The dominant garble was NOT the mic: T:RAWMIC/recorder ran full-duplex (start_loopback), so
the T-Deck played the decoded loopback round-trip out its OWN speaker, inches from the mic ->
acoustic feedback that built into "oscillating static that gets slowly louder". Gate the
playback ring-write off while the PSRAM recorder is active (i2s_playback.cpp) so raw-mic
captures are feedback-free -> speech becomes intelligible (Whisper recovers ~85% of words).
Also in this change:
- Recorder stores BOTH TDM channels (main.cpp) for offboard de-interleave; confirmed CH0 is
the mic, CH1 dead -> de-interleave was always correct.
- MCLK raised to an exact APLL-locked 12.288MHz (es7210.cpp MCLK_DIV_FRE=768 + the
{12288000,16000} coeff; i2s_capture.cpp use_apll=true/fixed_mclk) -- cleaner than the
fractional-N 4.096MHz, though the feedback (not jitter) was the real cause.
Residual (under investigation): signal-correlated distortion ("static reacts to speech") +
50Hz mains hum (USB-ground pickup, removed by the existing 300Hz HPF in the call path).
Co-Authored-By: Claude Opus 4.8 (1M context) <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01UWZuYkHBRqNb6BZHV8sTG5
Confirmed fixes:
- LVGL hybrid-allocator threshold 1024->256 (lib/lv_mem_hybrid.h): moves ~45KB of
small LVGL objects to PSRAM, fixing internal-heap starvation that made the
call/loopback pipeline intermittently fail to allocate and reboot mid-call
(internal free 71->116KB, largest contiguous block 61->106KB).
- ES7210 mic PGA gain 21dB->12dB (lxst_audio.cpp): 21dB saturated the ADC -- an
rms-7003 noise floor in silence + 0x8000 negative-rail spikes + a spectral-peak
shift that masqueraded as a "+17% pitch warp". At 12dB idle silence is clean (rms 48).
Diagnostic harness (PYXIS_TEST_HOOKS, test-only): T:RAWMIC[stage], T:REG, and
T:RECORD/T:DUMPREC add a raw-mic UDP tap plus a reliable PSRAM recorder with a
checksummed serial transfer, and T:LOOPBACK wiring, for objective mic-capture
analysis (offboard Whisper STT scoring).
Exploratory / still unresolved: captured speech remains garbled (oscillating static,
likely an ES7210 sigma-delta capture instability -- under investigation). Includes a
16kHz/main-PLL clock path + MICBIAS exploration; the "warp" was the gain artifact, not
a clock issue, so the 16kHz path can later be simplified back to 8kHz.
Co-Authored-By: Claude Opus 4.8 (1M context) <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01UWZuYkHBRqNb6BZHV8sTG5
Extends the LXST harness from frame-flow only (#62) and decoder QoS
only (#63) to a full content-fidelity test: a known 1kHz sine wave
flows in BOTH directions through the Codec2-3200 round-trip, and
the harness asserts the decoded RMS at each end matches expected
energy within tolerance (after Codec2's lossy speech-codec behavior).
Firmware additions (under PYXIS_TEST_HOOKS):
I2SCapture
setInjectSine(enabled, freq=1000, amp=0.5)
Replaces mic input with a phase-continuous synthesized sine.
Bypasses ES7210 capture and the voice filter chain so the
encoder sees pure samples — peer's decoded RMS validates that
pyxis's TX path delivers content.
I2SPlayback
pcmSampleCount(), pcmSumSquares()
Decoded-PCM energy accumulators, fed from each successful
Codec2 decode. uint64 sumsq holds ~2³⁴ frames before overflow,
far longer than any test call.
LXSTAudio + UIManager (test-only)
captureSetInjectSine, playbackPcmSampleCount, playbackPcmSumSquares
test_call_set_inject_sine, test_call_pcm_sample_count,
test_call_pcm_sum_squares
Serial T: hooks (main.cpp)
T:CALL_INJECT <on|off> [freq] [amp_pct]
Drive the capture-side injection from the harness.
