Index: sys/arm/allwinner/a64/sun50i_a64_acodec.c =================================================================== --- /dev/null +++ sys/arm/allwinner/a64/sun50i_a64_acodec.c @@ -0,0 +1,488 @@ +/*- + * SPDX-License-Identifier: BSD-2-Clause-FreeBSD + * + * Copyright (c) 2020 Oleksandr Tymoshenko + * Copyright (c) 2018 Jared McNeill + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions + * are met: + * 1. Redistributions of source code must retain the above copyright + * notice, this list of conditions and the following disclaimer. + * 2. Redistributions in binary form must reproduce the above copyright + * notice, this list of conditions and the following disclaimer in the + * documentation and/or other materials provided with the distribution. + * + * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR + * IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES + * OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. + * IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, + * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, + * BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; + * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED + * AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, + * OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY + * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF + * SUCH DAMAGE. + * + * $FreeBSD$ + */ + +#include +__FBSDID("$FreeBSD$"); + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include + +#include +#include +#include + +#include "syscon_if.h" + +#include "opt_snd.h" +#include +#include +#include "audio_dai_if.h" +#include "mixer_if.h" + +#define A64_PR_CFG 0x00 +#define A64_AC_PR_RST (1 << 28) +#define A64_AC_PR_RW (1 << 24) +#define A64_AC_PR_ADDR_MASK (0x1f << 16) +#define A64_AC_PR_ADDR(n) (((n) & 0x1f) << 16) +#define A64_ACDA_PR_WDAT_MASK (0xff << 8) +#define A64_ACDA_PR_WDAT(n) (((n) & 0xff) << 8) +#define A64_ACDA_PR_RDAT(n) ((n) & 0xff) + +#define A64_HP_CTRL 0x00 +#define A64_HPPA_EN (1 << 6) +#define A64_HPVOL_MASK 0x3f +#define A64_HPVOL(n) ((n) & 0x3f) +#define A64_OL_MIX_CTRL 0x01 +#define A64_LMIXMUTE_LDAC (1 << 1) +#define A64_OR_MIX_CTRL 0x02 +#define A64_RMIXMUTE_RDAC (1 << 1) +#define A64_LINEOUT_CTRL0 0x05 +#define A64_LINEOUT_LEFT_EN (1 << 7) +#define A64_LINEOUT_RIGHT_EN (1 << 6) +#define A64_LINEOUT_EN (A64_LINEOUT_LEFT_EN|A64_LINEOUT_RIGHT_EN) +#define A64_LINEOUT_CTRL1 0x06 +#define A64_LINEOUT_VOL __BITS(4,0) +#define A64_MIC1_CTRL 0x07 +#define A64_MIC1G __BITS(6,4) +#define A64_MIC1AMPEN (1 << 3) +#define A64_MIC1BOOST __BITS(2,0) +#define A64_MIC2_CTRL 0x08 +#define A64_MIC2_SEL (1 << 7) +#define A64_MIC2G_MASK (7 << 4) +#define A64_MIC2G(n) (((n) & 7) << 4) +#define A64_MIC2AMPEN (1 << 3) +#define A64_MIC2BOOST_MASK (7 << 0) +#define A64_MIC2BOOST(n) (((n) & 7) << 0) +#define A64_LINEIN_CTRL 0x09 +#define A64_LINEING __BITS(6,4) +#define A64_MIX_DAC_CTRL 0x0a +#define A64_DACAREN (1 << 7) +#define A64_DACALEN (1 << 6) +#define A64_RMIXEN (1 << 5) +#define A64_LMIXEN (1 << 4) +#define A64_RHPPAMUTE (1 << 3) +#define A64_LHPPAMUTE (1 << 2) +#define A64_RHPIS (1 << 1) +#define A64_LHPIS (1 << 0) +#define A64_L_ADCMIX_SRC 0x0b +#define A64_R_ADCMIX_SRC 0x0c +#define A64_ADCMIX_SRC_MIC1 (1 << 6) +#define A64_ADCMIX_SRC_MIC2 (1 << 5) +#define A64_ADCMIX_SRC_LINEIN (1 << 2) +#define A64_ADCMIX_SRC_OMIXER (1 << 1) +#define A64_ADC_CTRL 0x0d +#define A64_ADCREN (1 << 7) +#define A64_ADCLEN (1 << 6) +#define A64_ADCG __BITS(2,0) +#define A64_JACK_MIC_CTRL 0x1d +#define A64_JACKDETEN (1 << 7) +#define A64_INNERRESEN (1 << 6) +#define A64_HMICBIASEN (1 << 5) +#define A64_AUTOPLEN (1 << 1) + +#define A64CODEC_MIXER_DEVS ((1 << SOUND_MIXER_VOLUME) | \ + (1 << SOUND_MIXER_MIC)) + +static struct ofw_compat_data compat_data[] = { + { "allwinner,sun50i-a64-codec-analog", 1}, + { NULL, 0 } +}; + +struct a64codec_softc { + device_t dev; + struct resource *res; + struct mtx mtx; + u_int regaddr; /* address for the sysctl */ +}; + +#define A64CODEC_LOCK(sc) mtx_lock(&(sc)->mtx) +#define A64CODEC_UNLOCK(sc) mtx_unlock(&(sc)->mtx) +#define A64CODEC_READ(sc, reg) bus_read_4((sc)->res, (reg)) +#define A64CODEC_WRITE(sc, reg, val) bus_write_4((sc)->res, (reg), (val)) + +static int a64codec_probe(device_t dev); +static int a64codec_attach(device_t dev); +static int a64codec_detach(device_t dev); + +static u_int +a64_acodec_pr_read(struct a64codec_softc *sc, u_int addr) +{ + uint32_t val; + + /* Read current value */ + val = A64CODEC_READ(sc, A64_PR_CFG); + + /* De-assert reset */ + val |= A64_AC_PR_RST; + A64CODEC_WRITE(sc, A64_PR_CFG, val); + + /* Read mode */ + val &= ~A64_AC_PR_RW; + A64CODEC_WRITE(sc, A64_PR_CFG, val); + + /* Set address */ + val &= ~A64_AC_PR_ADDR_MASK; + val |= A64_AC_PR_ADDR(addr); + A64CODEC_WRITE(sc, A64_PR_CFG, val); + + /* Read data */ + val = A64CODEC_READ(sc, A64_PR_CFG); + return A64_ACDA_PR_RDAT(val); +} + +static void +a64_acodec_pr_write(struct a64codec_softc *sc, u_int addr, u_int data) +{ + uint32_t val; + + /* Read current value */ + val = A64CODEC_READ(sc, A64_PR_CFG); + + /* De-assert reset */ + val |= A64_AC_PR_RST; + A64CODEC_WRITE(sc, A64_PR_CFG, val); + + /* Set address */ + val &= ~A64_AC_PR_ADDR_MASK; + val |= A64_AC_PR_ADDR(addr); + A64CODEC_WRITE(sc, A64_PR_CFG, val); + + /* Write data */ + val &= ~A64_ACDA_PR_WDAT_MASK; + val |= A64_ACDA_PR_WDAT(data); + A64CODEC_WRITE(sc, A64_PR_CFG, val); + + /* Write mode */ + val |= A64_AC_PR_RW; + A64CODEC_WRITE(sc, A64_PR_CFG, val); + + /* Clear write mode */ + val &= ~A64_AC_PR_RW; + A64CODEC_WRITE(sc, A64_PR_CFG, val); +} + +static void +a64_acodec_pr_set_clear(struct a64codec_softc *sc, u_int addr, u_int set, u_int clr) +{ + u_int old, new; + + old = a64_acodec_pr_read(sc, addr); + new = set | (old & ~clr); + a64_acodec_pr_write(sc, addr, new); +} + +static int +a64codec_probe(device_t dev) +{ + if (!ofw_bus_status_okay(dev)) + return (ENXIO); + + if (!ofw_bus_search_compatible(dev, compat_data)->ocd_data) + return (ENXIO); + + device_set_desc(dev, "Allwinner A64 Analog Codec"); + return (BUS_PROBE_DEFAULT); +} + +static int +a64codec_attach(device_t dev) +{ + struct a64codec_softc *sc; + int error, rid; + phandle_t node; + regulator_t reg; + + sc = device_get_softc(dev); + sc->dev = dev; + + mtx_init(&sc->mtx, device_get_nameunit(dev), NULL, MTX_DEF); + + rid = 0; + sc->res = bus_alloc_resource_any(dev, SYS_RES_MEMORY, &rid, RF_ACTIVE); + if (!sc->res) { + device_printf(dev, "cannot allocate resource for device\n"); + error = ENXIO; + goto fail; + } + + if (regulator_get_by_ofw_property(dev, 0, "cpvdd-supply", ®) == 0) { + error = regulator_enable(reg); + if (error != 0) { + device_printf(dev, "cannot enable PHY regulator\n"); + goto fail; + } + } + + /* Right & Left Headphone PA enable */ + a64_acodec_pr_set_clear(sc, A64_HP_CTRL, + A64_HPPA_EN, 0); + + /* Microphone BIAS enable */ + a64_acodec_pr_set_clear(sc, A64_JACK_MIC_CTRL, + A64_HMICBIASEN | A64_INNERRESEN, 0); + + /* Unmute DAC to output mixer */ + a64_acodec_pr_set_clear(sc, A64_OL_MIX_CTRL, + A64_LMIXMUTE_LDAC, 0); + a64_acodec_pr_set_clear(sc, A64_OR_MIX_CTRL, + A64_RMIXMUTE_RDAC, 0); + + /* For now we work only with headphones */ + a64_acodec_pr_set_clear(sc, A64_LINEOUT_CTRL0, + 0, A64_LINEOUT_EN); + a64_acodec_pr_set_clear(sc, A64_HP_CTRL, + A64_HPPA_EN, 0); + + u_int val = a64_acodec_pr_read(sc, A64_HP_CTRL); + val &= ~(0x3f); + val |= 0x25; + a64_acodec_pr_write(sc, A64_HP_CTRL, val); + + a64_acodec_pr_set_clear(sc, A64_MIC2_CTRL, + A64_MIC2AMPEN | A64_MIC2_SEL | A64_MIC2G(0x3) | A64_MIC2BOOST(0x4), + A64_MIC2G_MASK | A64_MIC2BOOST_MASK); + + a64_acodec_pr_write(sc, A64_L_ADCMIX_SRC, + A64_ADCMIX_SRC_MIC2); + a64_acodec_pr_write(sc, A64_R_ADCMIX_SRC, + A64_ADCMIX_SRC_MIC2); + + /* Max out MIC2 gain */ + val = a64_acodec_pr_read(sc, A64_MIC2_CTRL); + val &= ~(0x7); + val |= (0x7); + val &= ~(7 << 4); + val |= (7 << 4); + a64_acodec_pr_write(sc, A64_MIC2_CTRL, val); + + node = ofw_bus_get_node(dev); + OF_device_register_xref(OF_xref_from_node(node), dev); + + return (0); + +fail: + a64codec_detach(dev); + return (error); +} + +static int +a64codec_detach(device_t dev) +{ + struct a64codec_softc *sc; + + sc = device_get_softc(dev); + + if (sc->res) + bus_release_resource(dev, SYS_RES_MEMORY, 0, sc->res); + mtx_destroy(&sc->mtx); + + return (0); +} + +static int +a64codec_mixer_init(struct snd_mixer *m) +{ + + mix_setdevs(m, A64CODEC_MIXER_DEVS); + + return (0); +} + +static int +a64codec_mixer_uninit(struct snd_mixer *m) +{ + + return (0); +} + +static int +a64codec_mixer_reinit(struct snd_mixer *m) +{ + + return (0); +} + +static int +a64codec_mixer_set(struct snd_mixer *m, unsigned dev, unsigned left, unsigned right) +{ + struct a64codec_softc *sc; + struct mtx *mixer_lock; + uint8_t do_unlock; + u_int val; + + sc = device_get_softc(mix_getdevinfo(m)); + mixer_lock = mixer_get_lock(m); + + if (mtx_owned(mixer_lock)) { + do_unlock = 0; + } else { + do_unlock = 1; + mtx_lock(mixer_lock); + } + + right = left; + + A64CODEC_LOCK(sc); + switch(dev) { + case SOUND_MIXER_VOLUME: + val = a64_acodec_pr_read(sc, A64_HP_CTRL); + val &= ~(A64_HPVOL_MASK); + val |= A64_HPVOL(left * 63 / 100); + a64_acodec_pr_write(sc, A64_HP_CTRL, val); + break; + + case SOUND_MIXER_MIC: + val = a64_acodec_pr_read(sc, A64_MIC2_CTRL); + val &= ~(A64_MIC2BOOST_MASK); + val |= A64_MIC2BOOST(left * 7 / 100); + a64_acodec_pr_write(sc, A64_MIC2_CTRL, val); + break; + default: + break; + } + A64CODEC_UNLOCK(sc); + + if (do_unlock) { + mtx_unlock(mixer_lock); + } + + return (left | (right << 8)); +} + +static unsigned +a64codec_mixer_setrecsrc(struct snd_mixer *m, unsigned src) +{ + + return (0); +} + +static kobj_method_t a64codec_mixer_methods[] = { + KOBJMETHOD(mixer_init, a64codec_mixer_init), + KOBJMETHOD(mixer_uninit, a64codec_mixer_uninit), + KOBJMETHOD(mixer_reinit, a64codec_mixer_reinit), + KOBJMETHOD(mixer_set, a64codec_mixer_set), + KOBJMETHOD(mixer_setrecsrc, a64codec_mixer_setrecsrc), + KOBJMETHOD_END +}; + +MIXER_DECLARE(a64codec_mixer); + +static int +a64codec_dai_init(device_t dev, uint32_t format) +{ + + return (0); +} + +static int +a64codec_dai_trigger(device_t dev, int go, int pcm_dir) +{ + struct a64codec_softc *sc = device_get_softc(dev); + + if ((pcm_dir != PCMDIR_PLAY) && (pcm_dir != PCMDIR_REC)) + return (EINVAL); + + switch (go) { + case PCMTRIG_START: + if (pcm_dir == PCMDIR_PLAY) { + /* Enable DAC analog l/r channels, HP PA, and output mixer */ + a64_acodec_pr_set_clear(sc, A64_MIX_DAC_CTRL, + A64_DACAREN | A64_DACALEN | A64_RMIXEN | A64_LMIXEN | + A64_RHPPAMUTE | A64_LHPPAMUTE, 0); + } + else if (pcm_dir == PCMDIR_REC) { + /* Enable ADC analog l/r channels */ + a64_acodec_pr_set_clear(sc, A64_ADC_CTRL, + A64_ADCREN | A64_ADCLEN, 0); + } + break; + + case PCMTRIG_STOP: + case PCMTRIG_ABORT: + if (pcm_dir == PCMDIR_PLAY) { + /* Disable DAC analog l/r channels, HP PA, and output mixer */ + a64_acodec_pr_set_clear(sc, A64_MIX_DAC_CTRL, + 0, A64_DACAREN | A64_DACALEN | A64_RMIXEN | A64_LMIXEN | + A64_RHPPAMUTE | A64_LHPPAMUTE); + } + else if (pcm_dir == PCMDIR_REC) { + /* Disable ADC analog l/r channels */ + a64_acodec_pr_set_clear(sc, A64_ADC_CTRL, + 0, A64_ADCREN | A64_ADCLEN); + } + break; + } + + return (0); +} + +static int +a64codec_dai_setup_mixer(device_t dev, device_t pcmdev) +{ + + mixer_init(pcmdev, &a64codec_mixer_class, dev); + + return (0); +} + +static device_method_t a64codec_methods[] = { + /* Device interface */ + DEVMETHOD(device_probe, a64codec_probe), + DEVMETHOD(device_attach, a64codec_attach), + DEVMETHOD(device_detach, a64codec_detach), + + DEVMETHOD(audio_dai_init, a64codec_dai_init), + DEVMETHOD(audio_dai_setup_mixer, a64codec_dai_setup_mixer), + DEVMETHOD(audio_dai_trigger, a64codec_dai_trigger), + + DEVMETHOD_END +}; + +static driver_t a64codec_driver = { + "a64codec", + a64codec_methods, + sizeof(struct a64codec_softc), +}; + +static devclass_t a64codec_devclass; + +DRIVER_MODULE(a64codec, simplebus, a64codec_driver, a64codec_devclass, 0, 0); +SIMPLEBUS_PNP_INFO(compat_data); Index: sys/arm/allwinner/aw_i2s.c =================================================================== --- /dev/null +++ sys/arm/allwinner/aw_i2s.c @@ -0,0 +1,813 @@ +/*- + * SPDX-License-Identifier: BSD-2-Clause-FreeBSD + * + * Copyright (c) 2020 Oleksandr Tymoshenko + * Copyright (c) 2018 Jared McNeill + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions + * are met: + * 1. Redistributions of source code must retain the above copyright + * notice, this list of conditions and the following disclaimer. + * 2. Redistributions in binary form must reproduce the above copyright + * notice, this list of conditions and the following disclaimer in the + * documentation and/or other materials provided with the distribution. + * + * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR + * IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES + * OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. + * IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, + * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, + * BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; + * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED + * AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, + * OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY + * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF + * SUCH DAMAGE. + * + * $FreeBSD$ + */ + +#include +__FBSDID("$FreeBSD$"); + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include + +#include +#include + +#include "syscon_if.h" + +#include "opt_snd.h" +#include +#include +#include "audio_dai_if.h" + +#define FIFO_LEVEL 0x40 + +#define DA_CTL 0x00 +#define DA_CTL_BCLK_OUT (1 << 18) /* sun8i */ +#define DA_CLK_LRCK_OUT (1 << 17) /* sun8i */ +#define DA_CTL_SDO_EN (1 << 8) +#define DA_CTL_MS (1 << 5) /* sun4i */ +#define DA_CTL_PCM (1 << 4) /* sun4i */ +#define DA_CTL_MODE_SEL_MASK (3 << 4) /* sun8i */ +#define DA_CTL_MODE_SEL_PCM (0 << 4) /* sun8i */ +#define DA_CTL_MODE_SEL_LJ (1 << 4) /* sun8i */ +#define DA_CTL_MODE_SEL_RJ (2 << 4) /* sun8i */ +#define DA_CTL_TXEN (1 << 2) +#define DA_CTL_RXEN (1 << 1) +#define DA_CTL_GEN (1 << 0) +#define DA_FAT0 0x04 +#define DA_FAT0_LRCK_PERIOD_MASK (0x3ff << 8) /* sun8i */ +#define DA_FAT0_LRCK_PERIOD(n) (((n) & 0x3fff) << 8) /* sun8i */ +#define DA_FAT0_LRCP_MASK (1 << 7) +#define DA_LRCP_NORMAL (0 << 7) +#define DA_LRCP_INVERTED (1 << 7) +#define DA_FAT0_BCP_MASK (1 << 6) +#define DA_BCP_NORMAL (0 << 6) +#define DA_BCP_INVERTED (1 << 6) +#define DA_FAT0_SR __BITS(5,4) +#define DA_FAT0_WSS __BITS(3,2) +#define DA_FAT0_FMT_MASK (3 << 0) +#define DA_FMT_I2S 0 +#define DA_FMT_LJ 1 +#define DA_FMT_RJ 2 +#define DA_FAT1 0x08 +#define DA_ISTA 0x0c +#define DA_ISTA_TXUI_INT (1 << 6) +#define DA_ISTA_TXEI_INT (1 << 4) +#define DA_ISTA_RXAI_INT (1 << 0) +#define DA_RXFIFO 0x10 +#define DA_FCTL 0x14 +#define DA_FCTL_HUB_EN (1 << 31) +#define DA_FCTL_FTX (1 << 25) +#define DA_FCTL_FRX (1 << 24) +#define DA_FCTL_TXTL_MASK (0x7f << 12) +#define DA_FCTL_TXTL(v) (((v) & 0x7f) << 12) +#define DA_FCTL_TXIM (1 << 2) +#define DA_FSTA 0x18 +#define DA_FSTA_TXE_CNT(v) (((v) >> 16) & 0xff) +#define DA_FSTA_RXA_CNT(v) ((v) & 0x3f) +#define DA_INT 0x1c +#define DA_INT_TX_DRQ (1 << 7) +#define DA_INT_TXUI_EN (1 << 6) +#define DA_INT_TXEI_EN (1 << 4) +#define DA_INT_RX_DRQ (1 << 3) +#define DA_INT_RXAI_EN (1 << 0) +#define DA_TXFIFO 0x20 +#define DA_CLKD 0x24 +#define DA_CLKD_MCLKO_EN_SUN8I (1 << 8) +#define DA_CLKD_MCLKO_EN_SUN4I (1 << 7) +#define DA_CLKD_BCLKDIV_SUN8I(n) (((n) & 0xf) << 4) +#define DA_CLKD_BCLKDIV_SUN8I_MASK (0xf << 4) +#define DA_CLKD_BCLKDIV_SUN4I(n) (((n) & 7) << 4) +#define DA_CLKD_BCLKDIV_SUN4I_MASK (7 << 4) +#define DA_CLKD_BCLKDIV_8 3 +#define DA_CLKD_BCLKDIV_16 5 +#define DA_CLKD_MCLKDIV(n) (((n) & 0xff) << 0) +#define DA_CLKD_MCLKDIV_MASK (0xf << 0) +#define DA_CLKD_MCLKDIV_1 0 +#define DA_TXCNT 0x28 +#define DA_RXCNT 0x2c +#define DA_CHCFG 0x30 /* sun8i */ +#define DA_CHCFG_TX_SLOT_HIZ (1 << 9) +#define DA_CHCFG_TXN_STATE (1 << 8) +#define DA_CHCFG_RX_SLOT_NUM_MASK (7 << 4) +#define DA_CHCFG_RX_SLOT_NUM(n) (((n) & 7) << 4) +#define DA_CHCFG_TX_SLOT_NUM_MASK (7 << 0) +#define DA_CHCFG_TX_SLOT_NUM(n) (((n) & 7) << 0) + +#define DA_CHSEL_OFFSET(n) (((n) & 3) << 12) /* sun8i */ +#define DA_CHSEL_OFFSET_MASK (3 << 12) /* sun8i */ +#define DA_CHSEL_EN(n) (((n) & 0xff) << 4) +#define DA_CHSEL_EN_MASK (0xff << 4) +#define DA_CHSEL_SEL(n) (((n) & 7) << 0) +#define DA_CHSEL_SEL_MASK (7 << 0) + +#define AUDIO_BUFFER_SIZE 48000 * 4 + +#define AW_I2S_SAMPLE_RATE 48000 +#define AW_I2S_CLK_RATE 24576000 + +enum sunxi_i2s_type { + SUNXI_I2S_SUN4I, + SUNXI_I2S_SUN8I, +}; + +struct sunxi_i2s_config { + const char *name; + enum sunxi_i2s_type type; + bus_size_t txchsel; + bus_size_t txchmap; + bus_size_t rxchsel; + bus_size_t rxchmap; +}; + +static const struct sunxi_i2s_config sun50i_a64_codec_config = { + .name = "Audio Codec (digital part)", + .type = SUNXI_I2S_SUN4I, + .txchsel = 0x30, + .txchmap = 0x34, + .rxchsel = 0x38, + .rxchmap = 0x3c, +}; + +static const struct sunxi_i2s_config sun8i_h3_config = { + .name = "I2S/PCM controller", + .type = SUNXI_I2S_SUN8I, + .txchsel = 0x34, + .txchmap = 0x44, + .rxchsel = 0x54, + .rxchmap = 0x58, +}; + +static const u_int sun4i_i2s_bclk_divmap[] = { + [0] = 2, + [1] = 4, + [2] = 6, + [3] = 8, + [4] = 12, + [5] = 16, +}; + +static const u_int sun4i_i2s_mclk_divmap[] = { + [0] = 1, + [1] = 2, + [2] = 4, + [3] = 6, + [4] = 8, + [5] = 12, + [6] = 16, + [7] = 24, +}; + +static const u_int sun8i_i2s_divmap[] = { + [1] = 1, + [2] = 2, + [3] = 4, + [4] = 6, + [5] = 8, + [6] = 12, + [7] = 16, + [8] = 24, + [9] = 32, + [10] = 48, + [11] = 64, + [12] = 