Monorepo for Aesthetic.Computer
aesthetic.computer
native: prefer plughw: for speaker PCM + log negotiated format/channels
Speakers produced crunchy quiet audio on G7 — likely format/channel
mismatch between our S16_LE stereo config and what the SSP1 BE DAI
expects. Switch to plughw: wrapper as first-choice device, which adds
automatic format/rate/channel conversion. Also log actual negotiated
format + channels for diagnostics.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
+19
-5
+19
-5
fedac/native/src/audio.c
+19
-5
fedac/native/src/audio.c
···
1897
1897
char ucm_speaker_plug[80] = "";
1898
1898
if (ucm_speaker_pcm[0]) {
1899
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snprintf(ucm_speaker_plug, sizeof(ucm_speaker_plug),
1900
-
"plug%s", ucm_speaker_pcm);
1900
+
"plughw%s", ucm_speaker_pcm + 2); /* hw:0,0 → plughw:0,0 */
1901
1901
int n = 0;
1902
-
devices_with_spk[n++] = ucm_speaker_pcm;
1902
+
/* Prefer plug-wrapped device FIRST — the plug layer handles
1903
+
* format/rate/channel conversion, which fixes the "crunchy
1904
+
* quiet" audio on SOF boards where the SSP1 BE DAI runs at
1905
+
* a different format than our S16_LE 48kHz stereo request. */
1903
1906
devices_with_spk[n++] = ucm_speaker_plug;
1907
+
devices_with_spk[n++] = ucm_speaker_pcm;
1904
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for (int i = 0; devices_default[i] && n < 15; i++)
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devices_with_spk[n++] = devices_default[i];
1906
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devices_with_spk[n] = NULL;
···
2164
2168
audio->room_pos = 0;
2165
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}
2166
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2167
-
fprintf(stderr, "[audio] ALSA: requested %dHz, got %uHz, period=%lu, buffer=%lu (%.1fms latency)\n",
2168
-
AUDIO_SAMPLE_RATE, rate, (unsigned long)period, (unsigned long)buffer_size,
2169
-
(double)period / rate * 1000.0);
2171
+
/* Log the actual negotiated params — channels and format are
2172
+
* particularly important for diagnosing the "crunchy quiet" bug
2173
+
* on SOF boards where SSP1 may expect different bit depth. */
2174
+
{
2175
+
snd_pcm_format_t fmt;
2176
+
unsigned int ch = 0;
2177
+
snd_pcm_hw_params_get_format(params, &fmt);
2178
+
snd_pcm_hw_params_get_channels(params, &ch);
2179
+
fprintf(stderr, "[audio] ALSA: %uHz %uch fmt=%s period=%lu buf=%lu (%.1fms)\n",
2180
+
rate, ch, snd_pcm_format_name(fmt),
2181
+
(unsigned long)period, (unsigned long)buffer_size,
2182
+
(double)period / rate * 1000.0);
2183
+
}
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2184
snprintf(audio->audio_status, sizeof(audio->audio_status),
2171
2185
"ok %uHz %lufrm", rate, (unsigned long)period);
2172
2186
if (rate != AUDIO_SAMPLE_RATE)