T:CALL_QOS now also returns pcm_n + pcm_ss
Harness divides + sqrts to RMS for content validation.
Validated with /tmp/lxst_call_harness.py + /tmp/lxst_call_bot.py
(scripts vault-local per the no-PII rule):
pyxis_rms = 6363, bot_rms = 1930, decode_fail = 0
PASS: bidirectional audio + content-fidelity validated
The empirical RMS floor is 800 (pycodec2 self-test on 1kHz amp 0.5
yields ~1400; pyxis decoder hits ~6300; bot decoder hits ~1900 —
all far above the ~5-50 silence floor). Codec2 is a speech codec
so pure-tone round-trip is naturally lossy; the test gates on
"audio bytes carry actual content energy", not lossless round-trip.
Co-Authored-By: Claude Opus 4.7 (1M context) <noreply@anthropic.com>
Adds the on-device test surface the LXST voice-call harness needs to
drive an end-to-end audio flow + QoS validation against a Mac-side
softphone bot. Both hooked behind PYXIS_TEST_HOOKS so production
firmware is unaffected.
T: serial commands (main.cpp):
T:CALL <hex> — initiate call (hex = peer's lxst.telephony dest)
T:CALL_STATE — current call FSM state name
T:CALL_HANGUP — tear down the active call
T:CALL_STATS — audio frame counters: tx, rx, state
T:CALL_QOS — Codec2 decoder QoS: decode_ok, decode_fail, state
T:HASIDENTITY <h> — bool, distinct from T:HASPATH because path_store
and known_destinations are populated by separate
code paths (announce can land in one before the
other; harness has to wait for both).
UIManager additions (under PYXIS_TEST_HOOKS):
test_call_initiate, test_call_hangup, test_call_state_name,
test_call_audio_tx_count, test_call_audio_rx_count,
test_call_decode_ok, test_call_decode_fail.
I2SPlayback / LXSTAudio additions (always on — counters are tiny):
decodeOkCount(), decodeFailCount(), resetCounters() on I2SPlayback
surface the Codec2 decode success/fail rate. LXSTAudio re-exports.
Each writeEncodedPacket call increments exactly one counter so the
ratio is "wire-level audio fidelity" of the peer's encoder.
Validated:
Frame-flow soak (12s call): pyxis tx=73 rx=72, bot tx=230 rx=72.
QoS soak (12s call): pyxis tx=73 rx=76, decode_ok=81 decode_fail=0,
bot tx=227 rx=73.
Harness scripts that drive these hooks live in the local Obsidian
vault (under 80 Assistant/Memory/pyxis/soak_scripts/) — they encode
LAN-specific state and aren't checked in. See the vault's
automated_soak_testing.md for the full procedure.
Co-Authored-By: Claude Opus 4.7 (1M context) <noreply@anthropic.com>
Both encoded ring buffer (capture) and PCM ring buffer (playback) are
SPSC (Single Producer Single Consumer) with lock-free atomics. The
overflow handlers were calling read() from the producer thread, racing
with the consumer thread on the read index. This caused frame
corruption, duplication, and skipping — resulting in distorted audio.
Fix: Drop new frames on overflow instead of evicting old ones. The
consumer (TX pump / playback task) will drain the buffer naturally.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Replace custom ES7210 register-level driver with the verbatim LilyGO
T-Deck Plus ES7210 library. The custom driver was writing MODE_CONFIG
and clock doubler registers (master mode path) which broke the ES7210's
clock chain, causing all-zero PCM output. The LilyGO library in slave
mode leaves those registers at power-on defaults, which is correct since
the ESP32 I2S master provides MCLK/BCLK/LRCK.
Also includes LXST voice call protocol improvements:
- LXST IN destination for receiving calls
- Announce handler for tracking voice-capable peers
- Path request before outgoing calls
- Throttled SPIFFS saves in microReticulum (dirty flag)
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Add ES7210 I2C address and I2S mic capture pin definitions
- Add ring/hangup tone helpers to Tone library
- Add lxst_audio library scaffold
- Add Codec2 dependency to platformio.ini
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
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