I don't want all the utils from sxmo, just this so I copied it. <https://git.sr.ht/~mil/sxmo-utils/tree/master/item/programs/sxmo_megiaudioroute.c>master
commit
6d84cfee06
4 changed files with 402 additions and 0 deletions
@ -0,0 +1,11 @@ |
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PREFIX = /usr/local
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.PHONY: install |
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megiaudioroute: megiaudioroute.c |
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gcc -o megiaudioroute megiaudioroute.c -lX11
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install: megiaudioroute |
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mkdir -p $(PREFIX)/bin
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cp -f megiaudioroute $(PREFIX)/bin/megiaudioroute
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chmod 755 $(PREFIX)/bin/megiaudioroute
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@ -0,0 +1,5 @@ |
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This is a tool for setting up audio on phone calls |
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Taken from the sxmo project. |
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<https://git.sr.ht/~mil/sxmo-utils/tree/master/item/programs/sxmo_megiaudioroute.c> |
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/*
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* Voice call audio setup tool |
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* |
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* Copyright (C) 2020 Ondřej Jirman <megous@megous.com> |
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* |
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* This program is free software: you can redistribute it and/or modify |
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* it under the terms of the GNU General Public License as published by |
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* the Free Software Foundation, either version 3 of the License, or |
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* (at your option) any later version. |
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* |
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* This program is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
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* GNU General Public License for more details. |
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* |
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* You should have received a copy of the GNU General Public License |
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* along with this program. If not, see <http://www.gnu.org/licenses/>.
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* |
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* 2020-09-29: Updated for the new Samuel's digital codec driver |
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*/ |
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#include <assert.h> |
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#include <stdlib.h> |
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#include <stdbool.h> |
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#include <stdio.h> |
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#include <stdarg.h> |
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#include <stdint.h> |
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#include <string.h> |
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#include <errno.h> |
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#include <unistd.h> |
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#include <inttypes.h> |
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#include <fcntl.h> |
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#include <sys/ioctl.h> |
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#include <sound/asound.h> |
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#include <sound/tlv.h> |
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#define ARRAY_SIZE(a) (sizeof((a)) / sizeof((a)[0])) |
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void syscall_error(int is_err, const char* fmt, ...) |
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{ |
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va_list ap; |
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if (!is_err) |
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return; |
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printf("ERROR: "); |
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va_start(ap, fmt); |
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vprintf(fmt, ap); |
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va_end(ap); |
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printf(": %s\n", strerror(errno)); |
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exit(1); |
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} |
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void error(const char* fmt, ...) |
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{ |
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va_list ap; |
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printf("ERROR: "); |
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va_start(ap, fmt); |
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vprintf(fmt, ap); |
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va_end(ap); |
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printf("\n"); |
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exit(1); |
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} |
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struct audio_control_state { |
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char name[128]; |
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union { |
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int64_t i[4]; |
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const char* e[4]; |
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} vals; |
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bool used; |
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}; |
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static bool audio_restore_state(struct audio_control_state* controls, int n_controls) |
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{ |
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int fd; |
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int ret; |
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fd = open("/dev/snd/controlC0", O_CLOEXEC | O_NONBLOCK); |
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if (fd < 0) |
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error("failed to open card\n"); |
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struct snd_ctl_elem_list el = { |
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.offset = 0, |
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.space = 0, |
