Audio on PinePhone
ContributeHardware
The codec inside A64 is basicaly identical to AC100 codec that's part of AXP813 PMIC that's paired with A83T. It has the same register values and general organization of analog and digital audio paths.
One difference is that AC100 is external to the SoC, while the codec on A64 is inside the SoC, so codec registers start at different offset, and are spaced more widely (aligned to 4 bytes) on A64.
On A83T the external codec is connected via I2S to the SoC's I2S controller on one side and to AIF1 on the codec side.
This is carried over to A64 so the A64 has DAI connected internally to AIF1 of the codec.
Linux kernel
The current (2020–02–08) state of the art are these 5 branches from smaeul:
- https://github.com/…dec-to-codec
- https://github.com/…i-codec-aif2
- https://github.com/…i-codec-aif3
- https://github.com/…-codec-fixes
- https://github.com/…i-codec-dapm
- these patches:
- https://github.com/…dc2301129b39
- https://github.com/…3f7945e0fdf5
- https://github.com/…55a627bf2b8c
- https://github.com/…467633b450fe
I have them all integrated into my tree.
Userspace
Sound
card in Linux has some controls accessible via /dev/snd/controlC0
and a PCM interfaces for streaming data in/out of the card (via
/dev/snd/pcmC0D0c and /dev/pcmC0D0p).
The
controls are basically just of two types in case of PP: integer,
enum (the rest, like boolean reduces to
integer internally).
On PinePhone, these controls control how the codec sets up analog and digital audio paths within itself. That is, where the audio signal will be enabled to go along multiple fixed paths, and how it will be mixed with other signals and amplified.
A64 audio codec controls diagram
See the larger image here
To have these controls you need to use the above patches.
Voice call audio routing setup
The current description for my specific audio card controls setup
for PP modem voice calls is in the next diagram. The diagram shows multiple
different paths, but only a few are improtant for connecting audio from the
mic1 to the modem (bb) and from
modem to the earphone on the top of
the phone.
The remaining 4 routes are meant for the CPU to be able to record me, record the other caller, play back audio to me, and to play back audio to the other caller. Each of these routes can be used separately by sending/reading audio data to/from the left (me) and the right (other caller) channels of the playback and capture PCM devices. (see above)
See the larger image here
The image contains the
routes in different colors and also the names of controls in pink that are
presented by ALSA on the /dev/snd/controlC0 device.
You can
dump the list of all controls using alsactl store -f -. You can
also set these controls up using alsamixer or by modifying and
loading the text file generated by alsactl via
alsactl restore -f <path>.
I wrote a simple program that takes some arguments describing the desired audio routing configuration for voice calls (like whether speaker or mic or earphone should be enabled, or whether to disable/enable the digital audio interface to the modem), and updates all the controls automatically, to match it.
/* * 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; bool hpmic_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 hpmic_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 = { s->hpmic_gain } }, // 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 = { !!s->hpmic_on, !!s->hpmic_on } }, { .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 = false, .ear_on = false, .spk_on = false, .hp_on = false, .hpmic_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, .hpmic_gain = 1, }; int main(int ac, char* av[]) { int opt; while ((opt = getopt(ac, av, "smhle2")) != -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 'l': audio_setup.hpmic_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] [-l] [-e] [-2]\n", av[0]); exit(EXIT_FAILURE); } } audio_set_controls(&audio_setup); return 0; }
Download the
code. You may want to modify it to your needs (change volumes, or add
command line parameter to set them from command line) or to use it to setup the
card, and dump the controls via the above alsactl commands.
Otherwise it's quite a lot of work to setup everything correctly, because many
controls control multiple channels separately, and you need to sometimes setup
each channel for a different use case, or disable one of the
channels, etc.
There's older version for kernels older than 5.9 here.