diff --git a/Documentation/devicetree/bindings/sound/sun4i-codec.txt b/Documentation/devicetree/bindings/sound/sun4i-codec.txt index 0dce690f78f5..3033bd8aab0f 100644 --- a/Documentation/devicetree/bindings/sound/sun4i-codec.txt +++ b/Documentation/devicetree/bindings/sound/sun4i-codec.txt @@ -1,8 +1,12 @@ * Allwinner A10 Codec Required properties: -- compatible: must be either "allwinner,sun4i-a10-codec" or - "allwinner,sun7i-a20-codec" +- compatible: must be one of the following compatibles: + - "allwinner,sun4i-a10-codec" + - "allwinner,sun6i-a31-codec" + - "allwinner,sun7i-a20-codec" + - "allwinner,sun8i-a23-codec" + - "allwinner,sun8i-h3-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, @@ -17,6 +21,43 @@ Required properties: Optional properties: - allwinner,pa-gpios: gpio to enable external amplifier +Required properties for the following compatibles: + - "allwinner,sun6i-a31-codec" + - "allwinner,sun8i-a23-codec" + - "allwinner,sun8i-h3-codec" +- resets: phandle to the reset control for this device +- allwinner,audio-routing: A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. Valid names include: + + Audio pins on the SoC: + "HP" + "HPCOM" + "LINEIN" + "LINEOUT" (not on sun8i-a23) + "MIC1" + "MIC2" + "MIC3" (sun6i-a31 only) + + Microphone biases from the SoC: + "HBIAS" + "MBIAS" + + Board connectors: + "Headphone" + "Headset Mic" + "Line In" + "Line Out" + "Mic" + "Speaker" + +Required properties for the following compatibles: + - "allwinner,sun8i-a23-codec" + - "allwinner,sun8i-h3-codec" +- allwinner,codec-analog-controls: A phandle to the codec analog controls + block in the PRCM. + Example: codec: codec@01c22c00 { #sound-dai-cells = <0>; @@ -28,3 +69,23 @@ codec: codec@01c22c00 { dmas = <&dma 0 19>, <&dma 0 19>; dma-names = "rx", "tx"; }; + +codec: codec@01c22c00 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun6i-a31-codec"; + reg = <0x01c22c00 0x98>; + interrupts = ; + clocks = <&ccu CLK_APB1_CODEC>, <&ccu CLK_CODEC>; + clock-names = "apb", "codec"; + resets = <&ccu RST_APB1_CODEC>; + dmas = <&dma 15>, <&dma 15>; + dma-names = "rx", "tx"; + allwinner,audio-routing = + "Headphone", "HP", + "Speaker", "LINEOUT", + "LINEIN", "Line In", + "MIC1", "MBIAS", + "MIC1", "Mic", + "MIC2", "HBIAS", + "MIC2", "Headset Mic"; +}; diff --git a/Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt b/Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt new file mode 100644 index 000000000000..779b735781ba --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt @@ -0,0 +1,16 @@ +* Allwinner Codec Analog Controls + +Required properties: +- compatible: must be one of the following compatibles: + - "allwinner,sun8i-a23-codec-analog" + - "allwinner,sun8i-h3-codec-analog" + +Required properties if not a sub-node of the PRCM node: +- reg: must contain the registers location and length + +Example: +prcm: prcm@01f01400 { + codec_analog: codec-analog { + compatible = "allwinner,sun8i-a23-codec-analog"; + }; +}; diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 27f30d352867..9de7fe8af255 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -26,31 +27,56 @@ #include #include -#include "stac9766.h" - #define STAC9766_VENDOR_ID 0x83847666 #define STAC9766_VENDOR_ID_MASK 0xffffffff -/* - * STAC9766 register cache - */ -static const u16 stac9766_reg[] = { - 0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */ - 0x0000, 0x0000, 0x8008, 0x8008, /* e */ - 0x8808, 0x8808, 0x8808, 0x8808, /* 16 */ - 0x8808, 0x0000, 0x8000, 0x0000, /* 1e */ - 0x0000, 0x0000, 0x0000, 0x000f, /* 26 */ - 0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */ - 0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */ - 0x0000, 0x2000, 0x0000, 0x0100, /* 3e */ - 0x0000, 0x0000, 0x0080, 0x0000, /* 46 */ - 0x0000, 0x0000, 0x0003, 0xffff, /* 4e */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */ - 0x4000, 0x0000, 0x0000, 0x0000, /* 5e */ - 0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */ - 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 7e */ +#define AC97_STAC_DA_CONTROL 0x6A +#define AC97_STAC_ANALOG_SPECIAL 0x6E +#define AC97_STAC_STEREO_MIC 0x78 + +static const struct reg_default stac9766_reg_defaults[] = { + { 0x02, 0x8000 }, + { 0x04, 0x8000 }, + { 0x06, 0x8000 }, + { 0x0a, 0x0000 }, + { 0x0c, 0x8008 }, + { 0x0e, 0x8008 }, + { 0x10, 0x8808 }, + { 0x12, 0x8808 }, + { 0x14, 0x8808 }, + { 0x16, 0x8808 }, + { 0x18, 0x8808 }, + { 0x1a, 0x0000 }, + { 0x1c, 0x8000 }, + { 0x20, 0x0000 }, + { 0x22, 0x0000 }, + { 0x28, 0x0a05 }, + { 0x2c, 0xbb80 }, + { 0x32, 0xbb80 }, + { 0x3a, 0x2000 }, + { 0x3e, 0x0100 }, + { 0x4c, 0x0300 }, + { 0x4e, 0xffff }, + { 0x50, 0x0000 }, + { 0x52, 0x0000 }, + { 0x54, 0x0000 }, + { 0x6a, 0x0000 }, + { 0x6e, 0x1000 }, + { 0x72, 0x0000 }, + { 0x78, 0x0000 }, +}; + +static const struct regmap_config stac9766_regmap_config = { + .reg_bits = 16, + .reg_stride = 2, + .val_bits = 16, + .max_register = 0x78, + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = regmap_ac97_default_volatile, + + .reg_defaults = stac9766_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(stac9766_reg_defaults), }; static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX", @@ -139,71 +165,22 @@ static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = { SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum), }; -static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int val) -{ - struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); - u16 *cache = codec->reg_cache; - - if (reg > AC97_STAC_PAGE0) { - stac9766_ac97_write(codec, AC97_INT_PAGING, 0); - soc_ac97_ops->write(ac97, reg, val); - stac9766_ac97_write(codec, AC97_INT_PAGING, 1); - return 0; - } - if (reg / 2 >= ARRAY_SIZE(stac9766_reg)) - return -EIO; - - soc_ac97_ops->write(ac97, reg, val); - cache[reg / 2] = val; - return 0; -} - -static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); - u16 val = 0, *cache = codec->reg_cache; - - if (reg > AC97_STAC_PAGE0) { - stac9766_ac97_write(codec, AC97_INT_PAGING, 0); - val = soc_ac97_ops->read(ac97, reg - AC97_STAC_PAGE0); - stac9766_ac97_write(codec, AC97_INT_PAGING, 1); - return val; - } - if (reg / 2 >= ARRAY_SIZE(stac9766_reg)) - return -EIO; - - if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || - reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 || - reg == AC97_VENDOR_ID2) { - - val = soc_ac97_ops->read(ac97, reg); - return val; - } - return cache[reg / 2]; -} - static int ac97_analog_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct snd_pcm_runtime *runtime = substream->runtime; - unsigned short reg, vra; + unsigned short reg; - vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); - - vra |= 0x1; /* enable variable rate audio */ - vra &= ~0x4; /* disable SPDIF output */ - - stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + /* enable variable rate audio, disable SPDIF output */ + snd_soc_update_bits(codec, AC97_EXTENDED_STATUS, 0x5, 0x1); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = AC97_PCM_FRONT_DAC_RATE; else reg = AC97_PCM_LR_ADC_RATE; - return stac9766_ac97_write(codec, reg, runtime->rate); + return snd_soc_write(codec, reg, runtime->rate); } static int ac97_digital_prepare(struct snd_pcm_substream *substream, @@ -211,18 +188,16 @@ static int ac97_digital_prepare(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct snd_pcm_runtime *runtime = substream->runtime; - unsigned short reg, vra; + unsigned short reg; - stac9766_ac97_write(codec, AC97_SPDIF, 0x2002); + snd_soc_write(codec, AC97_SPDIF, 0x2002); - vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); - vra |= 0x5; /* Enable VRA and SPDIF out */ - - stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + /* Enable VRA and SPDIF out */ + snd_soc_update_bits(codec, AC97_EXTENDED_STATUS, 0x5, 0x5); reg = AC97_PCM_FRONT_DAC_RATE; - return stac9766_ac97_write(codec, reg, runtime->rate); + return snd_soc_write(codec, reg, runtime->rate); } static int stac9766_set_bias_level(struct snd_soc_codec *codec, @@ -232,11 +207,11 @@ static int stac9766_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_ON: /* full On */ case SND_SOC_BIAS_PREPARE: /* partial On */ case SND_SOC_BIAS_STANDBY: /* Off, with power */ - stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000); + snd_soc_write(codec, AC97_POWERDOWN, 0x0000); break; case SND_SOC_BIAS_OFF: /* Off, without power */ /* disable everything including AC link */ - stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff); + snd_soc_write(codec, AC97_POWERDOWN, 0xffff); break; } return 0; @@ -300,21 +275,34 @@ static struct snd_soc_dai_driver stac9766_dai[] = { static int stac9766_codec_probe(struct snd_soc_codec *codec) { struct snd_ac97 *ac97; + struct regmap *regmap; + int ret; ac97 = snd_soc_new_ac97_codec(codec, STAC9766_VENDOR_ID, STAC9766_VENDOR_ID_MASK); if (IS_ERR(ac97)) return PTR_ERR(ac97); + regmap = regmap_init_ac97(ac97, &stac9766_regmap_config); + if (IS_ERR(regmap)) { + ret = PTR_ERR(regmap); + goto err_free_ac97; + } + + snd_soc_codec_init_regmap(codec, regmap); snd_soc_codec_set_drvdata(codec, ac97); return 0; +err_free_ac97: + snd_soc_free_ac97_codec(ac97); + return ret; } static int stac9766_codec_remove(struct snd_soc_codec *codec) { struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + snd_soc_codec_exit_regmap(codec); snd_soc_free_ac97_codec(ac97); return 0; } @@ -324,17 +312,11 @@ static struct snd_soc_codec_driver soc_codec_dev_stac9766 = { .controls = stac9766_snd_ac97_controls, .num_controls = ARRAY_SIZE(stac9766_snd_ac97_controls), }, - .write = stac9766_ac97_write, - .read = stac9766_ac97_read, .set_bias_level = stac9766_set_bias_level, .suspend_bias_off = true, .probe = stac9766_codec_probe, .remove = stac9766_codec_remove, .resume = stac9766_codec_resume, - .reg_cache_size = ARRAY_SIZE(stac9766_reg), - .reg_word_size = sizeof(u16), - .reg_cache_step = 2, - .reg_cache_default = stac9766_reg, }; static int stac9766_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/stac9766.h b/sound/soc/codecs/stac9766.h deleted file mode 100644 index c726f907e2c0..000000000000 --- a/sound/soc/codecs/stac9766.h +++ /dev/null @@ -1,17 +0,0 @@ -/* - * stac9766.h -- STAC9766 Soc Audio driver - */ - -#ifndef _STAC9766_H -#define _STAC9766_H - -#define AC97_STAC_PAGE0 0x1000 -#define AC97_STAC_DA_CONTROL (AC97_STAC_PAGE0 | 0x6A) -#define AC97_STAC_ANALOG_SPECIAL (AC97_STAC_PAGE0 | 0x6E) -#define AC97_STAC_STEREO_MIC 0x78 - -/* STAC9766 DAI ID's */ -#define STAC9766_DAI_AC97_ANALOG 0 -#define STAC9766_DAI_AC97_DIGITAL 1 - -#endif diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c index d6e00c77edcd..62c618765224 100644 --- a/sound/soc/codecs/sti-sas.c +++ b/sound/soc/codecs/sti-sas.c @@ -14,28 +14,8 @@ #include #include -/* chipID supported */ -#define CHIPID_STIH416 0 -#define CHIPID_STIH407 1 - /* DAC definitions */ -/* stih416 DAC registers */ -/* sysconf 2517: Audio-DAC-Control */ -#define STIH416_AUDIO_DAC_CTRL 0x00000814 -/* sysconf 2519: Audio-Gue-Control */ -#define STIH416_AUDIO_GLUE_CTRL 0x0000081C - -#define STIH416_DAC_NOT_STANDBY 0x3 -#define STIH416_DAC_SOFTMUTE 0x4 -#define STIH416_DAC_ANA_NOT_PWR 0x5 -#define STIH416_DAC_NOT_PNDBG 0x6 - -#define STIH416_DAC_NOT_STANDBY_MASK BIT(STIH416_DAC_NOT_STANDBY) -#define STIH416_DAC_SOFTMUTE_MASK BIT(STIH416_DAC_SOFTMUTE) -#define STIH416_DAC_ANA_NOT_PWR_MASK BIT(STIH416_DAC_ANA_NOT_PWR) -#define STIH416_DAC_NOT_PNDBG_MASK BIT(STIH416_DAC_NOT_PNDBG) - /* stih407 DAC registers */ /* sysconf 5041: Audio-Gue-Control */ #define STIH407_AUDIO_GLUE_CTRL 0x000000A4 @@ -63,14 +43,9 @@ enum { STI_SAS_DAI_ANALOG_OUT, }; -static const struct reg_default stih416_sas_reg_defaults[] = { - { STIH407_AUDIO_GLUE_CTRL, 0x00000040 }, - { STIH407_AUDIO_DAC_CTRL, 0x000000000 }, -}; - static const struct reg_default stih407_sas_reg_defaults[] = { - { STIH416_AUDIO_DAC_CTRL, 0x000000000 }, - { STIH416_AUDIO_GLUE_CTRL, 0x00000040 }, + { STIH407_AUDIO_DAC_CTRL, 0x000000000 }, + { STIH407_AUDIO_GLUE_CTRL, 0x00000040 }, }; struct sti_dac_audio { @@ -89,7 +64,6 @@ struct sti_spdif_audio { /* device data structure */ struct sti_sas_dev_data { - const int chipid; /* IC version */ const struct regmap_config *regmap; const struct snd_soc_dai_ops *dac_ops; /* DAC function callbacks */ const struct snd_soc_dapm_widget *dapm_widgets; /* dapms declaration */ @@ -150,51 +124,27 @@ static int sti_sas_init_sas_registers(struct snd_soc_codec *codec, ret = snd_soc_update_bits(codec, STIH407_AUDIO_GLUE_CTRL, SPDIF_BIPHASE_IDLE_MASK, 0); if (ret < 0) { - dev_err(codec->dev, "Failed to update SPDIF registers"); + dev_err(codec->dev, "Failed to update SPDIF registers\n"); return ret; } /* Init DAC configuration */ - switch (data->dev_data->chipid) { - case CHIPID_STIH407: - /* init configuration */ - ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL, - STIH407_DAC_STANDBY_MASK, - STIH407_DAC_STANDBY_MASK); + /* init configuration */ + ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL, + STIH407_DAC_STANDBY_MASK, + STIH407_DAC_STANDBY_MASK); - if (!ret) - ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL, - STIH407_DAC_STANDBY_ANA_MASK, - STIH407_DAC_STANDBY_ANA_MASK); - if (!ret) - ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL, - STIH407_DAC_SOFTMUTE_MASK, - STIH407_DAC_SOFTMUTE_MASK); - break; - case CHIPID_STIH416: - ret = snd_soc_update_bits(codec, STIH416_AUDIO_DAC_CTRL, - STIH416_DAC_NOT_STANDBY_MASK, 0); - if (!ret) - ret = snd_soc_update_bits(codec, - STIH416_AUDIO_DAC_CTRL, - STIH416_DAC_ANA_NOT_PWR, 0); - if (!ret) - ret = snd_soc_update_bits(codec, - STIH416_AUDIO_DAC_CTRL, - STIH416_DAC_NOT_PNDBG_MASK, - 0); - if (!ret) - ret = snd_soc_update_bits(codec, - STIH416_AUDIO_DAC_CTRL, - STIH416_DAC_SOFTMUTE_MASK, - STIH416_DAC_SOFTMUTE_MASK); - break; - default: - return -EINVAL; - } + if (!ret) + ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL, + STIH407_DAC_STANDBY_ANA_MASK, + STIH407_DAC_STANDBY_ANA_MASK); + if (!ret) + ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL, + STIH407_DAC_SOFTMUTE_MASK, + STIH407_DAC_SOFTMUTE_MASK); if (ret < 0) { - dev_err(codec->dev, "Failed to update DAC registers"); + dev_err(codec->dev, "Failed to update DAC registers\n"); return ret; } @@ -217,37 +167,6 @@ static int sti_sas_dac_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } -static int stih416_dac_probe(struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - struct sti_sas_data *drvdata = dev_get_drvdata(codec->dev); - struct sti_dac_audio *dac = &drvdata->dac; - - /* Get reset control */ - dac->rst = devm_reset_control_get(codec->dev, "dac_rst"); - if (IS_ERR(dac->rst)) { - dev_err(dai->codec->dev, - "%s: ERROR: DAC reset control not defined !\n", - __func__); - dac->rst = NULL; - return -EFAULT; - } - /* Put the DAC into reset */ - reset_control_assert(dac->rst); - - return 0; -} - -static const struct snd_soc_dapm_widget stih416_sas_dapm_widgets[] = { - SND_SOC_DAPM_PGA("DAC bandgap", STIH416_AUDIO_DAC_CTRL, - STIH416_DAC_NOT_PNDBG_MASK, 0, NULL, 0), - SND_SOC_DAPM_OUT_DRV("DAC standby ana", STIH416_AUDIO_DAC_CTRL, - STIH416_DAC_ANA_NOT_PWR, 0, NULL, 0), - SND_SOC_DAPM_DAC("DAC standby", "dac_p", STIH416_AUDIO_DAC_CTRL, - STIH416_DAC_NOT_STANDBY, 0), - SND_SOC_DAPM_OUTPUT("DAC Output"), -}; - static const struct snd_soc_dapm_widget stih407_sas_dapm_widgets[] = { SND_SOC_DAPM_OUT_DRV("DAC standby ana", STIH407_AUDIO_DAC_CTRL, STIH407_DAC_STANDBY_ANA, 1, NULL, 0), @@ -256,30 +175,11 @@ static const struct snd_soc_dapm_widget stih407_sas_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("DAC Output"), }; -static const struct snd_soc_dapm_route stih416_sas_route[] = { - {"DAC Output", NULL, "DAC bandgap"}, - {"DAC Output", NULL, "DAC standby ana"}, - {"DAC standby ana", NULL, "DAC standby"}, -}; - static const struct snd_soc_dapm_route stih407_sas_route[] = { {"DAC Output", NULL, "DAC standby ana"}, {"DAC standby ana", NULL, "DAC standby"}, }; -static int stih416_sas_dac_mute(struct snd_soc_dai *dai, int mute, int stream) -{ - struct snd_soc_codec *codec = dai->codec; - - if (mute) { - return snd_soc_update_bits(codec, STIH416_AUDIO_DAC_CTRL, - STIH416_DAC_SOFTMUTE_MASK, - STIH416_DAC_SOFTMUTE_MASK); - } else { - return snd_soc_update_bits(codec, STIH416_AUDIO_DAC_CTRL, - STIH416_DAC_SOFTMUTE_MASK, 0); - } -} static int stih407_sas_dac_mute(struct snd_soc_dai *dai, int mute, int stream) { @@ -392,13 +292,13 @@ static int sti_sas_prepare(struct snd_pcm_substream *substream, switch (dai->id) { case STI_SAS_DAI_SPDIF_OUT: if ((drvdata->spdif.mclk / runtime->rate) != 128) { - dev_err(codec->dev, "unexpected mclk-fs ratio"); + dev_err(codec->dev, "unexpected mclk-fs ratio\n"); return -EINVAL; } break; case STI_SAS_DAI_ANALOG_OUT: if ((drvdata->dac.mclk / runtime->rate) != 256) { - dev_err(codec->dev, "unexpected mclk-fs ratio"); + dev_err(codec->dev, "unexpected mclk-fs ratio\n"); return -EINVAL; } break; @@ -407,13 +307,6 @@ static int sti_sas_prepare(struct snd_pcm_substream *substream, return 0; } -static const struct snd_soc_dai_ops stih416_dac_ops = { - .set_fmt = sti_sas_dac_set_fmt, - .mute_stream = stih416_sas_dac_mute, - .prepare = sti_sas_prepare, - .set_sysclk = sti_sas_set_sysclk, -}; - static const struct snd_soc_dai_ops stih407_dac_ops = { .set_fmt = sti_sas_dac_set_fmt, .mute_stream = stih407_sas_dac_mute, @@ -434,31 +327,7 @@ static const struct regmap_config stih407_sas_regmap = { .reg_write = sti_sas_write_reg, }; -static const struct regmap_config stih416_sas_regmap = { - .reg_bits = 32, - .val_bits = 32, - - .max_register = STIH416_AUDIO_DAC_CTRL, - .reg_defaults = stih416_sas_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(stih416_sas_reg_defaults), - .volatile_reg = sti_sas_volatile_register, - .cache_type = REGCACHE_RBTREE, - .reg_read = sti_sas_read_reg, - .reg_write = sti_sas_write_reg, -}; - -static const struct sti_sas_dev_data stih416_data = { - .chipid = CHIPID_STIH416, - .regmap = &stih416_sas_regmap, - .dac_ops = &stih416_dac_ops, - .dapm_widgets = stih416_sas_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(stih416_sas_dapm_widgets), - .dapm_routes = stih416_sas_route, - .num_dapm_routes = ARRAY_SIZE(stih416_sas_route), -}; - static const struct sti_sas_dev_data stih407_data = { - .chipid = CHIPID_STIH407, .regmap = &stih407_sas_regmap, .dac_ops = &stih407_dac_ops, .dapm_widgets = stih407_sas_dapm_widgets, @@ -532,10 +401,6 @@ static struct snd_soc_codec_driver sti_sas_driver = { }; static const struct of_device_id sti_sas_dev_match[] = { - { - .compatible = "st,stih416-sas-codec", - .data = &stih416_data, - }, { .compatible = "st,stih407-sas-codec", .data = &stih407_data, @@ -558,7 +423,7 @@ static int sti_sas_driver_probe(struct platform_device *pdev) /* Populate data structure depending on compatibility */ of_id = of_match_node(sti_sas_dev_match, pnode); if (!of_id->data) { - dev_err(&pdev->dev, "data associated to device is missing"); + dev_err(&pdev->dev, "data associated to device is missing\n"); return -EINVAL; } @@ -584,10 +449,6 @@ static int sti_sas_driver_probe(struct platform_device *pdev) } drvdata->spdif.regmap = drvdata->dac.regmap; - /* Set DAC dai probe */ - if (drvdata->dev_data->chipid == CHIPID_STIH416) - sti_sas_dai[STI_SAS_DAI_ANALOG_OUT].probe = stih416_dac_probe; - sti_sas_dai[STI_SAS_DAI_ANALOG_OUT].ops = drvdata->dev_data->dac_ops; /* Set dapms*/ diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c index b2acd3293ea8..f200d1cfc4bd 100644 --- a/sound/soc/fsl/efika-audio-fabric.c +++ b/sound/soc/fsl/efika-audio-fabric.c @@ -27,7 +27,6 @@ #include "mpc5200_dma.h" #include "mpc5200_psc_ac97.h" -#include "../codecs/stac9766.h" #define DRV_NAME "efika-audio-fabric" diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index 549fac349fa0..98eb205a0b62 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -7,6 +7,7 @@ #include #include +#include #include "uniperif.h" @@ -97,6 +98,28 @@ static const struct of_device_id snd_soc_sti_match[] = { {}, }; +int sti_uniperiph_reset(struct uniperif *uni) +{ + int count = 10; + + /* Reset uniperipheral uni */ + SET_UNIPERIF_SOFT_RST_SOFT_RST(uni); + + if (uni->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) { + while (GET_UNIPERIF_SOFT_RST_SOFT_RST(uni) && count) { + udelay(5); + count--; + } + } + + if (!count) { + dev_err(uni->dev, "Failed to reset uniperif\n"); + return -EIO; + } + + return 0; +} + int sti_uniperiph_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) @@ -293,7 +316,7 @@ static int sti_uniperiph_dai_suspend(struct snd_soc_dai *dai) /* The uniperipheral should be in stopped state */ if (uni->state != UNIPERIF_STATE_STOPPED) { - dev_err(uni->dev, "%s: invalid uni state( %d)", + dev_err(uni->dev, "%s: invalid uni state( %d)\n", __func__, (int)uni->state); return -EBUSY; } @@ -301,7 +324,7 @@ static int sti_uniperiph_dai_suspend(struct snd_soc_dai *dai) /* Pinctrl: switch pinstate to sleep */ ret = pinctrl_pm_select_sleep_state(uni->dev); if (ret) - dev_err(uni->dev, "%s: failed to select pinctrl state", + dev_err(uni->dev, "%s: failed to select pinctrl state\n", __func__); return ret; @@ -322,7 +345,7 @@ static int sti_uniperiph_dai_resume(struct snd_soc_dai *dai) /* pinctrl: switch pinstate to default */ ret = pinctrl_pm_select_default_state(uni->dev); if (ret) - dev_err(uni->dev, "%s: failed to select pinctrl state", + dev_err(uni->dev, "%s: failed to select pinctrl state\n", __func__); return ret; @@ -366,11 +389,12 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node, const struct of_device_id *of_id; const struct sti_uniperiph_dev_data *dev_data; const char *mode; + int ret; /* Populate data structure depending on compatibility */ of_id = of_match_node(snd_soc_sti_match, node); if (!of_id->data) { - dev_err(dev, "data associated to device is missing"); + dev_err(dev, "data associated to device is missing\n"); return -EINVAL; } dev_data = (struct sti_uniperiph_dev_data *)of_id->data; @@ -389,7 +413,7 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node, uni->mem_region = platform_get_resource(priv->pdev, IORESOURCE_MEM, 0); if (!uni->mem_region) { - dev_err(dev, "Failed to get memory resource"); + dev_err(dev, "Failed to get memory resource\n"); return -ENODEV; } @@ -403,7 +427,7 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node, uni->irq = platform_get_irq(priv->pdev, 0); if (uni->irq < 0) { - dev_err(dev, "Failed to get IRQ resource"); + dev_err(dev, "Failed to get IRQ resource\n"); return -ENXIO; } @@ -421,12 +445,15 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node, dai_data->stream = dev_data->stream; if (priv->dai_data.stream == SNDRV_PCM_STREAM_PLAYBACK) { - uni_player_init(priv->pdev, uni); + ret = uni_player_init(priv->pdev, uni); stream = &dai->playback; } else { - uni_reader_init(priv->pdev, uni); + ret = uni_reader_init(priv->pdev, uni); stream = &dai->capture; } + if (ret < 0) + return ret; + dai->ops = uni->dai_ops; stream->stream_name = dai->name; diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index 1993c655fb79..d487dd2ef016 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -1397,6 +1397,8 @@ static inline int sti_uniperiph_get_unip_tdm_frame_size(struct uniperif *uni) return (uni->tdm_slot.slots * uni->tdm_slot.slot_width / 8); } +int sti_uniperiph_reset(struct uniperif *uni); + int sti_uniperiph_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index ad54d4cf58ad..60ae31a303ab 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -6,8 +6,6 @@ */ #include -#include -#include #include #include @@ -55,25 +53,6 @@ static const struct snd_pcm_hardware uni_player_pcm_hw = { .buffer_bytes_max = 256 * PAGE_SIZE }; -static inline int reset_player(struct uniperif *player) -{ - int count = 10; - - if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) { - while (GET_UNIPERIF_SOFT_RST_SOFT_RST(player) && count) { - udelay(5); - count--; - } - } - - if (!count) { - dev_err(player->dev, "Failed to reset uniperif"); - return -EIO; - } - - return 0; -} - /* * uni_player_irq_handler * In case of error audio stream is stopped; stop action is protected via PCM @@ -97,7 +76,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) /* Check for fifo error (underrun) */ if (unlikely(status & UNIPERIF_ITS_FIFO_ERROR_MASK(player))) { - dev_err(player->dev, "FIFO underflow error detected"); + dev_err(player->dev, "FIFO underflow error detected\n"); /* Interrupt is just for information when underflow recovery */ if (player->underflow_enabled) { @@ -119,7 +98,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) /* Check for dma error (overrun) */ if (unlikely(status & UNIPERIF_ITS_DMA_ERROR_MASK(player))) { - dev_err(player->dev, "DMA error detected"); + dev_err(player->dev, "DMA error detected\n"); /* Disable interrupt so doesn't continually fire */ SET_UNIPERIF_ITM_BCLR_DMA_ERROR(player); @@ -135,11 +114,14 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) /* Check for underflow recovery done */ if (unlikely(status & UNIPERIF_ITM_UNDERFLOW_REC_DONE_MASK(player))) { if (!player->underflow_enabled) { - dev_err(player->dev, "unexpected Underflow recovering"); + dev_err(player->dev, + "unexpected Underflow recovering\n"); return -EPERM; } /* Read the underflow recovery duration */ tmp = GET_UNIPERIF_STATUS_1_UNDERFLOW_DURATION(player); + dev_dbg(player->dev, "Underflow recovered (%d LR clocks max)\n", + tmp); /* Clear the underflow recovery duration */ SET_UNIPERIF_BIT_CONTROL_CLR_UNDERFLOW_DURATION(player); @@ -153,7 +135,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) /* Check if