96, + [13] = 128, + [14] = 176, + [15] = 192, +}; + + +static struct ofw_compat_data compat_data[] = { + { "allwinner,sun50i-a64-codec-i2s", (uintptr_t)&sun50i_a64_codec_config }, + { "allwinner,sun8i-h3-i2s", (uintptr_t)&sun8i_h3_config }, + { NULL, 0 } +}; + +static struct resource_spec aw_i2s_spec[] = { + { SYS_RES_MEMORY, 0, RF_ACTIVE }, + { SYS_RES_IRQ, 0, RF_ACTIVE | RF_SHAREABLE }, + { -1, 0 } +}; + +struct aw_i2s_softc { + device_t dev; + struct resource *res[2]; + struct mtx mtx; + clk_t clk; + struct sunxi_i2s_config *cfg; + void * intrhand; + /* pointers to playback/capture buffers */ + uint32_t play_ptr; + uint32_t rec_ptr; +}; + +#define I2S_LOCK(sc) mtx_lock(&(sc)->mtx) +#define I2S_UNLOCK(sc) mtx_unlock(&(sc)->mtx) +#define I2S_READ(sc, reg) bus_read_4((sc)->res[0], (reg)) +#define I2S_WRITE(sc, reg, val) bus_write_4((sc)->res[0], (reg), (val)) +#define I2S_TYPE(sc) ((sc)->cfg->type) + +static int aw_i2s_probe(device_t dev); +static int aw_i2s_attach(device_t dev); +static int aw_i2s_detach(device_t dev); + +static u_int +sunxi_i2s_div_to_regval(const u_int *divmap, u_int divmaplen, u_int div) +{ + u_int n; + + for (n = 0; n < divmaplen; n++) + if (divmap[n] == div) + return n; + + return -1; +} + +static uint32_t sc_fmt[] = { + SND_FORMAT(AFMT_S16_LE, 2, 0), + 0 +}; +static struct pcmchan_caps aw_i2s_caps = {AW_I2S_SAMPLE_RATE, AW_I2S_SAMPLE_RATE, sc_fmt, 0}; + + +static int +aw_i2s_init(struct aw_i2s_softc *sc) +{ + uint32_t val; + int error; + + error = clk_enable(sc->clk); + if (error != 0) { + device_printf(sc->dev, "cannot enable mod clock\n"); + return (ENXIO); + } + + /* Reset */ + val = I2S_READ(sc, DA_CTL); + val &= ~(DA_CTL_TXEN|DA_CTL_RXEN|DA_CTL_GEN); + I2S_WRITE(sc, DA_CTL, val); + + val = I2S_READ(sc, DA_FCTL); + val &= ~(DA_FCTL_FTX|DA_FCTL_FRX); + val &= ~(DA_FCTL_TXTL_MASK); + val |= DA_FCTL_TXTL(FIFO_LEVEL); + I2S_WRITE(sc, DA_FCTL, val); + + I2S_WRITE(sc, DA_TXCNT, 0); + I2S_WRITE(sc, DA_RXCNT, 0); + + /* Enable */ + val = I2S_READ(sc, DA_CTL); + val |= DA_CTL_GEN; + I2S_WRITE(sc, DA_CTL, val); + val |= DA_CTL_SDO_EN; + I2S_WRITE(sc, DA_CTL, val); + + /* Setup channels */ + I2S_WRITE(sc, sc->cfg->txchmap, 0x76543210); + val = I2S_READ(sc, sc->cfg->txchsel); + val &= ~DA_CHSEL_EN_MASK; + val |= DA_CHSEL_EN(3); + val &= ~DA_CHSEL_SEL_MASK; + val |= DA_CHSEL_SEL(1); + I2S_WRITE(sc, sc->cfg->txchsel, val); + I2S_WRITE(sc, sc->cfg->rxchmap, 0x76543210); + val = I2S_READ(sc, sc->cfg->rxchsel); + val &= ~DA_CHSEL_EN_MASK; + val |= DA_CHSEL_EN(3); + val &= ~DA_CHSEL_SEL_MASK; + val |= DA_CHSEL_SEL(1); + I2S_WRITE(sc, sc->cfg->rxchsel, val); + + if (I2S_TYPE(sc) == SUNXI_I2S_SUN8I) { + val = I2S_READ(sc, DA_CHCFG); + val &= ~DA_CHCFG_TX_SLOT_NUM_MASK; + val |= DA_CHCFG_TX_SLOT_NUM(1); + val &= ~DA_CHCFG_RX_SLOT_NUM_MASK; + val |= DA_CHCFG_RX_SLOT_NUM(1); + I2S_WRITE(sc, DA_CHCFG, val); + } + + return (0); +} + +static int +aw_i2s_probe(device_t dev) +{ + if (!ofw_bus_status_okay(dev)) + return (ENXIO); + + if (!ofw_bus_search_compatible(dev, compat_data)->ocd_data) + return (ENXIO); + + device_set_desc(dev, "Rockchip I2S"); + return (BUS_PROBE_DEFAULT); +} + +static int +aw_i2s_attach(device_t dev) +{ + struct aw_i2s_softc *sc; + int error; + phandle_t node; + hwreset_t rst; + clk_t clk; + + sc = device_get_softc(dev); + sc->dev = dev; + + sc->cfg = (void*)ofw_bus_search_compatible(dev, compat_data)->ocd_data; + + mtx_init(&sc->mtx, device_get_nameunit(dev), NULL, MTX_DEF); + + if (bus_alloc_resources(dev, aw_i2s_spec, sc->res) != 0) { + device_printf(dev, "cannot allocate resources for device\n"); + error = ENXIO; + goto fail; + } + + error = clk_get_by_ofw_name(dev, 0, "mod", &sc->clk); + if (error != 0) { + device_printf(dev, "cannot get i2s_clk clock\n"); + goto fail; + } + + error = clk_get_by_ofw_name(dev, 0, "apb", &clk); + if (error != 0) { + device_printf(dev, "cannot get APB clock\n"); + goto fail; + } + + error = clk_enable(clk); + if (error != 0) { + device_printf(dev, "cannot enable APB clock\n"); + goto fail; + } + + if (hwreset_get_by_ofw_idx(dev, 0, 0, &rst) == 0) { + error = hwreset_deassert(rst); + if (error != 0) { + device_printf(dev, "cannot de-assert reset\n"); + goto fail; + } + } + + aw_i2s_init(sc); + + node = ofw_bus_get_node(dev); + OF_device_register_xref(OF_xref_from_node(node), dev); + + return (0); + +fail: + aw_i2s_detach(dev); + return (error); +} + +static int +aw_i2s_detach(device_t dev) +{ + struct aw_i2s_softc *i2s; + + i2s = device_get_softc(dev); + + if (i2s->clk) + clk_release(i2s->clk); + + if (i2s->intrhand != NULL) + bus_teardown_intr(i2s->dev, i2s->res[1], i2s->intrhand); + + bus_release_resources(dev, aw_i2s_spec, i2s->res); + mtx_destroy(&i2s->mtx); + + return (0); +} + +static int +aw_i2s_dai_init(device_t dev, uint32_t format) +{ + struct aw_i2s_softc *sc; + int fmt, pol, clk; + uint32_t ctl, fat0, chsel; + u_int offset; + + sc = device_get_softc(dev); + + fmt = AUDIO_DAI_FORMAT_FORMAT(format); + pol = AUDIO_DAI_FORMAT_POLARITY(format); + clk = AUDIO_DAI_FORMAT_CLOCK(format); + + ctl = I2S_READ(sc, DA_CTL); + fat0 = I2S_READ(sc, DA_FAT0); + + if (I2S_TYPE(sc) == SUNXI_I2S_SUN4I) { + fat0 &= ~DA_FAT0_FMT_MASK; + switch (fmt) { + case AUDIO_DAI_FORMAT_I2S: + fat0 |= DA_FMT_I2S; + break; + case AUDIO_DAI_FORMAT_RJ: + fat0 |= DA_FMT_RJ; + break; + case AUDIO_DAI_FORMAT_LJ: + fat0 |= DA_FMT_LJ; + break; + default: + return EINVAL; + } + ctl &= ~DA_CTL_PCM; + } else { + ctl &= ~DA_CTL_MODE_SEL_MASK; + switch (fmt) { + case AUDIO_DAI_FORMAT_I2S: + ctl |= DA_CTL_MODE_SEL_LJ; + offset = 1; + break; + case AUDIO_DAI_FORMAT_LJ: + ctl |= DA_CTL_MODE_SEL_LJ; + offset = 0; + break; + case AUDIO_DAI_FORMAT_RJ: + ctl |= DA_CTL_MODE_SEL_RJ; + offset = 0; + break; + case AUDIO_DAI_FORMAT_DSPA: + ctl |= DA_CTL_MODE_SEL_PCM; + offset = 1; + break; + case AUDIO_DAI_FORMAT_DSPB: + ctl |= DA_CTL_MODE_SEL_PCM; + offset = 0; + break; + default: + return EINVAL; + } + + chsel = I2S_READ(sc, sc->cfg->txchsel); + chsel &= ~DA_CHSEL_OFFSET_MASK; + chsel |= DA_CHSEL_OFFSET(offset); + I2S_WRITE(sc, sc->cfg->txchsel, chsel); + + chsel = I2S_READ(sc, sc->cfg->rxchsel); + chsel &= ~DA_CHSEL_OFFSET_MASK; + chsel |= DA_CHSEL_OFFSET(offset); + I2S_WRITE(sc, sc->cfg->rxchsel, chsel); + } + + fat0 &= ~(DA_FAT0_LRCP_MASK|DA_FAT0_BCP_MASK); + if (I2S_TYPE(sc) == SUNXI_I2S_SUN4I) { + if (AUDIO_DAI_POLARITY_INVERTED_BCLK(pol)) + fat0 |= DA_BCP_INVERTED; + if (AUDIO_DAI_POLARITY_INVERTED_FRAME(pol)) + fat0 |= DA_LRCP_INVERTED; + } else { + if (AUDIO_DAI_POLARITY_INVERTED_BCLK(pol)) + fat0 |= DA_BCP_INVERTED; + if (!AUDIO_DAI_POLARITY_INVERTED_FRAME(pol)) + fat0 |= DA_LRCP_INVERTED; + + fat0 &= ~DA_FAT0_LRCK_PERIOD_MASK; + fat0 |= DA_FAT0_LRCK_PERIOD(32 - 1); + } + + I2S_WRITE(sc, DA_CTL, ctl); + I2S_WRITE(sc, DA_FAT0, fat0); + + return (0); +} + + +static int +aw_i2s_dai_intr(device_t dev, struct snd_dbuf *play_buf, struct snd_dbuf *rec_buf) +{ + struct aw_i2s_softc *sc; + int ret = 0; + uint32_t val, status; + + sc = device_get_softc(dev); + + I2S_LOCK(sc); + + status = I2S_READ(sc, DA_ISTA); + /* Clear interrupts */ + // device_printf(sc->dev, "status: %08x\n", status); + I2S_WRITE(sc, DA_ISTA, status); + + if (status & DA_ISTA_TXEI_INT) { + uint8_t *samples; + uint32_t count, size, readyptr, written, empty; + + val = I2S_READ(sc, DA_FSTA); + empty = DA_FSTA_TXE_CNT(val); + count = sndbuf_getready(play_buf); + size = sndbuf_getsize(play_buf); + readyptr = sndbuf_getreadyptr(play_buf); + + samples = (uint8_t*)sndbuf_getbuf(play_buf); + written = 0; + if (empty > count / 2) + empty = count / 2; + for (; empty > 0; empty--) { + val = (samples[readyptr++ % size] << 16); + val |= (samples[readyptr++ % size] << 24); + written += 2; + I2S_WRITE(sc, DA_TXFIFO, val); + } + sc->play_ptr += written; + sc->play_ptr %= size; + ret |= AUDIO_DAI_PLAY_INTR; + } + + if (status & DA_ISTA_RXAI_INT) { + uint8_t *samples; + uint32_t count, size, freeptr, recorded, available; + + val = I2S_READ(sc, DA_FSTA); + available = DA_FSTA_RXA_CNT(val); + + count = sndbuf_getfree(rec_buf); + size = sndbuf_getsize(rec_buf); + freeptr = sndbuf_getfreeptr(rec_buf); + samples = (uint8_t*)sndbuf_getbuf(rec_buf); + recorded = 0; + if (available > count / 2) + available = count / 2; + + for (; available > 0; available--) { + val = I2S_READ(sc, DA_RXFIFO); + samples[freeptr++ % size] = (val >> 16) & 0xff; + samples[freeptr++ % size] = (val >> 24) & 0xff; + recorded += 2; + } + sc->rec_ptr += recorded; + sc->rec_ptr %= size; + ret |= AUDIO_DAI_REC_INTR; + } + + I2S_UNLOCK(sc); + + return (ret); +} + +static struct pcmchan_caps * +aw_i2s_dai_get_caps(device_t dev) +{ + return (&aw_i2s_caps); +} + +static int +aw_i2s_dai_trigger(device_t dev, int go, int pcm_dir) +{ + struct aw_i2s_softc *sc = device_get_softc(dev); + uint32_t val; + + if ((pcm_dir != PCMDIR_PLAY) && (pcm_dir != PCMDIR_REC)) + return (EINVAL); + + switch (go) { + case PCMTRIG_START: + if (pcm_dir == PCMDIR_PLAY) { + /* Flush FIFO */ + val = I2S_READ(sc, DA_FCTL); + I2S_WRITE(sc, DA_FCTL, val | DA_FCTL_FTX); + I2S_WRITE(sc, DA_FCTL, val & ~DA_FCTL_FTX); + + /* Reset TX sample counter */ + I2S_WRITE(sc, DA_TXCNT, 0); + + /* Enable TX block */ + val = I2S_READ(sc, DA_CTL); + I2S_WRITE(sc, DA_CTL, val | DA_CTL_TXEN); + + /* Enable TX underrun interrupt */ + val = I2S_READ(sc, DA_INT); + I2S_WRITE(sc, DA_INT, val | DA_INT_TXEI_EN); + } + + if (pcm_dir == PCMDIR_REC) { + /* Flush FIFO */ + val = I2S_READ(sc, DA_FCTL); + I2S_WRITE(sc, DA_FCTL, val | DA_FCTL_FRX); + I2S_WRITE(sc, DA_FCTL, val & ~DA_FCTL_FRX); + + /* Reset RX sample counter */ + I2S_WRITE(sc, DA_RXCNT, 0); + + /* Enable RX block */ + val = I2S_READ(sc, DA_CTL); + I2S_WRITE(sc, DA_CTL, val | DA_CTL_RXEN); + + /* Enable RX data available interrupt */ + val = I2S_READ(sc, DA_INT); + I2S_WRITE(sc, DA_INT, val | DA_INT_RXAI_EN); + } + + break; + + case PCMTRIG_STOP: + case PCMTRIG_ABORT: + I2S_LOCK(sc); + + if (pcm_dir == PCMDIR_PLAY) { + /* Disable TX block */ + val = I2S_READ(sc, DA_CTL); + I2S_WRITE(sc, DA_CTL, val & ~DA_CTL_TXEN); + + /* Enable TX underrun interrupt */ + val = I2S_READ(sc, DA_INT); + I2S_WRITE(sc, DA_INT, val & ~DA_INT_TXEI_EN); + + sc->play_ptr = 0; + } else { + /* Disable RX block */ + val = I2S_READ(sc, DA_CTL); + I2S_WRITE(sc, DA_CTL, val & ~DA_CTL_RXEN); + + /* Disable RX data available interrupt */ + val = I2S_READ(sc, DA_INT); + I2S_WRITE(sc, DA_INT, val & ~DA_INT_RXAI_EN); + + sc->rec_ptr = 0; + } + + I2S_UNLOCK(sc); + break; + } + + return (0); +} + +static uint32_t +aw_i2s_dai_get_ptr(device_t dev, int pcm_dir) +{ + struct aw_i2s_softc *sc; + uint32_t ptr; + + sc = device_get_softc(dev); + + I2S_LOCK(sc); + if (pcm_dir == PCMDIR_PLAY) + ptr = sc->play_ptr; + else + ptr = sc->rec_ptr; + I2S_UNLOCK(sc); + + return ptr; +} + +static int +aw_i2s_dai_setup_intr(device_t dev, driver_intr_t intr_handler, void *intr_arg) +{ + struct aw_i2s_softc *sc = device_get_softc(dev); + + if (bus_setup_intr(dev, sc->res[1], + INTR_TYPE_MISC | INTR_MPSAFE, NULL, intr_handler, intr_arg, + &sc->intrhand)) { + device_printf(dev, "cannot setup interrupt handler\n"); + return (ENXIO); + } + + return (0); +} + +static uint32_t +aw_i2s_dai_set_chanformat(device_t dev, uint32_t format) +{ + + return (0); +} + +static int +aw_i2s_dai_set_sysclk(device_t dev, unsigned int rate, int dai_dir) +{ + struct aw_i2s_softc *sc; + int bclk_val, mclk_val; + uint32_t val; + int error; + + sc = device_get_softc(dev); + + error = clk_set_freq(sc->clk, AW_I2S_CLK_RATE, CLK_SET_ROUND_DOWN); + if (error != 0) { + device_printf(sc->dev, + "couldn't set mod clock rate to %u Hz: %d\n", AW_I2S_CLK_RATE, error); + return error; + } + error = clk_enable(sc->clk); + if (error != 0) { + device_printf(sc->dev, + "couldn't enable mod clock: %d\n", error); + return error; + } + + const u_int bclk_prate = I2S_TYPE(sc) == SUNXI_I2S_SUN4I ? rate : AW_I2S_CLK_RATE; + + const u_int bclk_div = bclk_prate / (2 * 32 * AW_I2S_SAMPLE_RATE); + const u_int mclk_div = AW_I2S_CLK_RATE / rate; + + if (I2S_TYPE(sc) == SUNXI_I2S_SUN4I) { + bclk_val = sunxi_i2s_div_to_regval(sun4i_i2s_bclk_divmap, + nitems(sun4i_i2s_bclk_divmap), bclk_div); + mclk_val = sunxi_i2s_div_to_regval(sun4i_i2s_mclk_divmap, + nitems(sun4i_i2s_mclk_divmap), mclk_div); + } else { + bclk_val = sunxi_i2s_div_to_regval(sun8i_i2s_divmap, + nitems(sun8i_i2s_divmap), bclk_div); + mclk_val = sunxi_i2s_div_to_regval(sun8i_i2s_divmap, + nitems(sun8i_i2s_divmap), mclk_div); + } + if (bclk_val == -1 || mclk_val == -1) { + device_printf(sc->dev, "couldn't configure bclk/mclk dividers\n"); + return EIO; + } + + val = I2S_READ(sc, DA_CLKD); + if (I2S_TYPE(sc) == SUNXI_I2S_SUN4I) { + val |= DA_CLKD_MCLKO_EN_SUN4I; + val &= ~DA_CLKD_BCLKDIV_SUN4I_MASK; + val |= DA_CLKD_BCLKDIV_SUN4I(bclk_val); + } else { + val |= DA_CLKD_MCLKO_EN_SUN8I; + val &= ~DA_CLKD_BCLKDIV_SUN8I_MASK; + val |= DA_CLKD_BCLKDIV_SUN8I(bclk_val); + } + val &= ~DA_CLKD_MCLKDIV_MASK; + val |= DA_CLKD_MCLKDIV(mclk_val); + I2S_WRITE(sc, DA_CLKD, val); + + + return (0); +} + +static uint32_t +aw_i2s_dai_set_chanspeed(device_t dev, uint32_t speed) +{ + + return (speed); +} + +static device_method_t aw_i2s_methods[] = { + /* Device interface */ + DEVMETHOD(device_probe, aw_i2s_probe), + DEVMETHOD(device_attach, aw_i2s_attach), + DEVMETHOD(device_detach, aw_i2s_detach), + + DEVMETHOD(audio_dai_init, aw_i2s_dai_init), + DEVMETHOD(audio_dai_setup_intr, aw_i2s_dai_setup_intr), + DEVMETHOD(audio_dai_set_sysclk, aw_i2s_dai_set_sysclk), + DEVMETHOD(audio_dai_set_chanspeed, aw_i2s_dai_set_chanspeed), + DEVMETHOD(audio_dai_set_chanformat, aw_i2s_dai_set_chanformat), + DEVMETHOD(audio_dai_intr, aw_i2s_dai_intr), + DEVMETHOD(audio_dai_get_caps, aw_i2s_dai_get_caps), + DEVMETHOD(audio_dai_trigger, aw_i2s_dai_trigger), + DEVMETHOD(audio_dai_get_ptr, aw_i2s_dai_get_ptr), + + DEVMETHOD_END +}; + +static driver_t aw_i2s_driver = { + "i2s", + aw_i2s_methods, + sizeof(struct aw_i2s_softc), +}; + +static devclass_t aw_i2s_devclass; + +DRIVER_MODULE(aw_i2s, simplebus, aw_i2s_driver, aw_i2s_devclass, 0, 0); +SIMPLEBUS_PNP_INFO(compat_data); Index: sys/arm/allwinner/sun8i_codec.c =================================================================== --- /dev/null +++ sys/arm/allwinner/sun8i_codec.c @@ -0,0 +1,417 @@ +/*- + * SPDX-License-Identifier: BSD-2-Clause-FreeBSD + * + * Copyright (c) 2020 Oleksandr Tymoshenko + * Copyright (c) 2018 Jared McNeill + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions + * are met: + * 1. Redistributions of source code must retain the above copyright + * notice, this list of conditions and the following disclaimer. + * 2. Redistributions in binary form must reproduce the above copyright + * notice, this list of conditions and the following disclaimer in the + * documentation and/or other materials provided with the distribution. + * + * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR + * IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES + * OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. + * IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, + * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, + * BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; + * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED + * AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, + * OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY + * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF + * SUCH DAMAGE. + * + * $FreeBSD$ + */ + +#include +__FBSDID("$FreeBSD$"); + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include + +#include +#include + +#include + +#include "opt_snd.h" +#include +#include +#include "audio_dai_if.h" + +#define SYSCLK_CTL 0x00c +#define AIF1CLK_ENA (1 << 11) +#define AIF1CLK_SRC_MASK (3 << 8) +#define AIF1CLK_SRC_PLL (2 << 8) +#define SYSCLK_ENA (1 << 3) +#define SYSCLK_SRC (1 << 0) + +#define MOD_CLK_ENA 0x010 +#define MOD_RST_CTL 0x014 +#define MOD_AIF1 (1 << 15) +#define MOD_ADC (1 << 3) +#define MOD_DAC (1 << 2) + +#define SYS_SR_CTRL 0x018 +#define AIF1_FS_MASK (0xf << 12) +#define AIF_FS_48KHZ (8 << 12) + +#define AIF1CLK_CTRL 0x040 +#define AIF1_MSTR_MOD (1 << 15) +#define AIF1_BCLK_INV (1 << 14) +#define AIF1_LRCK_INV (1 << 13) +#define AIF1_BCLK_DIV_MASK (0xf << 9) +#define AIF1_BCLK_DIV_16 (6 << 9) +#define AIF1_LRCK_DIV_MASK (7 << 6) +#define AIF1_LRCK_DIV_16 (0 << 6) +#define AIF1_LRCK_DIV_64 (2 << 6) +#define AIF1_WORD_SIZ_MASK (3 << 4) +#define AIF1_WORD_SIZ_16 (1 << 4) +#define AIF1_DATA_FMT_MASK (3 << 2) +#define AIF1_DATA_FMT_I2S (0 << 2) +#define AIF1_DATA_FMT_LJ (1 << 2) +#define AIF1_DATA_FMT_RJ (2 << 2) +#define AIF1_DATA_FMT_DSP (3 << 2) + +#define AIF1_ADCDAT_CTRL 0x044 +#define AIF1_ADC0L_ENA (1 << 15) +#define AIF1_ADC0R_ENA (1 << 14) + +#define AIF1_DACDAT_CTRL 0x048 +#define AIF1_DAC0L_ENA (1 << 15) +#define AIF1_DAC0R_ENA (1 << 14) + +#define AIF1_MXR_SRC 0x04c +#define AIF1L_MXR_SRC_MASK (0xf << 12) +#define AIF1L_MXR_SRC_AIF1 (0x8 << 12) +#define AIF1L_MXR_SRC_ADC (0x2 << 12) +#define AIF1R_MXR_SRC_MASK (0xf << 8) +#define AIF1R_MXR_SRC_AIF1 (0x8 << 8) +#define AIF1R_MXR_SRC_ADC (0x2 << 8) + +#define ADC_DIG_CTRL 0x100 +#define ADC_DIG_CTRL_ENAD (1 << 15) + +#define HMIC_CTRL1 0x110 +#define HMIC_CTRL1_N_MASK (0xf << 8) +#define HMIC_CTRL1_N(n) (((n) & 0xf) << 8) +#define HMIC_CTRL1_JACK_IN_IRQ_EN (1 << 4) +#define HMIC_CTRL1_JACK_OUT_IRQ_EN (1 << 3) +#define HMIC_CTRL1_MIC_DET_IRQ_EN (1 << 0) + +#define HMIC_CTRL2 0x114 +#define HMIC_CTRL2_MDATA_THRES __BITS(12,8) + +#define HMIC_STS 0x118 +#define HMIC_STS_MIC_PRESENT (1 << 6) +#define HMIC_STS_JACK_DET_OIRQ (1 << 4) +#define HMIC_STS_JACK_DET_IIRQ (1 << 3) +#define HMIC_STS_MIC_DET_ST (1 << 0) + +#define DAC_DIG_CTRL 0x120 +#define DAC_DIG_CTRL_ENDA (1 << 15) + +#define DAC_MXR_SRC 0x130 +#define DACL_MXR_SRC_MASK (0xf << 12) +#define DACL_MXR_SRC_AIF1_DAC0L (0x8 << 12) +#define DACR_MXR_SRC_MASK (0xf << 8) +#define DACR_MXR_SRC_AIF1_DAC0R (0x8 << 8) + +static struct ofw_compat_data compat_data[] = { + { "allwinner,sun8i-a33-codec", 1}, + { NULL, 0 } +}; + +static struct resource_spec sun8i_codec_spec[] = { + { SYS_RES_MEMORY, 0, RF_ACTIVE }, + { SYS_RES_IRQ, 0, RF_ACTIVE | RF_SHAREABLE }, + { -1, 0 } +}; + +struct sun8i_codec_softc { + device_t dev; + struct resource *res[2]; + struct mtx mtx; + clk_t clk_gate; + clk_t clk_mod; + void * intrhand; +}; + +#define CODEC_LOCK(sc) mtx_lock(&(sc)->mtx) +#define CODEC_UNLOCK(sc) mtx_unlock(&(sc)->mtx) +#define CODEC_READ(sc, reg) bus_read_4((sc)->res[0], (reg)) +#define CODEC_WRITE(sc, reg, val) bus_write_4((sc)->res[0], (reg), (val)) + +static int sun8i_codec_probe(device_t dev); +static int sun8i_codec_attach(device_t dev); +static int sun8i_codec_detach(device_t dev); + +static int +sun8i_codec_probe(device_t dev) +{ + if (!ofw_bus_status_okay(dev)) + return (ENXIO); + + if (!ofw_bus_search_compatible(dev, compat_data)->ocd_data) + return (ENXIO); + + device_set_desc(dev, "Allwinner Codec"); + return (BUS_PROBE_DEFAULT); +} + +static int +sun8i_codec_attach(device_t dev) +{ + struct sun8i_codec_softc *sc; + int error; + uint32_t val; + struct gpiobus_pin *pa_pin; + phandle_t node; + + sc = device_get_softc(dev); + sc->dev = dev; + node = ofw_bus_get_node(dev); + + mtx_init(&sc->mtx, device_get_nameunit(dev), NULL, MTX_DEF); + + if (bus_alloc_resources(dev, sun8i_codec_spec, sc->res) != 0) { + device_printf(dev, "cannot allocate resources for device\n"); + error = ENXIO; + goto fail; + } + + error = clk_get_by_ofw_name(dev, 0, "mod", &sc->clk_mod); + if (error != 0) { + device_printf(dev, "cannot get \"mod\" clock\n"); + goto fail; + } + + error = clk_get_by_ofw_name(dev, 0, "bus", &sc->clk_gate); + if (error != 0) { + device_printf(dev, "cannot get \"bus\" clock\n"); + goto fail; + } + + error = clk_enable(sc->clk_gate); + if (error != 0) { + device_printf(dev, "cannot enable \"bus\" clock\n"); + goto fail; + } + + /* Enable clocks */ + val = CODEC_READ(sc, SYSCLK_CTL); + val |= AIF1CLK_ENA; + val &= ~AIF1CLK_SRC_MASK; + val |= AIF1CLK_SRC_PLL; + val |= SYSCLK_ENA; + val &= ~SYSCLK_SRC; + CODEC_WRITE(sc, SYSCLK_CTL, val); + CODEC_WRITE(sc, MOD_CLK_ENA, MOD_AIF1 | MOD_ADC | MOD_DAC); + CODEC_WRITE(sc, MOD_RST_CTL, MOD_AIF1 | MOD_ADC | MOD_DAC); + + /* Enable digital parts */ + CODEC_WRITE(sc, DAC_DIG_CTRL, DAC_DIG_CTRL_ENDA); + CODEC_WRITE(sc, ADC_DIG_CTRL, ADC_DIG_CTRL_ENAD); + + /* Set AIF1 to 48 kHz */ + val = CODEC_READ(sc, SYS_SR_CTRL); + val &= ~AIF1_FS_MASK; + val |= AIF_FS_48KHZ; + CODEC_WRITE(sc, SYS_SR_CTRL, val); + + /* Set AIF1 to 16-bit */ + val = CODEC_READ(sc, AIF1CLK_CTRL); + val &= ~AIF1_WORD_SIZ_MASK; + val |= AIF1_WORD_SIZ_16; + CODEC_WRITE(sc, AIF1CLK_CTRL, val); + + /* Enable AIF1 DAC timelot 0 */ + val = CODEC_READ(sc, AIF1_DACDAT_CTRL); + val |= AIF1_DAC0L_ENA; + val |= AIF1_DAC0R_ENA; + CODEC_WRITE(sc, AIF1_DACDAT_CTRL, val); + + /* Enable AIF1 ADC timelot 0 */ + val = CODEC_READ(sc, AIF1_ADCDAT_CTRL); + val |= AIF1_ADC0L_ENA; + val |= AIF1_ADC0R_ENA; + CODEC_WRITE(sc, AIF1_ADCDAT_CTRL, val); + + /* DAC mixer source select */ + val = CODEC_READ(sc, DAC_MXR_SRC); + val &= ~DACL_MXR_SRC_MASK; + val |= DACL_MXR_SRC_AIF1_DAC0L; + val &= ~DACR_MXR_SRC_MASK; + val |= DACR_MXR_SRC_AIF1_DAC0R; + CODEC_WRITE(sc, DAC_MXR_SRC, val); + + /* ADC mixer source select */ + val = CODEC_READ(sc, AIF1_MXR_SRC); + val &= ~AIF1L_MXR_SRC_MASK; + val |= AIF1L_MXR_SRC_ADC; + val &= ~AIF1R_MXR_SRC_MASK; + val |= AIF1R_MXR_SRC_ADC; + CODEC_WRITE(sc, AIF1_MXR_SRC, val); + + /* Enable PA power */ + /* Unmute PA */ + if (gpio_pin_get_by_ofw_property(dev, node, "allwinner,pa-gpios", + &pa_pin) == 0) { + error = gpio_pin_set_active(pa_pin, 1); + if (error != 0) + device_printf(dev, "failed to unmute PA\n"); + } + + OF_device_register_xref(OF_xref_from_node(node), dev); + + return (0); + +fail: + sun8i_codec_detach(dev); + return (error); +} + +static int +sun8i_codec_detach(device_t dev) +{ + struct sun8i_codec_softc *sc; + + sc = device_get_softc(dev); + + if (sc->clk_gate) + clk_release(sc->clk_gate); + + if (sc->clk_mod) + clk_release(sc->clk_mod); + + if (sc->intrhand != NULL) + bus_teardown_intr(sc->dev, sc->res[1], sc->intrhand); + + bus_release_resources(dev, sun8i_codec_spec, sc->res); + mtx_destroy(&sc->mtx); + + return (0); +} + +static int +sun8i_codec_dai_init(device_t dev, uint32_t format) +{ + struct sun8i_codec_softc *sc; + int fmt, pol, clk; + uint32_t val; + + sc = device_get_softc(dev); + + fmt = AUDIO_DAI_FORMAT_FORMAT(format); + pol = AUDIO_DAI_FORMAT_POLARITY(format); + clk = AUDIO_DAI_FORMAT_CLOCK(format); + + val = CODEC_READ(sc, AIF1CLK_CTRL); + + val &= ~AIF1_DATA_FMT_MASK; + switch (fmt) { + case AUDIO_DAI_FORMAT_I2S: + val |= AIF1_DATA_FMT_I2S; + break; + case AUDIO_DAI_FORMAT_RJ: + val |= AIF1_DATA_FMT_RJ; + break; + case AUDIO_DAI_FORMAT_LJ: + val |= AIF1_DATA_FMT_LJ; + break; + case AUDIO_DAI_FORMAT_DSPA: + case AUDIO_DAI_FORMAT_DSPB: + val |= AIF1_DATA_FMT_DSP; + break; + default: + return EINVAL; + } + + val &= ~(AIF1_BCLK_INV|AIF1_LRCK_INV); + /* Codec LRCK polarity is inverted (datasheet is wrong) */ + if (!AUDIO_DAI_POLARITY_INVERTED_FRAME(pol)) + val |= AIF1_LRCK_INV; + if (AUDIO_DAI_POLARITY_INVERTED_BCLK(pol)) + val |= AIF1_BCLK_INV; + + switch (clk) { + case AUDIO_DAI_CLOCK_CBM_CFM: + val &= ~AIF1_MSTR_MOD; /* codec is master */ + break; + case AUDIO_DAI_CLOCK_CBS_CFS: + val |= AIF1_MSTR_MOD; /* codec is slave */ + break; + default: + return EINVAL; + } + + val &= ~AIF1_LRCK_DIV_MASK; + val |= AIF1_LRCK_DIV_64; + + val &= ~AIF1_BCLK_DIV_MASK; + val |= AIF1_BCLK_DIV_16; + + CODEC_WRITE(sc, AIF1CLK_CTRL, val); + + return (0); +} + +static int +sun8i_codec_dai_trigger(device_t dev, int go, int pcm_dir) +{ + + return (0); +} + +static int +sun8i_codec_dai_setup_mixer(device_t dev, device_t pcmdev) +{ + struct sun8i_codec_softc *sc; + + sc = device_get_softc(dev); + /* Do nothing for now */ + + return (0); +} + + +static device_method_t sun8i_codec_methods[] = { + /* Device interface */ + DEVMETHOD(device_probe, sun8i_codec_probe), + DEVMETHOD(device_attach, sun8i_codec_attach), + DEVMETHOD(device_detach, sun8i_codec_detach), + + DEVMETHOD(audio_dai_init, sun8i_codec_dai_init), + DEVMETHOD(audio_dai_setup_mixer, sun8i_codec_dai_setup_mixer), + DEVMETHOD(audio_dai_trigger, sun8i_codec_dai_trigger), + + DEVMETHOD_END +}; + +static driver_t sun8i_codec_driver = { + "sun8icodec", + sun8i_codec_methods, + sizeof(struct sun8i_codec_softc), +}; + +static devclass_t sun8i_codec_devclass; + +DRIVER_MODULE(sun8i_codec, simplebus, sun8i_codec_driver, sun8i_codec_devclass, 0, 0); +SIMPLEBUS_PNP_INFO(compat_data); Index: sys/conf/files.arm64 =================================================================== --- sys/conf/files.arm64 +++ sys/conf/files.arm64 @@ -28,9 +28,11 @@ arm/allwinner/a10_timer.c optional a10_timer fdt arm/allwinner/a10_codec.c optional sound a10_codec arm/allwinner/a31_dmac.c optional a31_dmac +arm/allwinner/a64/sun50i_a64_acodec.c optional fdt sound arm/allwinner/sunxi_dma_if.m optional a31_dmac arm/allwinner/aw_cir.c optional evdev aw_cir fdt arm/allwinner/aw_dwc3.c optional aw_dwc3 fdt +arm/allwinner/aw_i2s.c optional fdt sound arm/allwinner/aw_gpio.c optional gpio aw_gpio fdt arm/allwinner/aw_mmc.c optional mmc aw_mmc fdt | mmccam aw_mmc fdt arm/allwinner/aw_nmi.c optional aw_nmi fdt \ @@ -47,6 +49,7 @@ arm/allwinner/aw_wdog.c optional aw_wdog fdt arm/allwinner/axp81x.c optional axp81x fdt arm/allwinner/if_awg.c optional awg ext_resources syscon aw_sid nvmem fdt +arm/allwinner/sun8i_codec.c optional fdt sound # Allwinner clock driver arm/allwinner/clkng/aw_ccung.c optional aw_ccu fdt @@ -382,6 +385,10 @@ dev/safexcel/safexcel.c optional safexcel fdt dev/sdhci/sdhci_xenon.c optional sdhci_xenon sdhci fdt dev/uart/uart_cpu_arm64.c optional uart +dev/sound/fdt/audio_dai_if.m optional sound fdt +dev/sound/fdt/audio_soc.c optional sound fdt +dev/sound/fdt/dummy_codec.c optional sound fdt +dev/sound/fdt/simple_amplifier.c optional sound fdt dev/uart/uart_dev_mu.c optional uart uart_mu dev/uart/uart_dev_pl011.c optional uart pl011 dev/usb/controller/dwc_otg_hisi.c optional dwcotg fdt soc_hisi_hi6220 Index: sys/dev/sound/fdt/audio_dai.h =================================================================== --- /dev/null +++ sys/dev/sound/fdt/audio_dai.h @@ -0,0 +1,72 @@ +/*- + * Copyright (c) 2019 Oleksandr Tymoshenko + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions + * are met: + * 1. Redistributions of source code must retain the above copyright + * notice, this list of conditions and the following disclaimer. + * 2. Redistributions in binary form must reproduce