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}; |
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ret = ioctl(fd, SNDRV_CTL_IOCTL_ELEM_LIST, &el); |
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syscall_error(ret < 0, "SNDRV_CTL_IOCTL_ELEM_LIST failed"); |
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struct snd_ctl_elem_id ids[el.count]; |
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el.pids = ids; |
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el.space = el.count; |
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ret = ioctl(fd, SNDRV_CTL_IOCTL_ELEM_LIST, &el); |
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syscall_error(ret < 0, "SNDRV_CTL_IOCTL_ELEM_LIST failed"); |
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for (int i = 0; i < el.used; i++) { |
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struct snd_ctl_elem_info inf = { |
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.id = ids[i], |
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}; |
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ret = ioctl(fd, SNDRV_CTL_IOCTL_ELEM_INFO, &inf); |
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syscall_error(ret < 0, "SNDRV_CTL_IOCTL_ELEM_INFO failed"); |
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if ((inf.access & SNDRV_CTL_ELEM_ACCESS_READ) && (inf.access & SNDRV_CTL_ELEM_ACCESS_WRITE)) { |
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struct snd_ctl_elem_value val = { |
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.id = ids[i], |
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}; |
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int64_t cval = 0; |
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ret = ioctl(fd, SNDRV_CTL_IOCTL_ELEM_READ, &val); |
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syscall_error(ret < 0, "SNDRV_CTL_IOCTL_ELEM_READ failed"); |
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struct audio_control_state* cs = NULL; |
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for (int j = 0; j < n_controls; j++) { |
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if (!strcmp(controls[j].name, ids[i].name)) { |
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cs = &controls[j]; |
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break; |
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} |
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} |
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if (!cs) { |
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printf("Control \"%s\" si not defined in the controls state\n", ids[i].name); |
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continue; |
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} |
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cs->used = 1; |
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// check if value needs changing
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switch (inf.type) { |
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case SNDRV_CTL_ELEM_TYPE_BOOLEAN: |
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case SNDRV_CTL_ELEM_TYPE_INTEGER: |
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for (int j = 0; j < inf.count; j++) { |
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if (cs->vals.i[j] != val.value.integer.value[j]) { |
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// update
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//printf("%s <=[%d]= %"PRIi64"\n", ids[i].name, j, cs->vals.i[j]);
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val.value.integer.value[j] = cs->vals.i[j]; |
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ret = ioctl(fd, SNDRV_CTL_IOCTL_ELEM_WRITE, &val); |
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syscall_error(ret < 0, "SNDRV_CTL_IOCTL_ELEM_WRITE failed"); |
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} |
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} |
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break; |
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case SNDRV_CTL_ELEM_TYPE_INTEGER64: |
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for (int j = 0; j < inf.count; j++) { |
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if (cs->vals.i[j] != val.value.integer64.value[j]) { |
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// update
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//printf("%s <=[%d]= %"PRIi64"\n", ids[i].name, j, cs->vals.i[j]);
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val.value.integer64.value[j] = cs->vals.i[j]; |
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ret = ioctl(fd, SNDRV_CTL_IOCTL_ELEM_WRITE, &val); |
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syscall_error(ret < 0, "SNDRV_CTL_IOCTL_ELEM_WRITE failed"); |
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} |
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} |
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break; |
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case SNDRV_CTL_ELEM_TYPE_ENUMERATED: { |
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for (int k = 0; k < inf.count; k++) { |
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int eval = -1; |
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for (int j = 0; j < inf.value.enumerated.items; j++) { |
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inf.value.enumerated.item = j; |
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ret = ioctl(fd, SNDRV_CTL_IOCTL_ELEM_INFO, &inf); |
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syscall_error(ret < 0, "SNDRV_CTL_IOCTL_ELEM_INFO failed"); |
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if (!strcmp(cs->vals.e[k], inf.value.enumerated.name)) { |
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eval = j; |
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break; |
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} |
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} |
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if (eval < 0) |
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error("enum value %s not found\n", cs->vals.e[k]); |
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if (eval != val.value.enumerated.item[k]) { |
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// update
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//printf("%s <=%d= %s\n", ids[i].name, k, cs->vals.e[k]);
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val.value.enumerated.item[k] = eval; |
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ret = ioctl(fd, SNDRV_CTL_IOCTL_ELEM_WRITE, &val); |
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syscall_error(ret < 0, "SNDRV_CTL_IOCTL_ELEM_WRITE failed"); |
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} |
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} |
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break; |