underflow recovery failed */ if (unlikely(status & UNIPERIF_ITM_UNDERFLOW_REC_FAILED_MASK(player))) { - dev_err(player->dev, "Underflow recovery failed"); + dev_err(player->dev, "Underflow recovery failed\n"); /* Stop the player */ snd_pcm_stream_lock(player->substream); @@ -336,7 +318,7 @@ static int uni_player_prepare_iec958(struct uniperif *player, /* Oversampling must be multiple of 128 as iec958 frame is 32-bits */ if ((clk_div % 128) || (clk_div <= 0)) { - dev_err(player->dev, "%s: invalid clk_div %d", + dev_err(player->dev, "%s: invalid clk_div %d\n", __func__, clk_div); return -EINVAL; } @@ -359,7 +341,7 @@ static int uni_player_prepare_iec958(struct uniperif *player, SET_UNIPERIF_I2S_FMT_DATA_SIZE_24(player); break; default: - dev_err(player->dev, "format not supported"); + dev_err(player->dev, "format not supported\n"); return -EINVAL; } @@ -448,12 +430,12 @@ static int uni_player_prepare_pcm(struct uniperif *player, * for 16 bits must be a multiple of 64 */ if ((slot_width == 32) && (clk_div % 128)) { - dev_err(player->dev, "%s: invalid clk_div", __func__); + dev_err(player->dev, "%s: invalid clk_div\n", __func__); return -EINVAL; } if ((slot_width == 16) && (clk_div % 64)) { - dev_err(player->dev, "%s: invalid clk_div", __func__); + dev_err(player->dev, "%s: invalid clk_div\n", __func__); return -EINVAL; } @@ -471,7 +453,7 @@ static int uni_player_prepare_pcm(struct uniperif *player, SET_UNIPERIF_I2S_FMT_DATA_SIZE_16(player); break; default: - dev_err(player->dev, "subframe format not supported"); + dev_err(player->dev, "subframe format not supported\n"); return -EINVAL; } @@ -491,7 +473,7 @@ static int uni_player_prepare_pcm(struct uniperif *player, break; default: - dev_err(player->dev, "format not supported"); + dev_err(player->dev, "format not supported\n"); return -EINVAL; } @@ -504,7 +486,7 @@ static int uni_player_prepare_pcm(struct uniperif *player, /* Number of channelsmust be even*/ if ((runtime->channels % 2) || (runtime->channels < 2) || (runtime->channels > 10)) { - dev_err(player->dev, "%s: invalid nb of channels", __func__); + dev_err(player->dev, "%s: invalid nb of channels\n", __func__); return -EINVAL; } @@ -762,7 +744,7 @@ static int uni_player_prepare(struct snd_pcm_substream *substream, /* The player should be stopped */ if (player->state != UNIPERIF_STATE_STOPPED) { - dev_err(player->dev, "%s: invalid player state %d", __func__, + dev_err(player->dev, "%s: invalid player state %d\n", __func__, player->state); return -EINVAL; } @@ -791,7 +773,8 @@ static int uni_player_prepare(struct snd_pcm_substream *substream, /* Trigger limit must be an even number */ if ((!trigger_limit % 2) || (trigger_limit != 1 && transfer_size % 2) || (trigger_limit > UNIPERIF_CONFIG_DMA_TRIG_LIMIT_MASK(player))) { - dev_err(player->dev, "invalid trigger limit %d", trigger_limit); + dev_err(player->dev, "invalid trigger limit %d\n", + trigger_limit); return -EINVAL; } @@ -812,7 +795,7 @@ static int uni_player_prepare(struct snd_pcm_substream *substream, ret = uni_player_prepare_tdm(player, runtime); break; default: - dev_err(player->dev, "invalid player type"); + dev_err(player->dev, "invalid player type\n"); return -EINVAL; } @@ -852,16 +835,14 @@ static int uni_player_prepare(struct snd_pcm_substream *substream, SET_UNIPERIF_I2S_FMT_PADDING_SONY_MODE(player); break; default: - dev_err(player->dev, "format not supported"); + dev_err(player->dev, "format not supported\n"); return -EINVAL; } SET_UNIPERIF_I2S_FMT_NO_OF_SAMPLES_TO_READ(player, 0); - /* Reset uniperipheral player */ - SET_UNIPERIF_SOFT_RST_SOFT_RST(player); - return reset_player(player); + return sti_uniperiph_reset(player); } static int uni_player_start(struct uniperif *player) @@ -870,13 +851,13 @@ static int uni_player_start(struct uniperif *player) /* The player should be stopped */ if (player->state != UNIPERIF_STATE_STOPPED) { - dev_err(player->dev, "%s: invalid player state", __func__); + dev_err(player->dev, "%s: invalid player state\n", __func__); return -EINVAL; } ret = clk_prepare_enable(player->clk); if (ret) { - dev_err(player->dev, "%s: Failed to enable clock", __func__); + dev_err(player->dev, "%s: Failed to enable clock\n", __func__); return ret; } @@ -893,10 +874,7 @@ static int uni_player_start(struct uniperif *player) SET_UNIPERIF_ITM_BSET_UNDERFLOW_REC_FAILED(player); } - /* Reset uniperipheral player */ - SET_UNIPERIF_SOFT_RST_SOFT_RST(player); - - ret = reset_player(player); + ret = sti_uniperiph_reset(player); if (ret < 0) { clk_disable_unprepare(player->clk); return ret; @@ -938,17 +916,14 @@ static int uni_player_stop(struct uniperif *player) /* The player should not be in stopped state */ if (player->state == UNIPERIF_STATE_STOPPED) { - dev_err(player->dev, "%s: invalid player state", __func__); + dev_err(player->dev, "%s: invalid player state\n", __func__); return -EINVAL; } /* Turn the player off */ SET_UNIPERIF_CTRL_OPERATION_OFF(player); - /* Soft reset the player */ - SET_UNIPERIF_SOFT_RST_SOFT_RST(player); - - ret = reset_player(player); + ret = sti_uniperiph_reset(player); if (ret < 0) return ret; @@ -973,7 +948,7 @@ int uni_player_resume(struct uniperif *player) ret = regmap_field_write(player->clk_sel, 1); if (ret) { dev_err(player->dev, - "%s: Failed to select freq synth clock", + "%s: Failed to select freq synth clock\n", __func__); return ret; } @@ -1070,7 +1045,7 @@ int uni_player_init(struct platform_device *pdev, ret = uni_player_parse_dt_audio_glue(pdev, player); if (ret < 0) { - dev_err(player->dev, "Failed to parse DeviceTree"); + dev_err(player->dev, "Failed to parse DeviceTree\n"); return ret; } @@ -1085,15 +1060,17 @@ int uni_player_init(struct platform_device *pdev, /* Get uniperif resource */ player->clk = of_clk_get(pdev->dev.of_node, 0); - if (IS_ERR(player->clk)) + if (IS_ERR(player->clk)) { + dev_err(player->dev, "Failed to get clock\n"); ret = PTR_ERR(player->clk); + } /* Select the frequency synthesizer clock */ if (player->clk_sel) { ret = regmap_field_write(player->clk_sel, 1); if (ret) { dev_err(player->dev, - "%s: Failed to select freq synth clock", + "%s: Failed to select freq synth clock\n", __func__); return ret; } @@ -1105,7 +1082,7 @@ int uni_player_init(struct platform_device *pdev, ret = regmap_field_write(player->valid_sel, player->id); if (ret) { dev_err(player->dev, - "%s: unable to connect to tdm bus", __func__); + "%s: unable to connect to tdm bus\n", __func__); return ret; } } @@ -1113,8 +1090,10 @@ int uni_player_init(struct platform_device *pdev, ret = devm_request_irq(&pdev->dev, player->irq, uni_player_irq_handler, IRQF_SHARED, dev_name(&pdev->dev), player); - if (ret < 0) + if (ret < 0) { + dev_err(player->dev, "unable to request IRQ %d\n", player->irq); return ret; + } mutex_init(&player->ctrl_lock); diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index 0e1c3ee56675..5992c6ab3833 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -5,10 +5,6 @@ * License terms: GNU General Public License (GPL), version 2 */ -#include -#include -#include - #include #include "uniperif.h" @@ -52,7 +48,7 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id) if (reader->state == UNIPERIF_STATE_STOPPED) { /* Unexpected IRQ: do nothing */ - dev_warn(reader->dev, "unexpected IRQ "); + dev_warn(reader->dev, "unexpected IRQ\n"); return IRQ_HANDLED; } @@ -62,7 +58,7 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id) /* Check for fifo overflow error */ if (unlikely(status & UNIPERIF_ITS_FIFO_ERROR_MASK(reader))) { - dev_err(reader->dev, "FIFO error detected"); + dev_err(reader->dev, "FIFO error detected\n"); snd_pcm_stream_lock(reader->substream); snd_pcm_stop(reader->substream, SNDRV_PCM_STATE_XRUN); @@ -105,7 +101,7 @@ static int uni_reader_prepare_pcm(struct snd_pcm_runtime *runtime, SET_UNIPERIF_I2S_FMT_DATA_SIZE_16(reader); break; default: - dev_err(reader->dev, "subframe format not supported"); + dev_err(reader->dev, "subframe format not supported\n"); return -EINVAL; } @@ -125,14 +121,14 @@ static int uni_reader_prepare_pcm(struct snd_pcm_runtime *runtime, break; default: - dev_err(reader->dev, "format not supported"); + dev_err(reader->dev, "format not supported\n"); return -EINVAL; } /* Number of channels must be even */ if ((runtime->channels % 2) || (runtime->channels < 2) || (runtime->channels > 10)) { - dev_err(reader->dev, "%s: invalid nb of channels", __func__); + dev_err(reader->dev, "%s: invalid nb of channels\n", __func__); return -EINVAL; } @@ -186,11 +182,10 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream, struct uniperif *reader = priv->dai_data.uni; struct snd_pcm_runtime *runtime = substream->runtime; int transfer_size, trigger_limit, ret; - int count = 10; /* The reader should be stopped */ if (reader->state != UNIPERIF_STATE_STOPPED) { - dev_err(reader->dev, "%s: invalid reader state %d", __func__, + dev_err(reader->dev, "%s: invalid reader state %d\n", __func__, reader->state); return -EINVAL; } @@ -219,7 +214,8 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream, if ((!trigger_limit % 2) || (trigger_limit != 1 && transfer_size % 2) || (trigger_limit > UNIPERIF_CONFIG_DMA_TRIG_LIMIT_MASK(reader))) { - dev_err(reader->dev, "invalid trigger limit %d", trigger_limit); + dev_err(reader->dev, "invalid trigger limit %d\n", + trigger_limit); return -EINVAL; } @@ -246,7 +242,7 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream, SET_UNIPERIF_I2S_FMT_PADDING_SONY_MODE(reader); break; default: - dev_err(reader->dev, "format not supported"); + dev_err(reader->dev, "format not supported\n"); return -EINVAL; } @@ -287,25 +283,14 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream, } /* Reset uniperipheral reader */ - SET_UNIPERIF_SOFT_RST_SOFT_RST(reader); - - while (GET_UNIPERIF_SOFT_RST_SOFT_RST(reader)) { - udelay(5); - count--; - } - if (!count) { - dev_err(reader->dev, "Failed to reset uniperif"); - return -EIO; - } - - return 0; + return sti_uniperiph_reset(reader); } static int uni_reader_start(struct uniperif *reader) { /* The reader should be stopped */ if (reader->state != UNIPERIF_STATE_STOPPED) { - dev_err(reader->dev, "%s: invalid reader state", __func__); + dev_err(reader->dev, "%s: invalid reader state\n", __func__); return -EINVAL; } @@ -325,7 +310,7 @@ static int uni_reader_stop(struct uniperif *reader) { /* The reader should not be in stopped state */ if (reader->state == UNIPERIF_STATE_STOPPED) { - dev_err(reader->dev, "%s: invalid reader state", __func__); + dev_err(reader->dev, "%s: invalid reader state\n", __func__); return -EINVAL; } @@ -423,7 +408,7 @@ int uni_reader_init(struct platform_device *pdev, uni_reader_irq_handler, IRQF_SHARED, dev_name(&pdev->dev), reader); if (ret < 0) { - dev_err(&pdev->dev, "Failed to request IRQ"); + dev_err(&pdev->dev, "Failed to request IRQ\n"); return -EBUSY; } diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index dd2368297fd3..6c344e16aca4 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -9,6 +9,14 @@ config SND_SUN4I_CODEC Select Y or M to add support for the Codec embedded in the Allwinner A10 and affiliated SoCs. +config SND_SUN8I_CODEC_ANALOG + tristate "Allwinner sun8i Codec Analog Controls Support" + depends on MACH_SUN8I || COMPILE_TEST + select REGMAP + help + Say Y or M if you want to add support for the analog controls for + the codec embedded in newer Allwinner SoCs. + config SND_SUN4I_I2S tristate "Allwinner A10 I2S Support" select SND_SOC_GENERIC_DMAENGINE_PCM diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index 604c7b842837..241c0df9ca0c 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -1,3 +1,4 @@ obj-$(CONFIG_SND_SUN4I_CODEC) += sun4i-codec.o obj-$(CONFIG_SND_SUN4I_I2S) += sun4i-i2s.o obj-$(CONFIG_SND_SUN4I_SPDIF) += sun4i-spdif.o +obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 56ed9472e89f..848af01692a0 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -3,6 +3,7 @@ * Copyright 2014 Jon Smirl * Copyright 2015 Maxime Ripard * Copyright 2015 Adam Sampson + * Copyright 2016 Chen-Yu Tsai * * Based on the Allwinner SDK driver, released under the GPL. * @@ -24,10 +25,12 @@ #include #include #include -#include #include +#include +#include #include #include +#include #include #include @@ -38,7 +41,7 @@ #include #include -/* Codec DAC register offsets and bit fields */ +/* Codec DAC digital controls and FIFO registers */ #define SUN4I_CODEC_DAC_DPC (0x00) #define SUN4I_CODEC_DAC_DPC_EN_DA (31) #define SUN4I_CODEC_DAC_DPC_DVOL (12) @@ -55,6 +58,8 @@ #define SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH (0) #define SUN4I_CODEC_DAC_FIFOS (0x08) #define SUN4I_CODEC_DAC_TXDATA (0x0c) + +/* Codec DAC side analog signal controls */ #define SUN4I_CODEC_DAC_ACTL (0x10) #define SUN4I_CODEC_DAC_ACTL_DACAENR (31) #define