Marcus/Voice/wake_detector.py
kassam 5d839d4f4e Voice: finalise on faster-whisper + energy wake, remove Vosk
Full-day voice-stack refactor. Experiments run and reverted:
- Gemini Live HTTP microservice (Python 3.8 env incompat, latency)
- Vosk grammar STT (English lexicon can't decode 'Sanad'; big model
  cold-load too slow on Jetson CPU)

Kept architecture:
- Voice/wake_detector.py — pure-numpy energy state machine with
  adaptive baseline, burst-audio capture for post-hoc verify.
- Voice/marcus_voice.py — orchestrator with 3 modes
  (wake_and_command / always_on / always_on_gated), hysteretic VAD,
  pre-silence trim (300 ms pre-roll), DSP pipeline (DC remove,
  80 Hz HPF, 0.97 pre-emphasis, peak-normalize), faster-whisper
  base.en int8 with beam=8 + temperature fallback [0,0.2,0.4],
  fuzzy-match canonicalisation, GARBAGE_PATTERNS + length filter,
  /s-/ phonetic wake-verify, full-turn debug WAV recording.

Config-driven vocab (zero hardcoded strings in Python):
- stt.wake_words (33 variants of 'Sanad')
- stt.command_vocab (68 canonical phrases)
- stt.garbage_patterns (17 Whisper noise outputs)
- stt.min_transcription_length, stt.command_vocab_cutoff

Command parser widened (Brain/command_parser.py):
- _RE_SIMPLE_DIR — bare direction + verb+direction combos
  ('left', 'go back', 'move forward', 'step right', ...)
- _RE_STOP_SIMPLE — bare stop/halt/wait/pause/freeze/hold
- All motion constants sourced from config_Navigation.json
  (move_map + step_duration_sec) via API/zmq_api.py; no more
  hardcoded 0.3 / 2.0 magic numbers.

API/audio_api.py — _play_pcm now uses AudioClient.PlayStream with
automatic resampling to 16 kHz (matches Sanad's proven pattern).

Removed:
- Voice/vosk_stt.py (and all Vosk references in marcus_voice.py)
- Models/vosk-model-small-en-us-0.15/ (40 MB model + zip)
- All Vosk keys from Config/config_Voice.json

Documentation synced across README, Doc/architecture.md,
Doc/pipeline.md, Doc/functions.md, Doc/controlling.md,
Doc/MARCUS_API.md, Doc/environment.md changelog.

Known limitation: faster-whisper base.en on Jetson CPU + G1
far-field mic yields ~50% command-transcription accuracy due
to model capacity and mic reverberation. Wake + ack + recording
+ trim + Whisper + fuzzy + brain + motion all verified working
end-to-end. Future improvement path (unused): close-talking USB
mic via pactl_parec, or Gemini Live via HTTP microservice.

Co-Authored-By: Claude Opus 4.7 (1M context) <noreply@anthropic.com>
2026-04-24 14:32:28 +04:00

