/*************************************************************************** * __________ __ ___. * Open \______ \ ____ ____ | | _\_ |__ _______ ___ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ / * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < < * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \ * \/ \/ \/ \/ \/ * * * Copyright (c) 2018 Marcin Bukat * Copyright (c) 2020 Solomon Peachy * * 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. * ****************************************************************************/ // #define LOGF_ENABLE #include "config.h" #include "audio.h" #include "audiohw.h" #include "system.h" #include "panic.h" #include "sysfs.h" #include "alsa-controls.h" #include "pcm-alsa.h" #include "settings.h" #include "logf.h" /* PCM device hw:0,0 ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED FORMAT: S16_LE S24_LE SUBFORMAT: STD SAMPLE_BITS: [16 32] FRAME_BITS: [16 64] CHANNELS: [1 2] RATE: [8000 192000] PERIOD_TIME: (2666 8192000] PERIOD_SIZE: [512 65536] PERIOD_BYTES: [4096 131072] PERIODS: [4 128] BUFFER_TIME: (10666 32768000] BUFFER_SIZE: [2048 262144] BUFFER_BYTES: [4096 524288] TICK_TIME: ALL Mixer controls (v1): numid=1,iface=MIXER,name='Output Port Switch' ; type=INTEGER,access=rw------,values=1,min=0,max=5,step=0 : values=4 Mixer controls (v2+): numid=3,iface=MIXER,name='ES9018_K2M Digital Filter' ; type=INTEGER,access=rw------,values=1,min=0,max=4,step=0 : values=0 numid=1,iface=MIXER,name='Left Playback Volume' ; type=INTEGER,access=rw------,values=1,min=0,max=255,step=0 : values=0 numid=4,iface=MIXER,name='Output Port Switch' ; type=INTEGER,access=rw------,values=1,min=0,max=5,step=0 : values=0 numid=2,iface=MIXER,name='Right Playback Volume' ; type=INTEGER,access=rw------,values=1,min=0,max=255,step=0 : values=0 numid=5,iface=MIXER,name='isDSD' ; type=BOOLEAN,access=rw------,values=1 : values=off */ static int hw_init = 0; static long int vol_l_hw = 255; static long int vol_r_hw = 255; static long int last_ps = -1; static int muted = -1; extern int hwver; void audiohw_mute(int mute) { logf("mute %d", mute); if (hw_init < 0 || muted == mute) return; muted = mute; if(mute) { long int ps0 = 0; alsa_controls_set_ints("Output Port Switch", 1, &ps0); } else { last_ps = 0; erosq_get_outputs(); } } int erosq_get_outputs(void) { long int ps = 0; // Muted, if nothing is plugged in! int status = 0; if (!hw_init) return ps; const char * const sysfs_lo_switch = "/sys/class/switch/lineout/state"; const char * const sysfs_hs_switch = "/sys/class/switch/headset/state"; sysfs_get_int(sysfs_lo_switch, &status); if (status) ps = 1; // lineout sysfs_get_int(sysfs_hs_switch, &status); if (status) ps = 2; // headset erosq_set_output(ps); return ps; } void erosq_set_output(int ps) { if (!hw_init || muted) return; if (last_ps != ps) { logf("set out %d/%ld", ps, last_ps); /* Output port switch */ last_ps = ps; alsa_controls_set_ints("Output Port Switch", 1, &last_ps); audiohw_set_volume(vol_l_hw, vol_r_hw); } } void audiohw_preinit(void) { logf("hw preinit"); alsa_controls_init("default"); hw_init = 1; /* See if we have hw2 or later */ if (alsa_controls_find("Left Playback Volume") == -1) hwver = 1; else if (hwver == 1) hwver = 23; audiohw_mute(false); /* No need to stay muted */ } void audiohw_postinit(void) { logf("hw postinit"); } void audiohw_close(void) { logf("hw close"); hw_init = 0; muted = -1; alsa_controls_close(); } void audiohw_set_frequency(int fsel) { (void)fsel; } /* min/max for pcm volume */ const int min_pcm = -740; const int max_pcm = 0; static void audiohw_set_volume_v1(int vol_l, int vol_r) { long l,r; vol_l_hw = vol_l; vol_r_hw = vol_r; logf("set_volume_v1 %d, %d", vol_l, vol_r); if (lineout_inserted() && !headphones_inserted()) { /* On the EROS Q/K hardware, full scale line out is _very_ hot at ~5.8Vpp. As the hardware provides no way to reduce output gain, we have to back off on the PCM signal to avoid blowing out the signal. */ l = r = global_settings.volume_limit * 10; } else { l = vol_l_hw; r = vol_r_hw; } int sw_volume_l = l <= min_pcm ? min_pcm : MIN(l, max_pcm); int sw_volume_r = r <= min_pcm ? min_pcm : MIN(r, max_pcm); pcm_set_mixer_volume(sw_volume_l, sw_volume_r); } static void audiohw_set_volume_v2(int vol_l, int vol_r) { long l,r; vol_l_hw = vol_l; vol_r_hw = vol_r; logf("set_volume_v2 %d, %d", vol_l, vol_r); if (lineout_inserted() && !headphones_inserted()) { // was l = r = ... syntax. // for some reason r channel was not getting set // and was quiet...? l = -1 * (global_settings.volume_limit * 10) / 5; r = l; } else { // never save volume as positive, we will // just oscillate between positive and negative then l = -1 * vol_l_hw / 5; r = -1 * vol_r_hw / 5; } if (!hw_init) return; alsa_controls_set_ints("Left Playback Volume", 1, &l); alsa_controls_set_ints("Right Playback Volume", 1, &r); /* Dial back PCM mixer to avoid compression */ pcm_set_mixer_volume(global_settings.volume_limit, global_settings.volume_limit); } void audiohw_set_volume(int vol_l, int vol_r) { if (hwver >= 2) { audiohw_set_volume_v2(vol_l, vol_r); } else { audiohw_set_volume_v1(vol_l, vol_r); } } void audiohw_set_lineout_volume(int vol_l, int vol_r) { long l,r; (void)vol_l; (void)vol_r; if (lineout_inserted() && !headphones_inserted()) { // was l = r = ... syntax. // for some reason r channel was not getting set // and was quiet...? l = global_settings.volume_limit * 10; r = l; logf("lo vol %ld %ld", l, r); if (hw_init){ if (hwver >= 2) { l /= 5; l = l * -1; r /= 5; r = r * -1; alsa_controls_set_ints("Left Playback Volume", 1, &l); alsa_controls_set_ints("Right Playback Volume", 1, &r); } else { int sw_volume_l = l <= min_pcm ? min_pcm : MIN(l, max_pcm); int sw_volume_r = r <= min_pcm ? min_pcm : MIN(r, max_pcm); pcm_set_mixer_volume(sw_volume_l, sw_volume_r); } } } }