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audio: Move hosted audio "codec" drivers into their respective target dirs

Browse source 
They are nearly entirely generic wrappers around ALSA controls, unique
per target, and are ripe for further consolidation.

Change-Id: I05e4a450e3e89e03616906601c4f8fa46200dff5
This commit is contained in:
Solomon Peachy 2025-11-30 11:21:37 -05:00
parent d376e0afb7
commit a79bdaf462
20 changed files with 62 additions and 43 deletions
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93
firmware/target/hosted/agptek/rocker_codec.c Normal file
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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
*
* Copyright (c) 2018 Marcin Bukat
*
* 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 "config.h"
#include "audio.h"
#include "audiohw.h"
#include "system.h"
#include "panic.h"
#include "alsa-controls.h"
static int hw_init = 0;
static long int vol_l_hw = 255;
static long int vol_r_hw = 255;
static int muted = -1;
void audiohw_mute(int mute)
{
if (!hw_init || muted == mute)
return;
muted = mute;
if(mute)
{
long int ps0 = 0;
alsa_controls_set_ints("Output Port Switch", 1, &ps0);
}
else
{
long int ps2 = 2;
alsa_controls_set_ints("Output Port Switch", 1, &ps2);
}
}
void audiohw_preinit(void)
{
alsa_controls_init("default");
hw_init = 1;
#if defined(AUDIOHW_MUTE_ON_STOP) || defined(AUDIOHW_NEEDS_INITIAL_UNMUTE)
audiohw_mute(true); /* Start muted to avoid the POP */
#else
audiohw_mute(false);
#endif
}
void audiohw_postinit(void)
{
}
void audiohw_close(void)
{
hw_init = 0;
muted = -1;
alsa_controls_close();
}
void audiohw_set_frequency(int fsel)
{
(void)fsel;
}
void audiohw_set_volume(int vol_l, int vol_r)
{
vol_l_hw = -vol_l/5;
vol_r_hw = -vol_r/5;
if (!hw_init)
return;
alsa_controls_set_ints("Left Playback Volume", 1, &vol_l_hw);
alsa_controls_set_ints("Right Playback Volume", 1, &vol_r_hw);
}

13
firmware/target/hosted/agptek/rocker_codec.h Normal file
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#ifndef __ROCKER_CODEC__
#define __ROCKER_CODEC__
#define AUDIOHW_CAPS 0
AUDIOHW_SETTING(VOLUME, "dB", 1, 5, -115*10, 0, -30*10)
#endif
//#define AUDIOHW_MUTE_ON_STOP
//#define AUDIOHW_NEEDS_INITIAL_UNMUTE
/* Note: Due to Kernel bug, we can't use MUTE_ON_PAUSE with backlight fading */
void audiohw_mute(int mute);

281
firmware/target/hosted/aigo/erosqlinux_codec.c Normal file
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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/%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 / 20, sw_volume_r / 20);
}
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 / 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;
(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 / 20, sw_volume_r / 20);
}
}
}
}

23
firmware/target/hosted/aigo/erosqlinux_codec.h Normal file
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#ifndef __EROSQLINUX_CODEC__
#define __EROSQLINUX_CODEC__
#define AUDIOHW_CAPS (LINEOUT_CAP)
/* a small DC offset prevents play/pause clicking due to the DAC auto-muting */
#define PCM_DC_OFFSET_VALUE -1
/*
* Note: Maximum volume is set one step below unity in order to
* avoid overflowing pcm samples due to our DC Offset.
*
* The DAC's output is hot enough this should not be an issue.
*/
AUDIOHW_SETTING(VOLUME, "dB", 0, 2, -74, -2, -40)
//#define AUDIOHW_NEEDS_INITIAL_UNMUTE
void audiohw_mute(int mute);
void erosq_set_output(int ps);
int erosq_get_outputs(void);
#endif

40
firmware/target/hosted/android/android-codec.c Normal file
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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright © 2010 Thomas Martitz
*
* 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 "config.h"
#include "audiohw.h"
#include "pcm-android.h"
void audiohw_set_volume(int volume)
{
pcm_set_mixer_volume(volume);
}
void audiohw_set_balance(int balance)
{
(void)balance;
}
void audiohw_close(void)
{
pcm_shutdown();
}

