/*************************************************************************** * __________ __ ___. * Open \______ \ ____ ____ | | _\_ |__ _______ ___ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ / * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < < * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \ * \/ \/ \/ \/ \/ * * Copyright (c) 2016 Amaury Pouly * * 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 "logf.h" #include "system.h" #include "kernel.h" #include "string.h" #include "stdio.h" #include "audio.h" #include "sound.h" #include "audiohw.h" #include "nwzlinux_codec.h" #include "stdlib.h" #include "panic.h" #include #include "nwz_audio.h" #include "pcm-alsa.h" #include "alsa-controls.h" /* This driver handle the Sony linux audio drivers: despite using many differents * codecs, it appears that they all share a common interface and common controls. */ /* This is the alsa mixer interface exposed by Sony: numid=3,iface=MIXER,name='Capture Src Switch' ; type=ENUMERATED,access=rw------,values=1,items=4 ; Item #0 'None' ; Item #1 'Line' ; Item #2 'Fm' ; Item #3 'Mic' : values=0 numid=2,iface=MIXER,name='Playback Src Switch' ; type=ENUMERATED,access=rw------,values=1,items=7 ; Item #0 'None' ; Item #1 'Music' ; Item #2 'Video' ; Item #3 'Tv' ; Item #4 'Fm' ; Item #5 'Line' ; Item #6 'Mic' : values=1 numid=1,iface=MIXER,name='Playback Volume' ; type=INTEGER,access=rw------,values=2,min=0,max=100,step=1 : values=5,5 numid=7,iface=MIXER,name='CODEC Acoustic Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=8,iface=MIXER,name='CODEC Cue/Rev Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=9,iface=MIXER,name='CODEC Fade In Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=6,iface=MIXER,name='CODEC Mute Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=5,iface=MIXER,name='CODEC Power Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=10,iface=MIXER,name='CODEC Stanby Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=4,iface=MIXER,name='Output Switch' ; type=ENUMERATED,access=rw------,values=1,items=4 ; Item #0 'Headphone' ; Item #1 'LineVariable' ; Item #2 'LineFixed' ; Item #3 'Speaker' : values=0 numid=6,iface=MIXER,name='Sampling Rate' ; type=ENUMERATED,access=rw------,values=1,items=6 ; Item #0 '44100' ; Item #1 '48000' ; Item #2 '88200' ; Item #3 '96000' ; Item #4 '176400' ; Item #5 '192000' : values=0 */ /* List of various codecs used by Sony */ enum nwz_codec_t { NWZ_CS42L56, NWZ_R2A15602LG_D, NWZ_CS47L01_A, NWZ_CS47L01_D, NWZ_CXD3774GF_D, NWZ_UNK_CODEC, }; #define NWZ_LEVEL_MUTE -1000 /* Description of the volume curve implemented by the kernel driver */ struct nwz_vol_curve_t { int count; /* number of levels */ int level[]; /* levels in tenth-dB, level[0] is always mute */ }; /* file descriptor of the icx_noican device */ static int fd_noican; /* file descriptor of the hardware sound device */ static int fd_hw; /* Codec */ static enum nwz_codec_t nwz_codec; /* does the code support setting the sample rate? */ static bool has_sample_rate; static enum nwz_codec_t find_codec(void) { if(nwz_is_kernel_module_loaded("cs42L56_d")) return NWZ_CS42L56; if(nwz_is_kernel_module_loaded("r2A15602LG_d")) return NWZ_R2A15602LG_D; if(nwz_is_kernel_module_loaded("cs47L01_d")) return NWZ_CS47L01_D; if(nwz_is_kernel_module_loaded("cs47L01_a")) return NWZ_CS47L01_A; if(nwz_is_kernel_module_loaded("cxd3774gf_d")) return NWZ_CXD3774GF_D; return NWZ_UNK_CODEC; } const char *nwz_get_codec_name(void) { switch(nwz_codec) { case NWZ_CS42L56: return "cs42L56_d"; case NWZ_R2A15602LG_D: return "r2A15602LG_d"; case NWZ_CS47L01_D: return "cs47L01_d"; case NWZ_CS47L01_A: return "cs47L01_a"; case NWZ_CXD3774GF_D: return "cxd3774gf_d"; default: return "Unknown"; } } static struct nwz_vol_curve_t cxd3774gf_vol_curve = { .count = 31, /* Most Sonys seem to follow the convention of 3dB/step then 2dB/step then 1dB/step */ .level = {NWZ_LEVEL_MUTE, -550, -520, -490, -460, -430, -400, -370, -340, -310, -280, -250, /* 3dB/step */ -230, -210, -190, -170, -150, -130, -110, -90, /* 2dB/step */ -80, -70, -60, -50, -40, -30, -20, -10, 0, /* 1dB/step */ 15, 35, /* 1.5dB then 2dB */ } }; struct nwz_vol_curve_t *nwz_get_codec_vol_curve(void) { switch(nwz_codec) { case NWZ_CS47L01_A: case NWZ_CS47L01_D: /* there are 32 levels but the last two are the same so in fact it * is the same curve as the cxd3774gf_d */ case NWZ_CXD3774GF_D: return &cxd3774gf_vol_curve; default: /* return the safest curve (only 31 levels) */ return &cxd3774gf_vol_curve; } } static void noican_init(void) { fd_noican = open(NWZ_NC_DEV, O_RDWR); /* some targets don't have noise cancelling so silently fail */ } static void noican_close(void) { if(fd_noican >= 0) close(fd_noican); } /* Set NC switch */ static void noican_set_switch(int sw) { if(ioctl(fd_noican, NWZ_NC_SET_SWITCH, &sw) < 0) panicf("ioctl(NWZ_NC_SET_SWITCH) failed"); } /* Get NC switch */ static int noican_get_switch(void) { int val; if(ioctl(fd_noican, NWZ_NC_GET_SWITCH, &val) < 0) panicf("ioctl(NWZ_NC_GET_SWITCH) failed"); return val; } /* Get HP status */ static int noican_get_hp_status(void) { int val; if(ioctl(fd_noican, NWZ_NC_GET_HP_STATUS, &val) < 0) panicf("ioctl(NWZ_NC_GET_HP_STATUS) failed"); return val; } /* Set HP type */ static void noican_set_hp_type(int type) { if(ioctl(fd_noican, NWZ_NC_SET_HP_TYPE, &type) < 0) panicf("ioctl(NWZ_NC_SET_HP_TYPE) failed"); } /* Get HP type */ static int noican_get_hp_type(void) { int val; if(ioctl(fd_noican, NWZ_NC_GET_HP_TYPE, &val) < 0) panicf("ioctl(NWZ_NC_GET_HP_TYPE) failed"); return val; } /* Set gain */ static void noican_set_gain(int gain) { if(ioctl(fd_noican, NWZ_NC_SET_GAIN, &gain) < 0) panicf("ioctl(NWZ_NC_SET_GAIN) failed"); } /* Get gain */ static int noican_get_gain(void) { int val; if(ioctl(fd_noican, NWZ_NC_GET_GAIN, &val) < 0) panicf("ioctl(NWZ_NC_GET_GAIN) failed"); return val; } /* Set filter */ static void noican_set_filter(int filter) { if(ioctl(fd_noican, NWZ_NC_SET_FILTER, &filter) < 0) panicf("ioctl(NWZ_NC_SET_FILTER) failed"); } /* Get filter */ static int noican_get_filter(void) { int val; if(ioctl(fd_noican, NWZ_NC_GET_FILTER, &val) < 0) panicf("ioctl(NWZ_NC_GET_FILTER) failed"); return val; } static void hw_open(void) { fd_hw = open("/dev/snd/hwC0D0", O_RDWR); if(fd_hw < 0) panicf("Cannot open '/dev/snd/hwC0D0'"); } static void hw_close(void) { close(fd_hw); } /* Acoustic and Cue/Rev control how the volume curve, but it is not clear * what the intention of these modes are and the OF does not seem to use * them by default */ bool audiohw_acoustic_enabled(void) { return alsa_controls_get_bool("CODEC Acoustic Switch"); } void audiohw_enable_acoustic(bool en) { alsa_controls_set_bool("CODEC Acoustic Switch", en); } bool audiohw_cuerev_enabled(void) { return alsa_controls_get_bool("CODEC Cue/Rev Switch"); } void audiohw_enable_cuerev(bool en) { alsa_controls_set_bool("CODEC Cue/Rev Switch", en); } void audiohw_set_playback_src(enum nwz_src_t src) { switch(src) { case NWZ_RADIO: alsa_controls_set_enum("Playback Src Switch", "Fm"); break; case NWZ_MIC: alsa_controls_set_enum("Playback Src Switch", "Mic"); break; case NWZ_PLAYBACK: default: alsa_controls_set_enum("Playback Src Switch", "Music"); break; } } void audiohw_preinit(void) { alsa_controls_init("default"); /* turn on codec */ alsa_controls_set_bool("CODEC Power Switch", true); /* mute */ alsa_controls_set_bool("CODEC Mute Switch", true); /* Acoustic and Cue/Rev control how