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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
*
* 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/%d", 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;
if (lineout_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 / 20, sw_volume_r / 20);
}
static void audiohw_set_volume_v2(int vol_l, int vol_r)
{
long l,r;
if (lineout_inserted()) {
vol_l_hw = vol_r_hw = global_settings.volume_limit * 10;
} else {
vol_l_hw = -vol_l;
vol_r_hw = -vol_r;
}
if (!hw_init)
return;
l = vol_l_hw / 5;
r = vol_l_hw / 5;
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 / 2, global_settings.volume_limit / 2);
}
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;
logf("lo vol %d %d", vol_l, vol_r);
(void)vol_l;
(void)vol_r;
if (lineout_inserted()) {
l = r = global_settings.volume_limit * 10;
} else {
l = vol_l_hw;
r = vol_r_hw;
}
if (hwver >= 2) {
if (hw_init) {
l /= 5;
r /= 5;
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 / 20, sw_volume_r / 20);
}
}
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