Monorepo for Aesthetic.Computer
aesthetic.computer
fix: negotiate audio sample rate dynamically for hardware compat
Instead of hardcoding 192kHz (which causes XRUN storms on hardware that
can't sustain it), probe the hardware rate range and pick the best
supported rate. Period/buffer sizes scale to the negotiated rate.
Adds plughw fallback if hw: config fails entirely.
Fixes audio on MacBook Pro 2011 (Cirrus Logic CS4206, max 96kHz).
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
+69
-16
+67
-15
fedac/native/src/audio.c
+67
-15
fedac/native/src/audio.c
···
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static void *audio_thread_fn(void *arg) {
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ACAudio *audio = (ACAudio *)arg;
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-
int16_t buffer[AUDIO_PERIOD_SIZE * AUDIO_CHANNELS];
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const unsigned int period_frames = audio->actual_period ? audio->actual_period : AUDIO_PERIOD_SIZE;
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int16_t *buffer = calloc(period_frames * AUDIO_CHANNELS, sizeof(int16_t));
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if (!buffer) { fprintf(stderr, "[audio] thread: alloc failed\n"); return NULL; }
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const double rate = (double)(audio->actual_rate ? audio->actual_rate : AUDIO_SAMPLE_RATE);
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const double dt = 1.0 / rate;
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double mix_divisor = 1.0; // Smooth auto-mix (matches speaker.mjs)
···
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pthread_mutex_lock(&audio->lock);
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for (int i = 0; i < AUDIO_PERIOD_SIZE; i++) {
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for (unsigned int i = 0; i < period_frames; i++) {
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double mix_l = 0.0, mix_r = 0.0;
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double voice_sum = 0.0; // Total voice weight for auto-mix
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···
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}
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// BPM metronome
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audio->beat_elapsed += (double)AUDIO_PERIOD_SIZE * dt;
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audio->beat_elapsed += (double)period_frames * dt;
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double beat_interval = 60.0 / audio->bpm;
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if (audio->beat_elapsed >= beat_interval) {
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audio->beat_elapsed -= beat_interval;
···
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pthread_mutex_unlock(&audio->lock);
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audio->total_frames += AUDIO_PERIOD_SIZE;
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audio->total_frames += period_frames;
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audio->time = (double)audio->total_frames / rate;
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// Recording tap: send mixed PCM to recorder (if active)
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if (audio->rec_callback)
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audio->rec_callback(buffer, AUDIO_PERIOD_SIZE, audio->rec_userdata);
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audio->rec_callback(buffer, period_frames, audio->rec_userdata);
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// Write to ALSA (handle short writes to avoid dropped samples/clicks)
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snd_pcm_t *pcm = (snd_pcm_t *)audio->pcm;
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int remaining = AUDIO_PERIOD_SIZE;
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int remaining = (int)period_frames;
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int offset = 0;
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while (remaining > 0) {
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int frames = snd_pcm_writei(pcm, buffer + offset * AUDIO_CHANNELS, remaining);
···
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}
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}
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free(buffer);
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return NULL;
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}
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···
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audio->mic_last_error[0] = 0;
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seed_default_sample(audio);
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// TTS PCM ring buffer (5 seconds at output rate)
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audio->tts_buf_size = AUDIO_SAMPLE_RATE * 5;
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// TTS PCM ring buffer (5 seconds at max output rate)
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audio->tts_buf_size = AUDIO_SAMPLE_RATE * 5; // allocated at max, actual_rate adjusts usage
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audio->tts_buf = calloc(audio->tts_buf_size, sizeof(float));
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audio->tts_read_pos = 0;
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audio->tts_write_pos = 0;
···
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return audio;
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}
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// Configure ALSA
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// Configure ALSA — negotiate rate dynamically.
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// Try preferred rates from highest to lowest. The hardware decides what it
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// actually supports; we adapt period/buffer sizes to match the negotiated rate.