the above copyright + * notice, this list of conditions and the following disclaimer in the + * documentation and/or other materials provided with the distribution. + * + * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR + * IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES + * OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. + * IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, + * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT + * NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, + * DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY + * THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT + * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF + * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. + * + * $FreeBSD$ + */ + +#ifndef __DAI_H__ +#define __DAI_H__ + +#define AUDIO_DAI_FORMAT_I2S 0 +#define AUDIO_DAI_FORMAT_RJ 1 +#define AUDIO_DAI_FORMAT_LJ 2 +#define AUDIO_DAI_FORMAT_DSPA 3 +#define AUDIO_DAI_FORMAT_DSPB 4 +#define AUDIO_DAI_FORMAT_AC97 5 +#define AUDIO_DAI_FORMAT_PDM 6 + +/* + * Polarity: Normal/Inverted BCLK/Frame + */ +#define AUDIO_DAI_POLARITY_NB_NF 0 +#define AUDIO_DAI_POLARITY_NB_IF 1 +#define AUDIO_DAI_POLARITY_IB_NF 2 +#define AUDIO_DAI_POLARITY_IB_IF 3 +#define AUDIO_DAI_POLARITY_INVERTED_FRAME(n) ((n) & 0x01) +#define AUDIO_DAI_POLARITY_INVERTED_BCLK(n) ((n) & 0x2) + +#define AUDIO_DAI_CLOCK_CBM_CFM 0 +#define AUDIO_DAI_CLOCK_CBS_CFM 1 +#define AUDIO_DAI_CLOCK_CBM_CFS 2 +#define AUDIO_DAI_CLOCK_CBS_CFS 3 + +#define AUDIO_DAI_CLOCK_IN 0 +#define AUDIO_DAI_CLOCK_OUT 1 + +#define AUDIO_DAI_JACK_HP 0 +#define AUDIO_DAI_JACK_MIC 1 + +/* + * Signal to audio_soc that chn_intr required + * for either recording or playback + */ +#define AUDIO_DAI_REC_INTR (1 << 1) +#define AUDIO_DAI_PLAY_INTR (1 << 0) + +#define AUDIO_DAI_FORMAT(fmt, pol, clk) (((fmt) << 16) | ((pol) << 8) | (clk)) +#define AUDIO_DAI_FORMAT_FORMAT(format) (((format) >> 16) & 0xff) +#define AUDIO_DAI_FORMAT_POLARITY(format) (((format) >> 8) & 0xff) +#define AUDIO_DAI_FORMAT_CLOCK(format) (((format) >> 0) & 0xff) + + +#endif /* __DAI_H__ */ Index: sys/dev/sound/fdt/audio_dai_if.m =================================================================== --- /dev/null +++ sys/dev/sound/fdt/audio_dai_if.m @@ -0,0 +1,95 @@ +#- +# Copyright (c) 2019 Oleksandr Tymoshenko +# +# Redistribution and use in source and binary forms, with or without +# modification, are permitted provided that the following conditions +# are met: +# 1. Redistributions of source code must retain the above copyright +# notice, this list of conditions and the following disclaimer. +# 2. Redistributions in binary form must reproduce the above copyright +# notice, this list of conditions and the following disclaimer in the +# documentation and/or other materials provided with the distribution. +# +# THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND +# ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +# IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +# ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE +# FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL +# DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS +# OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) +# HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT +# LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY +# OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF +# SUCH DAMAGE. +# +# $FreeBSD$ +# + +CODE { + #include + #include + #include +} + +INTERFACE audio_dai; + +# set DAI format for communications between CPU/codec nodes +METHOD int init { + device_t dev; + uint32_t format; +} + +# Initialize DAI and set up interrrupt handler +METHOD int setup_intr { + device_t dev; + driver_intr_t intr_handler; + void *intr_arg; +} + +# Setup mixers for codec node +METHOD int setup_mixer { + device_t dev; + device_t ausocdev; +} + +# setup clock speed +METHOD int set_sysclk { + device_t dev; + uint32_t rate; + int dai_dir; +} + +METHOD int trigger { + device_t dev; + int go; + int pcm_dir; +} + +METHOD struct pcmchan_caps* get_caps { + device_t dev; +} + +METHOD uint32_t get_ptr { + device_t dev; + int pcm_dir; +} + +# Set PCM channel format +METHOD uint32_t set_chanformat { + device_t dev; + uint32_t format; +} + +# Set PCM channel sampling rate +METHOD uint32_t set_chanspeed { + device_t dev; + uint32_t speed; +} + +# call DAI interrupt handler +# returns 1 if call to chn_intr required, 0 otherwise +METHOD int intr { + device_t dev; + struct snd_dbuf *play_buf; + struct snd_dbuf *rec_buf; +} Index: sys/dev/sound/fdt/audio_soc.c =================================================================== --- /dev/null +++ sys/dev/sound/fdt/audio_soc.c @@ -0,0 +1,546 @@ +/*- + * Copyright (c) 2019 Oleksandr Tymoshenko + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions + * are met: + * 1. Redistributions of source code must retain the above copyright + * notice, this list of conditions and the following disclaimer. + * 2. Redistributions in binary form must reproduce the above copyright + * notice, this list of conditions and the following disclaimer in the + * documentation and/or other materials provided with the distribution. + * + * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR + * IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES + * OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. + * IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, + * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT + * NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, + * DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY + * THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT + * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF + * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. + * + */ + +#include +__FBSDID("$FreeBSD$"); + +#include "opt_platform.h" + +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include + +#include +#include +#include "audio_dai_if.h" + +#define AUDIO_BUFFER_SIZE 48000 * 4 + +struct audio_soc_aux_node { + SLIST_ENTRY(audio_soc_aux_node) link; + device_t dev; +}; + +struct audio_soc_channel { + struct audio_soc_softc *sc; /* parent device's softc */ + struct pcm_channel *pcm; /* PCM channel */ + struct snd_dbuf *buf; /* PCM buffer */ + int dir; /* direction */ +}; + +struct audio_soc_softc { + /* + * pcm_register assumes that sc is snddev_info, + * so this has to be first structure member for "compatiblity" + */ + struct snddev_info info; + device_t dev; + char *name; + struct intr_config_hook init_hook; + device_t cpu_dev; + device_t codec_dev; + SLIST_HEAD(, audio_soc_aux_node) aux_devs; + unsigned int mclk_fs; + struct audio_soc_channel play_channel; + struct audio_soc_channel rec_channel; + /* + * The format is from the CPU node, for CODEC node clock roles + * need to be reversed. + */ + uint32_t format; + uint32_t link_mclk_fs; +}; + +static struct ofw_compat_data compat_data[] = { + {"simple-audio-card", 1}, + {NULL, 0}, +}; + +static struct { + const char *name; + unsigned int fmt; +} ausoc_dai_formats[] = { + { "i2s", AUDIO_DAI_FORMAT_I2S }, + { "right_j", AUDIO_DAI_FORMAT_RJ }, + { "left_j", AUDIO_DAI_FORMAT_LJ }, + { "dsp_a", AUDIO_DAI_FORMAT_DSPA }, + { "dsp_b", AUDIO_DAI_FORMAT_DSPB }, + { "ac97", AUDIO_DAI_FORMAT_AC97 }, + { "pdm", AUDIO_DAI_FORMAT_PDM }, +}; + +static int audio_soc_probe(device_t dev); +static int audio_soc_attach(device_t dev); +static int audio_soc_detach(device_t dev); + +/* + * Invert master/slave roles for CODEC side of the node + */ +static uint32_t +audio_soc_reverse_clocks(uint32_t format) +{ + int fmt, pol, clk; + + fmt = AUDIO_DAI_FORMAT_FORMAT(format); + pol = AUDIO_DAI_FORMAT_POLARITY(format); + clk = AUDIO_DAI_FORMAT_CLOCK(format); + + switch (clk) { + case AUDIO_DAI_CLOCK_CBM_CFM: + clk = AUDIO_DAI_CLOCK_CBS_CFS; + break; + case AUDIO_DAI_CLOCK_CBS_CFM: + clk = AUDIO_DAI_CLOCK_CBM_CFS; + break; + case AUDIO_DAI_CLOCK_CBM_CFS: + clk = AUDIO_DAI_CLOCK_CBS_CFM; + break; + case AUDIO_DAI_CLOCK_CBS_CFS: + clk = AUDIO_DAI_CLOCK_CBM_CFM; + break; + } + + return AUDIO_DAI_FORMAT(fmt, pol, clk); +} + +static uint32_t +audio_soc_chan_setblocksize(kobj_t obj, void *data, uint32_t blocksz) +{ + + return (blocksz); +} + +static int +audio_soc_chan_setformat(kobj_t obj, void *data, uint32_t format) +{ + + struct audio_soc_softc *sc; + struct audio_soc_channel *ausoc_chan; + + ausoc_chan = data; + sc = ausoc_chan->sc; + + return AUDIO_DAI_SET_CHANFORMAT(sc->cpu_dev, format); +} + +static uint32_t +audio_soc_chan_setspeed(kobj_t obj, void *data, uint32_t speed) +{ + + struct audio_soc_softc *sc; + struct audio_soc_channel *ausoc_chan; + uint32_t rate; + struct audio_soc_aux_node *aux_node; + + ausoc_chan = data; + sc = ausoc_chan->sc; + + if (sc->link_mclk_fs) { + rate = speed * sc->link_mclk_fs; + if (AUDIO_DAI_SET_SYSCLK(sc->cpu_dev, rate, AUDIO_DAI_CLOCK_IN)) + device_printf(sc->dev, "failed to set sysclk for CPU node\n"); + + if (AUDIO_DAI_SET_SYSCLK(sc->codec_dev, rate, AUDIO_DAI_CLOCK_OUT)) + device_printf(sc->dev, "failed to set sysclk for codec node\n"); + + SLIST_FOREACH(aux_node, &sc->aux_devs, link) { + if (AUDIO_DAI_SET_SYSCLK(aux_node->dev, rate, AUDIO_DAI_CLOCK_OUT)) + device_printf(sc->dev, "failed to set sysclk for aux node\n"); + } + } + + /* + * Let CPU node determine speed + */ + speed = AUDIO_DAI_SET_CHANSPEED(sc->cpu_dev, speed); + AUDIO_DAI_SET_CHANSPEED(sc->codec_dev, speed); + SLIST_FOREACH(aux_node, &sc->aux_devs, link) { + AUDIO_DAI_SET_CHANSPEED(aux_node->dev, speed); + } + + return (speed); +} + +static uint32_t +audio_soc_chan_getptr(kobj_t obj, void *data) +{ + struct audio_soc_softc *sc; + struct audio_soc_channel *ausoc_chan; + + ausoc_chan = data; + sc = ausoc_chan->sc; + + return AUDIO_DAI_GET_PTR(sc->cpu_dev, ausoc_chan->dir); +} + +static void * +audio_soc_chan_init(kobj_t obj, void *devinfo, struct snd_dbuf *b, + struct pcm_channel *c, int dir) +{ + struct audio_soc_softc *sc; + struct audio_soc_channel *ausoc_chan; + void *buffer; + + ausoc_chan = devinfo; + sc = ausoc_chan->sc; + buffer = malloc(AUDIO_BUFFER_SIZE, M_DEVBUF, M_WAITOK | M_ZERO); + + if (sndbuf_setup(b, buffer, AUDIO_BUFFER_SIZE) != 0) { + free(buffer, M_DEVBUF); + return NULL; + } + + ausoc_chan->dir = dir; + ausoc_chan->buf = b; + ausoc_chan->pcm = c; + + return (devinfo); +} + +static int +audio_soc_chan_trigger(kobj_t obj, void *data, int go) +{ + struct audio_soc_softc *sc; + struct audio_soc_channel *ausoc_chan; + struct audio_soc_aux_node *aux_node; + + ausoc_chan = (struct audio_soc_channel *)data; + sc = ausoc_chan->sc; + AUDIO_DAI_TRIGGER(sc->codec_dev, go, ausoc_chan->dir); + SLIST_FOREACH(aux_node, &sc->aux_devs, link) { + AUDIO_DAI_TRIGGER(aux_node->dev, go, ausoc_chan->dir); + } + + return AUDIO_DAI_TRIGGER(sc->cpu_dev, go, ausoc_chan->dir); +} + +static int +audio_soc_chan_free(kobj_t obj, void *data) +{ + + struct audio_soc_softc *sc; + struct audio_soc_channel *ausoc_chan; + void *buffer; + + ausoc_chan = (struct audio_soc_channel *)data; + sc = ausoc_chan->sc; + + buffer = sndbuf_getbuf(ausoc_chan->buf); + if (buffer) + free(buffer, M_DEVBUF); + + return (0); +} + +static struct pcmchan_caps * +audio_soc_chan_getcaps(kobj_t obj, void *data) +{ + struct audio_soc_softc *sc; + struct audio_soc_channel *ausoc_chan; + + ausoc_chan = data; + sc = ausoc_chan->sc; + + return AUDIO_DAI_GET_CAPS(sc->cpu_dev); +} + +static kobj_method_t audio_soc_chan_methods[] = { + KOBJMETHOD(channel_init, audio_soc_chan_init), + KOBJMETHOD(channel_free, audio_soc_chan_free), + KOBJMETHOD(channel_setformat, audio_soc_chan_setformat), + KOBJMETHOD(channel_setspeed, audio_soc_chan_setspeed), + KOBJMETHOD(channel_setblocksize,audio_soc_chan_setblocksize), + KOBJMETHOD(channel_trigger, audio_soc_chan_trigger), + KOBJMETHOD(channel_getptr, audio_soc_chan_getptr), + KOBJMETHOD(channel_getcaps, audio_soc_chan_getcaps), + KOBJMETHOD_END +}; +CHANNEL_DECLARE(audio_soc_chan); + +static void +audio_soc_intr(void *arg) +{ + struct audio_soc_softc *sc; + int channel_intr_required; + + sc = (struct audio_soc_softc *)arg; + channel_intr_required = AUDIO_DAI_INTR(sc->cpu_dev, sc->play_channel.buf, sc->rec_channel.buf); + if (channel_intr_required & AUDIO_DAI_PLAY_INTR) + chn_intr(sc->play_channel.pcm); + if (channel_intr_required & AUDIO_DAI_REC_INTR) + chn_intr(sc->rec_channel.pcm); +} + +static int +audio_soc_probe(device_t dev) +{ + + if (!ofw_bus_status_okay(dev)) + return (ENXIO); + + if (ofw_bus_search_compatible(dev, compat_data)->ocd_data != 0) { + device_set_desc(dev, "simple-audio-card"); + return (BUS_PROBE_DEFAULT); + } + + return (ENXIO); +} + +static void +audio_soc_init(void *arg) +{ + struct audio_soc_softc *sc; + phandle_t node, child; + device_t daidev, auxdev; + uint32_t xref; + uint32_t *aux_devs; + int ncells, i; + struct audio_soc_aux_node *aux_node; + + sc = (struct audio_soc_softc *)arg; + config_intrhook_disestablish(&sc->init_hook); + + node = ofw_bus_get_node(sc->dev); + /* TODO: handle multi-link nodes */ + child = ofw_bus_find_child(node, "simple-audio-card,cpu"); + if (child == 0) { + device_printf(sc->dev, "cpu node is missing\n"); + return; + } + if ((OF_getencprop(child, "sound-dai", &xref, sizeof(xref))) <= 0) { + device_printf(sc->dev, "missing sound-dai property in cpu node\n"); + return; + } + daidev = OF_device_from_xref(xref); + if (daidev == NULL) { + device_printf(sc->dev, "no driver attached to cpu node\n"); + return; + } + sc->cpu_dev = daidev; + + child = ofw_bus_find_child(node, "simple-audio-card,codec"); + if (child == 0) { + device_printf(sc->dev, "codec node is missing\n"); + return; + } + if ((OF_getencprop(child, "sound-dai", &xref, sizeof(xref))) <= 0) { + device_printf(sc->dev, "missing sound-dai property in codec node\n"); + return; + } + daidev = OF_device_from_xref(xref); + if (daidev == NULL) { + device_printf(sc->dev, "no driver attached to codec node\n"); + return; + } + sc->codec_dev = daidev; + + /* Add AUX devices */ + aux_devs = NULL; + ncells = OF_getencprop_alloc_multi(node, "simple-audio-card,aux-devs", sizeof(*aux_devs), + (void **)&aux_devs); + + for (i = 0; i < ncells; i++) { + auxdev = OF_device_from_xref(aux_devs[i]); + if (auxdev == NULL) + device_printf(sc->dev, "warning: no driver attached to aux node\n"); + aux_node = (struct audio_soc_aux_node *)malloc(sizeof(*aux_node), M_DEVBUF, M_NOWAIT); + if (aux_node == NULL) { + device_printf(sc->dev, "failed to allocate aux node struct\n"); + return; + } + aux_node->dev = auxdev; + SLIST_INSERT_HEAD(&sc->aux_devs, aux_node, link); + } + + if (aux_devs) + OF_prop_free(aux_devs); + + if (AUDIO_DAI_INIT(sc->cpu_dev, sc->format)) { + device_printf(sc->dev, "failed to initialize cpu node\n"); + return; + } + + /* Reverse clock roles for CODEC */ + if (AUDIO_DAI_INIT(sc->codec_dev, audio_soc_reverse_clocks(sc->format))) { + device_printf(sc->dev, "failed to initialize codec node\n"); + return; + } + + SLIST_FOREACH(aux_node, &sc->aux_devs, link) { + if (AUDIO_DAI_INIT(aux_node->dev, audio_soc_reverse_clocks(sc->format))) { + device_printf(sc->dev, "failed to initialize aux node\n"); + return; + } + } + + if (pcm_register(sc->dev, sc, 1, 1)) { + device_printf(sc->dev, "failed to register PCM\n"); + return; + } + + pcm_getbuffersize(sc->dev, AUDIO_BUFFER_SIZE, AUDIO_BUFFER_SIZE, + AUDIO_BUFFER_SIZE); + sc->play_channel.sc = sc; + sc->rec_channel.sc = sc; + + pcm_addchan(sc->dev, PCMDIR_PLAY, &audio_soc_chan_class, &sc->play_channel); + pcm_addchan(sc->dev, PCMDIR_REC, &audio_soc_chan_class, &sc->rec_channel); + + pcm_setstatus(sc->dev, "at EXPERIMENT"); + + AUDIO_DAI_SETUP_INTR(sc->cpu_dev, audio_soc_intr, sc); + AUDIO_DAI_SETUP_MIXER(sc->codec_dev, sc->dev); + SLIST_FOREACH(aux_node, &sc->aux_devs, link) { + AUDIO_DAI_SETUP_MIXER(aux_node->dev, sc->dev); + } +} + +static int +audio_soc_attach(device_t dev) +{ + struct audio_soc_softc *sc; + char *name; + phandle_t node, cpu_child; + uint32_t xref; + int i, ret; + char tmp[32]; + unsigned int fmt, pol, clk; + bool frame_master, bitclock_master; + + sc = device_get_softc(dev); + sc->dev = dev; + node = ofw_bus_get_node(dev); + + ret = OF_getprop_alloc(node, "name", (void **)&name); + if (ret == -1) + name = "SoC audio"; + + sc->name = strdup(name, M_DEVBUF); + device_set_desc(dev, sc->name); + + if (ret != -1) + OF_prop_free(name); + + SLIST_INIT(&sc->aux_devs); + + ret = OF_getprop(node, "simple-audio-card,format", tmp, sizeof(tmp)); + if (ret == 0) { + for (i = 0; i < nitems(ausoc_dai_formats); i++) { + if (strcmp(tmp, ausoc_dai_formats[i].name) == 0) { + fmt = ausoc_dai_formats[i].fmt; + break; + } + } + if (i == nitems(ausoc_dai_formats)) + return (EINVAL); + } else + fmt = AUDIO_DAI_FORMAT_I2S; + + if (OF_getencprop(node, "simple-audio-card,mclk-fs", + &sc->link_mclk_fs, sizeof(sc->link_mclk_fs)) <= 0) + sc->link_mclk_fs = 0; + + /* Unless specified otherwise, CPU node is the master */ + frame_master = bitclock_master = true; + + cpu_child = ofw_bus_find_child(node, "simple-audio-card,cpu"); + + if ((OF_getencprop(node, "simple-audio-card,frame-master", &xref, sizeof(xref))) > 0) + frame_master = cpu_child == OF_node_from_xref(xref); + + if ((OF_getencprop(node, "simple-audio-card,bitclock-master", &xref, sizeof(xref))) > 0) + bitclock_master = cpu_child == OF_node_from_xref(xref); + + if (frame_master) { + clk = bitclock_master ? + AUDIO_DAI_CLOCK_CBM_CFM : AUDIO_DAI_CLOCK_CBS_CFM; + } else { + clk = bitclock_master ? + AUDIO_DAI_CLOCK_CBM_CFS : AUDIO_DAI_CLOCK_CBS_CFS; + } + + bool bitclock_inversion = OF_hasprop(node, "simple-audio-card,bitclock-inversion"); + bool frame_inversion = OF_hasprop(node, "simple-audio-card,frame-inversion"); + if (bitclock_inversion) { + pol = frame_inversion ? + AUDIO_DAI_POLARITY_IB_IF : AUDIO_DAI_POLARITY_IB_NF; + } else { + pol = frame_inversion ? + AUDIO_DAI_POLARITY_NB_IF : AUDIO_DAI_POLARITY_NB_NF; + } + + sc->format = AUDIO_DAI_FORMAT(fmt, pol, clk); + + sc->init_hook.ich_func = audio_soc_init; + sc->init_hook.ich_arg = sc; + if (config_intrhook_establish(&sc->init_hook) != 0) + return (ENOMEM); + + return (0); +} + +static int +audio_soc_detach(device_t dev) +{ + struct audio_soc_softc *sc; + struct audio_soc_aux_node *aux; + + sc = device_get_softc(dev); + if (sc->name) + free(sc->name, M_DEVBUF); + + while ((aux = SLIST_FIRST(&sc->aux_devs)) != NULL) { + SLIST_REMOVE_HEAD(&sc->aux_devs, link); + free(aux, M_DEVBUF); + } + + return (0); +} + +static device_method_t audio_soc_methods[] = { + /* device_if methods */ + DEVMETHOD(device_probe, audio_soc_probe), + DEVMETHOD(device_attach, audio_soc_attach), + DEVMETHOD(device_detach, audio_soc_detach), + + DEVMETHOD_END, +}; + +static driver_t audio_soc_driver = { + "pcm", + audio_soc_methods, + sizeof(struct audio_soc_softc), +}; + +DRIVER_MODULE(audio_soc, simplebus, audio_soc_driver, pcm_devclass, NULL, NULL); +MODULE_VERSION(audio_soc, 1); Index: sys/dev/sound/fdt/dummy_codec.c =================================================================== --- /dev/null +++ sys/dev/sound/fdt/dummy_codec.c @@ -0,0 +1,127 @@ +/*- + * SPDX-License-Identifier: BSD-2-Clause-FreeBSD + * + * Copyright (c) 2020 Oleksandr Tymoshenko + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions + * are met: + * 1. Redistributions of source code must retain the above copyright + * notice, this list of conditions and the following disclaimer. + * 2. Redistributions in binary form must reproduce the above copyright + * notice, this list of conditions and the following disclaimer in the + * documentation and/or other materials provided with the distribution. + * + * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR + * IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES + * OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. + * IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, + * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, + * BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; + * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED + * AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, + * OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY + * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF + * SUCH DAMAGE. + * + * $FreeBSD$ + */ + +#include +__FBSDID("$FreeBSD$"); + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include + +#include "opt_snd.h" +#include +#include +#include "audio_dai_if.h" + +static struct ofw_compat_data compat_data[] = { + { "dummy-codec", 1}, + { NULL, 0 } +}; + +struct dummy_codec_softc { + device_t dev; +}; + +static int dummy_codec_probe(device_t dev); +static int dummy_codec_attach(device_t dev); +static int dummy_codec_detach(device_t dev); + +static int +dummy_codec_probe(device_t dev) +{ + if (!ofw_bus_status_okay(dev)) + return (ENXIO); + + if (!ofw_bus_search_compatible(dev, compat_data)->ocd_data) + return (ENXIO); + + device_set_desc(dev, "Dummy Codec"); + return (BUS_PROBE_DEFAULT); +} + +static int +dummy_codec_attach(device_t dev) +{ + struct dummy_codec_softc *sc; + phandle_t node; + + sc = device_get_softc(dev); + sc->dev = dev; + + node = ofw_bus_get_node(dev); + OF_device_register_xref(OF_xref_from_node(node), dev); + + return (0); +} + +static int +dummy_codec_detach(device_t dev) +{ + + return (0); +} + +static int +dummy_codec_dai_init(device_t dev, uint32_t format) +{ + + return (0); +} + +static device_method_t dummy_codec_methods[] = { + /* Device interface */ + DEVMETHOD(device_probe, dummy_codec_probe), + DEVMETHOD(device_attach, dummy_codec_attach), + DEVMETHOD(device_detach, dummy_codec_detach), + + DEVMETHOD(audio_dai_init, dummy_codec_dai_init), + + DEVMETHOD_END +}; + +static driver_t dummy_codec_driver = { + "dummycodec", + dummy_codec_methods, + sizeof(struct dummy_codec_softc), +}; + +static devclass_t dummy_codec_devclass; + +DRIVER_MODULE(dummy_codec, simplebus, dummy_codec_driver, dummy_codec_devclass, 0, 0); +SIMPLEBUS_PNP_INFO(compat_data); Index: sys/dev/sound/fdt/simple_amplifier.c =================================================================== --- /dev/null +++ sys/dev/sound/fdt/simple_amplifier.c @@ -0,0 +1,206 @@ +/*- + * SPDX-License-Identifier: BSD-2-Clause-FreeBSD + * + * Copyright (c) 2020 Oleksandr Tymoshenko + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions + * are met: + * 1. Redistributions of source code must retain the above copyright + * notice, this list of conditions and the following disclaimer. + * 2. Redistributions in binary form must reproduce the above copyright + * notice, this list of conditions and the following disclaimer in the + * documentation and/or other materials provided with the distribution. + * + * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR + * IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES + * OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. + * IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, + * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, + * BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; + * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED + * AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, + * OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY + * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF + * SUCH DAMAGE. + * + * $FreeBSD$ + */ + +#include +__FBSDID("$FreeBSD$"); + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include + +#include +#include + +#include "opt_snd.h" +#include +#include +#include "audio_dai_if.h" + +static struct ofw_compat_data compat_data[] = { + { "simple-audio-amplifier", 1}, + { NULL, 0} +}; + +struct simple_amp_softc { + device_t dev; + regulator_t supply_vcc; + gpio_pin_t gpio_enable; + bool gpio_is_valid; +}; + +static int simple_amp_probe(device_t dev); +static int simple_amp_attach(device_t dev); +static int simple_amp_detach(device_t dev); + +static int +simple_amp_probe(device_t dev) +{ + if (!ofw_bus_status_okay(dev)) + return (ENXIO); + + if (!ofw_bus_search_compatible(dev, compat_data)->ocd_data) + return (ENXIO); + + device_set_desc(dev, "Simple Amplifier"); + return (BUS_PROBE_DEFAULT); +} + +static int +simple_amp_attach(device_t dev) +{ + struct simple_amp_softc *sc; + phandle_t node; + int error; + + sc = device_get_softc(dev); + sc->dev = dev; + node = ofw_bus_get_node(dev); + + error = gpio_pin_get_by_ofw_property(dev, node, + "enable-gpios", &sc->gpio_enable); + if (error != 0) + sc->gpio_is_valid = false; + else + sc->gpio_is_valid = true; + + error = regulator_get_by_ofw_property(dev, 0, "VCC-supply", + &sc->supply_vcc); + if (error != 0) + device_printf(dev, "no VCC supply"); + + OF_device_register_xref(OF_xref_from_node(node), dev); + + return (0); +} + +static int +simple_amp_detach(device_t dev) +{ + + return (0); +} + +static int +simple_amp_dai_init(device_t dev, uint32_t format) +{ + + return (0); +} + +static int +simple_amp_dai_trigger(device_t dev, int go, int pcm_dir) +{ + struct simple_amp_softc *sc; + int error; + + if ((pcm_dir != PCMDIR_PLAY) && (pcm_dir != PCMDIR_REC)) + return (EINVAL); + + sc = device_get_softc(dev); + error = 0; + switch (go) { + case PCMTRIG_START: + if (sc->supply_vcc != NULL) { + error = regulator_enable(sc->supply_vcc); + if (error != 0) { + device_printf(sc->dev, + "could not enable 'VCC' regulator\n"); + break; + } + } + + if (sc->gpio_is_valid) { + error = gpio_pin_set_active(sc->gpio_enable, 1); + if (error != 0) { + device_printf(sc->dev, + "could not set 'gpio-enable' gpio\n"); + break; + } + } + + break; + + case PCMTRIG_STOP: + case PCMTRIG_ABORT: + if (sc->gpio_is_valid) { + error = gpio_pin_set_active(sc->gpio_enable, 0); + if (error != 0) { + device_printf(sc->dev, + "could not clear 'gpio-enable' gpio\n"); + break; + } + } + + if (sc->supply_vcc != NULL) { + error = regulator_disable(sc->supply_vcc); + if (error != 0) { + device_printf(sc->dev, + "could not disable 'VCC' regulator\n"); + break; + } + } + + break; + } + + return (error); +} + +static device_method_t simple_amp_methods[] = { + /* Device interface */ + DEVMETHOD(device_probe, simple_amp_probe), + DEVMETHOD(device_attach, simple_amp_attach), + DEVMETHOD(device_detach, simple_amp_detach), + + DEVMETHOD(audio_dai_init, simple_amp_dai_init), + DEVMETHOD(audio_dai_trigger, simple_amp_dai_trigger), + + DEVMETHOD_END +}; + +static driver_t simple_amp_driver = { + "simpleamp", + simple_amp_methods, + sizeof(struct simple_amp_softc), +}; + +static devclass_t simple_amp_devclass; + +DRIVER_MODULE(simple_amp, simplebus, simple_amp_driver, simple_amp_devclass, 0, 0); +SIMPLEBUS_PNP_INFO(compat_data);