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audio: Move hosted audio "codec" drivers into their respective target dirs
They are nearly entirely generic wrappers around ALSA controls, unique per target, and are ripe for further consolidation. Change-Id: I05e4a450e3e89e03616906601c4f8fa46200dff5
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parent
d376e0afb7
commit
a79bdaf462
20 changed files with 62 additions and 43 deletions
281
firmware/target/hosted/aigo/erosqlinux_codec.c
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281
firmware/target/hosted/aigo/erosqlinux_codec.c
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/***************************************************************************
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* __________ __ ___.
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* Open \______ \ ____ ____ | | _\_ |__ _______ ___
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* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
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* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
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* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
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* \/ \/ \/ \/ \/
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* $Id$
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*
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*
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* Copyright (c) 2018 Marcin Bukat
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* Copyright (c) 2020 Solomon Peachy
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* as published by the Free Software Foundation; either version 2
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* of the License, or (at your option) any later version.
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*
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* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
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* KIND, either express or implied.
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*
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****************************************************************************/
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// #define LOGF_ENABLE
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#include "config.h"
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#include "audio.h"
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#include "audiohw.h"
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#include "system.h"
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#include "panic.h"
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#include "sysfs.h"
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#include "alsa-controls.h"
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#include "pcm-alsa.h"
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#include "settings.h"
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#include "logf.h"
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/*
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PCM device hw:0,0
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ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED
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FORMAT: S16_LE S24_LE
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SUBFORMAT: STD
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SAMPLE_BITS: [16 32]
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FRAME_BITS: [16 64]
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CHANNELS: [1 2]
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RATE: [8000 192000]
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PERIOD_TIME: (2666 8192000]
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PERIOD_SIZE: [512 65536]
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PERIOD_BYTES: [4096 131072]
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PERIODS: [4 128]
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BUFFER_TIME: (10666 32768000]
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BUFFER_SIZE: [2048 262144]
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BUFFER_BYTES: [4096 524288]
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TICK_TIME: ALL
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Mixer controls (v1):
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numid=1,iface=MIXER,name='Output Port Switch'
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; type=INTEGER,access=rw------,values=1,min=0,max=5,step=0
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: values=4
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Mixer controls (v2+):
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numid=3,iface=MIXER,name='ES9018_K2M Digital Filter'
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; type=INTEGER,access=rw------,values=1,min=0,max=4,step=0
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: values=0
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numid=1,iface=MIXER,name='Left Playback Volume'
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; type=INTEGER,access=rw------,values=1,min=0,max=255,step=0
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: values=0
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numid=4,iface=MIXER,name='Output Port Switch'
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; type=INTEGER,access=rw------,values=1,min=0,max=5,step=0
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: values=0
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numid=2,iface=MIXER,name='Right Playback Volume'
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; type=INTEGER,access=rw------,values=1,min=0,max=255,step=0
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: values=0
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numid=5,iface=MIXER,name='isDSD'
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; type=BOOLEAN,access=rw------,values=1
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: values=off
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*/
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static int hw_init = 0;
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static long int vol_l_hw = 255;
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static long int vol_r_hw = 255;
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static long int last_ps = -1;
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static int muted = -1;
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extern int hwver;
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void audiohw_mute(int mute)
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{
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logf("mute %d", mute);
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if (hw_init < 0 || muted == mute)
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return;
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muted = mute;
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if(mute)
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{
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long int ps0 = 0;
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alsa_controls_set_ints("Output Port Switch", 1, &ps0);
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}
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else
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{
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last_ps = 0;
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erosq_get_outputs();
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}
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}
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int erosq_get_outputs(void) {
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long int ps = 0; // Muted, if nothing is plugged in!
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int status = 0;
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if (!hw_init) return ps;
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const char * const sysfs_lo_switch = "/sys/class/switch/lineout/state";
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const char * const sysfs_hs_switch = "/sys/class/switch/headset/state";
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sysfs_get_int(sysfs_lo_switch, &status);
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if (status) ps = 1; // lineout
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sysfs_get_int(sysfs_hs_switch, &status);
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if (status) ps = 2; // headset
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erosq_set_output(ps);
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return ps;
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}
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void erosq_set_output(int ps)
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{
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if (!hw_init || muted) return;
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if (last_ps != ps)
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{
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logf("set out %d/%ld", ps, last_ps);
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/* Output port switch */
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last_ps = ps;
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alsa_controls_set_ints("Output Port Switch", 1, &last_ps);
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audiohw_set_volume(vol_l_hw, vol_r_hw);
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}
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}
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void audiohw_preinit(void)
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{
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logf("hw preinit");
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alsa_controls_init("default");
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hw_init = 1;
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/* See if we have hw2 or later */
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if (alsa_controls_find("Left Playback Volume") == -1)
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hwver = 1;
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else if (hwver == 1)
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hwver = 23;
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audiohw_mute(false); /* No need to stay muted */
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}
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void audiohw_postinit(void)
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{
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logf("hw postinit");
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}
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void audiohw_close(void)
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{
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logf("hw close");
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hw_init = 0;
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muted = -1;
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alsa_controls_close();
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}
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void audiohw_set_frequency(int fsel)
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{
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(void)fsel;
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}
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/* min/max for pcm volume */
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const int min_pcm = -740;
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const int max_pcm = 0;
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static void audiohw_set_volume_v1(int vol_l, int vol_r)
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{
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long l,r;
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vol_l_hw = vol_l;
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vol_r_hw = vol_r;
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logf("set_volume_v1 %d, %d", vol_l, vol_r);
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if (lineout_inserted() && !headphones_inserted()) {
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/* On the EROS Q/K hardware, full scale line out is _very_ hot
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at ~5.8Vpp. As the hardware provides no way to reduce
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output gain, we have to back off on the PCM signal
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to avoid blowing out the signal.