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} |
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} |
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} |
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} |
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for (int j = 0; j < n_controls; j++) |
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if (!controls[j].used) |
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printf("Control \"%s\" is defined in state but not present on the card\n", controls[j].name); |
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close(fd); |
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return true; |
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} |
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struct audio_setup { |
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bool mic_on; |
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bool spk_on; |
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bool hp_on; |
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bool ear_on; |
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// when sending audio to modem from AIF1 R, also play that back
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// to me locally (just like AIF1 L plays just to me)
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//
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// this is to monitor what SW is playing to the modem (so that
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// I can hear my robocaller talking)
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bool modem_playback_monitor; |
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// enable modem routes to DAC/from ADC (spk/mic)
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// digital paths to AIF1 are always on
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bool to_modem_on; |
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bool from_modem_on; |
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// shut off/enable all digital paths to the modem:
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// keep this off until the call starts, then turn it on
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bool dai2_en; |
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int mic_gain; |
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int spk_vol; |
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int ear_vol; |
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int hp_vol; |
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}; |
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static void audio_set_controls(struct audio_setup* s) |
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{ |
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struct audio_control_state controls[] = { |
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//
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// Analog input:
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//
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// Mic 1 (daughterboard)
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{ .name = "Mic1 Boost Volume", .vals.i = { s->mic_gain } }, |
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// Mic 2 (headphones)
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{ .name = "Mic2 Boost Volume", .vals.i = { 0 } }, |
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// Line in (unused on PP)
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// no controls yet
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// Input mixers before ADC
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{ .name = "Mic1 Capture Switch", .vals.i = { !!s->mic_on, !!s->mic_on } }, |
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{ .name = "Mic2 Capture Switch", .vals.i = { 0, 0 } }, |
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{ .name = "Line In Capture Switch", .vals.i = { 0, 0 } }, // Out Mix -> In Mix
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{ .name = "Mixer Capture Switch", .vals.i = { 0, 0 } }, |
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{ .name = "Mixer Reversed Capture Switch", .vals.i = { 0, 0 } }, |
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// ADC
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{ .name = "ADC Gain Capture Volume", .vals.i = { 0 } }, |
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{ .name = "ADC Capture Volume", .vals.i = { 160, 160 } }, // digital gain
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//
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// Digital paths:
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//
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// AIF1 (SoC)
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// AIF1 slot0 capture mixer sources
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{ .name = "AIF1 Data Digital ADC Capture Switch", .vals.i = { 1, 0 } }, |
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{ .name = "AIF1 Slot 0 Digital ADC Capture Switch", .vals.i = { 0, 0 } }, |
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{ .name = "AIF2 Digital ADC Capture Switch", .vals.i = { 0, 1 } }, |
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{ .name = "AIF2 Inv Digital ADC Capture Switch", .vals.i = { 0, 0 } }, //XXX: capture right from the left AIF2?
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// AIF1 slot0 capture/playback mono mixing/digital volume
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{ .name = "AIF1 AD0 Capture Volume", .vals.i = { 160, 160 } }, |
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{ .name = "AIF1 AD0 Stereo Capture Route", .vals.e = { "Stereo", "Stereo" } }, |
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{ .name = "AIF1 DA0 Playback Volume", .vals.i = { 160, 160 } }, |
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{ .name = "AIF1 DA0 Stereo Playback Route", .vals.e = { "Stereo", "Stereo" } }, |
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// AIF2 (modem)
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// AIF2 capture mixer sources
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{ .name = "AIF2 ADC Mixer ADC Capture Switch", .vals.i = { !!s->to_modem_on && !!s->dai2_en, 0 } }, // from adc/mic
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{ .name = "AIF2 ADC Mixer AIF1 DA0 Capture Switch", .vals.i = { 0, 1 } }, // from aif1 R
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{ .name = "AIF2 ADC Mixer AIF2 DAC Rev Capture Switch", .vals.i = { 0, 0 } }, |
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// AIF2 capture/playback mono mixing/digital volume
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{ .name = "AIF2 ADC Capture Volume", .vals.i = { 160, 160 } }, |
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{ .name = "AIF2 DAC Playback Volume", .vals.i = { 160, 160 } }, |
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{ .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)