43
firmware/target/hosted/android/android_codec.h Normal file
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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2010 by Thomas Martitz
*
* 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.
*
****************************************************************************/
#ifndef __ANDROID_CODEC_H
#define __ANDROID_CODEC_H
/* Bass and treble tone controls */
#ifdef AUDIOHW_HAVE_BASS
AUDIOHW_SETTING(BASS, "dB", 0, 1, -24, 24, 0)
#endif
#ifdef AUDIOHW_HAVE_TREBLE
AUDIOHW_SETTING(TREBLE, "dB", 0, 1, -24, 24, 0)
#endif
#if defined(HAVE_RECORDING)
AUDIOHW_SETTING(LEFT_GAIN, "dB", 1, 1,-128, 96, 0)
AUDIOHW_SETTING(RIGHT_GAIN, "dB", 1, 1,-128, 96, 0)
AUDIOHW_SETTING(MIC_GAIN, "dB", 1, 1,-128, 108, 16)
#endif
#if defined(AUDIOHW_HAVE_BASS_CUTOFF)
AUDIOHW_SETTING(BASS_CUTOFF, "", 0, 1, 1, 4, 1)
#endif
#if defined(AUDIOHW_HAVE_TREBLE_CUTOFF)
AUDIOHW_SETTING(TREBLE_CUTOFF, "", 0, 1, 1, 4, 1)
#endif
#endif /* __ANDROID_CODEC_H */

7
firmware/target/hosted/android/pcm-android.h Normal file
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#ifndef __PCM_ANDROID_H
#define __PCM_ANDROID_H
void pcm_set_mixer_volume(int volume);
void pcm_shutdown(void);
#endif /* __PCM_ANDROID_H */

169
firmware/target/hosted/fiio/fiiolinux_codec.c Normal file
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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
*
* Copyright (c) 2018 Marcin Bukat
* Copyright (c) 2019 Roman Stolyarov
*
* 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 "config.h"
#include "audio.h"
#include "audiohw.h"
#include "system.h"
#include "kernel.h"
#include "panic.h"
#include "sysfs.h"
#include "alsa-controls.h"
#include "pcm-alsa.h"
#include <sys/ioctl.h>
static int ak_hw = -1;
static int vol_sw[2] = {0};
static long int vol_hw[2] = {0};
static void hw_open(void)
{
ak_hw = open("/dev/ak4376", O_RDWR);
if(ak_hw < 0)
panicf("Cannot open '/dev/ak4376'");
if(ioctl(ak_hw, 0x20003424, 0) < 0)
{
panicf("Call cmd AK4376_POWER_ON fail");
}
}
static void hw_close(void)
{
if(ioctl(ak_hw, 0x20003425, 0) < 0)
{
panicf("Call cmd AK4376_POWER_OFF fail");
}
close(ak_hw);
ak_hw = -1;
}
void audiohw_preinit(void)
{
alsa_controls_init("default");
hw_open();
// NOTE:
// Of the exported controls, only these do anything:
// 10 DACL Playback Volume
// 11 DACR Playback Volume
// 12 Low Mode Switch (see table 25 in datasheet, not simple..)
audiohw_mute(false);
}
void audiohw_postinit(void)
{
}
void audiohw_close(void)
{
hw_close();
alsa_controls_close();
}
void audiohw_set_frequency(int fsel)
{
(void)fsel;
}
static int muted = -1;
void audiohw_set_volume(int vol_l, int vol_r)
{
int vol[2];
if (ak_hw < 0)
return;
vol[0] = vol_l / 20;
vol[1] = vol_r / 20;
for (int i = 0; i < 2; i++)
{
if (vol[i] > -56)
{
if (vol[i] < -12)
{
vol_hw[i] = 1;
vol_sw[i] = vol[i] + 12;
}
else
{
vol_hw[i] = 25 - (-vol[i] * 2);
vol_sw[i] = 0;
}
}
else
{
// Mute
vol_hw[i] = 0;
vol_sw[i] = 0;
}
}
if (!muted) {
alsa_controls_set_ints("DACL Playback Volume", 1, &vol_hw[0]);
alsa_controls_set_ints("DACR Playback Volume", 1, &vol_hw[1]);
pcm_set_mixer_volume(vol_sw[0], vol_sw[1]);
}
}
void audiohw_mute(int mute)
{
long int vol0 = 0;
if (ak_hw < 0 || muted == mute)
return;
muted = mute;
if(mute)
{
alsa_controls_set_ints("DACL Playback Volume", 1, &vol0);
alsa_controls_set_ints("DACR Playback Volume", 1, &vol0);
pcm_set_mixer_volume(0, 0);
}
else
{
alsa_controls_set_ints("DACL Playback Volume", 1, &vol_hw[0]);
alsa_controls_set_ints("DACR Playback Volume", 1, &vol_hw[1]);
pcm_set_mixer_volume(vol_sw[0], vol_sw[1]);
}
}
void audiohw_set_filter_roll_off(int value)
{
#if 0 // defined(FIIO_M3K_LINUX)
if (ak_hw < 0)
return;
/* 0 = Sharp;
1 = Slow;
2 = Short Sharp
3 = Short Slow */
// AK4376 supports this but the control isn't wired into ALSA!
long int value_hw = value;
alsa_controls_set_ints("AK4376 Digital Filter", 1, &value_hw);
#endif
(void)value;
}

