Clone of file from sxmo project It handles audio routing when in a phone call
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/*
* Voice call audio setup tool
*
* Copyright (C) 2020 Ondřej Jirman <megous@megous.com>
*
* 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 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program. If not, see <http://www.gnu.org/licenses/>.
*
* 2020-09-29: Updated for the new Samuel's digital codec driver
*/
#include <assert.h>
#include <stdlib.h>
#include <stdbool.h>
#include <stdio.h>
#include <stdarg.h>
#include <stdint.h>
#include <string.h>
#include <errno.h>
#include <unistd.h>
#include <inttypes.h>
#include <fcntl.h>
#include <sys/ioctl.h>
#include <sound/asound.h>
#include <sound/tlv.h>
#define ARRAY_SIZE(a) (sizeof((a)) / sizeof((a)[0]))
void syscall_error(int is_err, const char* fmt, ...)
{
va_list ap;
if (!is_err)
return;
printf("ERROR: ");
va_start(ap, fmt);
vprintf(fmt, ap);
va_end(ap);
printf(": %s\n", strerror(errno));
exit(1);
}
void error(const char* fmt, ...)
{
va_list ap;
printf("ERROR: ");
va_start(ap, fmt);
vprintf(fmt, ap);
va_end(ap);
printf("\n");
exit(1);
}
struct audio_control_state {
char name[128];
union {
int64_t i[4];
const char* e[4];
} vals;
bool used;
};
static bool audio_restore_state(struct audio_control_state* controls, int n_controls)
{
int fd;
int ret;
fd = open("/dev/snd/controlC0", O_CLOEXEC | O_NONBLOCK);
if (fd < 0)
error("failed to open card\n");
struct snd_ctl_elem_list el = {
.offset = 0,
.space = 0,
};
ret = ioctl(fd, SNDRV_CTL_IOCTL_ELEM_LIST, &el);
syscall_error(ret < 0, "SNDRV_CTL_IOCTL_ELEM_LIST failed");
struct snd_ctl_elem_id ids[el.count];
el.pids = ids;
el.space = el.count;
ret = ioctl(fd, SNDRV_CTL_IOCTL_ELEM_LIST, &el);
syscall_error(ret < 0, "SNDRV_CTL_IOCTL_ELEM_LIST failed");
for (int i = 0; i < el.used; i++) {
struct snd_ctl_elem_info inf = {
.id = ids[i],
};
ret = ioctl(fd, SNDRV_CTL_IOCTL_ELEM_INFO, &inf);
syscall_error(ret < 0, "SNDRV_CTL_IOCTL_ELEM_INFO failed");
if ((inf.access & SNDRV_CTL_ELEM_ACCESS_READ) && (inf.access & SNDRV_CTL_ELEM_ACCESS_WRITE)) {
struct snd_ctl_elem_value val = {
.id = ids[i],
};
int64_t cval = 0;
ret = ioctl(fd, SNDRV_CTL_IOCTL_ELEM_READ, &val);
syscall_error(ret < 0, "SNDRV_CTL_IOCTL_ELEM_READ failed");
struct audio_control_state* cs = NULL;
for (int j = 0; j < n_controls; j++) {
if (!strcmp(controls[j].name, ids[i].name)) {
cs = &controls[j];
break;
}
}
if (!cs) {
printf("Control \"%s\" si not defined in the controls state\n", ids[i].name);
continue;
}
cs->used = 1;
// check if value needs changing
switch (inf.type) {
case SNDRV_CTL_ELEM_TYPE_BOOLEAN:
case SNDRV_CTL_ELEM_TYPE_INTEGER:
for (int j = 0; j < inf.count; j++) {
if (cs->vals.i[j] != val.value.integer.value[j]) {
// update
//printf("%s <=[%d]= %"PRIi64"\n", ids[i].name, j, cs->vals.i[j]);
val.value.integer.value[j] = cs->vals.i[j];
ret = ioctl(fd, SNDRV_CTL_IOCTL_ELEM_WRITE, &val);
syscall_error(ret < 0, "SNDRV_CTL_IOCTL_ELEM_WRITE failed");
}
}
break;
case SNDRV_CTL_ELEM_TYPE_INTEGER64:
