", "lfall_rrise"] - dlg,dmic-samplephase : When to sample audio from DMIC. ["on_clkedge", "between_clkedge"] -- dlg,dmic-clkrate : DMIC clock frequency (MHz). +- dlg,dmic-clkrate : DMIC clock frequency (Hz). [<1500000>, <3000000>] ====== -- cgit v1.2.3 From 51e5084e718f990e88aeb0a9219adef15f847dc8 Mon Sep 17 00:00:00 2001 From: Sjoerd Simons Date: Thu, 8 Oct 2015 15:31:12 +0200 Subject: ASoC: dt-bindings: add rockchip tranceiver bindings Add devicetree bindings for the spdif tranceiver found on found on rk3066, rk3188 and rk3288 SoCs Signed-off-by: Sjoerd Simons Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/rockchip-spdif.txt | 44 ++++++++++++++++++++++ 1 file changed, 44 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.txt (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt new file mode 100644 index 000000000000..33dd82c7820e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt @@ -0,0 +1,44 @@ +* Rockchip SPDIF transceiver + +The S/PDIF audio block is a stereo transceiver that allows the +processor to receive and transmit digital audio via an coaxial cable or +a fibre cable. + +Required properties: + +- compatible: should be one of the following: + - "rockchip,rk3288-spdif", "rockchip,rk3188-spdif" or + "rockchip,rk3066-spdif" +- reg: physical base address of the controller and length of memory mapped + region. +- interrupts: should contain the SPDIF interrupt. +- #address-cells: should be 1. +- #size-cells: should be 0. +- dmas: DMA specifiers for tx dma. See the DMA client binding, + Documentation/devicetree/bindings/dma/dma.txt +- dma-names: should be "tx" +- clocks: a list of phandle + clock-specifier pairs, one for each entry + in clock-names. +- clock-names: should contain following: + - "hclk": clock for SPDIF controller + - "mclk" : clock for SPDIF bus + +Required properties on RK3288: + - rockchip,grf: the phandle of the syscon node for the general register + file (GRF) + +Example for the rk3188 SPDIF controller: + +spdif: spdif@0x1011e000 { + compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif"; + reg = <0x1011e000 0x2000>; + interrupts = ; + #address-cells = <1>; + #size-cells = <0>; + dmas = <&dmac1_s 8>; + dma-names = "tx"; + clock-names = "hclk", "mclk"; + clocks = <&cru HCLK_SPDIF>, <&cru SCLK_SPDIF>; + status = "disabled"; + #sound-dai-cells = <0>; +}; -- cgit v1.2.3 From 083117c0ca8359ac82854a809120694be6375eb4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 17 Oct 2015 18:32:27 +0200 Subject: ALSA: hda - Remove obsoleted documentation It's totally outdated. We need a revised version later, maybe better integrated into kernel doc. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/hda_codec.txt | 322 --------------------------------- 1 file changed, 322 deletions(-) delete mode 100644 Documentation/sound/alsa/hda_codec.txt (limited to 'Documentation') diff --git a/Documentation/sound/alsa/hda_codec.txt b/Documentation/sound/alsa/hda_codec.txt deleted file mode 100644 index de8efbc7e4bd..000000000000 --- a/Documentation/sound/alsa/hda_codec.txt +++ /dev/null @@ -1,322 +0,0 @@ -Notes on Universal Interface for Intel High Definition Audio Codec ------------------------------------------------------------------- - -Takashi Iwai - - -[Still a draft version] - - -General -======= - -The snd-hda-codec module supports the generic access function for the -High Definition (HD) audio codecs. It's designed to be independent -from the controller code like ac97 codec module. The real accessors -from/to the controller must be implemented in the lowlevel driver. - -The structure of this module is similar with ac97_codec module. -Each codec chip belongs to a bus class which communicates with the -controller. - - -Initialization of Bus Instance -============================== - -The card driver has to create struct hda_bus at first. The template -struct should be filled and passed to the constructor: - -struct hda_bus_template { - void *private_data; - struct pci_dev *pci; - const char *modelname; - struct hda_bus_ops ops; -}; - -The card driver can set and use the private_data field to retrieve its -own data in callback functions. The pci field is used when the patch -needs to check the PCI subsystem IDs, so on. For non-PCI system, it -doesn't have to be set, of course. -The modelname field specifies the board's specific configuration. The -string is passed to the codec parser, and it depends on the parser how -the string is used. -These fields, private_data, pci and modelname are all optional. - -The ops field contains the callback functions as the following: - -struct hda_bus_ops { - int (*command)(struct hda_codec *codec, hda_nid_t nid, int direct, - unsigned int verb, unsigned int parm); - unsigned int (*get_response)(struct hda_codec *codec); - void (*private_free)(struct hda_bus *); -#ifdef CONFIG_SND_HDA_POWER_SAVE - void (*pm_notify)(struct hda_codec *codec); -#endif -}; - -The