12
firmware/target/hosted/fiio/fiiolinux_codec.h Normal file
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#ifndef __FIIOLINUX_CODEC__
#define __FIIOLINUX_CODEC__
#define AUDIOHW_CAPS (FILTER_ROLL_OFF_CAP)
#define AUDIOHW_HAVE_SHORT2_ROLL_OFF
AUDIOHW_SETTING(VOLUME, "dB", 0, 1, -100, 0, -30)
AUDIOHW_SETTING(FILTER_ROLL_OFF, "", 0, 1, 0, 4, 0)
#endif
#define AUDIOHW_MUTE_ON_STOP
void audiohw_mute(int mute);

55
firmware/target/hosted/ibasso/dx50/codec-dx50.h Normal file
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/***************************************************************************
* __________ __ ___
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
*
* Copyright (C) 2014 by Ilia Sergachev: Initial Rockbox port to iBasso DX50
* Copyright (C) 2014 by Mario Basister: iBasso DX90 port
* Copyright (C) 2014 by Simon Rothen: Initial Rockbox repository submission, additional features
* Copyright (C) 2014 by Udo Schläpfer: Code clean up, additional features
*
* 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.
*
****************************************************************************/
#ifndef _CODEC_DX50_H_
#define _CODEC_DX50_H_
#define AUDIOHW_CAPS (MONO_VOL_CAP | FILTER_ROLL_OFF_CAP)
/*
http://www.wolfsonmicro.com/media/76425/WM8740.pdf
0.5 * ( x - 255 ) = ydB 1 <= x <= 255
mute x = 0
x = 255 -> 0dB
.
.
.
x = 2 -> -126.5dB
x = 1 -> -127dB
x = 0 -> -128dB
See audiohw.h, sound.c.
*/
AUDIOHW_SETTING(VOLUME, "dB", 0, 1, -128, 0, -30)
/* 1: slow roll off, 0: sharp roll off, sharp roll off default */
AUDIOHW_SETTING(FILTER_ROLL_OFF, "", 0, 1, 0, 1, 0)
#endif

35
firmware/target/hosted/ibasso/dx90/codec-dx90.h Normal file
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/***************************************************************************
* __________ __ ___
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
*
* Copyright (C) 2014 by Ilia Sergachev: Initial Rockbox port to iBasso DX50
* Copyright (C) 2014 by Mario Basister: iBasso DX90 port
* Copyright (C) 2014 by Simon Rothen: Initial Rockbox repository submission, additional features
* Copyright (C) 2014 by Udo Schläpfer: Code clean up, additional features
*
* 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.
*
****************************************************************************/
#ifndef _CODEC_DX90_H_
#define _CODEC_DX90_H_
#define AUDIOHW_CAPS MONO_VOL_CAP
AUDIOHW_SETTING(VOLUME, "", 0, 1, -255, 0, -128)
#endif

