Monorepo for Aesthetic.Computer
aesthetic.computer
fix: restore hot-mic with correct prerequisites
Now that we know the EIO root causes (HDMI audio + 6-period playback
buffer), hot-mic can work safely with:
- No HDMI PCM open in audio_init
- 3-period playback buffer (original value)
- Capture mixer enabled in capture thread (not audio_init)
- ALSA default period/buffer (no explicit setting)
This gives instant recording with zero device-open latency.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
+86
-39
+86
-39
fedac/native/src/audio.c
+86
-39
fedac/native/src/audio.c
···
1004
1004
audio->target_fx_mix = mix;
1005
1005
}
1006
1006
1007
-
// --- Thread-per-recording capture (original working approach) ---
1008
-
// Opens device fresh each recording, reads directly, closes on stop.
1009
-
// Hot-mic was tried but causes EIO on some HDA codecs (11e Yoga Gen 5).
1007
+
// --- Hot-mic capture thread ---
1008
+
// Device opens once (on wave-enter), stays running. Always reads to keep
1009
+
// ALSA happy and the level meter live. Only writes to sample_buf when
1010
+
// recording flag is set. Instant recording with zero device-open latency.
1011
+
//
1012
+
// IMPORTANT: HDMI audio must be DISABLED in audio_init and playback buffer
1013
+
// must be 3 periods (not 6) — otherwise the HDA controller runs out of
1014
+
// streams and capture gets EIO.
1010
1015
static void *capture_thread_func(void *arg) {
1011
1016
ACAudio *audio = (ACAudio *)arg;
1012
1017
snd_pcm_t *cap = NULL;
1013
1018
1014
-
// Try to open capture device
1015
1019
const char *devices[] = {"hw:0,0", "hw:1,0", "hw:0,6", "hw:0,7",
1016
1020
"plughw:0,0", "plughw:1,0", "default", NULL};
1017
1021
for (int i = 0; devices[i]; i++) {
···
1026
1030
snprintf(audio->mic_last_error, sizeof(audio->mic_last_error),
1027
1031
"no capture device found");
1028
1032
ac_log("[mic] no capture device found\n");
1029
-
audio->recording = 0;
1033
+
audio->mic_hot = 0;
1030
1034
return NULL;
1031
1035
}
1032
1036
1033
-
// Configure capture — NO period/buffer setting, use ALSA defaults
1037
+
// Configure — NO period/buffer setting (ALSA defaults work on this HDA)
1034
1038
snd_pcm_hw_params_t *hw;
1035
1039
snd_pcm_hw_params_alloca(&hw);
1036
1040
snd_pcm_hw_params_any(cap, hw);
···
1051
1055
"failed to configure capture");
1052
1056
ac_log("[mic] failed to configure capture\n");
1053
1057
snd_pcm_close(cap);
1054
-
audio->recording = 0;
1058
+
audio->mic_hot = 0;
1055
1059
return NULL;
1056
1060
}
1057
1061
1058
-
// Enable capture mixer controls NOW (after PCM open, in the capture thread)
1059
-
// Must NOT be done in audio_init — that causes EIO on some HDA codecs.
1062
+
// Enable capture mixer (safe here — after PCM open, in capture thread)
1060
1063
{
1061
1064
int cnum = 0;
1062
1065
const char *d = audio->mic_device;
···
1090
1093
}
1091
1094
1092
1095
audio->sample_rate = rate;
1093
-
audio->sample_write_pos = 0;
1094
1096
audio->mic_connected = 1;
1095
-
ac_log("[mic] recording at %u Hz, %u ch\n", rate, channels);
1097
+
ac_log("[mic] hot-mic running at %u Hz, %u ch\n", rate, channels);
1096
1098
1097
-
int16_t buf[1024 * 2]; // max stereo
1098
-
float peak = 0.0f;
1099
-
while (audio->recording && audio->sample_write_pos < audio->sample_max_len) {
1099
+
int16_t buf[1024 * 2];
1100
+
while (audio->mic_hot) {
1100
1101
int n = snd_pcm_readi(cap, buf, 512);
1101
1102
if (n < 0) {
1102
1103
n = snd_pcm_recover(cap, n, 0);
1103
-
if (n < 0) break;
1104
+
if (n < 0) {
1105
+
snprintf(audio->mic_last_error, sizeof(audio->mic_last_error),
1106
+
"capture read failed: %s", snd_strerror(n));
1107
+
ac_log("[mic] capture read failed: %s\n", snd_strerror(n));
1108
+
break;
1109
+
}
1104
1110
continue;
1105
1111
}
1106
-
for (int s = 0; s < n && audio->sample_write_pos < audio->sample_max_len; s++) {
1112
+
1113
+
float peak = 0.0f;
1114
+
for (int s = 0; s < n; s++) {
1107
1115
float sample;
1108
1116
if (channels == 1) {
1109
1117
sample = buf[s] / 32768.0f;
···
1112
1120
}
1113
1121
float abs_s = fabsf(sample);
1114
1122
if (abs_s > peak) peak = abs_s;
1115
-
audio->sample_buf[audio->sample_write_pos++] = sample;
1123
+
1124
+
// Always write to ring buffer
1125
+
audio->mic_ring[audio->mic_ring_pos % audio->sample_max_len] = sample;
1126
+
audio->mic_ring_pos++;
1127
+
1128
+
// Direct-write when recording
1129
+