command callback is called when the codec module needs to send a -VERB to the controller. It's always a single command. -The get_response callback is called when the codec requires the answer -for the last command. These two callbacks are mandatory and have to -be given. -The third, private_free callback, is optional. It's called in the -destructor to release any necessary data in the lowlevel driver. - -The pm_notify callback is available only with -CONFIG_SND_HDA_POWER_SAVE kconfig. It's called when the codec needs -to power up or may power down. The controller should check the all -belonging codecs on the bus whether they are actually powered off -(check codec->power_on), and optionally the driver may power down the -controller side, too. - -The bus instance is created via snd_hda_bus_new(). You need to pass -the card instance, the template, and the pointer to store the -resultant bus instance. - -int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp, - struct hda_bus **busp); - -It returns zero if successful. A negative return value means any -error during creation. - - -Creation of Codec Instance -========================== - -Each codec chip on the board is then created on the BUS instance. -To create a codec instance, call snd_hda_codec_new(). - -int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, - struct hda_codec **codecp); - -The first argument is the BUS instance, the second argument is the -address of the codec, and the last one is the pointer to store the -resultant codec instance (can be NULL if not needed). - -The codec is stored in a linked list of bus instance. You can follow -the codec list like: - - struct hda_codec *codec; - list_for_each_entry(codec, &bus->codec_list, list) { - ... - } - -The codec isn't initialized at this stage properly. The -initialization sequence is called when the controls are built later. - - -Codec Access -============ - -To access codec, use snd_hda_codec_read() and snd_hda_codec_write(). -snd_hda_param_read() is for reading parameters. -For writing a sequence of verbs, use snd_hda_sequence_write(). - -There are variants of cached read/write, snd_hda_codec_write_cache(), -snd_hda_sequence_write_cache(). These are used for recording the -register states for the power-management resume. When no PM is needed, -these are equivalent with non-cached version. - -To retrieve the number of sub nodes connected to the given node, use -snd_hda_get_sub_nodes(). The connection list can be obtained via -snd_hda_get_connections() call. - -When an unsolicited event happens, pass the event via -snd_hda_queue_unsol_event() so that the codec routines will process it -later. - - -(Mixer) Controls -================ - -To create mixer controls of all codecs, call -snd_hda_build_controls(). It then builds the mixers and does -initialization stuff on each codec. - - -PCM Stuff -========= - -snd_hda_build_pcms() gives the necessary information to create PCM -streams. When it's called, each codec belonging to the bus stores -codec->num_pcms and codec->pcm_info fields. The num_pcms indicates -the number of elements in pcm_info array. The card driver is supposed -to traverse the codec linked list, read the pcm information in -pcm_info array, and build pcm instances according to them. - -The pcm_info array contains the following record: - -/* PCM information for each substream */ -struct hda_pcm_stream { - unsigned int substreams; /* number of substreams, 0 = not exist */ - unsigned int channels_min; /* min. number of channels */ - unsigned int channels_max; /* max. number of channels */ - hda_nid_t nid; /* default NID to query rates/formats/bps, or set up */ - u32 rates; /* supported rates */ - u64 formats; /* supported formats (SNDRV_PCM_FMTBIT_) */ - unsigned int maxbps; /* supported max. bit per sample */ - struct hda_pcm_ops ops; -}; - -/* for PCM creation */ -struct hda_pcm { - char *name; - struct hda_pcm_stream stream[2]; -}; - -The name can be passed to snd_pcm_new(). The stream field contains -the information for playback (SNDRV_PCM_STREAM_PLAYBACK = 0) and -capture (SNDRV_PCM_STREAM_CAPTURE = 1) directions. The card driver -should pass substreams to snd_pcm_new() for the number of substreams -to create. - -The channels_min, channels_max, rates and formats should be copied to -runtime->hw record. They and maxbps fields are used also to compute -the format value for the HDA codec and controller. Call -snd_hda_calc_stream_format() to get the format value. - -The ops field contains the following callback functions: - -struct hda_pcm_ops { - int (*open)(struct hda_pcm_stream *info, struct hda_codec *codec, - struct snd_pcm_substream *substream); - int (*close)(struct hda_pcm_stream *info, struct hda_codec *codec, - struct snd_pcm_substream *substream); - int (*prepare)(struct hda_pcm_stream *info, struct hda_codec *codec, - unsigned int stream_tag, unsigned int format, - struct snd_pcm_substream *substream); - int (*cleanup)(struct hda_pcm_stream *info, struct hda_codec *codec, - struct snd_pcm_substream *substream); -}; - -All are non-NULL, so you can call them safely without NULL check. - -The open callback should be called in PCM open after runtime->hw is -set up. It may override some setting and constraints additionally. -Similarly, the close callback should be called in the PCM close. - -The prepare callback should be called in PCM prepare. This will set -up the codec chip properly for the operation. The cleanup should be -called in hw_free to clean up the configuration. - -The caller should check the return value, at least for open and -prepare callbacks. When a negative value is returned, some error -occurred. - - -Proc Files -========== - -Each codec dumps the widget node information in -/proc/asound/card*/codec#* file. This information would be really -helpful for debugging. Please provide its contents together with the -bug report. - - -Power Management -================ - -It's simple: -Call snd_hda_suspend() in the PM suspend callback. -Call snd_hda_resume() in the PM resume callback. - - -Codec Preset (Patch) -==================== - -To set up and handle the codec functionality fully, each codec may -have a codec preset (patch). It's defined in struct hda_codec_preset: - - struct hda_codec_preset { - unsigned int id; - unsigned int mask; - unsigned int subs; - unsigned int subs_mask; - unsigned int rev; - const char *name; - int (*patch)(struct hda_codec *codec); - }; - -When the codec id and codec subsystem id match with the given id and -subs fields bitwise (with bitmask mask and subs_mask), the callback -patch is called. The patch callback should initialize the codec and -set the codec->patch_ops field. This is defined as below: - - struct hda_codec_ops { - int (*build_controls)(struct hda_codec *codec); - int (*build_pcms)(struct hda_codec *codec); - int (*init)(struct hda_codec *codec); - void (*free)(struct hda_codec *codec); - void (*unsol_event)(struct hda_codec *codec, unsigned int res); - #ifdef CONFIG_PM - int (*suspend)(struct hda_codec *codec, pm_message_t state); - int (*resume)(struct hda_codec *codec); - #endif - #ifdef CONFIG_SND_HDA_POWER_SAVE - int (*check_power_status)(struct hda_codec *codec, - hda_nid_t nid); - #endif - }; - -The build_controls callback is called from snd_hda_build_controls(). -Similarly, the build_pcms callback is called from -snd_hda_build_pcms(). The init callback is called after -build_controls to initialize the hardware. -The free callback is called as a destructor. - -The unsol_event callback is called when an unsolicited event is -received. - -The suspend and resume callbacks are for power management. -They can be NULL if no special sequence is required. When the resume -callback is NULL, the driver calls the init callback and resumes the -registers from the cache. If other handling is needed, you'd need to -write your own resume callback. There, the amp values can be resumed -via - void snd_hda_codec_resume_amp(struct hda_codec *codec); -and the other codec registers via - void snd_hda_codec_resume_cache(struct hda_codec *codec); - -The check_power_status callback is called when the amp value of the -given widget NID is changed. The codec code can turn on/off the power -appropriately from this information. - -Each entry can be NULL if not necessary to be called. - - -Generic Parser -============== - -When the device doesn't match with any given presets, the widgets are -parsed via th generic parser (hda_generic.c). Its support is -limited: no multi-channel support, for example. - - -Digital I/O -=========== - -Call snd_hda_create_spdif_out_ctls() from the patch to create controls -related with SPDIF out. - - -Helper Functions -================ - -snd_hda_get_codec_name() stores the codec name on the given string. - -snd_hda_check_board_config() can be used to obtain the configuration -information matching with the device. Define the model string table -and the table with struct snd_pci_quirk entries (zero-terminated), -and pass it to the function. The function checks the modelname given -as a module parameter, and PCI subsystem IDs. If the matching entry -is found, it returns the config field value. - -snd_hda_add_new_ctls() can be used to create and add control entries. -Pass the zero-terminated array of struct snd_kcontrol_new - -Macros HDA_CODEC_VOLUME(), HDA_CODEC_MUTE() and their variables can be -used for the entry of struct snd_kcontrol_new. - -The input MUX helper callbacks for such a control are provided, too: -snd_hda_input_mux_info() and snd_hda_input_mux_put(). See -patch_realtek.c for example. -- cgit v1.2.3 From 13531520e3106e73474225b68b889e9dc7da329e Mon Sep 17 00:00:00 2001 From: Sjoerd Simons Date: Mon, 19 Oct 2015 10:15:39 +0200 Subject: ASoC: rockchip: Drop unneeded properties rockchip i2s/spdif bindings Neither the rockchip i2s