Monorepo for Aesthetic.Computer
aesthetic.computer
fix: set period=1024 for sample-accurate capture timing
ALSA defaults (period=32768 = 682ms) caused massive timing drift.
Period=1024 gives ~21ms latency. The EIO root causes (HDMI open,
6-period playback, boot mixer) are all fixed so this should work now.
Removed first-chunk skip hack — with 21ms periods the direct write
path is precise enough.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
+11
-8
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-8
fedac/native/src/audio.c
+11
-8
fedac/native/src/audio.c
···
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return NULL;
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}
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-
// Configure — NO period/buffer setting (ALSA defaults work on this HDA)
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snd_pcm_hw_params_t *hw;
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snd_pcm_hw_params_alloca(&hw);
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snd_pcm_hw_params_any(cap, hw);
···
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unsigned int rate = 48000;
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snd_pcm_hw_params_set_rate_near(cap, hw, &rate, NULL);
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+
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// Set period=1024 (~21ms) for low-latency capture. This previously
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// caused EIO but the root causes were: HDMI audio open (exhausting
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// HDA streams), 6-period playback buffer, and capture mixer in
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// audio_init. All three are now fixed.
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snd_pcm_uframes_t period_frames = 1024;
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snd_pcm_hw_params_set_period_size_near(cap, hw, &period_frames, NULL);
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snd_pcm_uframes_t buffer_frames = 8192;
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snd_pcm_hw_params_set_buffer_size_near(cap, hw, &buffer_frames);
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if (snd_pcm_hw_params(cap, hw) < 0) {
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snprintf(audio->mic_last_error, sizeof(audio->mic_last_error),
···
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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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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 (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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+
// Direct-write when recording
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+
if (audio->recording && 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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}