SUN4I_CODEC_DAC_ACTL_DACAENL (30) @@ -69,7 +74,7 @@ #define SUN4I_CODEC_DAC_TUNE (0x14) #define SUN4I_CODEC_DAC_DEBUG (0x18) -/* Codec ADC register offsets and bit fields */ +/* Codec ADC digital controls and FIFO registers */ #define SUN4I_CODEC_ADC_FIFOC (0x1c) #define SUN4I_CODEC_ADC_FIFOC_ADC_FS (29) #define SUN4I_CODEC_ADC_FIFOC_EN_AD (28) @@ -81,6 +86,8 @@ #define SUN4I_CODEC_ADC_FIFOC_FIFO_FLUSH (0) #define SUN4I_CODEC_ADC_FIFOS (0x20) #define SUN4I_CODEC_ADC_RXDATA (0x24) + +/* Codec ADC side analog signal controls */ #define SUN4I_CODEC_ADC_ACTL (0x28) #define SUN4I_CODEC_ADC_ACTL_ADC_R_EN (31) #define SUN4I_CODEC_ADC_ACTL_ADC_L_EN (30) @@ -93,19 +100,141 @@ #define SUN4I_CODEC_ADC_ACTL_DDE (3) #define SUN4I_CODEC_ADC_DEBUG (0x2c) -/* Other various ADC registers */ +/* FIFO counters */ #define SUN4I_CODEC_DAC_TXCNT (0x30) #define SUN4I_CODEC_ADC_RXCNT (0x34) + +/* Calibration register (sun7i only) */ #define SUN7I_CODEC_AC_DAC_CAL (0x38) + +/* Microphone controls (sun7i only) */ #define SUN7I_CODEC_AC_MIC_PHONE_CAL (0x3c) +/* + * sun6i specific registers + * + * sun6i shares the same digital control and FIFO registers as sun4i, + * but only the DAC digital controls are at the same offset. The others + * have been moved around to accommodate extra analog controls. + */ + +/* Codec DAC digital controls and FIFO registers */ +#define SUN6I_CODEC_ADC_FIFOC (0x10) +#define SUN6I_CODEC_ADC_FIFOC_EN_AD (28) +#define SUN6I_CODEC_ADC_FIFOS (0x14) +#define SUN6I_CODEC_ADC_RXDATA (0x18) + +/* Output mixer and gain controls */ +#define SUN6I_CODEC_OM_DACA_CTRL (0x20) +#define SUN6I_CODEC_OM_DACA_CTRL_DACAREN (31) +#define SUN6I_CODEC_OM_DACA_CTRL_DACALEN (30) +#define SUN6I_CODEC_OM_DACA_CTRL_RMIXEN (29) +#define SUN6I_CODEC_OM_DACA_CTRL_LMIXEN (28) +#define SUN6I_CODEC_OM_DACA_CTRL_RMIX_MIC1 (23) +#define SUN6I_CODEC_OM_DACA_CTRL_RMIX_MIC2 (22) +#define SUN6I_CODEC_OM_DACA_CTRL_RMIX_PHONE (21) +#define SUN6I_CODEC_OM_DACA_CTRL_RMIX_PHONEP (20) +#define SUN6I_CODEC_OM_DACA_CTRL_RMIX_LINEINR (19) +#define SUN6I_CODEC_OM_DACA_CTRL_RMIX_DACR (18) +#define SUN6I_CODEC_OM_DACA_CTRL_RMIX_DACL (17) +#define SUN6I_CODEC_OM_DACA_CTRL_LMIX_MIC1 (16) +#define SUN6I_CODEC_OM_DACA_CTRL_LMIX_MIC2 (15) +#define SUN6I_CODEC_OM_DACA_CTRL_LMIX_PHONE (14) +#define SUN6I_CODEC_OM_DACA_CTRL_LMIX_PHONEN (13) +#define SUN6I_CODEC_OM_DACA_CTRL_LMIX_LINEINL (12) +#define SUN6I_CODEC_OM_DACA_CTRL_LMIX_DACL (11) +#define SUN6I_CODEC_OM_DACA_CTRL_LMIX_DACR (10) +#define SUN6I_CODEC_OM_DACA_CTRL_RHPIS (9) +#define SUN6I_CODEC_OM_DACA_CTRL_LHPIS (8) +#define SUN6I_CODEC_OM_DACA_CTRL_RHPPAMUTE (7) +#define SUN6I_CODEC_OM_DACA_CTRL_LHPPAMUTE (6) +#define SUN6I_CODEC_OM_DACA_CTRL_HPVOL (0) +#define SUN6I_CODEC_OM_PA_CTRL (0x24) +#define SUN6I_CODEC_OM_PA_CTRL_HPPAEN (31) +#define SUN6I_CODEC_OM_PA_CTRL_HPCOM_CTL (29) +#define SUN6I_CODEC_OM_PA_CTRL_COMPTEN (28) +#define SUN6I_CODEC_OM_PA_CTRL_MIC1G (15) +#define SUN6I_CODEC_OM_PA_CTRL_MIC2G (12) +#define SUN6I_CODEC_OM_PA_CTRL_LINEING (9) +#define SUN6I_CODEC_OM_PA_CTRL_PHONEG (6) +#define SUN6I_CODEC_OM_PA_CTRL_PHONEPG (3) +#define SUN6I_CODEC_OM_PA_CTRL_PHONENG (0) + +/* Microphone, line out and phone out controls */ +#define SUN6I_CODEC_MIC_CTRL (0x28) +#define SUN6I_CODEC_MIC_CTRL_HBIASEN (31) +#define SUN6I_CODEC_MIC_CTRL_MBIASEN (30) +#define SUN6I_CODEC_MIC_CTRL_MIC1AMPEN (28) +#define SUN6I_CODEC_MIC_CTRL_MIC1BOOST (25) +#define SUN6I_CODEC_MIC_CTRL_MIC2AMPEN (24) +#define SUN6I_CODEC_MIC_CTRL_MIC2BOOST (21) +#define SUN6I_CODEC_MIC_CTRL_MIC2SLT (20) +#define SUN6I_CODEC_MIC_CTRL_LINEOUTLEN (19) +#define SUN6I_CODEC_MIC_CTRL_LINEOUTREN (18) +#define SUN6I_CODEC_MIC_CTRL_LINEOUTLSRC (17) +#define SUN6I_CODEC_MIC_CTRL_LINEOUTRSRC (16) +#define SUN6I_CODEC_MIC_CTRL_LINEOUTVC (11) +#define SUN6I_CODEC_MIC_CTRL_PHONEPREG (8) + +/* ADC mixer controls */ +#define SUN6I_CODEC_ADC_ACTL (0x2c) +#define SUN6I_CODEC_ADC_ACTL_ADCREN (31) +#define SUN6I_CODEC_ADC_ACTL_ADCLEN (30) +#define SUN6I_CODEC_ADC_ACTL_ADCRG (27) +#define SUN6I_CODEC_ADC_ACTL_ADCLG (24) +#define SUN6I_CODEC_ADC_ACTL_RADCMIX_MIC1 (13) +#define SUN6I_CODEC_ADC_ACTL_RADCMIX_MIC2 (12) +#define SUN6I_CODEC_ADC_ACTL_RADCMIX_PHONE (11) +#define SUN6I_CODEC_ADC_ACTL_RADCMIX_PHONEP (10) +#define SUN6I_CODEC_ADC_ACTL_RADCMIX_LINEINR (9) +#define SUN6I_CODEC_ADC_ACTL_RADCMIX_OMIXR (8) +#define SUN6I_CODEC_ADC_ACTL_RADCMIX_OMIXL (7) +#define SUN6I_CODEC_ADC_ACTL_LADCMIX_MIC1 (6) +#define SUN6I_CODEC_ADC_ACTL_LADCMIX_MIC2 (5) +#define SUN6I_CODEC_ADC_ACTL_LADCMIX_PHONE (4) +#define SUN6I_CODEC_ADC_ACTL_LADCMIX_PHONEN (3) +#define SUN6I_CODEC_ADC_ACTL_LADCMIX_LINEINL (2) +#define SUN6I_CODEC_ADC_ACTL_LADCMIX_OMIXL (1) +#define SUN6I_CODEC_ADC_ACTL_LADCMIX_OMIXR (0) + +/* Analog performance tuning controls */ +#define SUN6I_CODEC_ADDA_TUNE (0x30) + +/* Calibration controls */ +#define SUN6I_CODEC_CALIBRATION (0x34) + +/* FIFO counters */ +#define SUN6I_CODEC_DAC_TXCNT (0x40) +#define SUN6I_CODEC_ADC_RXCNT (0x44) + +/* headset jack detection and button support registers */ +#define SUN6I_CODEC_HMIC_CTL (0x50) +#define SUN6I_CODEC_HMIC_DATA (0x54) + +/* TODO sun6i DAP (Digital Audio Processing) bits */ + +/* FIFO counters moved on A23 */ +#define SUN8I_A23_CODEC_DAC_TXCNT (0x1c) +#define SUN8I_A23_CODEC_ADC_RXCNT (0x20) + +/* TX FIFO moved on H3 */ +#define SUN8I_H3_CODEC_DAC_TXDATA (0x20) +#define SUN8I_H3_CODEC_DAC_DBG (0x48) +#define SUN8I_H3_CODEC_ADC_DBG (0x4c) + +/* TODO H3 DAP (Digital Audio Processing) bits */ + struct sun4i_codec { struct device *dev; struct regmap *regmap; struct clk *clk_apb; struct clk *clk_module; + struct reset_control *rst; struct gpio_desc *gpio_pa; + /* ADC_FIFOC register is at different offset on different SoCs */ + struct regmap_field *reg_adc_fifoc; + struct snd_dmaengine_dai_dma_data capture_dma_data; struct snd_dmaengine_dai_dma_data playback_dma_data; }; @@ -134,16 +263,16 @@ static void sun4i_codec_stop_playback(struct sun4i_codec *scodec) static void sun4i_codec_start_capture(struct sun4i_codec *scodec) { /* Enable ADC DRQ */ - regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, - BIT(SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN), - BIT(SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN)); + regmap_field_update_bits(scodec->reg_adc_fifoc, + BIT(SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN), + BIT(SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN)); } static void sun4i_codec_stop_capture(struct sun4i_codec *scodec) { /* Disable ADC DRQ */ - regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, - BIT(SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN), 0); + regmap_field_update_bits(scodec->reg_adc_fifoc, + BIT(SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN), 0); } static int sun4i_codec_trigger(struct snd_pcm_substream *substream, int cmd, @@ -186,24 +315,29 @@ static int sun4i_codec_prepare_capture(struct snd_pcm_substream *substream, /* Flush RX FIFO */ - regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, - BIT(SUN4I_CODEC_ADC_FIFOC_FIFO_FLUSH), - BIT(SUN4I_CODEC_ADC_FIFOC_FIFO_FLUSH)); + regmap_field_update_bits(scodec->reg_adc_fifoc, + BIT(SUN4I_CODEC_ADC_FIFOC_FIFO_FLUSH), + BIT(SUN4I_CODEC_ADC_FIFOC_FIFO_FLUSH)); /* Set RX FIFO trigger level */ - regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, - 0xf << SUN4I_CODEC_ADC_FIFOC_RX_TRIG_LEVEL, - 0x7 << SUN4I_CODEC_ADC_FIFOC_RX_TRIG_LEVEL); + regmap_field_update_bits(scodec->reg_adc_fifoc, + 0xf << SUN4I_CODEC_ADC_FIFOC_RX_TRIG_LEVEL, + 0x7 << SUN4I_CODEC_ADC_FIFOC_RX_TRIG_LEVEL); /* * FIXME: Undocumented in the datasheet, but * Allwinner's code mentions that it is related * related to microphone gain */ - regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_ACTL, - 0x3 << 25, - 0x1 << 25); + if (of_device_is_compatible(scodec->dev->of_node, + "allwinner,sun4i-a10-codec") || + of_device_is_compatible(scodec->dev->of_node, + "allwinner,sun7i-a20-codec")) { + regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_ACTL, + 0x3 << 25, + 0x1 << 25); + } if (of_device_is_compatible(scodec->dev->of_node, "allwinner,sun7i-a20-codec")) @@ -213,9 +347,9 @@ static int sun4i_codec_prepare_capture(struct snd_pcm_substream *substream, 0x1 << 8); /* Fill most significant bits with valid data MSB */ - regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, - BIT(SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE), - BIT(SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE)); + regmap_field_update_bits(scodec->reg_adc_fifoc, + BIT(SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE), + BIT(SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE)); return 0; } @@ -342,18 +476,19 @@ static int sun4i_codec_hw_params_capture(struct sun4i_codec *scodec, unsigned int hwrate) { /* Set ADC sample rate */ - regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, - 7 << SUN4I_CODEC_ADC_FIFOC_ADC_FS, - hwrate << SUN4I_CODEC_ADC_FIFOC_ADC_FS); + regmap_field_update_bits(scodec->reg_adc_fifoc, + 7 << SUN4I_CODEC_ADC_FIFOC_ADC_FS, + hwrate << SUN4I_CODEC_ADC_FIFOC_ADC_FS); /* Set the number of channels we want to use */ if (params_channels(params) == 1) - regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, - BIT(SUN4I_CODEC_ADC_FIFOC_MONO_EN), - BIT(SUN4I_CODEC_ADC_FIFOC_MONO_EN)); + regmap_field_update_bits(scodec->reg_adc_fifoc, + BIT(SUN4I_CODEC_ADC_FIFOC_MONO_EN), + BIT(SUN4I_CODEC_ADC_FIFOC_MONO_EN)); else - regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, - BIT(SUN4I_CODEC_ADC_FIFOC_MONO_EN), 0); + regmap_field_update_bits(scodec->reg_adc_fifoc, + BIT(SUN4I_CODEC_ADC_FIFOC_MONO_EN), + 0); return 0; } @@ -502,7 +637,7 @@ static struct snd_soc_dai_driver sun4i_codec_dai = { }, }; -/*** Codec ***/ +/*** sun4i Codec ***/ static const struct snd_kcontrol_new sun4i_codec_pa_mute = SOC_DAPM_SINGLE("Switch", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_PA_MUTE, 1, 0); @@ -638,6 +773,337 @@ static struct snd_soc_codec_driver sun4i_codec_codec = { }, }; +/*** sun6i Codec ***/ + +/* mixer controls */ +static const struct snd_kcontrol_new sun6i_codec_mixer_controls[] = { + SOC_DAPM_DOUBLE("DAC Playback Switch", + SUN6I_CODEC_OM_DACA_CTRL, + SUN6I_CODEC_OM_DACA_CTRL_LMIX_DACL, + SUN6I_CODEC_OM_DACA_CTRL_RMIX_DACR, 1, 0), + SOC_DAPM_DOUBLE("DAC Reversed Playback Switch", + SUN6I_CODEC_OM_DACA_CTRL, + SUN6I_CODEC_OM_DACA_CTRL_LMIX_DACR, + SUN6I_CODEC_OM_DACA_CTRL_RMIX_DACL, 1, 0), + SOC_DAPM_DOUBLE("Line In Playback Switch", + SUN6I_CODEC_OM_DACA_CTRL, + SUN6I_CODEC_OM_DACA_CTRL_LMIX_LINEINL, + SUN6I_CODEC_OM_DACA_CTRL_RMIX_LINEINR, 1, 0), + SOC_DAPM_DOUBLE("Mic1 Playback Switch", + SUN6I_CODEC_OM_DACA_CTRL, + SUN6I_CODEC_OM_DACA_CTRL_LMIX_MIC1, + SUN6I_CODEC_OM_DACA_CTRL_RMIX_MIC1, 1, 0), + SOC_DAPM_DOUBLE("Mic2 Playback Switch", + SUN6I_CODEC_OM_DACA_CTRL, + SUN6I_CODEC_OM_DACA_CTRL_LMIX_MIC2, + SUN6I_CODEC_OM_DACA_CTRL_RMIX_MIC2, 1, 0), +}; + +/* ADC mixer controls */ +static const struct snd_kcontrol_new sun6i_codec_adc_mixer_controls[] = { + SOC_DAPM_DOUBLE("Mixer Capture Switch", + SUN6I_CODEC_ADC_ACTL, + SUN6I_CODEC_ADC_ACTL_LADCMIX_OMIXL, + SUN6I_CODEC_ADC_ACTL_RADCMIX_OMIXR, 1, 0), + SOC_DAPM_DOUBLE("Mixer Reversed Capture Switch", + SUN6I_CODEC_ADC_ACTL, + SUN6I_CODEC_ADC_ACTL_LADCMIX_OMIXR, + SUN6I_CODEC_ADC_ACTL_RADCMIX_OMIXL, 1, 0), + SOC_DAPM_DOUBLE("Line In Capture Switch", + SUN6I_CODEC_ADC_ACTL, + SUN6I_CODEC_ADC_ACTL_LADCMIX_LINEINL, + SUN6I_CODEC_ADC_ACTL_RADCMIX_LINEINR, 1, 0), + SOC_DAPM_DOUBLE("Mic1 Capture Switch", + SUN6I_CODEC_ADC_ACTL, + SUN6I_CODEC_ADC_ACTL_LADCMIX_MIC1, + SUN6I_CODEC_ADC_ACTL_RADCMIX_MIC1, 1, 0), + SOC_DAPM_DOUBLE("Mic2 Capture Switch", + SUN6I_CODEC_ADC_ACTL, + SUN6I_CODEC_ADC_ACTL_LADCMIX_MIC2, + SUN6I_CODEC_ADC_ACTL_RADCMIX_MIC2, 1, 0), +}; + +/* headphone controls */ +static const char * const sun6i_codec_hp_src_enum_text[] = { + "DAC", "Mixer", +}; + +static SOC_ENUM_DOUBLE_DECL(sun6i_codec_hp_src_enum, + SUN6I_CODEC_OM_DACA_CTRL, + SUN6I_CODEC_OM_DACA_CTRL_LHPIS, + SUN6I_CODEC_OM_DACA_CTRL_RHPIS, + sun6i_codec_hp_src_enum_text); + +static const struct snd_kcontrol_new sun6i_codec_hp_src[] = { + SOC_DAPM_ENUM("Headphone Source Playback Route", + sun6i_codec_hp_src_enum), +}; + +/* microphone controls */ +static const char * const sun6i_codec_mic2_src_enum_text[] = { + "Mic2", "Mic3", +}; + +static SOC_ENUM_SINGLE_DECL(sun6i_codec_mic2_src_enum, + SUN6I_CODEC_MIC_CTRL, + SUN6I_CODEC_MIC_CTRL_MIC2SLT, + sun6i_codec_mic2_src_enum_text); + +static const struct snd_kcontrol_new sun6i_codec_mic2_src[] = { + SOC_DAPM_ENUM("Mic2 Amplifier Source Route", + sun6i_codec_mic2_src_enum), +}; + +/* line out controls */ +static const char * const sun6i_codec_lineout_src_enum_text[] = { + "Stereo", "Mono Differential", +}; + +static