nor the rockchip spdif binding support child devices so #address-cells and #size-cells properties aren't required. Remove these from the bindings. Signed-off-by: Sjoerd Simons Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/rockchip-i2s.txt | 4 ---- Documentation/devicetree/bindings/sound/rockchip-spdif.txt | 4 ---- 2 files changed, 8 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt index 9b82c20b306b..085e0bc1f5d5 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt @@ -12,8 +12,6 @@ Required properties: - 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". @@ -28,8 +26,6 @@ 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 = "tx", "rx"; clock-names = "i2s_hclk", "i2s_clk"; diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt index 33dd82c7820e..e64dbdea7db9 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt @@ -12,8 +12,6 @@ Required properties: - reg: physical base address of the controller and length of memory mapped region. - interrupts: should contain the SPDIF interrupt. -- #address-cells: should be 1. -- #size-cells: should be 0. - dmas: DMA specifiers for tx dma. See the DMA client binding, Documentation/devicetree/bindings/dma/dma.txt - dma-names: should be "tx" @@ -33,8 +31,6 @@ spdif: spdif@0x1011e000 { compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif"; reg = <0x1011e000 0x2000>; interrupts = ; - #address-cells = <1>; - #size-cells = <0>; dmas = <&dmac1_s 8>; dma-names = "tx"; clock-names = "hclk", "mclk"; -- cgit v1.2.3 From 841fdde143a84cb71e168b4131e58e613d978e2a Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Wed, 21 Oct 2015 09:46:05 +0800 Subject: ASoC: rt5640: Revise the input pin name of IN1 and IN2 in document of the devicetree Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/rt5640.txt | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt index bac4d9ac1edc..5d062a567996 100644 --- a/Documentation/devicetree/bindings/sound/rt5640.txt +++ b/Documentation/devicetree/bindings/sound/rt5640.txt @@ -24,9 +24,9 @@ Pins on the device (for linking into audio routes) for RT5639/RT5640: * DMIC2 * MICBIAS1 * IN1P - * IN1R + * IN1N * IN2P - * IN2R + * IN2N * HPOL * HPOR * LOUTL -- cgit v1.2.3 From 16566e47098211e30b3d8a0bc6a3576871ada8e8 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Wed, 21 Oct 2015 09:46:05 +0800 Subject: ASoC: rt5640: Fill up the IN3's support Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/rt5640.txt | 5 ++++- include/sound/rt5640.h | 3 ++- sound/soc/codecs/rt5640.c | 22 +++++++++++++++++++++- 3 files changed, 27 insertions(+), 3 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt index 5d062a567996..9e62f6eb348f 100644 --- a/Documentation/devicetree/bindings/sound/rt5640.txt +++ b/Documentation/devicetree/bindings/sound/rt5640.txt @@ -14,7 +14,8 @@ Optional properties: - realtek,in1-differential - realtek,in2-differential - Boolean. Indicate MIC1/2 input are differential, rather than single-ended. +- realtek,in3-differential + Boolean. Indicate MIC1/2/3 input are differential, rather than single-ended. - realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. @@ -27,6 +28,8 @@ Pins on the device (for linking into audio routes) for RT5639/RT5640: * IN1N * IN2P * IN2N + * IN3P + * IN3N * HPOL * HPOR * LOUTL diff --git a/include/sound/rt5640.h b/include/sound/rt5640.h index 59d26dd81e45..e3c84b92ff70 100644 --- a/include/sound/rt5640.h +++ b/include/sound/rt5640.h @@ -12,9 +12,10 @@ #define __LINUX_SND_RT5640_H struct rt5640_platform_data { - /* IN1 & IN2 can optionally be differential */ + /* IN1 & IN2 & IN3 can optionally be differential */ bool in1_diff; bool in2_diff; + bool in3_diff; bool dmic_en; bool dmic1_data_pin; /* 0 = IN1P; 1 = GPIO3 */ diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index e1ceeb885f7d..f2beb1aa5763 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -405,11 +405,14 @@ static const struct snd_kcontrol_new rt5640_snd_controls[] = { SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5640_DAC1_DIG_VOL, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 175, 0, dac_vol_tlv), - /* IN1/IN2 Control */ + /* IN1/IN2/IN3 Control */ SOC_SINGLE_TLV("IN1 Boost", RT5640_IN1_IN2, RT5640_BST_SFT1, 8, 0, bst_tlv), SOC_SINGLE_TLV("IN2 Boost", RT5640_IN3_IN4, RT5640_BST_SFT2, 8, 0, bst_tlv), + SOC_SINGLE_TLV("IN3 Boost", RT5640_IN1_IN2, + RT5640_BST_SFT2, 8, 0, bst_tlv), + /* INL/INR Volume Control */ SOC_DOUBLE_TLV("IN Capture Volume", RT5640_INL_INR_VOL, RT5640_INL_VOL_SFT, RT5640_INR_VOL_SFT, @@ -598,6 +601,8 @@ static const struct snd_kcontrol_new rt5640_rec_l_mix[] = { RT5640_M_HP_L_RM_L_SFT, 1, 1), SOC_DAPM_SINGLE("INL Switch", RT5640_REC_L2_MIXER, RT5640_M_IN_L_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST3 Switch", RT5640_REC_L2_MIXER, + RT5640_M_BST2_RM_L_SFT, 1, 1), SOC_DAPM_SINGLE("BST2 Switch", RT5640_REC_L2_MIXER, RT5640_M_BST4_RM_L_SFT, 1, 1), SOC_DAPM_SINGLE("BST1 Switch", RT5640_REC_L2_MIXER, @@ -611,6 +616,8 @@ static const struct snd_kcontrol_new rt5640_rec_r_mix[] = { RT5640_M_HP_R_RM_R_SFT, 1, 1), SOC_DAPM_SINGLE("INR Switch", RT5640_REC_R2_MIXER, RT5640_M_IN_R_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST3 Switch", RT5640_REC_R2_MIXER, + RT5640_M_BST2_RM_R_SFT, 1, 1), SOC_DAPM_SINGLE("BST2 Switch", RT5640_REC_R2_MIXER, RT5640_M_BST4_RM_R_SFT, 1, 1), SOC_DAPM_SINGLE("BST1 Switch", RT5640_REC_R2_MIXER, @@ -1065,6 +1072,8 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN1N"), SND_SOC_DAPM_INPUT("IN2P"), SND_SOC_DAPM_INPUT("IN2N"), + SND_SOC_DAPM_INPUT("IN3P"), + SND_SOC_DAPM_INPUT("IN3N"), SND_SOC_DAPM_PGA("DMIC L1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("DMIC R1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("DMIC L2", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -1081,6 +1090,8 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { RT5640_PWR_BST1_BIT, 0, NULL, 0), SND_SOC_DAPM_PGA("BST2", RT5640_PWR_ANLG2, RT5640_PWR_BST4_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("BST3", RT5640_PWR_ANLG2, + RT5640_PWR_BST2_BIT, 0, NULL, 0), /* Input Volume */ SND_SOC_DAPM_PGA("INL VOL", RT5640_PWR_VOL, RT5640_PWR_IN_L_BIT, 0, NULL, 0), @@ -1310,6 +1321,7 @@ static const struct snd_soc_dapm_widget rt5639_specific_dapm_widgets[] = { static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"IN1P", NULL, "LDO2"}, {"IN2P", NULL, "LDO2"}, + {"IN3P", NULL, "LDO2"}, {"DMIC L1", NULL, "DMIC1"}, {"DMIC R1", NULL, "DMIC1"}, @@ -1320,18 +1332,22 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"BST1", NULL, "IN1N"}, {"BST2", NULL, "IN2P"}, {"BST2", NULL, "IN2N"}, + {"BST3", NULL, "IN3P"}, + {"BST3", NULL, "IN3N"}, {"INL VOL", NULL, "IN2P"}, {"INR VOL", NULL, "IN2N"}, {"RECMIXL", "HPOL Switch", "HPOL"}, {"RECMIXL", "INL Switch", "INL VOL"}, + {"RECMIXL", "BST3 Switch", "BST3"}, {"RECMIXL", "BST2 Switch", "BST2"}, {"RECMIXL", "BST1 Switch", "BST1"}, {"RECMIXL", "OUT MIXL Switch", "OUT MIXL"}, {"RECMIXR", "HPOR Switch", "HPOR"}, {"RECMIXR", "INR Switch", "INR VOL"}, + {"RECMIXR", "BST3 Switch", "BST3"}, {"RECMIXR", "BST2 Switch", "BST2"}, {"RECMIXR", "BST1 Switch", "BST1"}, {"RECMIXR", "OUT MIXR Switch", "OUT MIXR"}, @@ -2260,6 +2276,10 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5640->regmap, RT5640_IN3_IN4, RT5640_IN_DF2, RT5640_IN_DF2); + if (rt5640->pdata.in3_diff) + regmap_update_bits(rt5640->regmap, RT5640_IN1_IN2, + RT5640_IN_DF2, RT5640_IN_DF2); + rt5640->hp_mute = 1; return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, -- cgit v1.2.3 From 53e597b1d194910bef53ed0632da329fef497904 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 22 Oct 2015 13:11:56 +0200 Subject: ALSA: Remove transfer_ack_{begin,end} callbacks from struct snd_pcm_runtime While there is nothing wrong with the transfer_ack_begin and transfer_ack_end callbacks per-se, the last documented user was part of the alsa-driver 0.5.12a package, which was released 14 years ago and even predates the upstream integration of the ALSA core and has subsequently been superseded by newer alsa-driver releases. This seems to indicate that there is no need for having these callbacks and they are just cruft that can be removed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Takashi Iwai --- Documentation/DocBook/writing-an-alsa-driver.tmpl | 19 ++----------------- include/sound/pcm.h | 4 ---- sound/core/pcm_lib.c | 5 ----- 3 files changed, 2 insertions(+), 26 deletions(-) (limited to 'Documentation') diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl index 84ef6a90131c..a27ab9f53fb6 100644 --- a/Documentation/DocBook/writing-an-alsa-driver.tmpl +++ b/Documentation/DocBook/writing-an-alsa-driver.tmpl @@ -2181,10 +2181,6 @@ struct _snd_pcm_runtime { struct snd_pcm_hardware hw; struct snd_pcm_hw_constraints hw_constraints; - /* -- interrupt callbacks -- */ - void (*transfer_ack_begin)(struct snd_pcm_substream *substream); - void (*transfer_ack_end)(struct snd_pcm_substream *substream); - /* -- timer -- */ unsigned int timer_resolution; /* timer resolution */ @@ -2209,9 +2205,8 @@ struct _snd_pcm_runtime { For the operators (callbacks) of each sound driver, most of these records are supposed to be read-only. Only the PCM middle-layer changes / updates them. The exceptions are - the hardware description (hw), interrupt callbacks - (transfer_ack_xxx), DMA buffer information, and the private - data. Besides, if you use the standard buffer allocation + the hardware description (hw) DMA buffer information and the + private data. Besides, if you use the standard buffer allocation method via snd_pcm_lib_malloc_pages(), you don't need to set the DMA buffer information by yourself. @@ -2538,16 +2533,6 @@ struct _snd_pcm_runtime { -
- Interrupt Callbacks - - The field transfer_ack_begin and - transfer_ack_end are called at - the beginning and at the end of - snd_pcm_period_elapsed(), respectively. - -