33
firmware/target/hosted/sdl/sdl_codec.h Normal file
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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2010 by Thomas Martitz
*
* 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.
*
****************************************************************************/
#ifndef _SDL_CODEC_H
#define _SDL_CODEC_H
#if (defined(HAVE_SDL_AUDIO) \
&& !(CONFIG_PLATFORM & PLATFORM_MAEMO5)) \
|| (CONFIG_PLATFORM & PLATFORM_CTRU)
AUDIOHW_SETTING(VOLUME, "dB", 0, 1, -80, 0, 0)
#else
#define AUDIOHW_CAPS (MONO_VOL_CAP)
AUDIOHW_SETTING(VOLUME, "dB", 0, 1, -99, 0, 0)
#endif /* CONFIG_PLATFORM & PLATFORM_SDL */
#endif /* _SDL_CODEC_H */

437
firmware/target/hosted/sonynwz/nwzlinux-codec.c Normal file
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/***************************************************************************
* __________ __ ___.
* 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 <sys/ioctl.h>
#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 / 10, vol / 10);
}
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);
}
}

45
firmware/target/hosted/sonynwz/nwzlinux_codec.h Normal file
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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
*
* Copyright (C) 2016 by 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.
*
****************************************************************************/
#ifndef __NWZLINUX_CODEC_H__
#define __NWZLINUX_CODEC_H__
#define AUDIOHW_CAPS 0
/* Ranges from -100dB to 4dB */
AUDIOHW_SETTING(VOLUME, "dB", 0, 1, -100, 4, -10)
enum nwz_src_t
{
NWZ_PLAYBACK,
NWZ_RADIO,
NWZ_MIC,
};
/* enable/disable Sony's "acoustic" mode */
bool audiohw_acoustic_enabled(void);
void audiohw_enable_acoustic(bool en);
/* enable/disable Sony's "cuerev" mode */
bool audiohw_cuerev_enabled(void);
void audiohw_enable_cuerev(bool en);
/* select playback source */
void audiohw_set_playback_src(enum nwz_src_t src);
#endif /* __NWZLINUX_CODEC_H__ */

190
firmware/target/hosted/surfans/surfanslinux_codec.c Normal file
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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
*
* Copyright (c) 2018 Marcin Bukat
* Copyright (c) 2025 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 "button.h"
#include "system.h"
#include "kernel.h"
#include "panic.h"
#include "sysfs.h"
#include "alsa-controls.h"
#include "pcm-alsa.h"
#include "logf.h"
/*
f28:
**** List of PLAYBACK Hardware Devices ****
card 0: tyinx1 [tyin_x1], device 0: tyin_x1-es9018_k2m-i2s es9018_k2m-hifi-0 []
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: tyinx1 [tyin_x1], device 1: tyin_x1-es9018_k2m-pcm es9018_k2m-hifi-1 []
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: saspdif [sa_spdif], device 0: SA SPDIF Dummy spdif dump dai-0 []
Subdevices: 1/1
Subdevice #0: subdevice #0
Mixer controls:
numid=3,iface=MIXER,name='ES9018_K2M Digital Filter'
; type=INTEGER,access=rw------,values=1,min=0,max=4,step=0
: values=0
numid=5,iface=MIXER,name='Hardware Mute'
; type=BOOLEAN,access=rw------,values=1
: values=off
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=6,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;
void audiohw_mute(int mute)
{
if (hw_init < 0 || muted == mute)
return;
muted = mute;
alsa_controls_set_bool("Hardware Mute", !!mute);
}
int surfans_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_hs_switch = "/sys/class/switch/headset/state";
const char * const sysfs_bal_switch = "/sys/class/switch/balance/state";
sysfs_get_int(sysfs_hs_switch, &status);
if (status) ps = 2; // headset
sysfs_get_int(sysfs_bal_switch, &status);
if (status) ps = 3; // balanced output
surfans_set_output(ps);
return ps;
}
void surfans_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;
audiohw_mute(false); /* No need ? */
alsa_controls_set_bool("isDSD", 0);
}
void audiohw_postinit(void)
{
logf("hw postinit");
}
void audiohw_close(void)
{
logf("hw close");
hw_init = 0;
alsa_controls_close();
}
void audiohw_set_frequency(int fsel)
{
(void)fsel;
}
void audiohw_set_volume(int vol_l, int vol_r)
{
logf("hw vol %d %d", vol_l, vol_r);
long l,r;
vol_l_hw = vol_l;
vol_r_hw = vol_r;
l = -vol_l/5;
r = -vol_r/5;
if (!hw_init)
return;
alsa_controls_set_ints("Left Playback Volume", 1, &l);
alsa_controls_set_ints("Right Playback Volume", 1, &r);
}
void audiohw_set_filter_roll_off(int value)
{
logf("rolloff %d", value);
/* 0 = Sharp;
1 = Slow;
2 = Short Sharp
3 = Short Slow
4 = Super Slow */
#if defined(XDUOO_X20)
long int value_hw = value;
alsa_controls_set_ints("ES9018_K2M Digital Filter", 1, &value_hw);
#else
(void)value;
#endif
}

