Monorepo for Aesthetic.Computer
aesthetic.computer
fix: skip stale first chunk on rec start + add mic compressor
- Skip first snd_pcm_readi chunk after recording starts (contains
pre-buffered audio from before Home was pressed)
- Simple 4:1 compressor with fast attack / slow release to prevent
clipping and auto-level the mic input
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
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fedac/native/src/audio.c
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fedac/native/src/audio.c
···
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continue;
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}
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// Skip first chunk after recording starts — it contains stale
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// pre-buffered audio from before the user pressed rec.
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int is_first_rec_chunk = (audio->recording && audio->sample_write_pos == 0);
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float peak = 0.0f;
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// Simple compressor: track envelope, apply gain reduction
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static float env = 0.0f; // envelope follower
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static float comp_gain = 1.0f; // current gain
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const float threshold = 0.5f; // compress above this
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const float ratio = 4.0f; // 4:1 compression
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const float attack = 0.002f; // fast attack
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const float release = 0.0001f; // slow release
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for (int s = 0; s < n; s++) {
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float sample;
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if (channels == 1) {
···
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} else {
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sample = (buf[s * 2] + buf[s * 2 + 1]) / 65536.0f;
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}
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// Envelope follower
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float abs_s = fabsf(sample);
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if (abs_s > env)
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env += attack * (abs_s - env);
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else
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env += release * (abs_s - env);
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// Compute gain reduction
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if (env > threshold) {
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float over = env - threshold;
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float reduced = threshold + over / ratio;
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comp_gain = reduced / env;
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} else {
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// Slowly return to unity
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comp_gain += 0.0001f * (1.0f - comp_gain);
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}
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sample *= comp_gain;
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if (abs_s > peak) peak = abs_s;
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// Always write to ring buffer
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audio->mic_ring[audio->mic_ring_pos % audio->sample_max_len] = sample;
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audio->mic_ring_pos++;
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// Direct-write when recording
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if (audio->recording && audio->sample_write_pos < audio->sample_max_len) {
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// Direct-write when recording (skip first stale chunk)
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if (audio->recording && !is_first_rec_chunk &&
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audio->sample_write_pos < audio->sample_max_len) {
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audio->sample_buf[audio->sample_write_pos++] = sample;
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}
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}
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// If we skipped the first chunk, mark that we've consumed it
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// by writing at least 0 (sample_write_pos stays 0, next chunk writes)
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audio->mic_level = peak;
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if (audio->recording && audio->sample_write_pos >= audio->sample_max_len) {