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snd_pcm_hw_params_t *params;
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snd_pcm_hw_params_alloca(¶ms);
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snd_pcm_hw_params_any(pcm, params);
···
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snd_pcm_hw_params_set_format(pcm, params, SND_PCM_FORMAT_S16_LE);
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snd_pcm_hw_params_set_channels(pcm, params, AUDIO_CHANNELS);
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unsigned int rate = AUDIO_SAMPLE_RATE;
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// Query hardware rate range
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unsigned int rate_min = 0, rate_max = 0;
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snd_pcm_hw_params_get_rate_min(params, &rate_min, NULL);
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snd_pcm_hw_params_get_rate_max(params, &rate_max, NULL);
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fprintf(stderr, "[audio] Hardware rate range: %u–%u Hz\n", rate_min, rate_max);
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// Try rates from highest to lowest — pick the best the hardware supports
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unsigned int preferred_rates[] = { 192000, 96000, 48000, 44100, 0 };
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unsigned int rate = 0;
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for (int i = 0; preferred_rates[i]; i++) {
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unsigned int try_rate = preferred_rates[i];
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if (try_rate > rate_max || try_rate < rate_min) continue;
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// Test if this exact rate works
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if (snd_pcm_hw_params_test_rate(pcm, params, try_rate, 0) == 0) {
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rate = try_rate;
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fprintf(stderr, "[audio] Selected rate: %u Hz\n", rate);
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break;
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}
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}
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if (rate == 0) {
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// Fallback: let ALSA pick the nearest to 48kHz
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rate = 48000;
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fprintf(stderr, "[audio] No preferred rate supported, falling back to nearest-48kHz\n");
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}
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snd_pcm_hw_params_set_rate_near(pcm, params, &rate, 0);
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snd_pcm_uframes_t period = AUDIO_PERIOD_SIZE;
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// Scale period and buffer to ~1ms latency at the negotiated rate
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snd_pcm_uframes_t period = rate / 1000; // ~1ms worth of frames
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if (period < 64) period = 64;
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snd_pcm_hw_params_set_period_size_near(pcm, params, &period, 0);
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snd_pcm_uframes_t buffer_size = AUDIO_PERIOD_SIZE * 3; // 3 periods (~3ms total at 192kHz)
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snd_pcm_uframes_t buffer_size = period * 4; // 4 periods (~4ms total)
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snd_pcm_hw_params_set_buffer_size_near(pcm, params, &buffer_size);
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err = snd_pcm_hw_params(pcm, params);
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if (err < 0) {
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fprintf(stderr, "[audio] Cannot configure ALSA: %s\n", snd_strerror(err));
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fprintf(stderr, "[audio] Cannot configure ALSA at %uHz: %s\n", rate, snd_strerror(err));
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// Last resort: try plughw with default params
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fprintf(stderr, "[audio] Trying plughw fallback...\n");
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snd_pcm_close(pcm);
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audio->pcm = NULL;
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return audio;
835
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err = snd_pcm_open(&pcm, "plughw:0,0", SND_PCM_STREAM_PLAYBACK, 0);
836
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if (err >= 0) {
837
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snd_pcm_hw_params_any(pcm, params);
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snd_pcm_hw_params_set_access(pcm, params, SND_PCM_ACCESS_RW_INTERLEAVED);
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snd_pcm_hw_params_set_format(pcm, params, SND_PCM_FORMAT_S16_LE);
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snd_pcm_hw_params_set_channels(pcm, params, AUDIO_CHANNELS);
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rate = 48000;
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snd_pcm_hw_params_set_rate_near(pcm, params, &rate, 0);
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period = 256;
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snd_pcm_hw_params_set_period_size_near(pcm, params, &period, 0);
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buffer_size = 1024;
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snd_pcm_hw_params_set_buffer_size_near(pcm, params, &buffer_size);
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err = snd_pcm_hw_params(pcm, params);
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}
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if (err < 0) {
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fprintf(stderr, "[audio] All ALSA config attempts failed: %s\n", snd_strerror(err));
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snd_pcm_close(pcm);
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audio->pcm = NULL;
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return audio;
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}
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snprintf(audio->audio_device, sizeof(audio->audio_device), "plughw:0,0");
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}
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snd_pcm_prepare(pcm);
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audio->pcm = pcm;
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audio->actual_rate = rate;
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audio->actual_period = (unsigned int)period;
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// Update glitch rate for actual sample rate
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audio->glitch_rate = rate / 1600;
+2
-1
fedac/native/src/audio.h
+2
-1
fedac/native/src/audio.h
···
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// System mixer volume (0-100 percent)
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int system_volume;
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int card_index; // ALSA card number (0 or 1)
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-
unsigned int actual_rate; // Negotiated ALSA sample rate (may differ from requested)
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+
unsigned int actual_rate; // Negotiated ALSA sample rate (may differ from requested)
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unsigned int actual_period; // Negotiated ALSA period size in frames
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// TTS PCM buffer (resampled to output rate, mono → stereo in mix)
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float *tts_buf; // ring buffer of mono float samples at output rate