-
diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 2882dddfc91c..3e0ffd21901f 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -402,10 +402,6 @@ struct snd_pcm_runtime { struct snd_pcm_hardware hw; struct snd_pcm_hw_constraints hw_constraints; - /* -- interrupt callbacks -- */ - void (*transfer_ack_begin)(struct snd_pcm_substream *substream); - void (*transfer_ack_end)(struct snd_pcm_substream *substream); - /* -- timer -- */ unsigned int timer_resolution; /* timer resolution */ int tstamp_type; /* timestamp type */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 6dc4277937b8..05a3ca93c647 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1875,9 +1875,6 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) return; runtime = substream->runtime; - if (runtime->transfer_ack_begin) - runtime->transfer_ack_begin(substream); - snd_pcm_stream_lock_irqsave(substream, flags); if (!snd_pcm_running(substream) || snd_pcm_update_hw_ptr0(substream, 1) < 0) @@ -1889,8 +1886,6 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) #endif _end: snd_pcm_stream_unlock_irqrestore(substream, flags); - if (runtime->transfer_ack_end) - runtime->transfer_ack_end(substream); kill_fasync(&runtime->fasync, SIGIO, POLL_IN); } -- cgit v1.2.3 From 50760cad9de969fe85b24465afe6396b8bbc6a3f Mon Sep 17 00:00:00 2001 From: "Maciej S. Szmigiero" Date: Sat, 19 Sep 2015 02:00:25 +0200 Subject: ASoC: fsl-asoc-card: add AC'97 support Add AC'97 support to fsl-asoc-card using generic ASoC AC'97 CODEC. The SSI controller will silently enable any TX AC'97 slots that have their bits set in SLOTREQ received from CODEC and then will redirect some of playback samples there. That's why it is important to make sure that any of CODEC playback slots that can pull samples are set to slots 3/4 (standard PCM playback slots). Currently, this applies to S/PDIF slots as they were seen to pull samples sometimes even with S/PDIF output being disabled. Signed-off-by: Maciej Szmigiero Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/fsl-asoc-card.txt | 10 +- sound/soc/fsl/fsl-asoc-card.c | 140 ++++++++++++++++----- 2 files changed, 116 insertions(+), 34 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index a96774c194c8..ce55c0a6f757 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -13,13 +13,15 @@ So having this generic sound card allows all Freescale SoC users to benefit from the simplification of a new card support and the capability of the wide sample rates support through ASRC. -Note: The card is initially designed for those sound cards who use I2S and - PCM DAI formats. However, it'll be also possible to support those non - I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as long - as the driver has been properly upgraded. +Note: The card is initially designed for those sound cards who use AC'97, I2S + and PCM DAI formats. However, it'll be also possible to support those non + AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as + long as the driver has been properly upgraded. The compatible list for this generic sound card currently: + "fsl,imx-audio-ac97" + "fsl,imx-audio-cs42888" "fsl,imx-audio-wm8962" diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 5aeb6ed4827e..33628a09fcf5 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -14,6 +14,9 @@ #include #include #include +#if IS_ENABLED(CONFIG_SND_AC97_CODEC) +#include +#endif #include #include @@ -115,6 +118,11 @@ static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), }; +static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv) +{ + return priv->dai_fmt == SND_SOC_DAIFMT_AC97; +} + static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -133,7 +141,9 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, * set_bias_level(), bypass the remaining settings in hw_params(). * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS. */ - if (priv->card.set_bias_level && priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) + if ((priv->card.set_bias_level && + priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) || + fsl_asoc_card_is_ac97(priv)) return 0; /* Specific configurations of DAIs starts from here */ @@ -300,7 +310,7 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, ext_port--; /* - * Use asynchronous mode (6 wires) for all cases. + * Use asynchronous mode (6 wires) for all cases except AC97. * If only 4 wires are needed, just set SSI into * synchronous mode and enable 4 PADs in IOMUX. */ @@ -346,15 +356,30 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, IMX_AUDMUX_V2_PTCR_TCLKDIR; break; default: - return -EINVAL; + if (!fsl_asoc_card_is_ac97(priv)) + return -EINVAL; + } + + if (fsl_asoc_card_is_ac97(priv)) { + int_ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR; } /* Asynchronous mode can not be set along with RCLKDIR */ - ret = imx_audmux_v2_configure_port(int_port, 0, - IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); - if (ret) { - dev_err(dev, "audmux internal port setup failed\n"); - return ret; + if (!fsl_asoc_card_is_ac97(priv)) { + unsigned int pdcr = + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port); + + ret = imx_audmux_v2_configure_port(int_port, 0, + pdcr); + if (ret) { + dev_err(dev, "audmux internal port setup failed\n"); + return ret; + } } ret = imx_audmux_v2_configure_port(int_port, int_ptcr, @@ -364,11 +389,16 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, return ret; } - ret = imx_audmux_v2_configure_port(ext_port, 0, - IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); - if (ret) { - dev_err(dev, "audmux external port setup failed\n"); - return ret; + if (!fsl_asoc_card_is_ac97(priv)) { + unsigned int pdcr = + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port); + + ret = imx_audmux_v2_configure_port(ext_port, 0, + pdcr); + if (ret) { + dev_err(dev, "audmux external port setup failed\n"); + return ret; + } } ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr, @@ -389,6 +419,23 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) struct device *dev = card->dev; int ret; + if (fsl_asoc_card_is_ac97(priv)) { +#if IS_ENABLED(CONFIG_SND_AC97_CODEC) + struct snd_soc_codec *codec = card->rtd[0].codec; + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + + /* + * Use slots 3/4 for S/PDIF so SSI won't try to enable + * other slots and send some samples there + * due to SLOTREQ bits for S/PDIF received from codec + */ + snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS, + AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4); +#endif + + return 0; + } + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, codec_priv->mclk_freq, SND_SOC_CLOCK_IN); if (ret) { @@ -407,7 +454,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) struct platform_device *cpu_pdev; struct fsl_asoc_card_priv *priv; struct i2c_client *codec_dev; - struct clk *codec_clk; const char *codec_dai_name; u32 width; int ret; @@ -420,9 +466,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Give a chance to old DT binding */ if (!cpu_np) cpu_np = of_parse_phandle(np, "ssi-controller", 0); - codec_np = of_parse_phandle(np, "audio-codec", 0); - if (!cpu_np || !codec_np) { - dev_err(&pdev->dev, "phandle missing or invalid\n"); + if (!cpu_np) { + dev_err(&pdev->dev, "CPU phandle missing or invalid\n"); ret = -EINVAL; goto fail; } @@ -434,22 +479,24 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) goto fail; } - codec_dev = of_find_i2c_device_by_node(codec_np); - if (!codec_dev) { - dev_err(&pdev->dev, "failed to find codec platform device\n"); - ret = -EINVAL; - goto fail; - } + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (codec_np) + codec_dev = of_find_i2c_device_by_node(codec_np); + else + codec_dev = NULL; asrc_np = of_parse_phandle(np, "audio-asrc", 0); if (asrc_np) asrc_pdev = of_find_device_by_node(asrc_np); /* Get the MCLK rate only, and leave it controlled by CODEC drivers */ - codec_clk = clk_get(&codec_dev->dev, NULL); - if (!IS_ERR(codec_clk)) { - priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); - clk_put(codec_clk); + if (codec_dev) { + struct clk *codec_clk = clk_get(&codec_dev->dev, NULL); + + if (!IS_ERR(codec_clk)) { + priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); + clk_put(codec_clk); + } } /* Default sample rate and format, will be updated in hw_params() */ @@ -486,11 +533,21 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { + codec_dai_name = "ac97-hifi"; + priv->card.set_bias_level = NULL; + priv->dai_fmt = SND_SOC_DAIFMT_AC97; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); return -EINVAL; } + if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) { + dev_err(&pdev->dev, "failed to find codec device\n"); + ret = -EINVAL; + goto asrc_fail; + } + /* Common settings for corresponding Freescale CPU DAI driver */ if (strstr(cpu_np->name, "ssi")) { /* Only SSI needs to configure AUDMUX */ @@ -507,7 +564,9 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1; } - sprintf(priv->name, "%s-audio", codec_dev->name); + snprintf(priv->name, sizeof(priv->name), "%s-audio", + fsl_asoc_card_is_ac97(priv) ? "ac97" : + codec_dev->name); /* Initialize sound card */ priv->pdev = pdev; @@ -531,8 +590,26 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; - priv->dai_link[0].codec_of_node = codec_np; priv->dai_link[0].codec_dai_name = codec_dai_name; + + if (!fsl_asoc_card_is_ac97(priv)) + priv->dai_link[0].codec_of_node = codec_np; + else { + u32 idx; + + ret = of_property_read_u32(cpu_np, "cell-index", &idx); + if (ret) { + dev_err(&pdev->dev, + "cannot get CPU index property\n"); + goto asrc_fail; + } + + priv->dai_link[0].codec_name = + devm_kasprintf(&pdev->dev, GFP_KERNEL, + "ac97-codec.