the volume curve, but it is not clear * what the intention of these modes are and the OF does not seem to use * them by default */ audiohw_enable_acoustic(false); audiohw_enable_cuerev(false); /* select playback source */ audiohw_set_playback_src(NWZ_PLAYBACK); /* use headphone output */ alsa_controls_set_enum("Output Switch", "Headphone"); /* unmute */ alsa_controls_set_bool("CODEC Mute Switch", false); /* sample rate */ has_sample_rate = alsa_has_control("Sampling Rate"); /* init noican */ noican_init(); if(fd_noican >= 0) { /* dump configuration, for debug purposes */ printf("nc hp status: %d\n", noican_get_hp_status()); printf("nc type: %d\n", noican_get_hp_type()); printf("nc switch: %d\n", noican_get_switch()); printf("nc gain: %d\n", noican_get_gain()); printf("nc filter: %d\n", noican_get_filter()); /* make sure we start in a clean state */ noican_set_switch(NWZ_NC_SWITCH_OFF); noican_set_hp_type(NC_HP_TYPE_DEFAULT); noican_set_filter(NWZ_NC_FILTER_INDEX_0); noican_set_gain(NWZ_NC_GAIN_CENTER); } /* init hw */ hw_open(); nwz_codec = find_codec(); printf("Codec: %s\n", nwz_get_codec_name()); } void audiohw_postinit(void) { } /* volume must be driver unit */ static void nwz_set_driver_vol(int vol) { long vols[2]; /* the driver expects percent, convert from centibel in range 0...x */ vols[0] = vols[1] = vol; /* on some recent players like A10, Sony decided to merge left/right volume * into one, thus we need to make sure we write the correct number of values */ int vol_cnt; alsa_controls_get_info("Playback Volume", &vol_cnt); alsa_controls_set_ints("Playback Volume", vol_cnt, vols); } /* volume is in tenth-dB */ void audiohw_set_volume(int vol_l, int vol_r) { /* FIXME at the moment we don't support balance and just average left and right. * But this could be implemented using pcm alsa digital volume */ /* the Sony drivers expect vol_l = vol_r */ int vol = (vol_l + vol_r) / 2; printf("request volume %d dB\n", vol / 10); struct nwz_vol_curve_t *curve = nwz_get_codec_vol_curve(); /* min/max for pcm volume */ int min_pcm = -430; int max_pcm = 0; /* On some codecs (like cs47L01), Sony clear overdrives the DAC which produces * massive clipping at any level (since they fix the DAC volume at around +6dB * and then adjust HP volume in negative at the top of range !!). The only * solution around this problem is to use the digital volume first so that * very quickly the digital volume compensate for the DAC overdrive and we * avoid clipping. */ int sony_clip_level = -80; /* any volume above this will cause massive clipping the DAC */ /* to avoid the clipping problem, virtually decrease requested volume by the * clipping threshold, so that we will compensate in digital later by * at least this amount if possibly */ vol -= sony_clip_level; int drv_vol = curve->count - 1; /* pick driver level just above request volume */ while(drv_vol > 0 && curve->level[drv_vol - 1] >= vol) drv_vol--; /* now remove the artifical volume change */ vol += sony_clip_level; /* now adjust digital volume */ vol -= curve->level[drv_vol]; if(vol < min_pcm) { vol = min_pcm; /* digital cannot do <43dB */ drv_vol = 0; /* mute */ } else if(vol > max_pcm) vol = max_pcm; /* digital cannot do >0dB */ printf(" set driver volume %d (%d dB)\n", drv_vol, curve->level[drv_vol] / 10); nwz_set_driver_vol(drv_vol); printf(" set digital volume %d dB\n", vol / 10); pcm_set_mixer_volume(vol, vol); } void audiohw_close(void) { hw_close(); alsa_controls_close(); noican_close(); } void audiohw_set_frequency(int fsel) { if(has_sample_rate) { /* it's slightly annoying that Sony put the value in an enum with strings... */ char freq_str[16]; sprintf(freq_str, "%d", fsel); alsa_controls_set_enum("Sampling Rate", freq_str); } }