for (int j = 0; j < inf.count; j++) {
if (cs->vals.i[j] != val.value.integer64.value[j]) {
// update
//printf("%s <=[%d]= %"PRIi64"\n", ids[i].name, j, cs->vals.i[j]);
val.value.integer64.value[j] = cs->vals.i[j];
ret = ioctl(fd, SNDRV_CTL_IOCTL_ELEM_WRITE, &val);
syscall_error(ret < 0, "SNDRV_CTL_IOCTL_ELEM_WRITE failed");
}
}
break;
case SNDRV_CTL_ELEM_TYPE_ENUMERATED: {
for (int k = 0; k < inf.count; k++) {
int eval = -1;
for (int j = 0; j < inf.value.enumerated.items; j++) {
inf.value.enumerated.item = j;
ret = ioctl(fd, SNDRV_CTL_IOCTL_ELEM_INFO, &inf);
syscall_error(ret < 0, "SNDRV_CTL_IOCTL_ELEM_INFO failed");
if (!strcmp(cs->vals.e[k], inf.value.enumerated.name)) {
eval = j;
break;
}
}
if (eval < 0)
error("enum value %s not found\n", cs->vals.e[k]);
if (eval != val.value.enumerated.item[k]) {
// update
//printf("%s <=%d= %s\n", ids[i].name, k, cs->vals.e[k]);
val.value.enumerated.item[k] = eval;
ret = ioctl(fd, SNDRV_CTL_IOCTL_ELEM_WRITE, &val);
syscall_error(ret < 0, "SNDRV_CTL_IOCTL_ELEM_WRITE failed");
}
}
break;
}
}
}
}
for (int j = 0; j < n_controls; j++)
if (!controls[j].used)
printf("Control \"%s\" is defined in state but not present on the card\n", controls[j].name);
close(fd);
return true;
}
struct audio_setup {
bool mic_on;
bool spk_on;
bool hp_on;
bool ear_on;
// when sending audio to modem from AIF1 R, also play that back
// to me locally (just like AIF1 L plays just to me)
//
// this is to monitor what SW is playing to the modem (so that
// I can hear my robocaller talking)
bool modem_playback_monitor;
// enable modem routes to DAC/from ADC (spk/mic)
// digital paths to AIF1 are always on
bool to_modem_on;
bool from_modem_on;
// shut off/enable all digital paths to the modem:
// keep this off until the call starts, then turn it on
bool dai2_en;
int mic_gain;
int spk_vol;
int ear_vol;
int hp_vol;
};
static void audio_set_controls(struct audio_setup* s)
{
struct audio_control_state controls[] = {
//
// Analog input:
//
// Mic 1 (daughterboard)
{ .name = "Mic1 Boost Volume", .vals.i = { s->mic_gain } },
// Mic 2 (headphones)
{ .name = "Mic2 Boost Volume", .vals.i = { 0 } },
// Line in (unused on PP)
// no controls yet
// Input mixers before ADC
{ .name = "Mic1 Capture Switch", .vals.i = { !!s->mic_on, !!s->mic_on } },
{ .name = "Mic2 Capture Switch", .vals.i = { 0, 0 } },
{ .name = "Line In Capture Switch", .vals.i = { 0, 0 } }, // Out Mix -> In Mix
{ .name = "Mixer Capture Switch", .vals.i = { 0, 0 } },
{ .name = "Mixer Reversed Capture Switch", .vals.i = { 0, 0 } },
// ADC
{ .name = "ADC Gain Capture Volume", .vals.i = { 0 } },
{ .name = "ADC Capture Volume", .vals.i = { 160, 160 } }, // digital gain
//
// Digital paths:
//
// AIF1 (SoC)
// AIF1 slot0 capture mixer sources
{ .name = "AIF1 Data Digital ADC Capture Switch", .vals.i = { 1, 0 } },
{ .name = "AIF1 Slot 0 Digital ADC Capture Switch", .vals.i = { 0, 0 } },
{ .name = "AIF2 Digital ADC Capture Switch", .vals.i = { 0, 1 } },
{ .name = "AIF2 Inv Digital ADC Capture Switch", .vals.i = { 0, 0 } }, //XXX: capture right from the left AIF2?