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{ .name = "AIF2 DAC Stereo Playback Route", .vals.e = { "Sum Mono", "Sum Mono" } }, // we sum because modem is sending a single channel
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// AIF3 (bluetooth)
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{ .name = "AIF3 ADC Source Capture Route", .vals.e = { "None" } }, |
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{ .name = "AIF2 DAC Source Playback Route", .vals.e = { "AIF2" } }, |
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// DAC
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// DAC input mixers (sources from ADC, and AIF1/2)
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{ .name = "ADC Digital DAC Playback Switch", .vals.i = { 0, 0 } }, // we don't play our mic to ourselves
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{ .name = "AIF1 Slot 0 Digital DAC Playback Switch", .vals.i = { 1, !!s->modem_playback_monitor } }, |
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{ .name = "AIF2 Digital DAC Playback Switch", .vals.i = { 0, !!s->dai2_en && !!s->from_modem_on } }, |
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//
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// Analog output:
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//
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// Output mixer after DAC
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{ .name = "DAC Playback Switch", .vals.i = { 1, 1 } }, |
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{ .name = "DAC Reversed Playback Switch", .vals.i = { 1, 1 } }, |
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{ .name = "DAC Playback Volume", .vals.i = { 160, 160 } }, |
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{ .name = "Mic1 Playback Switch", .vals.i = { 0, 0 } }, |
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{ .name = "Mic1 Playback Volume", .vals.i = { 0 } }, |
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{ .name = "Mic2 Playback Switch", .vals.i = { 0, 0 } }, |
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{ .name = "Mic2 Playback Volume", .vals.i = { 0 } }, |
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{ .name = "Line In Playback Switch", .vals.i = { 0, 0 } }, |
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{ .name = "Line In Playback Volume", .vals.i = { 0 } }, |
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// Outputs
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{ .name = "Earpiece Source Playback Route", .vals.e = { "Left Mixer" } }, |
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{ .name = "Earpiece Playback Switch", .vals.i = { !!s->ear_on } }, |
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{ .name = "Earpiece Playback Volume", .vals.i = { s->ear_vol } }, |
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{ .name = "Headphone Source Playback Route", .vals.e = { "Mixer", "Mixer" } }, |
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{ .name = "Headphone Playback Switch", .vals.i = { !!s->hp_on, !!s->hp_on } }, |
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{ .name = "Headphone Playback Volume", .vals.i = { s->hp_vol } }, |
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// Loudspeaker
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{ .name = "Line Out Source Playback Route", .vals.e = { "Mono Differential", "Mono Differential" } }, |
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{ .name = "Line Out Playback Switch", .vals.i = { !!s->spk_on, !!s->spk_on } }, |
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{ .name = "Line Out Playback Volume", .vals.i = { s->spk_vol } }, |
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}; |
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audio_restore_state(controls, ARRAY_SIZE(controls)); |
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} |
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static struct audio_setup audio_setup = { |
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.mic_on = true, |
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.ear_on = true, |
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.spk_on = false, |
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.hp_on = false, |
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.from_modem_on = true, |
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.to_modem_on = true, |
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.modem_playback_monitor = false, |
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.dai2_en = false, |
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.hp_vol = 15, |
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.spk_vol = 15, |
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.ear_vol = 31, |
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.mic_gain = 1, |
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}; |
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int main(int ac, char* av[]) |
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{ |
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int opt; |
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while ((opt = getopt(ac, av, "smhe2")) != -1) { |
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switch (opt) { |
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case 's': |
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audio_setup.spk_on = 1; |
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break; |
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case 'm': |
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audio_setup.mic_on = 1; |
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break; |
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case 'h': |
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audio_setup.hp_on = 1; |
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break; |
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case 'e': |
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audio_setup.ear_on = 1; |
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break; |
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case '2': |
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audio_setup.dai2_en = 1; |
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break; |
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default: /* '?' */ |
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fprintf(stderr, "Usage: %s [-s] [-m] [-h] [-e] [-2]\n", av[0]); |
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exit(EXIT_FAILURE); |
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} |
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} |
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audio_set_controls(&audio_setup); |
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return 0; |
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} |
Loading…
Reference in new issue