SOC_ENUM_DOUBLE_DECL(sun6i_codec_lineout_src_enum, + SUN6I_CODEC_MIC_CTRL, + SUN6I_CODEC_MIC_CTRL_LINEOUTLSRC, + SUN6I_CODEC_MIC_CTRL_LINEOUTRSRC, + sun6i_codec_lineout_src_enum_text); + +static const struct snd_kcontrol_new sun6i_codec_lineout_src[] = { + SOC_DAPM_ENUM("Line Out Source Playback Route", + sun6i_codec_lineout_src_enum), +}; + +/* volume / mute controls */ +static const DECLARE_TLV_DB_SCALE(sun6i_codec_dvol_scale, -7308, 116, 0); +static const DECLARE_TLV_DB_SCALE(sun6i_codec_hp_vol_scale, -6300, 100, 1); +static const DECLARE_TLV_DB_SCALE(sun6i_codec_out_mixer_pregain_scale, + -450, 150, 0); +static const DECLARE_TLV_DB_RANGE(sun6i_codec_lineout_vol_scale, + 0, 1, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + 2, 31, TLV_DB_SCALE_ITEM(-4350, 150, 0), +); +static const DECLARE_TLV_DB_RANGE(sun6i_codec_mic_gain_scale, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 7, TLV_DB_SCALE_ITEM(2400, 300, 0), +); + +static const struct snd_kcontrol_new sun6i_codec_codec_widgets[] = { + SOC_SINGLE_TLV("DAC Playback Volume", SUN4I_CODEC_DAC_DPC, + SUN4I_CODEC_DAC_DPC_DVOL, 0x3f, 1, + sun6i_codec_dvol_scale), + SOC_SINGLE_TLV("Headphone Playback Volume", + SUN6I_CODEC_OM_DACA_CTRL, + SUN6I_CODEC_OM_DACA_CTRL_HPVOL, 0x3f, 0, + sun6i_codec_hp_vol_scale), + SOC_SINGLE_TLV("Line Out Playback Volume", + SUN6I_CODEC_MIC_CTRL, + SUN6I_CODEC_MIC_CTRL_LINEOUTVC, 0x1f, 0, + sun6i_codec_lineout_vol_scale), + SOC_DOUBLE("Headphone Playback Switch", + SUN6I_CODEC_OM_DACA_CTRL, + SUN6I_CODEC_OM_DACA_CTRL_LHPPAMUTE, + SUN6I_CODEC_OM_DACA_CTRL_RHPPAMUTE, 1, 0), + SOC_DOUBLE("Line Out Playback Switch", + SUN6I_CODEC_MIC_CTRL, + SUN6I_CODEC_MIC_CTRL_LINEOUTLEN, + SUN6I_CODEC_MIC_CTRL_LINEOUTREN, 1, 0), + /* Mixer pre-gains */ + SOC_SINGLE_TLV("Line In Playback Volume", + SUN6I_CODEC_OM_PA_CTRL, SUN6I_CODEC_OM_PA_CTRL_LINEING, + 0x7, 0, sun6i_codec_out_mixer_pregain_scale), + SOC_SINGLE_TLV("Mic1 Playback Volume", + SUN6I_CODEC_OM_PA_CTRL, SUN6I_CODEC_OM_PA_CTRL_MIC1G, + 0x7, 0, sun6i_codec_out_mixer_pregain_scale), + SOC_SINGLE_TLV("Mic2 Playback Volume", + SUN6I_CODEC_OM_PA_CTRL, SUN6I_CODEC_OM_PA_CTRL_MIC2G, + 0x7, 0, sun6i_codec_out_mixer_pregain_scale), + + /* Microphone Amp boost gains */ + SOC_SINGLE_TLV("Mic1 Boost Volume", SUN6I_CODEC_MIC_CTRL, + SUN6I_CODEC_MIC_CTRL_MIC1BOOST, 0x7, 0, + sun6i_codec_mic_gain_scale), + SOC_SINGLE_TLV("Mic2 Boost Volume", SUN6I_CODEC_MIC_CTRL, + SUN6I_CODEC_MIC_CTRL_MIC2BOOST, 0x7, 0, + sun6i_codec_mic_gain_scale), + SOC_DOUBLE_TLV("ADC Capture Volume", + SUN6I_CODEC_ADC_ACTL, SUN6I_CODEC_ADC_ACTL_ADCLG, + SUN6I_CODEC_ADC_ACTL_ADCRG, 0x7, 0, + sun6i_codec_out_mixer_pregain_scale), +}; + +static const struct snd_soc_dapm_widget sun6i_codec_codec_dapm_widgets[] = { + /* Microphone inputs */ + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("MIC3"), + + /* Microphone Bias */ + SND_SOC_DAPM_SUPPLY("HBIAS", SUN6I_CODEC_MIC_CTRL, + SUN6I_CODEC_MIC_CTRL_HBIASEN, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MBIAS", SUN6I_CODEC_MIC_CTRL, + SUN6I_CODEC_MIC_CTRL_MBIASEN, 0, NULL, 0), + + /* Mic input path */ + SND_SOC_DAPM_MUX("Mic2 Amplifier Source Route", + SND_SOC_NOPM, 0, 0, sun6i_codec_mic2_src), + SND_SOC_DAPM_PGA("Mic1 Amplifier", SUN6I_CODEC_MIC_CTRL, + SUN6I_CODEC_MIC_CTRL_MIC1AMPEN, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mic2 Amplifier", SUN6I_CODEC_MIC_CTRL, + SUN6I_CODEC_MIC_CTRL_MIC2AMPEN, 0, NULL, 0), + + /* Line In */ + SND_SOC_DAPM_INPUT("LINEIN"), + + /* Digital parts of the ADCs */ + SND_SOC_DAPM_SUPPLY("ADC Enable", SUN6I_CODEC_ADC_FIFOC, + SUN6I_CODEC_ADC_FIFOC_EN_AD, 0, + NULL, 0), + + /* Analog parts of the ADCs */ + SND_SOC_DAPM_ADC("Left ADC", "Codec Capture", SUN6I_CODEC_ADC_ACTL, + SUN6I_CODEC_ADC_ACTL_ADCLEN, 0), + SND_SOC_DAPM_ADC("Right ADC", "Codec Capture", SUN6I_CODEC_ADC_ACTL, + SUN6I_CODEC_ADC_ACTL_ADCREN, 0), + + /* ADC Mixers */ + SOC_MIXER_ARRAY("Left ADC Mixer", SND_SOC_NOPM, 0, 0, + sun6i_codec_adc_mixer_controls), + SOC_MIXER_ARRAY("Right ADC Mixer", SND_SOC_NOPM, 0, 0, + sun6i_codec_adc_mixer_controls), + + /* Digital parts of the DACs */ + SND_SOC_DAPM_SUPPLY("DAC Enable", SUN4I_CODEC_DAC_DPC, + SUN4I_CODEC_DAC_DPC_EN_DA, 0, + NULL, 0), + + /* Analog parts of the DACs */ + SND_SOC_DAPM_DAC("Left DAC", "Codec Playback", + SUN6I_CODEC_OM_DACA_CTRL, + SUN6I_CODEC_OM_DACA_CTRL_DACALEN, 0), + SND_SOC_DAPM_DAC("Right DAC", "Codec Playback", + SUN6I_CODEC_OM_DACA_CTRL, + SUN6I_CODEC_OM_DACA_CTRL_DACAREN, 0), + + /* Mixers */ + SOC_MIXER_ARRAY("Left Mixer", SUN6I_CODEC_OM_DACA_CTRL, + SUN6I_CODEC_OM_DACA_CTRL_LMIXEN, 0, + sun6i_codec_mixer_controls), + SOC_MIXER_ARRAY("Right Mixer", SUN6I_CODEC_OM_DACA_CTRL, + SUN6I_CODEC_OM_DACA_CTRL_RMIXEN, 0, + sun6i_codec_mixer_controls), + + /* Headphone output path */ + SND_SOC_DAPM_MUX("Headphone Source Playback Route", + SND_SOC_NOPM, 0, 0, sun6i_codec_hp_src), + SND_SOC_DAPM_OUT_DRV("Headphone Amp", SUN6I_CODEC_OM_PA_CTRL, + SUN6I_CODEC_OM_PA_CTRL_HPPAEN, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HPCOM Protection", SUN6I_CODEC_OM_PA_CTRL, + SUN6I_CODEC_OM_PA_CTRL_COMPTEN, 0, NULL, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPCOM", SUN6I_CODEC_OM_PA_CTRL, + SUN6I_CODEC_OM_PA_CTRL_HPCOM_CTL, 0x3, 0x3, 0), + SND_SOC_DAPM_OUTPUT("HP"), + + /* Line Out path */ + SND_SOC_DAPM_MUX("Line Out Source Playback Route", + SND_SOC_NOPM, 0, 0, sun6i_codec_lineout_src), + SND_SOC_DAPM_OUTPUT("LINEOUT"), +}; + +static const struct snd_soc_dapm_route sun6i_codec_codec_dapm_routes[] = { + /* DAC Routes */ + { "Left DAC", NULL, "DAC Enable" }, + { "Right DAC", NULL, "DAC Enable" }, + + /* Microphone Routes */ + { "Mic1 Amplifier", NULL, "MIC1"}, + { "Mic2 Amplifier Source Route", "Mic2", "MIC2" }, + { "Mic2 Amplifier Source Route", "Mic3", "MIC3" }, + { "Mic2 Amplifier", NULL, "Mic2 Amplifier Source Route"}, + + /* Left Mixer Routes */ + { "Left Mixer", "DAC Playback Switch", "Left DAC" }, + { "Left Mixer", "DAC Reversed Playback Switch", "Right DAC" }, + { "Left Mixer", "Line In Playback Switch", "LINEIN" }, + { "Left Mixer", "Mic1 Playback Switch", "Mic1 Amplifier" }, + { "Left Mixer", "Mic2 Playback Switch", "Mic2 Amplifier" }, + + /* Right Mixer Routes */ + { "Right Mixer", "DAC Playback Switch", "Right DAC" }, + { "Right Mixer", "DAC Reversed Playback Switch", "Left DAC" }, + { "Right Mixer", "Line In Playback Switch", "LINEIN" }, + { "Right Mixer", "Mic1 Playback Switch", "Mic1 Amplifier" }, + { "Right Mixer", "Mic2 Playback Switch", "Mic2 Amplifier" }, + + /* Left ADC Mixer Routes */ + { "Left ADC Mixer", "Mixer Capture Switch", "Left Mixer" }, + { "Left ADC Mixer", "Mixer Reversed Capture Switch", "Right Mixer" }, + { "Left ADC Mixer", "Line In Capture Switch", "LINEIN" }, + { "Left ADC Mixer", "Mic1 Capture Switch", "Mic1 Amplifier" }, + { "Left ADC Mixer", "Mic2 Capture Switch", "Mic2 Amplifier" }, + + /* Right ADC Mixer Routes */ + { "Right ADC Mixer", "Mixer Capture Switch", "Right Mixer" }, + { "Right ADC Mixer", "Mixer Reversed Capture Switch", "Left Mixer" }, + { "Right ADC Mixer", "Line In Capture Switch", "LINEIN" }, + { "Right ADC Mixer", "Mic1 Capture Switch", "Mic1 Amplifier" }, + { "Right ADC Mixer", "Mic2 Capture Switch", "Mic2 Amplifier" }, + + /* Headphone Routes */ + { "Headphone Source Playback Route", "DAC", "Left DAC" }, + { "Headphone Source Playback Route", "DAC", "Right DAC" }, + { "Headphone Source Playback Route", "Mixer", "Left Mixer" }, + { "Headphone Source Playback Route", "Mixer", "Right Mixer" }, + { "Headphone Amp", NULL, "Headphone Source Playback Route" }, + { "HP", NULL, "Headphone Amp" }, + { "HPCOM", NULL, "HPCOM Protection" }, + + /* Line Out Routes */ + { "Line Out Source Playback Route", "Stereo", "Left Mixer" }, + { "Line Out Source Playback Route", "Stereo", "Right Mixer" }, + { "Line Out Source Playback Route", "Mono Differential", "Left Mixer" }, + { "LINEOUT", NULL, "Line Out Source Playback Route" }, + + /* ADC Routes */ + { "Left ADC", NULL, "ADC Enable" }, + { "Right ADC", NULL, "ADC Enable" }, + { "Left ADC", NULL, "Left ADC Mixer" }, + { "Right ADC", NULL, "Right ADC Mixer" }, +}; + +static struct snd_soc_codec_driver sun6i_codec_codec = { + .component_driver = { + .controls = sun6i_codec_codec_widgets, + .num_controls = ARRAY_SIZE(sun6i_codec_codec_widgets), + .dapm_widgets = sun6i_codec_codec_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sun6i_codec_codec_dapm_widgets), + .dapm_routes = sun6i_codec_codec_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(sun6i_codec_codec_dapm_routes), + }, +}; + +/* sun8i A23 codec */ +static const struct snd_kcontrol_new sun8i_a23_codec_codec_controls[] = { + SOC_SINGLE_TLV("DAC Playback Volume", SUN4I_CODEC_DAC_DPC, + SUN4I_CODEC_DAC_DPC_DVOL, 0x3f, 1, + sun6i_codec_dvol_scale), +}; + +static const struct snd_soc_dapm_widget sun8i_a23_codec_codec_widgets[] = { + /* Digital parts of the ADCs */ + SND_SOC_DAPM_SUPPLY("ADC Enable", SUN6I_CODEC_ADC_FIFOC, + SUN6I_CODEC_ADC_FIFOC_EN_AD, 0, NULL, 0), + /* Digital parts of the DACs */ + SND_SOC_DAPM_SUPPLY("DAC Enable", SUN4I_CODEC_DAC_DPC, + SUN4I_CODEC_DAC_DPC_EN_DA, 0, NULL, 0), + +}; + +static struct snd_soc_codec_driver sun8i_a23_codec_codec = { + .component_driver = { + .controls = sun8i_a23_codec_codec_controls, + .num_controls = ARRAY_SIZE(sun8i_a23_codec_codec_controls), + .dapm_widgets = sun8i_a23_codec_codec_widgets, + .num_dapm_widgets = ARRAY_SIZE(sun8i_a23_codec_codec_widgets), + }, +}; + static const struct snd_soc_component_driver sun4i_codec_component = { .name = "sun4i-codec", }; @@ -678,45 +1144,6 @@ static struct snd_soc_dai_driver dummy_cpu_dai = { }, }; -static const struct regmap_config sun4i_codec_regmap_config = { - .reg_bits = 32, - .reg_stride = 4, - .val_bits = 32, - .max_register = SUN4I_CODEC_ADC_RXCNT, -}; - -static const struct regmap_config sun7i_codec_regmap_config = { - .reg_bits = 32, - .reg_stride = 4, - .val_bits = 32, - .max_register = SUN7I_CODEC_AC_MIC_PHONE_CAL, -}; - -struct sun4i_codec_quirks { - const struct regmap_config *regmap_config; -}; - -static const struct sun4i_codec_quirks sun4i_codec_quirks = { - .regmap_config = &sun4i_codec_regmap_config, -}; - -static const struct sun4i_codec_quirks sun7i_codec_quirks = { - .regmap_config = &sun7i_codec_regmap_config, -}; - -static const struct of_device_id sun4i_codec_of_match[] = { - { - .compatible = "allwinner,sun4i-a10-codec", - .data = &sun4i_codec_quirks, - }, - { - .compatible = "allwinner,sun7i-a20-codec", - .data = &sun7i_codec_quirks, - }, - {} -}; -MODULE_DEVICE_TABLE(of, sun4i_codec_of_match); - static struct snd_soc_dai_link *sun4i_codec_create_link(struct device *dev, int *num_links) { @@ -781,6 +1208,259 @@ static struct snd_soc_card *sun4i_codec_create_card(struct device *dev) return card; }; +static const struct snd_soc_dapm_widget sun6i_codec_card_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Mic", NULL), + SND_SOC_DAPM_SPK("Speaker", sun4i_codec_spk_event), +}; + +static struct snd_soc_card *sun6i_codec_create_card(struct device *dev) +{ + struct snd_soc_card *card; + int ret; + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return ERR_PTR(-ENOMEM); + + card->dai_link = sun4i_codec_create_link(dev, &card->num_links); + if (!card->dai_link) + return ERR_PTR(-ENOMEM); + + card->dev = dev; + card->name = "A31 Audio Codec"; + card->dapm_widgets = sun6i_codec_card_dapm_widgets; + card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); + card->fully_routed = true; + + ret = snd_soc_of_parse_audio_routing(card, "allwinner,audio-routing"); + if (ret) + dev_warn(dev, "failed to parse audio-routing: %d\n", ret); + + return card; +}; + +/* Connect digital side enables to analog side widgets */ +static const struct snd_soc_dapm_route sun8i_codec_card_routes[] = { + /* ADC Routes */ + { "Left ADC", NULL, "ADC Enable" }, + { "Right ADC", NULL, "ADC Enable" }, + { "Codec Capture", NULL, "Left ADC" }, + { "Codec Capture", NULL, "Right ADC" }, + + /* DAC Routes */ + { "Left DAC", NULL, "DAC Enable" }, + { "Right DAC", NULL, "DAC Enable" }, + { "Left DAC", NULL, "Codec Playback" }, + { "Right DAC", NULL, "Codec Playback" }, +}; + +static struct snd_soc_aux_dev aux_dev = { + .name = "Codec Analog Controls", +}; + +static struct snd_soc_card *sun8i_a23_codec_create_card(struct device *dev) +{ + struct snd_soc_card *card; + int ret; + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return ERR_PTR(-ENOMEM); + + aux_dev.codec_of_node = of_parse_phandle(dev->of_node, + "allwinner,codec-analog-controls", + 0); + if (!aux_dev.codec_of_node) { + dev_err(dev, "Can't find analog controls for codec.