// AIF1 slot0 capture/playback mono mixing/digital volume
{ .name = "AIF1 AD0 Capture Volume", .vals.i = { 160, 160 } },
{ .name = "AIF1 AD0 Stereo Capture Route", .vals.e = { "Stereo", "Stereo" } },
{ .name = "AIF1 DA0 Playback Volume", .vals.i = { 160, 160 } },
{ .name = "AIF1 DA0 Stereo Playback Route", .vals.e = { "Stereo", "Stereo" } },
// AIF2 (modem)
// AIF2 capture mixer sources
{ .name = "AIF2 ADC Mixer ADC Capture Switch", .vals.i = { !!s->to_modem_on && !!s->dai2_en, 0 } }, // from adc/mic
{ .name = "AIF2 ADC Mixer AIF1 DA0 Capture Switch", .vals.i = { 0, 1 } }, // from aif1 R
{ .name = "AIF2 ADC Mixer AIF2 DAC Rev Capture Switch", .vals.i = { 0, 0 } },
// AIF2 capture/playback mono mixing/digital volume
{ .name = "AIF2 ADC Capture Volume", .vals.i = { 160, 160 } },
{ .name = "AIF2 DAC Playback Volume", .vals.i = { 160, 160 } },
{ .name = "AIF2 ADC Stereo Capture Route", .vals.e = { "Mix Mono", "Mix Mono" } }, // we mix because we're sending two channels (from mic and AIF1 R)
{ .name = "AIF2 DAC Stereo Playback Route", .vals.e = { "Sum Mono", "Sum Mono" } }, // we sum because modem is sending a single channel
// AIF3 (bluetooth)
{ .name = "AIF3 ADC Source Capture Route", .vals.e = { "None" } },
{ .name = "AIF2 DAC Source Playback Route", .vals.e = { "AIF2" } },
// DAC
// DAC input mixers (sources from ADC, and AIF1/2)
{ .name = "ADC Digital DAC Playback Switch", .vals.i = { 0, 0 } }, // we don't play our mic to ourselves
{ .name = "AIF1 Slot 0 Digital DAC Playback Switch", .vals.i = { 1, !!s->modem_playback_monitor } },
{ .name = "AIF2 Digital DAC Playback Switch", .vals.i = { 0, !!s->dai2_en && !!s->from_modem_on } },
//
// Analog output:
//
// Output mixer after DAC
{ .name = "DAC Playback Switch", .vals.i = { 1, 1 } },
{ .name = "DAC Reversed Playback Switch", .vals.i = { 1, 1 } },
{ .name = "DAC Playback Volume", .vals.i = { 160, 160 } },
{ .name = "Mic1 Playback Switch", .vals.i = { 0, 0 } },
{ .name = "Mic1 Playback Volume", .vals.i = { 0 } },
{ .name = "Mic2 Playback Switch", .vals.i = { 0, 0 } },
{ .name = "Mic2 Playback Volume", .vals.i = { 0 } },
{ .name = "Line In Playback Switch", .vals.i = { 0, 0 } },
{ .name = "Line In Playback Volume", .vals.i = { 0 } },
// Outputs
{ .name = "Earpiece Source Playback Route", .vals.e = { "Left Mixer" } },
{ .name = "Earpiece Playback Switch", .vals.i = { !!s->ear_on } },
{ .name = "Earpiece Playback Volume", .vals.i = { s->ear_vol } },
{ .name = "Headphone Source Playback Route", .vals.e = { "Mixer", "Mixer" } },
{ .name = "Headphone Playback Switch", .vals.i = { !!s->hp_on, !!s->hp_on } },
{ .name = "Headphone Playback Volume", .vals.i = { s->hp_vol } },
// Loudspeaker
{ .name = "Line Out Source Playback Route", .vals.e = { "Mono Differential", "Mono Differential" } },
{ .name = "Line Out Playback Switch", .vals.i = { !!s->spk_on, !!s->spk_on } },
{ .name = "Line Out Playback Volume", .vals.i = { s->spk_vol } },
};
audio_restore_state(controls, ARRAY_SIZE(controls));
}
static struct audio_setup audio_setup = {
.mic_on = true,
.ear_on = true,
.spk_on = false,
.hp_on = false,
.from_modem_on = true,
.to_modem_on = true,
.modem_playback_monitor = false,
.dai2_en = false,
.hp_vol = 15,
.spk_vol = 15,
.ear_vol = 31,
.mic_gain = 1,
};
int main(int ac, char* av[])
{
int opt;
while ((opt = getopt(ac, av, "smhe2")) != -1) {
switch (opt) {
case 's':
audio_setup.spk_on = 1;
break;
case 'm':
audio_setup.mic_on = 1;
break;
case 'h':
audio_setup.hp_on = 1;
break;
case 'e':
audio_setup.ear_on = 1;
break;
case '2':
audio_setup.dai2_en = 1;
break;
default: /* '?' */
fprintf(stderr, "Usage: %s [-s] [-m] [-h] [-e] [-2]\n", av[0]);
exit(EXIT_FAILURE);
}
}
audio_set_controls(&audio_setup);
return 0;
}