17
firmware/target/hosted/surfans/surfanslinux_codec.h Normal file
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#ifndef __SURFANSLINUX_CODEC__
#define __SURFANSLINUX_CODEC__
#define AUDIOHW_CAPS (FILTER_ROLL_OFF_CAP)
AUDIOHW_SETTING(VOLUME, "dB", 1, 5, -102*10, 0, -30*10)
#endif
//#define AUDIOHW_MUTE_ON_STOP
#define AUDIOHW_MUTE_ON_SRATE_CHANGE
//#define AUDIOHW_NEEDS_INITIAL_UNMUTE
AUDIOHW_SETTING(FILTER_ROLL_OFF, "", 0, 1, 0, 4, 0)
#define AUDIOHW_HAVE_SHORT2_ROLL_OFF
void audiohw_mute(int mute);
void surfans_set_output(int ps);
int surfans_get_outputs(void);

252
firmware/target/hosted/xduoo/xduoolinux_codec.c Normal file
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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
*
* Copyright (c) 2018 Marcin Bukat
* Copyright (c) 2018 Roman Stolyarov
*
* 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 "button.h"
#include "system.h"
#include "kernel.h"
#include "panic.h"
#include "sysfs.h"
#include "alsa-controls.h"
#include "pcm-alsa.h"
#include "logf.h"
/*
X3ii:
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 384000]
PERIOD_TIME: (1333 16384000]
PERIOD_SIZE: [512 131072]
PERIOD_BYTES: [4096 262144]
PERIODS: [4 256]
BUFFER_TIME: (5333 65536000]
BUFFER_SIZE: [2048 524288]
BUFFER_BYTES: [4096 1048576]
TICK_TIME: ALL
Mixer controls:
numid=1,iface=MIXER,name='Left Playback Volume'
; type=INTEGER,access=rw------,values=1,min=0,max=255,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=3,iface=MIXER,name='AK4490 Digital Filter'
; type=INTEGER,access=rw------,values=1,min=0,max=4,step=0
: values=0
numid=4,iface=MIXER,name='AK4490 Soft Mute'
; type=BOOLEAN,access=rw------,values=1
: values=off
numid=5,iface=MIXER,name='Output Port Switch'
; type=INTEGER,access=rw------,values=1,min=0,max=5,step=0
: values=0
*/
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;
void audiohw_mute(int mute)
{
logf("mute %d", mute);
if (!hw_init || muted == mute)
return;
muted = mute;
if(mute)
{
long int ps0 = 0;
alsa_controls_set_ints("Output Port Switch", 1, &ps0);
}
else
{
last_ps = 0;
xduoo_get_outputs();
}
}
int xduoo_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";
#if defined(XDUOO_X20)
const char * const sysfs_bal_switch = "/sys/class/switch/balance/state";
#endif
sysfs_get_int(sysfs_lo_switch, &status);
if (status) ps = 1; // lineout
sysfs_get_int(sysfs_hs_switch, &status);
if (status) ps = 2; // headset
#if defined(XDUOO_X20)
sysfs_get_int(sysfs_bal_switch, &status);
if (status) ps = 3; // balanced output
#endif
xduoo_set_output(ps);
return ps;
}
void xduoo_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);
