/*************************************************************************** * __________ __ ___. * Open \______ \ ____ ____ | | _\_ |__ _______ ___ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ / * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < < * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \ * \/ \/ \/ \/ \/ * $Id$ * * Copyright (C) 2021 Aidan MacDonald * * 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 software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY * KIND, either express or implied. * ****************************************************************************/ #include "audiohw.h" #include "sound.h" #include "panic.h" #include "pcm_sampr.h" #include "pcm_sw_volume.h" #include "system.h" #include "i2c-async.h" #ifndef HAVE_SW_VOLUME_CONTROL # error "AK4376 requires HAVE_SW_VOLUME_CONTROL!" #endif /* NOTE: At present, only the FiiO M3K uses this driver so the handling of * the clock / audio interface is limited to I2S slave, 16-bit samples, with * DAC master clock provided directly on the MCLK input pin, fitting the * clock setup of the M3K. * * Feel free to expand upon this if another target ever needs this driver. */ /* Converts HW_FREQ_XX constants to register values */ static const int ak4376_fsel_to_hw[] = { HW_HAVE_192_(AK4376_FS_192,) HW_HAVE_176_(AK4376_FS_176,) HW_HAVE_96_(AK4376_FS_96,) HW_HAVE_88_(AK4376_FS_88,) HW_HAVE_64_(AK4376_FS_64,) HW_HAVE_48_(AK4376_FS_48,) HW_HAVE_44_(AK4376_FS_44,) HW_HAVE_32_(AK4376_FS_32,) HW_HAVE_24_(AK4376_FS_24,) HW_HAVE_22_(AK4376_FS_22,) HW_HAVE_16_(AK4376_FS_16,) HW_HAVE_12_(AK4376_FS_12,) HW_HAVE_11_(AK4376_FS_11,) HW_HAVE_8_(AK4376_FS_8,) }; static struct ak4376 { int fsel; int low_mode; int regs[AK4376_NUM_REGS]; } ak4376; void ak4376_init(void) { /* Initialize DAC state */ ak4376.fsel = HW_FREQ_48; ak4376.low_mode = 0; for(int i = 0; i < AK4376_NUM_REGS; ++i) ak4376.regs[i] = -1; /* Initial reset after power-on */ ak4376_set_pdn_pin(0); mdelay(1); ak4376_set_pdn_pin(1); mdelay(1); static const int init_config[] = { /* Ensure HPRHZ, HPLHZ are 0 */ AK4376_REG_OUTPUT_MODE, 0x00, /* Mute all volume controls */ AK4376_REG_MIXER, 0x00, AK4376_REG_LCH_VOLUME, 0x80, AK4376_REG_RCH_VOLUME, 0x00, AK4376_REG_AMP_VOLUME, 0x00, /* Clock source = MCLK, divider = 1 */ AK4376_REG_DAC_CLK_SRC, 0x00, AK4376_REG_DAC_CLK_DIV, 0x00, /* I2S slave mode, 16-bit samples */ AK4376_REG_AUDIO_IF_FMT, 0x03, /* Recommended by datasheet */ AK4376_REG_ADJUST1, 0x20, AK4376_REG_ADJUST2, 0x05, /* Power controls */ AK4376_REG_PWR2, 0x33, AK4376_REG_PWR3, 0x01, AK4376_REG_PWR4, 0x03, }; /* Write initial configuration prior to power-up */ for(size_t i = 0; i < ARRAYLEN(init_config); i += 2) ak4376_write(init_config[i], init_config[i+1]); /* Initial frequency setting, also handles DAC/amp power-up */ audiohw_set_frequency(HW_FREQ_48); } void ak4376_close(void) { /* Shut off power */ ak4376_write(AK4376_REG_PWR3, 0x00); ak4376_write(AK4376_REG_PWR4, 0x00); ak4376_write(AK4376_REG_PWR2, 0x00); /* PDN pin low */ ak4376_set_pdn_pin(0); } void ak4376_write(int reg, int value) { /* Ensure value is sensible and differs from the last set value */ if((value & 0xff) == value && ak4376.regs[reg] != value) { int r = i2c_reg_write1(AK4376_BUS, AK4376_ADDR, reg, value); if(r == I2C_STATUS_OK) ak4376.regs[reg] = value; else ak4376.regs[reg] = -1; } } int ak4376_read(int reg) { /* Only read from I2C if we don't already know the value */ if(ak4376.regs[reg] < 0) ak4376.regs[reg] = i2c_reg_read1(AK4376_BUS, AK4376_ADDR, reg); return