\n"); + return ERR_PTR(-EINVAL); + }; + + card->dai_link = sun4i_codec_create_link(dev, &card->num_links); + if (!card->dai_link) + return ERR_PTR(-ENOMEM); + + card->dev = dev; + card->name = "A23 Audio Codec"; + card->dapm_widgets = sun6i_codec_card_dapm_widgets; + card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); + card->dapm_routes = sun8i_codec_card_routes; + card->num_dapm_routes = ARRAY_SIZE(sun8i_codec_card_routes); + card->aux_dev = &aux_dev; + card->num_aux_devs = 1; + card->fully_routed = true; + + ret = snd_soc_of_parse_audio_routing(card, "allwinner,audio-routing"); + if (ret) + dev_warn(dev, "failed to parse audio-routing: %d\n", ret); + + return card; +}; + +static struct snd_soc_card *sun8i_h3_codec_create_card(struct device *dev) +{ + struct snd_soc_card *card; + int ret; + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return ERR_PTR(-ENOMEM); + + aux_dev.codec_of_node = of_parse_phandle(dev->of_node, + "allwinner,codec-analog-controls", + 0); + if (!aux_dev.codec_of_node) { + dev_err(dev, "Can't find analog controls for codec.\n"); + return ERR_PTR(-EINVAL); + }; + + card->dai_link = sun4i_codec_create_link(dev, &card->num_links); + if (!card->dai_link) + return ERR_PTR(-ENOMEM); + + card->dev = dev; + card->name = "H3 Audio Codec"; + card->dapm_widgets = sun6i_codec_card_dapm_widgets; + card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); + card->dapm_routes = sun8i_codec_card_routes; + card->num_dapm_routes = ARRAY_SIZE(sun8i_codec_card_routes); + card->aux_dev = &aux_dev; + card->num_aux_devs = 1; + card->fully_routed = true; + + ret = snd_soc_of_parse_audio_routing(card, "allwinner,audio-routing"); + if (ret) + dev_warn(dev, "failed to parse audio-routing: %d\n", ret); + + return card; +}; + +static const struct regmap_config sun4i_codec_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN4I_CODEC_ADC_RXCNT, +}; + +static const struct regmap_config sun6i_codec_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN6I_CODEC_HMIC_DATA, +}; + +static const struct regmap_config sun7i_codec_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN7I_CODEC_AC_MIC_PHONE_CAL, +}; + +static const struct regmap_config sun8i_a23_codec_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN8I_A23_CODEC_ADC_RXCNT, +}; + +static const struct regmap_config sun8i_h3_codec_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN8I_H3_CODEC_ADC_DBG, +}; + +struct sun4i_codec_quirks { + const struct regmap_config *regmap_config; + const struct snd_soc_codec_driver *codec; + struct snd_soc_card * (*create_card)(struct device *dev); + struct reg_field reg_adc_fifoc; /* used for regmap_field */ + unsigned int reg_dac_txdata; /* TX FIFO offset for DMA config */ + unsigned int reg_adc_rxdata; /* RX FIFO offset for DMA config */ + bool has_reset; +}; + +static const struct sun4i_codec_quirks sun4i_codec_quirks = { + .regmap_config = &sun4i_codec_regmap_config, + .codec = &sun4i_codec_codec, + .create_card = sun4i_codec_create_card, + .reg_adc_fifoc = REG_FIELD(SUN4I_CODEC_ADC_FIFOC, 0, 31), + .reg_dac_txdata = SUN4I_CODEC_DAC_TXDATA, + .reg_adc_rxdata = SUN4I_CODEC_ADC_RXDATA, +}; + +static const struct sun4i_codec_quirks sun6i_a31_codec_quirks = { + .regmap_config = &sun6i_codec_regmap_config, + .codec = &sun6i_codec_codec, + .create_card = sun6i_codec_create_card, + .reg_adc_fifoc = REG_FIELD(SUN6I_CODEC_ADC_FIFOC, 0, 31), + .reg_dac_txdata = SUN4I_CODEC_DAC_TXDATA, + .reg_adc_rxdata = SUN6I_CODEC_ADC_RXDATA, + .has_reset = true, +}; + +static const struct sun4i_codec_quirks sun7i_codec_quirks = { + .regmap_config = &sun7i_codec_regmap_config, + .codec = &sun4i_codec_codec, + .create_card = sun4i_codec_create_card, + .reg_adc_fifoc = REG_FIELD(SUN4I_CODEC_ADC_FIFOC, 0, 31), + .reg_dac_txdata = SUN4I_CODEC_DAC_TXDATA, + .reg_adc_rxdata = SUN4I_CODEC_ADC_RXDATA, +}; + +static const struct sun4i_codec_quirks sun8i_a23_codec_quirks = { + .regmap_config = &sun8i_a23_codec_regmap_config, + .codec = &sun8i_a23_codec_codec, + .create_card = sun8i_a23_codec_create_card, + .reg_adc_fifoc = REG_FIELD(SUN6I_CODEC_ADC_FIFOC, 0, 31), + .reg_dac_txdata = SUN4I_CODEC_DAC_TXDATA, + .reg_adc_rxdata = SUN6I_CODEC_ADC_RXDATA, + .has_reset = true, +}; + +static const struct sun4i_codec_quirks sun8i_h3_codec_quirks = { + .regmap_config = &sun8i_h3_codec_regmap_config, + /* + * TODO Share the codec structure with A23 for now. + * This should be split out when adding digital audio + * processing support for the H3. + */ + .codec = &sun8i_a23_codec_codec, + .create_card = sun8i_h3_codec_create_card, + .reg_adc_fifoc = REG_FIELD(SUN6I_CODEC_ADC_FIFOC, 0, 31), + .reg_dac_txdata = SUN8I_H3_CODEC_DAC_TXDATA, + .reg_adc_rxdata = SUN6I_CODEC_ADC_RXDATA, + .has_reset = true, +}; + +static const struct of_device_id sun4i_codec_of_match[] = { + { + .compatible = "allwinner,sun4i-a10-codec", + .data = &sun4i_codec_quirks, + }, + { + .compatible = "allwinner,sun6i-a31-codec", + .data = &sun6i_a31_codec_quirks, + }, + { + .compatible = "allwinner,sun7i-a20-codec", + .data = &sun7i_codec_quirks, + }, + { + .compatible = "allwinner,sun8i-a23-codec", + .data = &sun8i_a23_codec_quirks, + }, + { + .compatible = "allwinner,sun8i-h3-codec", + .data = &sun8i_h3_codec_quirks, + }, + {} +}; +MODULE_DEVICE_TABLE(of, sun4i_codec_of_match); + static int sun4i_codec_probe(struct platform_device *pdev) { struct snd_soc_card *card; @@ -829,6 +1509,14 @@ static int sun4i_codec_probe(struct platform_device *pdev) return PTR_ERR(scodec->clk_module); } + if (quirks->has_reset) { + scodec->rst = devm_reset_control_get(&pdev->dev, NULL); + if (IS_ERR(scodec->rst)) { + dev_err(&pdev->dev, "Failed to get reset control\n"); + return PTR_ERR(scodec->rst); + } + } + scodec->gpio_pa = devm_gpiod_get_optional(&pdev->dev, "allwinner,pa", GPIOD_OUT_LOW); if (IS_ERR(scodec->gpio_pa)) { @@ -838,27 +1526,48 @@ static int sun4i_codec_probe(struct platform_device *pdev) return ret; } + /* reg_field setup */ + scodec->reg_adc_fifoc = devm_regmap_field_alloc(&pdev->dev, + scodec->regmap, + quirks->reg_adc_fifoc); + if (IS_ERR(scodec->reg_adc_fifoc)) { + ret = PTR_ERR(scodec->reg_adc_fifoc); + dev_err(&pdev->dev, "Failed to create regmap fields: %d\n", + ret); + return ret; + } + /* Enable the bus clock */ if (clk_prepare_enable(scodec->clk_apb)) { dev_err(&pdev->dev, "Failed to enable the APB clock\n"); return -EINVAL; } + /* Deassert the reset control */ + if (scodec->rst) { + ret = reset_control_deassert(scodec->rst); + if (ret) { + dev_err(&pdev->dev, + "Failed to deassert the reset control\n"); + goto err_clk_disable; + } + } + /* DMA configuration for TX FIFO */ - scodec->playback_dma_data.addr = res->start + SUN4I_CODEC_DAC_TXDATA; - scodec->playback_dma_data.maxburst = 4; + scodec->playback_dma_data.addr = res->start + quirks->reg_dac_txdata; + scodec->playback_dma_data.maxburst = 8; scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; /* DMA configuration for RX FIFO */ - scodec->capture_dma_data.addr = res->start + SUN4I_CODEC_ADC_RXDATA; - scodec->capture_dma_data.maxburst = 4; + scodec->capture_dma_data.addr = res->start + quirks->reg_adc_rxdata; + scodec->capture_dma_data.maxburst = 8; scodec->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; - ret = snd_soc_register_codec(&pdev->dev, &sun4i_codec_codec, + ret = snd_soc_register_codec(&pdev->dev, quirks->codec, &sun4i_codec_dai, 1); if (ret) { dev_err(&pdev->dev, "Failed to register our codec\n"); - goto err_clk_disable; + goto err_assert_reset; } ret = devm_snd_soc_register_component(&pdev->dev, @@ -875,7 +1584,7 @@ static int sun4i_codec_probe(struct platform_device *pdev) goto err_unregister_codec; } - card = sun4i_codec_create_card(&pdev->dev); + card = quirks->create_card(&pdev->dev); if (IS_ERR(card)) { ret = PTR_ERR(card); dev_err(&pdev->dev, "Failed to create our card\n"); @@ -895,6 +1604,9 @@ static int sun4i_codec_probe(struct platform_device *pdev) err_unregister_codec: snd_soc_unregister_codec(&pdev->dev); +err_assert_reset: + if (scodec->rst) + reset_control_assert(scodec->rst); err_clk_disable: clk_disable_unprepare(scodec->clk_apb); return ret; @@ -907,6 +1619,8 @@ static int sun4i_codec_remove(struct platform_device *pdev) snd_soc_unregister_card(card); snd_soc_unregister_codec(&pdev->dev); + if (scodec->rst) + reset_control_assert(scodec->rst); clk_disable_unprepare(scodec->clk_apb); return 0; @@ -926,4 +1640,5 @@ MODULE_DESCRIPTION("Allwinner A10 codec driver"); MODULE_AUTHOR("Emilio López "); MODULE_AUTHOR("Jon Smirl "); MODULE_AUTHOR("Maxime Ripard "); +MODULE_AUTHOR("Chen-Yu Tsai "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index 687a8f83dbe5..f24d19526603 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -93,6 +93,9 @@ struct sun4i_i2s { struct clk *mod_clk; struct regmap *regmap; + unsigned int mclk_freq; + + struct snd_dmaengine_dai_dma_data capture_dma_data; struct snd_dmaengine_dai_dma_data playback_dma_data; }; @@ -157,14 +160,24 @@ static int sun4i_i2s_get_mclk_div(struct sun4i_i2s *i2s, } static int sun4i_i2s_oversample_rates[] = { 128, 192, 256, 384, 512, 768 }; +static bool sun4i_i2s_oversample_is_valid(unsigned int oversample) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(sun4i_i2s_oversample_rates); i++) + if (sun4i_i2s_oversample_rates[i] == oversample) + return true; + + return false; +} static int sun4i_i2s_set_clk_rate(struct sun4i_i2s *i2s, unsigned int rate, unsigned int word_size) { - unsigned int clk_rate; + unsigned int oversample_rate, clk_rate; int bclk_div, mclk_div; - int ret, i; + int ret; switch (rate) { case 176400: @@ -196,21 +209,18 @@ static int sun4i_i2s_set_clk_rate(struct sun4i_i2s *i2s, if (ret) return ret; - /* Always favor the highest oversampling rate */ - for (i = (ARRAY_SIZE(sun4i_i2s_oversample_rates) - 1); i >= 0; i--) { - unsigned int oversample_rate = sun4i_i2s_oversample_rates[i]; + oversample_rate = i2s->mclk_freq / rate; + if (!sun4i_i2s_oversample_is_valid(oversample_rate)) + return -EINVAL; - bclk_div = sun4i_i2s_get_bclk_div(i2s, oversample_rate, - word_size); - mclk_div = sun4i_i2s_get_mclk_div(i2s, oversample_rate, - clk_rate, - rate); + bclk_div = sun4i_i2s_get_bclk_div(i2s, oversample_rate, + word_size); + if (bclk_div < 0) + return -EINVAL; - if ((bclk_div >= 0) && (mclk_div >= 0)) - break; - } - - if ((bclk_div < 0) || (mclk_div < 0)) + mclk_div = sun4i_i2s_get_mclk_div(i2s, oversample_rate, + clk_rate, rate); + if (mclk_div < 0) return -EINVAL; regmap_write(i2s->regmap, SUN4I_I2S_CLK_DIV_REG, @@ -341,6 +351,27 @@ static int sun4i_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } +static void sun4i_i2s_start_capture(struct sun4i_i2s *i2s) +{ + /* Flush RX FIFO */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_FIFO_CTRL_REG, + SUN4I_I2S_FIFO_CTRL_FLUSH_RX, + SUN4I_I2S_FIFO_CTRL_FLUSH_RX); + + /* Clear RX counter */ + regmap_write(i2s->regmap, SUN4I_I2S_RX_CNT_REG, 0); + + /* Enable RX Block */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_RX_EN, + SUN4I_I2S_CTRL_RX_EN); + + /* Enable RX DRQ */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_DMA_INT_CTRL_REG, + SUN4I_I2S_DMA_INT_CTRL_RX_DRQ_EN, + SUN4I_I2S_DMA_INT_CTRL_RX_DRQ_EN); +} + static void sun4i_i2s_start_playback(struct sun4i_i2s *i2s) { /* Flush TX FIFO */ @@ -362,6 +393,18 @@ static void sun4i_i2s_start_playback(struct sun4i_i2s *i2s) SUN4I_I2S_DMA_INT_CTRL_TX_DRQ_EN); } +static void sun4i_i2s_stop_capture(struct sun4i_i2s *i2s) +{ + /* Disable RX Block */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_RX_EN, + 0); + + /* Disable RX DRQ */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_DMA_INT_CTRL_REG, + SUN4I_I2S_DMA_INT_CTRL_RX_DRQ_EN, + 0); +} static void sun4i_i2s_stop_playback(struct sun4i_i2s *i2s) { @@ -388,7 +431,7 @@ static int sun4i_i2s_trigger(struct snd_pcm_substream *substream, int cmd, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) sun4i_i2s_start_playback(i2s); else - return -EINVAL; + sun4i_i2s_start_capture(i2s); break; case SNDRV_PCM_TRIGGER_STOP: @@ -397,7 +440,7 @@ static int sun4i_i2s_trigger(struct snd_pcm_substream *substream, int cmd, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) sun4i_i2s_stop_playback(i2s); else - return -EINVAL; + sun4i_i2s_stop_capture(i2s); break; default: @@ -447,9 +490,23 @@ static void sun4i_i2s_shutdown(struct snd_pcm_substream *substream, regmap_write(i2s->regmap, SUN4I_I2S_CTRL_REG, 0); } +static int sun4i_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + if (clk_id != 0) + return -EINVAL; + + i2s->mclk_freq = freq; + + return 0; +} + static const struct snd_soc_dai_ops sun4i_i2s_dai_ops = { .hw_params = sun4i_i2s_hw_params, .set_fmt = sun4i_i2s_set_fmt, + .set_sysclk = sun4i_i2s_set_sysclk, .shutdown = sun4i_i2s_shutdown, .startup = sun4i_i2s_startup, .trigger = sun4i_i2s_trigger, @@ -459,7 +516,9 @@ static int sun4i_i2s_dai_probe(struct snd_soc_dai *dai) { struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai); - snd_soc_dai_init_dma_data(dai, &i2s->playback_dma_data, NULL); + snd_soc_dai_init_dma_data(dai, + &i2s->playback_dma_data, + &i2s->capture_dma_data); snd_soc_dai_set_drvdata(dai, i2s); @@ -468,6 +527,13 @@ static int sun4i_i2s_dai_probe(struct snd_soc_dai *dai) static struct snd_soc_dai_driver sun4i_i2s_dai = { .probe = sun4i_i2s_dai_probe, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, .playback = { .stream_name = "Playback", .channels_min = 2, @@ -630,6 +696,9 @@ static int sun4i_i2s_probe(struct platform_device *pdev) i2s->playback_dma_data.addr = res->start + SUN4I_I2S_FIFO_TX_REG; i2s->playback_dma_data.maxburst = 4; + i2s->capture_dma_data.addr = res->start + SUN4I_I2S_FIFO_RX_REG; + i2s->capture_dma_data.maxburst = 4; + pm_runtime_enable(&pdev->dev); if (!pm_runtime_enabled(&pdev->dev)) { ret = sun4i_i2s_runtime_resume(&pdev->dev); diff --git a/sound/soc/sunxi/sun8i-codec-analog.c b/sound/soc/sunxi/sun8i-codec-analog.c new file mode 100644 index 000000000000..af02290ebe49 --- /dev/null +++ b/sound/soc/sunxi/sun8i-codec-analog.c @@ -0,0 +1,665 @@ +/* + * This driver supports the analog controls for the internal codec + * found in Allwinner's A31s, A23, A33 and H3 SoCs. + * + * Copyright 2016 Chen-Yu Tsai + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include + +/* Codec analog control register offsets and bit fields */ +#define SUN8I_ADDA_HP_VOLC 0x00 +#define SUN8I_ADDA_HP_VOLC_PA_CLK_GATE 7 +#define SUN8I_ADDA_HP_VOLC_HP_VOL 0 +#define SUN8I_ADDA_LOMIXSC 0x01 +#define SUN8I_ADDA_LOMIXSC_MIC1 6 +#define SUN8I_ADDA_LOMIXSC_MIC2 5 +#define SUN8I_ADDA_LOMIXSC_PHONE 4 +#define SUN8I_ADDA_LOMIXSC_PHONEN 3 +#define SUN8I_ADDA_LOMIXSC_LINEINL 2 +#define SUN8I_ADDA_LOMIXSC_DACL 1 +#define SUN8I_ADDA_LOMIXSC_DACR 0 +#define SUN8I_ADDA_ROMIXSC 0x02 +#define SUN8I_ADDA_ROMIXSC_MIC1 6 +#define SUN8I_ADDA_ROMIXSC_MIC2 5 +#define SUN8I_ADDA_ROMIXSC_PHONE 4 +#define SUN8I_ADDA_ROMIXSC_PHONEP 3 +#define SUN8I_ADDA_ROMIXSC_LINEINR 2 +#define SUN8I_ADDA_ROMIXSC_DACR 1 +#define SUN8I_ADDA_ROMIXSC_DACL 0 +#define SUN8I_ADDA_DAC_PA_SRC 0x03 +#define SUN8I_ADDA_DAC_PA_SRC_DACAREN 7 +#define SUN8I_ADDA_DAC_PA_SRC_DACALEN 6 +#define SUN8I_ADDA_DAC_PA_SRC_RMIXEN 5 +#define SUN8I_ADDA_DAC_PA_SRC_LMIXEN 4 +#define SUN8I_ADDA_DAC_PA_SRC_RHPPAMUTE 3 +#define SUN8I_ADDA_DAC_PA_SRC_LHPPAMUTE 2 +#define SUN8I_ADDA_DAC_PA_SRC_RHPIS 1 +#define SUN8I_ADDA_DAC_PA_SRC_LHPIS 0 +#define SUN8I_ADDA_PHONEIN_GCTRL 0x04 +#define SUN8I_ADDA_PHONEIN_GCTRL_PHONEPG 4 +#define SUN8I_ADDA_PHONEIN_GCTRL_PHONENG 0 +#define SUN8I_ADDA_LINEIN_GCTRL 0x05 +#define SUN8I_ADDA_LINEIN_GCTRL_LINEING 4 +#define SUN8I_ADDA_LINEIN_GCTRL_PHONEG 0 +#define SUN8I_ADDA_MICIN_GCTRL 0x06 +#define SUN8I_ADDA_MICIN_GCTRL_MIC1G 4 +#define SUN8I_ADDA_MICIN_GCTRL_MIC2G 0 +#define SUN8I_ADDA_PAEN_HP_CTRL 0x07 +#define SUN8I_ADDA_PAEN_HP_CTRL_HPPAEN 7 +#define SUN8I_ADDA_PAEN_HP_CTRL_LINEOUTEN 7 /* H3 specific */ +#define SUN8I_ADDA_PAEN_HP_CTRL_HPCOM_FC 5 +#define SUN8I_ADDA_PAEN_HP_CTRL_COMPTEN 4 +#define SUN8I_ADDA_PAEN_HP_CTRL_PA_ANTI_POP_CTRL 2 +#define SUN8I_ADDA_PAEN_HP_CTRL_LTRNMUTE 1 +#define SUN8I_ADDA_PAEN_HP_CTRL_RTLNMUTE 0 +#define SUN8I_ADDA_PHONEOUT_CTRL 0x08 +#define SUN8I_ADDA_PHONEOUT_CTRL_PHONEOUTG 5 +#define SUN8I_ADDA_PHONEOUT_CTRL_PHONEOUTEN 4 +#define SUN8I_ADDA_PHONEOUT_CTRL_PHONEOUT_MIC1 3 +#define SUN8I_ADDA_PHONEOUT_CTRL_PHONEOUT_MIC2 2 +#define SUN8I_ADDA_PHONEOUT_CTRL_PHONEOUT_RMIX 1 +#define SUN8I_ADDA_PHONEOUT_CTRL_PHONEOUT_LMIX 0 +#define SUN8I_ADDA_PHONE_GAIN_CTRL 0x09 +#define SUN8I_ADDA_PHONE_GAIN_CTRL_LINEOUT_VOL 3 +#define SUN8I_ADDA_PHONE_GAIN_CTRL_PHONEPREG 0 +#define SUN8I_ADDA_MIC2G_CTRL 0x0a +#define SUN8I_ADDA_MIC2G_CTRL_MIC2AMPEN 7 +#define SUN8I_ADDA_MIC2G_CTRL_MIC2BOOST 4 +#define SUN8I_ADDA_MIC2G_CTRL_LINEOUTLEN 3 +#define SUN8I_ADDA_MIC2G_CTRL_LINEOUTREN 2 +#define SUN8I_ADDA_MIC2G_CTRL_LINEOUTLSRC 1 +#define SUN8I_ADDA_MIC2G_CTRL_LINEOUTRSRC 0 +#define SUN8I_ADDA_MIC1G_MICBIAS_CTRL 0x0b +#define SUN8I_ADDA_MIC1G_MICBIAS_CTRL_HMICBIASEN 7 +#define SUN8I_ADDA_MIC1G_MICBIAS_CTRL_MMICBIASEN 6 +#define SUN8I_ADDA_MIC1G_MICBIAS_CTRL_HMICBIAS_MODE 5 +#define SUN8I_ADDA_MIC1G_MICBIAS_CTRL_MIC1AMPEN 3 +#define SUN8I_ADDA_MIC1G_MICBIAS_CTRL_MIC1BOOST 0 +#define SUN8I_ADDA_LADCMIXSC 0x0c +#define SUN8I_ADDA_LADCMIXSC_MIC1 6 +#define SUN8I_ADDA_LADCMIXSC_MIC2 5 +#define SUN8I_ADDA_LADCMIXSC_PHONE 4 +#define SUN8I_ADDA_LADCMIXSC_PHONEN 3 +#define SUN8I_ADDA_LADCMIXSC_LINEINL 2 +#define SUN8I_ADDA_LADCMIXSC_OMIXRL 1 +#define SUN8I_ADDA_LADCMIXSC_OMIXRR 0 +#define SUN8I_ADDA_RADCMIXSC 0x0d +#define SUN8I_ADDA_RADCMIXSC_MIC1 6 +#define SUN8I_ADDA_RADCMIXSC_MIC2 5 +#define SUN8I_ADDA_RADCMIXSC_PHONE 4 +#define SUN8I_ADDA_RADCMIXSC_PHONEP 3 +#define SUN8I_ADDA_RADCMIXSC_LINEINR 2 +#define SUN8I_ADDA_RADCMIXSC_OMIXR 1 +#define SUN8I_ADDA_RADCMIXSC_OMIXL 0 +#define SUN8I_ADDA_RES 0x0e +#define SUN8I_ADDA_RES_MMICBIAS_SEL 4 +#define SUN8I_ADDA_RES_PA_ANTI_POP_CTRL 0 +#define SUN8I_ADDA_ADC_AP_EN 0x0f +#define SUN8I_ADDA_ADC_AP_EN_ADCREN 7 +#define SUN8I_ADDA_ADC_AP_EN_ADCLEN 6 +#define SUN8I_ADDA_ADC_AP_EN_ADCG 0 + +/* Analog control register access bits */ +#define ADDA_PR 0x0 /* PRCM base + 0x1c0 */ +#define ADDA_PR_RESET BIT(28) +#define ADDA_PR_WRITE BIT(24) +#define ADDA_PR_ADDR_SHIFT 16 +#define ADDA_PR_ADDR_MASK GENMASK(4, 0) +#define ADDA_PR_DATA_IN_SHIFT 8 +#define ADDA_PR_DATA_IN_MASK GENMASK(7, 0) +#define ADDA_PR_DATA_OUT_SHIFT 0 +#define ADDA_PR_DATA_OUT_MASK GENMASK(7, 0) + +/* regmap access bits */ +static int adda_reg_read(void *context, unsigned int reg, unsigned int *val) +{ + void __iomem *base = (void __iomem *)context; + u32 tmp; + + /* De-assert reset */ + writel(readl(base) | ADDA_PR_RESET, base); + + /* Clear write bit */ + writel(readl(base) & ~ADDA_PR_WRITE, base); + + /* Set register address */ + tmp = readl(base); + tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT); + tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT; + writel(tmp, base); + + /* Read back value */ + *val = readl(base) & ADDA_PR_DATA_OUT_MASK; + + return 0; +} + +static int adda_reg_write(void *context, unsigned int reg, unsigned int val) +{ + void __iomem *base = (void __iomem *)context; + u32 tmp; + + /* De-assert reset */ + writel(readl(base) | ADDA_PR_RESET, base); + + /* Set register address */ + tmp = readl(base); + tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT); + tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT; + writel(tmp, base); + + /* Set data to write */ + tmp = readl(base); + tmp &= ~(ADDA_PR_DATA_IN_MASK << ADDA_PR_DATA_IN_SHIFT); + tmp |= (val & ADDA_PR_DATA_IN_MASK) << ADDA_PR_DATA_IN_SHIFT; + writel(tmp, base); + + /* Set write bit to signal a write */ + writel(readl(base) | ADDA_PR_WRITE, base); + + /* Clear write bit */ + writel(readl(base) & ~ADDA_PR_WRITE, base); + + return 0; +} + +static const struct regmap_config adda_pr_regmap_cfg = { + .name = "adda-pr", + .reg_bits = 5, + .reg_stride = 1, + .val_bits = 8, + .reg_read = adda_reg_read, + .reg_write = adda_reg_write, + .fast_io = true, + .max_register = 24, +}; + +/* mixer controls */ +static const struct snd_kcontrol_new sun8i_codec_mixer_controls[] = { + SOC_DAPM_DOUBLE_R("DAC Playback Switch", + SUN8I_ADDA_LOMIXSC, + SUN8I_ADDA_ROMIXSC, + SUN8I_ADDA_LOMIXSC_DACL, 1, 0), + SOC_DAPM_DOUBLE_R("DAC Reversed Playback Switch", + SUN8I_ADDA_LOMIXSC, + SUN8I_ADDA_ROMIXSC, + SUN8I_ADDA_LOMIXSC_DACR, 1, 0), + SOC_DAPM_DOUBLE_R("Line In Playback Switch", + SUN8I_ADDA_LOMIXSC, + SUN8I_ADDA_ROMIXSC, + SUN8I_ADDA_LOMIXSC_LINEINL, 1, 0), + SOC_DAPM_DOUBLE_R("Mic1 Playback Switch", + SUN8I_ADDA_LOMIXSC, + SUN8I_ADDA_ROMIXSC, + SUN8I_ADDA_LOMIXSC_MIC1, 1, 0), + SOC_DAPM_DOUBLE_R("Mic2 Playback Switch", + SUN8I_ADDA_LOMIXSC, + SUN8I_ADDA_ROMIXSC, + SUN8I_ADDA_LOMIXSC_MIC2, 1, 0), +}; + +/* ADC mixer controls */ +static const struct snd_kcontrol_new sun8i_codec_adc_mixer_controls[] = { + SOC_DAPM_DOUBLE_R("Mixer Capture Switch", + SUN8I_ADDA_LADCMIXSC, + SUN8I_ADDA_RADCMIXSC, + SUN8I_ADDA_LADCMIXSC_OMIXRL, 1, 0), + SOC_DAPM_DOUBLE_R("Mixer Reversed Capture Switch", + SUN8I_ADDA_LADCMIXSC, + SUN8I_ADDA_RADCMIXSC, + SUN8I_ADDA_LADCMIXSC_OMIXRR, 1, 0), + SOC_DAPM_DOUBLE_R("Line In Capture Switch", + SUN8I_ADDA_LADCMIXSC, + SUN8I_ADDA_RADCMIXSC, + SUN8I_ADDA_LADCMIXSC_LINEINL, 1, 0), + SOC_DAPM_DOUBLE_R("Mic1 Capture Switch", + SUN8I_ADDA_LADCMIXSC, + SUN8I_ADDA_RADCMIXSC, + SUN8I_ADDA_LADCMIXSC_MIC1, 1, 0), + SOC_DAPM_DOUBLE_R("Mic2 Capture Switch", + SUN8I_ADDA_LADCMIXSC, + SUN8I_ADDA_RADCMIXSC, + SUN8I_ADDA_LADCMIXSC_MIC2, 1, 0), +}; + +/* volume / mute controls */ +static const DECLARE_TLV_DB_SCALE(sun8i_codec_out_mixer_pregain_scale, + -450, 150, 0); +static const DECLARE_TLV_DB_RANGE(sun8i_codec_mic_gain_scale, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 7, TLV_DB_SCALE_ITEM(2400, 300, 0), +); + +static const struct snd_kcontrol_new sun8i_codec_common_controls[] = { + /* Mixer pre-gains */ + SOC_SINGLE_TLV("Line In Playback Volume", SUN8I_ADDA_LINEIN_GCTRL, + SUN8I_ADDA_LINEIN_GCTRL_LINEING, + 0x7, 0, sun8i_codec_out_mixer_pregain_scale), + SOC_SINGLE_TLV("Mic1 Playback Volume", SUN8I_ADDA_MICIN_GCTRL, + SUN8I_ADDA_MICIN_GCTRL_MIC1G, + 0x7, 0, sun8i_codec_out_mixer_pregain_scale), + SOC_SINGLE_TLV("Mic2 Playback Volume", + SUN8I_ADDA_MICIN_GCTRL, SUN8I_ADDA_MICIN_GCTRL_MIC2G, + 0x7, 0, sun8i_codec_out_mixer_pregain_scale), + + /* Microphone Amp boost gains */ + SOC_SINGLE_TLV("Mic1 Boost Volume", SUN8I_ADDA_MIC1G_MICBIAS_CTRL, + SUN8I_ADDA_MIC1G_MICBIAS_CTRL_MIC1BOOST, 0x7, 0, + sun8i_codec_mic_gain_scale), + SOC_SINGLE_TLV("Mic2 Boost Volume", SUN8I_ADDA_MIC2G_CTRL, + SUN8I_ADDA_MIC2G_CTRL_MIC2BOOST, 0x7, 0, + sun8i_codec_mic_gain_scale), + + /* ADC */ + SOC_SINGLE_TLV("ADC Gain Capture Volume", SUN8I_ADDA_ADC_AP_EN, + SUN8I_ADDA_ADC_AP_EN_ADCG, 0x7, 0, + sun8i_codec_out_mixer_pregain_scale), +}; + +static const struct snd_soc_dapm_widget sun8i_codec_common_widgets[] = { + /* ADC */ + SND_SOC_DAPM_ADC("Left ADC", NULL, SUN8I_ADDA_ADC_AP_EN, + SUN8I_ADDA_ADC_AP_EN_ADCLEN, 0), + SND_SOC_DAPM_ADC("Right ADC", NULL, SUN8I_ADDA_ADC_AP_EN, + SUN8I_ADDA_ADC_AP_EN_ADCREN, 0), + + /* DAC */ + SND_SOC_DAPM_DAC("Left DAC", NULL, SUN8I_ADDA_DAC_PA_SRC, + SUN8I_ADDA_DAC_PA_SRC_DACALEN, 0), + SND_SOC_DAPM_DAC("Right DAC", NULL, SUN8I_ADDA_DAC_PA_SRC, + SUN8I_ADDA_DAC_PA_SRC_DACAREN, 0), + /* + * Due to this component and the codec belonging to separate DAPM + * contexts, we need to manually link the above widgets to their + * stream widgets at the card level. + */ + + /* Line In */ + SND_SOC_DAPM_INPUT("LINEIN"), + + /* Microphone inputs */ + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + + /* Microphone Bias */ + SND_SOC_DAPM_SUPPLY("MBIAS", SUN8I_ADDA_MIC1G_MICBIAS_CTRL, + SUN8I_ADDA_MIC1G_MICBIAS_CTRL_MMICBIASEN, + 0, NULL, 0), + + /* Mic input path */ + SND_SOC_DAPM_PGA("Mic1 Amplifier", SUN8I_ADDA_MIC1G_MICBIAS_CTRL, + SUN8I_ADDA_MIC1G_MICBIAS_CTRL_MIC1AMPEN, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mic2 Amplifier", SUN8I_ADDA_MIC2G_CTRL, + SUN8I_ADDA_MIC2G_CTRL_MIC2AMPEN, 0, NULL, 0), + + /* Mixers */ + SND_SOC_DAPM_MIXER("Left Mixer", SUN8I_ADDA_DAC_PA_SRC, + SUN8I_ADDA_DAC_PA_SRC_LMIXEN, 0, + sun8i_codec_mixer_controls, + ARRAY_SIZE(sun8i_codec_mixer_controls)), + SND_SOC_DAPM_MIXER("Right Mixer", SUN8I_ADDA_DAC_PA_SRC, + SUN8I_ADDA_DAC_PA_SRC_RMIXEN, 0, + sun8i_codec_mixer_controls, + ARRAY_SIZE(sun8i_codec_mixer_controls)), + SND_SOC_DAPM_MIXER("Left ADC Mixer", SUN8I_ADDA_ADC_AP_EN, + SUN8I_ADDA_ADC_AP_EN_ADCLEN, 0, + sun8i_codec_adc_mixer_controls, + ARRAY_SIZE(sun8i_codec_adc_mixer_controls)), + SND_SOC_DAPM_MIXER("Right ADC Mixer", SUN8I_ADDA_ADC_AP_EN, + SUN8I_ADDA_ADC_AP_EN_ADCREN, 0, + sun8i_codec_adc_mixer_controls, + ARRAY_SIZE(sun8i_codec_adc_mixer_controls)), +}; + +static const struct snd_soc_dapm_route sun8i_codec_common_routes[] = { + /* Microphone Routes */ + { "Mic1 Amplifier", NULL, "MIC1"}, + { "Mic2 Amplifier", NULL, "MIC2"}, + + /* Left Mixer Routes */ + { "Left Mixer", "DAC Playback Switch", "Left DAC" }, + { "Left Mixer", "DAC Reversed Playback Switch", "Right DAC" }, + { "Left Mixer", "Line In Playback Switch", "LINEIN" }, + { "Left Mixer", "Mic1 Playback Switch", "Mic1 Amplifier" }, + { "Left Mixer", "Mic2 Playback Switch", "Mic2 Amplifier" }, + + /* Right Mixer Routes */ + { "Right Mixer", "DAC Playback Switch", "Right DAC" }, + { "Right Mixer", "DAC Reversed Playback Switch", "Left DAC" }, + { "Right Mixer", "Line In Playback Switch", "LINEIN" }, + { "Right Mixer", "Mic1 Playback Switch", "Mic1 Amplifier" }, + { "Right Mixer", "Mic2 Playback Switch", "Mic2 Amplifier" }, + + /* Left ADC Mixer Routes */ + { "Left ADC Mixer", "Mixer Capture Switch", "Left