#if defined(XDUOO_X3II)
/* Enable/disable headphone remote ADC */
sysfs_set_string("/sys/devices/platform/earpods_adc.0/earpods_adc/earpods_adc_sw", (ps == 2) ? "on" : "off");
#endif
}
}
void audiohw_preinit(void)
{
logf("hw preinit");
alsa_controls_init("default");
hw_init = 1;
#if defined(XDUOO_X3II)
audiohw_mute(true); /* Start muted to avoid the POP */
#else
audiohw_mute(false); /* No need */
#endif
// const char * const codec_pmdown = "/sys/devices/platform/ingenic-x3ii.0/x3ii-ak4490-i2s/pmdown_time"; // in ms, defaults 5000
}
void audiohw_postinit(void)
{
logf("hw postinit");
}
void audiohw_close(void)
{
logf("hw close");
hw_init = 0;
alsa_controls_close();
}
void audiohw_set_frequency(int fsel)
{
(void)fsel;
}
void audiohw_set_volume(int vol_l, int vol_r)
{
logf("hw vol %d %d", vol_l, vol_r);
long l,r;
vol_l_hw = vol_l;
vol_r_hw = vol_r;
if (lineout_inserted()) {
l = 0;
r = 0;
} else {
l = -vol_l/5;
r = -vol_r/5;
}
if (!hw_init)
return;
alsa_controls_set_ints("Left Playback Volume", 1, &l);
alsa_controls_set_ints("Right Playback Volume", 1, &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 = 0;
r = 0;
} else {
l = -vol_l_hw/5;
r = -vol_r_hw/5;
}
alsa_controls_set_ints("Left Playback Volume", 1, &l);
alsa_controls_set_ints("Right Playback Volume", 1, &r);
}
void audiohw_set_filter_roll_off(int value)
{
logf("rolloff %d", value);
/* 0 = Sharp;
1 = Slow;
2 = Short Sharp
3 = Short Slow
4 = Super Slow */
#if defined(XDUOO_X3II)
long int value_hw = value;
alsa_controls_set_ints("AK4490 Digital Filter", 1, &value_hw);
#elif defined(XDUOO_X20)
long int value_hw = value;
alsa_controls_set_ints("ES9018_K2M Digital Filter", 1, &value_hw);
#else
(void)value;
#endif
}

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firmware/target/hosted/xduoo/xduoolinux_codec.h Normal file
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#ifndef __XDUOOLINUX_CODEC__
#define __XDUOOLINUX_CODEC__
#define AUDIOHW_CAPS (LINEOUT_CAP | FILTER_ROLL_OFF_CAP)
AUDIOHW_SETTING(VOLUME, "dB", 1, 5, -102*10, 0, -30*10)
#endif
// We want this, but the codec takes over a second to unmute!
//#define AUDIOHW_MUTE_ON_STOP
#if defined(XDUOO_X3II)
/* The AK4490 glitches when switching sample rates */
#define AUDIOHW_MUTE_ON_SRATE_CHANGE
AUDIOHW_SETTING(FILTER_ROLL_OFF, "", 0, 1, 0, 5, 0)
#define AUDIOHW_HAVE_SS_ROLL_OFF
#endif
#if defined(XDUOO_X20)
//#define AUDIOHW_NEEDS_INITIAL_UNMUTE
AUDIOHW_SETTING(FILTER_ROLL_OFF, "", 0, 1, 0, 4, 0)
#define AUDIOHW_HAVE_SHORT2_ROLL_OFF
#endif
void audiohw_mute(int mute);
void xduoo_set_output(int ps);
int xduoo_get_outputs(void);