%u", + (unsigned int)idx); + } + priv->dai_link[0].platform_of_node = cpu_np; priv->dai_link[0].dai_fmt = priv->dai_fmt; priv->card.num_links = 1; @@ -543,6 +620,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_link[1].platform_of_node = asrc_np; priv->dai_link[2].codec_dai_name = codec_dai_name; priv->dai_link[2].codec_of_node = codec_np; + priv->dai_link[2].codec_name = + priv->dai_link[0].codec_name; priv->dai_link[2].cpu_of_node = cpu_np; priv->dai_link[2].dai_fmt = priv->dai_fmt; priv->card.num_links = 3; @@ -578,14 +657,15 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_fail: of_node_put(asrc_np); -fail: of_node_put(codec_np); +fail: of_node_put(cpu_np); return ret; } static const struct of_device_id fsl_asoc_card_dt_ids[] = { + { .compatible = "fsl,imx-audio-ac97", }, { .compatible = "fsl,imx-audio-cs42888", }, { .compatible = "fsl,imx-audio-sgtl5000", }, { .compatible = "fsl,imx-audio-wm8962", }, -- cgit v1.2.3 From 391ac3ef50b9ec575c4d5c6f055efe5096ac1957 Mon Sep 17 00:00:00 2001 From: Songjun Wu Date: Thu, 8 Oct 2015 18:13:32 +0800 Subject: ASoC: atmel-classd: DT binding for Class D audio amplifier driver DT binding documentation for this new ASoC driver. Signed-off-by: Songjun Wu Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/atmel-classd.txt | 52 ++++++++++++++++++++++ 1 file changed, 52 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-classd.txt (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/atmel-classd.txt b/Documentation/devicetree/bindings/sound/atmel-classd.txt new file mode 100644 index 000000000000..0018451c4351 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel-classd.txt @@ -0,0 +1,52 @@ +* Atmel ClassD driver under ALSA SoC architecture + +Required properties: +- compatible + Should be "atmel,sama5d2-classd". +- reg + Should contain ClassD registers location and length. +- interrupts + Should contain the IRQ line for the ClassD. +- dmas + One DMA specifiers as described in atmel-dma.txt and dma.txt files. +- dma-names + Must be "tx". +- clock-names + Tuple listing input clock names. + Required elements: "pclk", "gclk" and "aclk". +- clocks + Please refer to clock-bindings.txt. + +Optional properties: +- pinctrl-names, pinctrl-0 + Please refer to pinctrl-bindings.txt. +- atmel,model + The user-visible name of this sound complex. + The default value is "CLASSD". +- atmel,pwm-type + PWM modulation type, "single" or "diff". + The default value is "single". +- atmel,non-overlap-time + Set non-overlapping time, the unit is nanosecond(ns). + There are four values, + <5>, <10>, <15>, <20>, the default value is <10>. + Non-overlapping will be disabled if not specified. + +Example: +classd: classd@fc048000 { + compatible = "atmel,sama5d2-classd"; + reg = <0xfc048000 0x100>; + interrupts = <59 IRQ_TYPE_LEVEL_HIGH 7>; + dmas = <&dma0 + (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) + | AT91_XDMAC_DT_PERID(47))>; + dma-names = "tx"; + clocks = <&classd_clk>, <&classd_gclk>, <&audio_pll_pmc>; + clock-names = "pclk", "gclk", "aclk"; + + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_classd_default>; + atmel,model = "classd @ SAMA5D2-Xplained"; + atmel,pwm-type = "diff"; + atmel,non-overlap-time = <10>; +}; -- cgit v1.2.3 From d307e01e41e830adac15e91d8cea38d8a53060a5 Mon Sep 17 00:00:00 2001 From: Sugar Zhang Date: Thu, 8 Oct 2015 20:40:08 +0800 Subject: ASoC: rockchip: add capture property rockchip,capture-channels: max capture channels, 2 channels default. Signed-off-by: Sugar Zhang Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/rockchip-i2s.txt | 2 ++ 1 file changed, 2 insertions(+) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt index 085e0bc1f5d5..2267d249ca0e 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt @@ -19,6 +19,7 @@ Required properties: - clock-names: should contain followings: - "i2s_hclk": clock for I2S BUS - "i2s_clk" : clock for I2S controller +- rockchip,capture-channels: max capture channels, if not set, 2 channels default. Example for rk3288 I2S controller: @@ -30,4 +31,5 @@ i2s@ff890000 { dma-names = "tx", "rx"; clock-names = "i2s_hclk", "i2s_clk"; clocks = <&cru HCLK_I2S0>, <&cru SCLK_I2S0>; + rockchip,capture-channels = <2>; }; -- cgit v1.2.3 From a5804dc7cf986cc99689ef54e577f9efb4f1c455 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 28 Oct 2015 12:33:48 +0100 Subject: ALSA: DocBook: Add soc-ops.c and soc-compress.c These have been missing in the template file although they provide good kernel doc comments. Let's add them. Signed-off-by: Takashi Iwai --- Documentation/DocBook/alsa-driver-api.tmpl | 2 ++ 1 file changed, 2 insertions(+) (limited to 'Documentation') diff --git a/Documentation/DocBook/alsa-driver-api.tmpl b/Documentation/DocBook/alsa-driver-api.tmpl index e94a10bb4a9e..53f439dcc94b 100644 --- a/Documentation/DocBook/alsa-driver-api.tmpl +++ b/Documentation/DocBook/alsa-driver-api.tmpl @@ -112,6 +112,8 @@ !Esound/soc/soc-devres.c !Esound/soc/soc-io.c !Esound/soc/soc-pcm.c +!Esound/soc/soc-ops.c +!Esound/soc/soc-compress.c ASoC DAPM API !Esound/soc/soc-dapm.c -- cgit v1.2.3