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*/
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l = r = global_settings.volume_limit * 10;
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} else {
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l = vol_l_hw;
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r = vol_r_hw;
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}
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int sw_volume_l = l <= min_pcm ? min_pcm : MIN(l, max_pcm);
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int sw_volume_r = r <= min_pcm ? min_pcm : MIN(r, max_pcm);
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pcm_set_mixer_volume(sw_volume_l / 20, sw_volume_r / 20);
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}
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static void audiohw_set_volume_v2(int vol_l, int vol_r)
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{
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long l,r;
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vol_l_hw = vol_l;
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vol_r_hw = vol_r;
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logf("set_volume_v2 %d, %d", vol_l, vol_r);
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if (lineout_inserted() && !headphones_inserted()) {
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// was l = r = ... syntax.
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// for some reason r channel was not getting set
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// and was quiet...?
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l = -1 * (global_settings.volume_limit * 10) / 5;
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r = l;
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} else {
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// never save volume as positive, we will
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// just oscillate between positive and negative then
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l = -1 * vol_l_hw / 5;
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r = -1 * vol_r_hw / 5;
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}
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if (!hw_init)
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return;
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alsa_controls_set_ints("Left Playback Volume", 1, &l);
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alsa_controls_set_ints("Right Playback Volume", 1, &r);
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/* Dial back PCM mixer to avoid compression */
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pcm_set_mixer_volume(global_settings.volume_limit / 2, global_settings.volume_limit / 2);
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}
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void audiohw_set_volume(int vol_l, int vol_r)
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{
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if (hwver >= 2) {
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audiohw_set_volume_v2(vol_l, vol_r);
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} else {
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audiohw_set_volume_v1(vol_l, vol_r);
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}
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}
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void audiohw_set_lineout_volume(int vol_l, int vol_r)
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{
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long l,r;
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(void)vol_l;
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(void)vol_r;
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if (lineout_inserted() && !headphones_inserted()) {
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// was l = r = ... syntax.
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// for some reason r channel was not getting set
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// and was quiet...?
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l = global_settings.volume_limit * 10;
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r = l;
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logf("lo vol %ld %ld", l, r);
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if (hw_init){
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if (hwver >= 2) {
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l /= 5;
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l = l * -1;
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r /= 5;
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r = r * -1;
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alsa_controls_set_ints("Left Playback Volume", 1, &l);
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alsa_controls_set_ints("Right Playback Volume", 1, &r);
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} else {
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int sw_volume_l = l <= min_pcm ? min_pcm : MIN(l, max_pcm);
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int sw_volume_r = r <= min_pcm ? min_pcm : MIN(r, max_pcm);
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pcm_set_mixer_volume(sw_volume_l / 20, sw_volume_r / 20);
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}
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}
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}
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}
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23
firmware/target/hosted/aigo/erosqlinux_codec.h
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23
firmware/target/hosted/aigo/erosqlinux_codec.h
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@ -0,0 +1,23 @@
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#ifndef __EROSQLINUX_CODEC__
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#define __EROSQLINUX_CODEC__
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#define AUDIOHW_CAPS (LINEOUT_CAP)
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/* a small DC offset prevents play/pause clicking due to the DAC auto-muting */
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#define PCM_DC_OFFSET_VALUE -1
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/*
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* Note: Maximum volume is set one step below unity in order to
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* avoid overflowing pcm samples due to our DC Offset.
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*
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* The DAC's output is hot enough this should not be an issue.
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*/
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AUDIOHW_SETTING(VOLUME, "dB", 0, 2, -74, -2, -40)
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//#define AUDIOHW_NEEDS_INITIAL_UNMUTE
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void audiohw_mute(int mute);
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void erosq_set_output(int ps);
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int erosq_get_outputs(void);
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#endif
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