Mixer" }, + { "Left ADC Mixer", "Mixer Reversed Capture Switch", "Right Mixer" }, + { "Left ADC Mixer", "Line In Capture Switch", "LINEIN" }, + { "Left ADC Mixer", "Mic1 Capture Switch", "Mic1 Amplifier" }, + { "Left ADC Mixer", "Mic2 Capture Switch", "Mic2 Amplifier" }, + + /* Right ADC Mixer Routes */ + { "Right ADC Mixer", "Mixer Capture Switch", "Right Mixer" }, + { "Right ADC Mixer", "Mixer Reversed Capture Switch", "Left Mixer" }, + { "Right ADC Mixer", "Line In Capture Switch", "LINEIN" }, + { "Right ADC Mixer", "Mic1 Capture Switch", "Mic1 Amplifier" }, + { "Right ADC Mixer", "Mic2 Capture Switch", "Mic2 Amplifier" }, + + /* ADC Routes */ + { "Left ADC", NULL, "Left ADC Mixer" }, + { "Right ADC", NULL, "Right ADC Mixer" }, +}; + +/* headphone specific controls, widgets, and routes */ +static const DECLARE_TLV_DB_SCALE(sun8i_codec_hp_vol_scale, -6300, 100, 1); +static const struct snd_kcontrol_new sun8i_codec_headphone_controls[] = { + SOC_SINGLE_TLV("Headphone Playback Volume", + SUN8I_ADDA_HP_VOLC, + SUN8I_ADDA_HP_VOLC_HP_VOL, 0x3f, 0, + sun8i_codec_hp_vol_scale), + SOC_DOUBLE("Headphone Playback Switch", + SUN8I_ADDA_DAC_PA_SRC, + SUN8I_ADDA_DAC_PA_SRC_LHPPAMUTE, + SUN8I_ADDA_DAC_PA_SRC_RHPPAMUTE, 1, 0), +}; + +static const char * const sun8i_codec_hp_src_enum_text[] = { + "DAC", "Mixer", +}; + +static SOC_ENUM_DOUBLE_DECL(sun8i_codec_hp_src_enum, + SUN8I_ADDA_DAC_PA_SRC, + SUN8I_ADDA_DAC_PA_SRC_LHPIS, + SUN8I_ADDA_DAC_PA_SRC_RHPIS, + sun8i_codec_hp_src_enum_text); + +static const struct snd_kcontrol_new sun8i_codec_hp_src[] = { + SOC_DAPM_ENUM("Headphone Source Playback Route", + sun8i_codec_hp_src_enum), +}; + +static const struct snd_soc_dapm_widget sun8i_codec_headphone_widgets[] = { + SND_SOC_DAPM_MUX("Headphone Source Playback Route", + SND_SOC_NOPM, 0, 0, sun8i_codec_hp_src), + SND_SOC_DAPM_OUT_DRV("Headphone Amp", SUN8I_ADDA_PAEN_HP_CTRL, + SUN8I_ADDA_PAEN_HP_CTRL_HPPAEN, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HPCOM Protection", SUN8I_ADDA_PAEN_HP_CTRL, + SUN8I_ADDA_PAEN_HP_CTRL_COMPTEN, 0, NULL, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPCOM", SUN8I_ADDA_PAEN_HP_CTRL, + SUN8I_ADDA_PAEN_HP_CTRL_HPCOM_FC, 0x3, 0x3, 0), + SND_SOC_DAPM_OUTPUT("HP"), +}; + +static const struct snd_soc_dapm_route sun8i_codec_headphone_routes[] = { + { "Headphone Source Playback Route", "DAC", "Left DAC" }, + { "Headphone Source Playback Route", "DAC", "Right DAC" }, + { "Headphone Source Playback Route", "Mixer", "Left Mixer" }, + { "Headphone Source Playback Route", "Mixer", "Right Mixer" }, + { "Headphone Amp", NULL, "Headphone Source Playback Route" }, + { "HPCOM", NULL, "HPCOM Protection" }, + { "HP", NULL, "Headphone Amp" }, +}; + +static int sun8i_codec_add_headphone(struct snd_soc_component *cmpnt) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(cmpnt); + struct device *dev = cmpnt->dev; + int ret; + + ret = snd_soc_add_component_controls(cmpnt, + sun8i_codec_headphone_controls, + ARRAY_SIZE(sun8i_codec_headphone_controls)); + if (ret) { + dev_err(dev, "Failed to add Headphone controls: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_new_controls(dapm, sun8i_codec_headphone_widgets, + ARRAY_SIZE(sun8i_codec_headphone_widgets)); + if (ret) { + dev_err(dev, "Failed to add Headphone DAPM widgets: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(dapm, sun8i_codec_headphone_routes, + ARRAY_SIZE(sun8i_codec_headphone_routes)); + if (ret) { + dev_err(dev, "Failed to add Headphone DAPM routes: %d\n", ret); + return ret; + } + + return 0; +} + +/* hmic specific widget */ +static const struct snd_soc_dapm_widget sun8i_codec_hmic_widgets[] = { + SND_SOC_DAPM_SUPPLY("HBIAS", SUN8I_ADDA_MIC1G_MICBIAS_CTRL, + SUN8I_ADDA_MIC1G_MICBIAS_CTRL_HMICBIASEN, + 0, NULL, 0), +}; + +static int sun8i_codec_add_hmic(struct snd_soc_component *cmpnt) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(cmpnt); + struct device *dev = cmpnt->dev; + int ret; + + ret = snd_soc_dapm_new_controls(dapm, sun8i_codec_hmic_widgets, + ARRAY_SIZE(sun8i_codec_hmic_widgets)); + if (ret) + dev_err(dev, "Failed to add Mic3 DAPM widgets: %d\n", ret); + + return ret; +} + +/* line out specific controls, widgets and routes */ +static const DECLARE_TLV_DB_RANGE(sun8i_codec_lineout_vol_scale, + 0, 1, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + 2, 31, TLV_DB_SCALE_ITEM(-4350, 150, 0), +); +static const struct snd_kcontrol_new sun8i_codec_lineout_controls[] = { + SOC_SINGLE_TLV("Line Out Playback Volume", + SUN8I_ADDA_PHONE_GAIN_CTRL, + SUN8I_ADDA_PHONE_GAIN_CTRL_LINEOUT_VOL, 0x1f, 0, + sun8i_codec_lineout_vol_scale), + SOC_DOUBLE("Line Out Playback Switch", + SUN8I_ADDA_MIC2G_CTRL, + SUN8I_ADDA_MIC2G_CTRL_LINEOUTLEN, + SUN8I_ADDA_MIC2G_CTRL_LINEOUTREN, 1, 0), +}; + +static const char * const sun8i_codec_lineout_src_enum_text[] = { + "Stereo", "Mono Differential", +}; + +static SOC_ENUM_DOUBLE_DECL(sun8i_codec_lineout_src_enum, + SUN8I_ADDA_MIC2G_CTRL, + SUN8I_ADDA_MIC2G_CTRL_LINEOUTLSRC, + SUN8I_ADDA_MIC2G_CTRL_LINEOUTRSRC, + sun8i_codec_lineout_src_enum_text); + +static const struct snd_kcontrol_new sun8i_codec_lineout_src[] = { + SOC_DAPM_ENUM("Line Out Source Playback Route", + sun8i_codec_lineout_src_enum), +}; + +static const struct snd_soc_dapm_widget sun8i_codec_lineout_widgets[] = { + SND_SOC_DAPM_MUX("Line Out Source Playback Route", + SND_SOC_NOPM, 0, 0, sun8i_codec_lineout_src), + /* It is unclear if this is a buffer or gate, model it as a supply */ + SND_SOC_DAPM_SUPPLY("Line Out Enable", SUN8I_ADDA_PAEN_HP_CTRL, + SUN8I_ADDA_PAEN_HP_CTRL_LINEOUTEN, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("LINEOUT"), +}; + +static const struct snd_soc_dapm_route sun8i_codec_lineout_routes[] = { + { "Line Out Source Playback Route", "Stereo", "Left Mixer" }, + { "Line Out Source Playback Route", "Stereo", "Right Mixer" }, + { "Line Out Source Playback Route", "Mono Differential", "Left Mixer" }, + { "Line Out Source Playback Route", "Mono Differential", "Right Mixer" }, + { "LINEOUT", NULL, "Line Out Source Playback Route" }, + { "LINEOUT", NULL, "Line Out Enable", }, +}; + +static int sun8i_codec_add_lineout(struct snd_soc_component *cmpnt) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(cmpnt); + struct device *dev = cmpnt->dev; + int ret; + + ret = snd_soc_add_component_controls(cmpnt, + sun8i_codec_lineout_controls, + ARRAY_SIZE(sun8i_codec_lineout_controls)); + if (ret) { + dev_err(dev, "Failed to add Line Out controls: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_new_controls(dapm, sun8i_codec_lineout_widgets, + ARRAY_SIZE(sun8i_codec_lineout_widgets)); + if (ret) { + dev_err(dev, "Failed to add Line Out DAPM widgets: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(dapm, sun8i_codec_lineout_routes, + ARRAY_SIZE(sun8i_codec_lineout_routes)); + if (ret) { + dev_err(dev, "Failed to add Line Out DAPM routes: %d\n", ret); + return ret; + } + + return 0; +} + +struct sun8i_codec_analog_quirks { + bool has_headphone; + bool has_hmic; + bool has_lineout; +}; + +static const struct sun8i_codec_analog_quirks sun8i_a23_quirks = { + .has_headphone = true, + .has_hmic = true, +}; + +static const struct sun8i_codec_analog_quirks sun8i_h3_quirks = { + .has_lineout = true, +}; + +static int sun8i_codec_analog_cmpnt_probe(struct snd_soc_component *cmpnt) +{ + struct device *dev = cmpnt->dev; + const struct sun8i_codec_analog_quirks *quirks; + int ret; + + /* + * This would never return NULL unless someone directly registers a + * platform device matching this driver's name, without specifying a + * device tree node. + */ + quirks = of_device_get_match_data(dev); + + /* Add controls, widgets, and routes for individual features */ + + if (quirks->has_headphone) { + ret = sun8i_codec_add_headphone(cmpnt); + if (ret) + return ret; + } + + if (quirks->has_hmic) { + ret = sun8i_codec_add_hmic(cmpnt); + if (ret) + return ret; + } + + if (quirks->has_lineout) { + ret = sun8i_codec_add_lineout(cmpnt); + if (ret) + return ret; + } + + return 0; +} + +static const struct snd_soc_component_driver sun8i_codec_analog_cmpnt_drv = { + .controls = sun8i_codec_common_controls, + .num_controls = ARRAY_SIZE(sun8i_codec_common_controls), + .dapm_widgets = sun8i_codec_common_widgets, + .num_dapm_widgets = ARRAY_SIZE(sun8i_codec_common_widgets), + .dapm_routes = sun8i_codec_common_routes, + .num_dapm_routes = ARRAY_SIZE(sun8i_codec_common_routes), + .probe = sun8i_codec_analog_cmpnt_probe, +}; + +static const struct of_device_id sun8i_codec_analog_of_match[] = { + { + .compatible = "allwinner,sun8i-a23-codec-analog", + .data = &sun8i_a23_quirks, + }, + { + .compatible = "allwinner,sun8i-h3-codec-analog", + .data = &sun8i_h3_quirks, + }, + {} +}; +MODULE_DEVICE_TABLE(of, sun8i_codec_analog_of_match); + +static int sun8i_codec_analog_probe(struct platform_device *pdev) +{ + struct resource *res; + struct regmap *regmap; + void __iomem *base; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) { + dev_err(&pdev->dev, "Failed to map the registers\n"); + return PTR_ERR(base); + } + + regmap = devm_regmap_init(&pdev->dev, NULL, base, &adda_pr_regmap_cfg); + if (IS_ERR(regmap)) { + dev_err(&pdev->dev, "Failed to create regmap\n"); + return PTR_ERR(regmap); + } + + return devm_snd_soc_register_component(&pdev->dev, + &sun8i_codec_analog_cmpnt_drv, + NULL, 0); +} + +static struct platform_driver sun8i_codec_analog_driver = { + .driver = { + .name = "sun8i-codec-analog", + .of_match_table = sun8i_codec_analog_of_match, + }, + .probe = sun8i_codec_analog_probe, +}; +module_platform_driver(sun8i_codec_analog_driver); + +MODULE_DESCRIPTION("Allwinner internal codec analog controls driver"); +MODULE_AUTHOR("Chen-Yu Tsai "); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sun8i-codec-analog"); diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index deb597f7c302..eead6e7f205b 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -65,7 +65,7 @@ static int tegra_alc5632_asoc_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops tegra_alc5632_asoc_ops = { +static const struct snd_soc_ops tegra_alc5632_asoc_ops = { .hw_params = tegra_alc5632_asoc_hw_params, }; diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index 902da36581d1..a403db6d563e 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -93,7 +93,7 @@ static int tegra_max98090_asoc_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops tegra_max98090_ops = { +static const struct snd_soc_ops tegra_max98090_ops = { .hw_params = tegra_max98090_asoc_hw_params, }; diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index e5ef4e9c4ac5..25b9fc03ba62 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -76,7 +76,7 @@ static int tegra_rt5640_asoc_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops tegra_rt5640_ops = { +static const struct snd_soc_ops tegra_rt5640_ops = { .hw_params = tegra_rt5640_asoc_hw_params, }; diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c index 1470873ecde6..ebf58d0e0f10 100644 --- a/sound/soc/tegra/tegra_rt5677.c +++ b/sound/soc/tegra/tegra_rt5677.c @@ -93,7 +93,7 @@ static int tegra_rt5677_event_hp(struct snd_soc_dapm_widget *w, return 0; } -static struct snd_soc_ops tegra_rt5677_ops = { +static const struct snd_soc_ops tegra_rt5677_ops = { .hw_params = tegra_rt5677_asoc_hw_params, }; diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c index 1e76869dd488..4bbab098f50b 100644 --- a/sound/soc/tegra/tegra_sgtl5000.c +++ b/sound/soc/tegra/tegra_sgtl5000.c @@ -82,7 +82,7 @@ static int tegra_sgtl5000_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops tegra_sgtl5000_ops = { +static const struct snd_soc_ops tegra_sgtl5000_ops = { .hw_params = tegra_sgtl5000_hw_params, }; diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c index f0cd01dbfc38..bdedd1028569 100644 --- a/sound/soc/tegra/tegra_wm8753.c +++ b/sound/soc/tegra/tegra_wm8753.c @@ -89,7 +89,7 @@ static int tegra_wm8753_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops tegra_wm8753_ops = { +static const struct snd_soc_ops tegra_wm8753_ops = { .hw_params = tegra_wm8753_hw_params, }; diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index e485278e027a..2013e9c4bba0 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -96,7 +96,7 @@ static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops tegra_wm8903_ops = { +static const struct snd_soc_ops tegra_wm8903_ops = { .hw_params = tegra_wm8903_hw_params, }; diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 2cea203c4f5f..870f84ab5005 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -74,7 +74,7 @@ static int trimslice_asoc_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops trimslice_asoc_ops = { +static const struct snd_soc_ops trimslice_asoc_ops = { .hw_params = trimslice_asoc_hw_params, };