if (audio->recording && audio->sample_write_pos < audio->sample_max_len) {
1130
+
audio->sample_buf[audio->sample_write_pos++] = sample;
1131
+
}
1132
+
}
1133
+
audio->mic_level = peak;
1134
+
1135
+
if (audio->recording && audio->sample_write_pos >= audio->sample_max_len) {
1136
+
audio->sample_len = audio->sample_write_pos;
1137
+
audio->recording = 0;
1138
+
ac_log("[mic] recording buffer full (%d samples)\n", audio->sample_len);
1116
1139
}
1117
-
audio->mic_level = peak; // update level meter live
1118
1140
}
1119
1141
1120
-
audio->sample_len = audio->sample_write_pos;
1121
-
ac_log("[mic] recorded %d samples (%.2fs) peak=%.4f\n",
1122
-
audio->sample_len, (double)audio->sample_len / audio->sample_rate, peak);
1142
+
ac_log("[mic] hot-mic thread exiting, device=%s\n", audio->mic_device);
1123
1143
snd_pcm_close(cap);
1124
1144
audio->mic_connected = 0;
1125
1145
audio->recording = 0;
···
1127
1147
}
1128
1148
1129
1149
int audio_mic_open(ACAudio *audio) {
1130
-
// No-op in thread-per-recording mode — device opens on rec start
1131
-
if (!audio) return -1;
1150
+
if (!audio || audio->mic_hot || audio->capture_thread_running) return -1;
1132
1151
audio->mic_hot = 1;
1133
-
ac_log("[mic] mic ready (thread-per-recording mode)\n");
1152
+
audio->capture_thread_running = 1;
1153
+
audio->mic_last_error[0] = 0;
1154
+
ac_log("[mic] opening hot-mic\n");
1155
+
if (pthread_create(&audio->capture_thread, NULL, capture_thread_func, audio) != 0) {
1156
+
audio->mic_hot = 0;
1157
+
audio->capture_thread_running = 0;
1158
+
ac_log("[mic] failed to create capture thread\n");
1159
+
return -1;
1160
+
}
1134
1161
return 0;
1135
1162
}
1136
1163
···
1138
1165
if (!audio) return;
1139
1166
audio->recording = 0;
1140
1167
audio->mic_hot = 0;
1141
-
ac_log("[mic] mic closed\n");
1168
+
if (audio->capture_thread_running) {
1169
+
pthread_join(audio->capture_thread, NULL);
1170
+
audio->capture_thread_running = 0;
1171
+
}
1172
+
ac_log("[mic] hot-mic closed\n");
1142
1173
}
1143
1174
1144
1175
int audio_mic_start(ACAudio *audio) {
1145
1176
if (!audio || audio->recording) return -1;
1146
-
audio->recording = 1;
1147
-
audio->sample_len = 0;
1177
+
if (!audio->mic_hot) {
1178
+
int rc = audio_mic_open(audio);
1179
+
if (rc != 0) return rc;
1180
+
}
1148
1181
// Kill any playing sample voices
1149
1182
for (int i = 0; i < AUDIO_MAX_SAMPLE_VOICES; i++)
1150
1183
audio->sample_voices[i].active = 0;
1151
-
if (pthread_create(&audio->capture_thread, NULL, capture_thread_func, audio) != 0) {
1152
-
audio->recording = 0;
1153
-
return -1;
1154
-
}
1155
-
pthread_detach(audio->capture_thread);
1156
-
ac_log("[mic] recording started\n");
1184
+
audio->rec_start_ring_pos = audio->mic_ring_pos;
1185
+
audio->sample_len = 0;
1186
+
audio->sample_write_pos = 0;
1187
+
__sync_synchronize();
1188
+
audio->recording = 1;
1189
+
ac_log("[mic] recording started (instant), ring_pos=%d\n", audio->rec_start_ring_pos);
1157
1190
return 0;
1158
1191
}
1159
1192
1160
1193
int audio_mic_stop(ACAudio *audio) {
1161
1194
if (!audio) return 0;
1162
1195
audio->recording = 0;
1163
-
// Wait for capture thread to finish writing sample_len (ALSA defaults
1164
-
// can have large periods so the last snd_pcm_readi may block ~700ms)
1165
-
for (int i = 0; i < 500 && audio->mic_connected; i++) {
1166
-
usleep(2000); // 2ms per iter, max 1s
1196
+
__sync_synchronize();
1197
+
1198
+
int direct_len = audio->sample_write_pos;
1199
+
if (direct_len > 0) {
1200
+
audio->sample_len = direct_len;
1201
+
ac_log("[mic] recording stopped (direct), sample_len=%d sample_rate=%u\n",
1202
+
audio->sample_len, audio->sample_rate);
1203
+
} else {
1204
+
// Fallback: extract from ring buffer
1205
+
int start = audio->rec_start_ring_pos;
1206
+
int end = audio->mic_ring_pos;
1207
+
int len = end - start;
1208
+
if (len < 0) len = 0;
1209
+
if (len > audio->sample_max_len) len = audio->sample_max_len;
1210
+
for (int i = 0; i < len; i++) {
1211
+
audio->sample_buf[i] = audio->mic_ring[(start + i) % audio->sample_max_len];
1212
+
}
1213
+
audio->sample_len = len;
1214
+
ac_log("[mic] recording stopped (ring), sample_len=%d ring_span=%d sample_rate=%u\n",
1215
+
audio->sample_len, end - start, audio->sample_rate);
1167
1216
}
1168
-
ac_log("[mic] recording stopped, sample_len=%d sample_rate=%u\n",
1169
-
audio->sample_len, audio->sample_rate);
1170
1217
return audio->sample_len;
1171
1218
}
1172
1219