ak4376.regs[reg]; } static int round_step_up(int x, int step) { int rem = x % step; if(rem > 0) rem -= step; return x - rem; } static void calc_volumes(int vol, int* mix, int* dig, int* sw) { /* Mixer can divide by 2, which gives an extra -6 dB adjustment */ if(vol < AK4376_DIG_VOLUME_MIN) { *mix |= AK4376_MIX_HALF; vol += 60; } *dig = round_step_up(vol, AK4376_DIG_VOLUME_STEP); *dig = MIN(*dig, AK4376_DIG_VOLUME_MAX); *dig = MAX(*dig, AK4376_DIG_VOLUME_MIN); vol -= *dig; /* Seems that this is the allowable range for software volume */ *sw = MIN(vol, 60); *sw = MAX(*sw, -730); vol -= *sw; } static int dig_vol_to_hw(int vol) { if(vol < AK4376_DIG_VOLUME_MIN) return 0; if(vol > AK4376_DIG_VOLUME_MAX) return 31; return (vol - AK4376_DIG_VOLUME_MIN) / AK4376_DIG_VOLUME_STEP + 1; } static int amp_vol_to_hw(int vol) { if(vol < AK4376_AMP_VOLUME_MIN) return 0; if(vol > AK4376_AMP_VOLUME_MAX) return 14; return (vol - AK4376_AMP_VOLUME_MIN) / AK4376_AMP_VOLUME_STEP + 1; } void audiohw_set_volume(int vol_l, int vol_r) { int amp; int mix_l = AK4376_MIX_LCH, dig_l, sw_l; int mix_r = AK4376_MIX_RCH, dig_r, sw_r; if(vol_l <= AK4376_MIN_VOLUME && vol_r <= AK4376_MIN_VOLUME) { /* Special case for full mute */ amp = AK4376_AMP_VOLUME_MUTE; dig_l = dig_r = AK4376_DIG_VOLUME_MUTE; sw_l = sw_r = PCM_MUTE_LEVEL; } else { /* Amp is a mono control -- calculate based on the loudest channel. * The quieter channel then gets reduced more by digital controls. */ amp = round_step_up(MAX(vol_l, vol_r), AK4376_AMP_VOLUME_STEP); amp = MIN(amp, AK4376_AMP_VOLUME_MAX); amp = MAX(amp, AK4376_AMP_VOLUME_MIN); /* Other controls are stereo */ calc_volumes(vol_l - amp, &mix_l, &dig_l, &sw_l); calc_volumes(vol_r - amp, &mix_r, &dig_r, &sw_r); } ak4376_write(AK4376_REG_MIXER, (mix_l & 0xf) | ((mix_r & 0xf) << 4)); ak4376_write(AK4376_REG_LCH_VOLUME, dig_vol_to_hw(dig_l) | (1 << 7)); ak4376_write(AK4376_REG_RCH_VOLUME, dig_vol_to_hw(dig_r)); ak4376_write(AK4376_REG_AMP_VOLUME, amp_vol_to_hw(amp)); pcm_set_master_volume(sw_l, sw_r); } void audiohw_set_filter_roll_off(int val) { int reg = ak4376_read(AK4376_REG_FILTER); reg &= ~0xc0; reg |= (val & 3) << 6; ak4376_write(AK4376_REG_FILTER, reg); } void audiohw_set_frequency(int fsel) { /* Determine master clock multiplier */ int mult = ak4376_set_mclk_freq(fsel, false); /* Calculate clock mode for frequency. Multipliers of 32/64 are only * for rates >= 256 KHz which are not supported by Rockbox, so they * are commented out -- but they're in the correct place. */ int clock_mode = ak4376_fsel_to_hw[fsel]; switch(mult) { /* case 32: */ case 256: break; /* case 64: */ case 512: clock_mode |= 0x20; break; case 1024: clock_mode |= 0x40; break; case 128: clock_mode |= 0x60; break; default: panicf("ak4376: bad master clock multiple %d", mult); return; } /* Handle the DSMLP bit in the MODE_CTRL register */ int mode_ctrl = 0x00; if(ak4376.low_mode || hw_freq_sampr[fsel] <= SAMPR_12) mode_ctrl |= 0x40; /* Program the new settings */ ak4376_write(AK4376_REG_CLOCK_MODE, clock_mode); ak4376_write(AK4376_REG_MODE_CTRL, mode_ctrl); ak4376_write(AK4376_REG_PWR3, ak4376.low_mode ? 0x11 : 0x01); /* Enable the master clock */ ak4376_set_mclk_freq(fsel, true); /* Remember the frequency */ ak4376.fsel = fsel; } void audiohw_set_power_mode(int mode) { /* This is handled via audiohw_set_frequency() since changing LPMODE * bit requires power-down/power-up & changing other bits as well */ if(ak4376.low_mode != mode) { ak4376.low_mode = mode; audiohw_set_frequency(ak4376.fsel); } }