264 lines
11 KiB
Python

#!/usr/bin/env python3
"""
Voice/wake_detector.py — custom wake-word detector (no ML, no Vosk, no Whisper).
Energy-envelope state machine. Monitors raw PCM audio and fires a wake
event when it sees a short speech burst (sized to match a single spoken
word like "Sanad") followed by a clear silence.
Why this exists:
Vosk's small English lexicon doesn't contain the word "sanad" and
substitutes arbitrary English words ("us", "of", "senate"). Whisper on
this Jetson's torch-aarch64 produces "!!!!!" garbage. Both are broken
for this specific hardware + wake word. An acoustic detector using
only numpy doesn't care what the word actually is — it detects the
*shape* of a single spoken word in the audio energy envelope.
Algorithm (state machine):
SILENCE ──(rms > speech_threshold)──> SPEAKING
SPEAKING ──(rms < silence_threshold for N chunks)──> ANALYZE
ANALYZE: if 0.2 s < speech_duration < 1.5 s → fire WAKE
else → reset to SILENCE (too short = cough, too long = sentence)
after fire → COOLDOWN for 1.5 s before next detection
What it does NOT do:
- Does not identify which word was spoken (anything in the
duration range triggers)
- Does not transcribe follow-on commands (you type those at the
terminal)
- Does not protect against loud non-speech (clapping, door slam)
Usage:
from Voice.wake_detector import WakeDetector
det = WakeDetector(sample_rate=16000)
while True:
chunk = mic.read_chunk(1024) # bytes of int16 PCM
if det.process(chunk):
print("Wake!")
"""
from __future__ import annotations
import time
from dataclasses import dataclass
from typing import Optional
import numpy as np
@dataclass
class WakeConfig:
sample_rate: int = 16_000
# RMS (int16 units) FLOOR for "this chunk is speech". The effective
# threshold is max(speech_threshold, ambient_baseline * adaptive_mult)
# so this is only a minimum guarantee — the detector adapts upward
# in noisy rooms but never below this floor.
# G1 far-field mic at normal speaking distance has rms ~ 80-400 for
# quiet speech, 400-1500 for clear speech. 80 catches quiet speech;
# raise to 120-150 if fan/typing noise triggers false wakes.
speech_threshold: float = 80.0
# How long a burst of speech must last to count as a "word".
min_word_duration_s: float = 0.20
max_word_duration_s: float = 1.50
# How long of continuous silence we need to consider the word ended.
post_silence_s: float = 0.30
# Minimum gap between two consecutive wake fires. Prevents a single
# spoken word from triggering twice.
cooldown_s: float = 1.50
# RMS window size — we analyze this many ms of audio per step.
chunk_ms: int = 50
# Adaptive: how many *recent silent* chunks to average for the noise
# floor, and the multiplier applied on top. effective_threshold =
# max(speech_threshold, baseline * adaptive_mult).
adaptive_window_n: int = 50 # ~2.5 s at 50 ms chunks
adaptive_mult: float = 3.0
# Periodic diagnostic log cadence (seconds). 0 disables.
diag_log_sec: float = 3.0
class WakeDetector:
"""Streaming acoustic wake detector — no language model required."""
STATE_SILENCE = "SILENCE"
STATE_SPEAKING = "SPEAKING"
def __init__(self, cfg: Optional[WakeConfig] = None):
self.cfg = cfg or WakeConfig()
self._chunk_samples = int(self.cfg.sample_rate * self.cfg.chunk_ms / 1000)
self._min_speech = int(self.cfg.min_word_duration_s * self.cfg.sample_rate)
self._max_speech = int(self.cfg.max_word_duration_s * self.cfg.sample_rate)
self._post_silence = int(self.cfg.post_silence_s * self.cfg.sample_rate)
self._state = self.STATE_SILENCE
self._speech_start = 0 # sample index where current burst began
self._silence_run = 0 # consecutive silent samples inside SPEAKING
self._sample_cursor = 0 # running sample count since start
self._cooldown_until = 0.0 # wall-clock time after which we can fire again
# A small rolling buffer of leftover samples (when the caller's
# chunks don't align with our internal analysis window).
self._carry = np.zeros(0, dtype=np.int16)
# Audio of the most-recent wake-triggering burst. Saved when the
# detector fires so callers (marcus_voice) can run Whisper on it
# and verify the word was actually "Sanad" rather than a cough.
self._burst_samples: list = [] # accumulated during SPEAKING
self._last_burst_audio: Optional[np.ndarray] = None
# Adaptive noise floor (rolling mean of RMS during SILENCE chunks).
self._baseline_buf = [] # last N silent-window RMS values
self._baseline = 0.0 # current estimate
self._peak_since_diag = 0.0 # max rms since last diag log
self._last_diag = time.time()
# Logger is optional — if the host app set up logging, use it.
try:
import logging
self._log = logging.getLogger("wake_detector")
except Exception:
self._log = None
# ── public API ────────────────────────────────────────────────
def process(self, pcm_bytes: bytes) -> bool:
"""
Feed int16 PCM bytes. Returns True once per spoken "word"
(short speech burst followed by silence).
"""
if not pcm_bytes:
return False
incoming = np.frombuffer(pcm_bytes, dtype=np.int16)
samples = np.concatenate([self._carry, incoming]) if self._carry.size else incoming
fired = False
n = self._chunk_samples
i = 0
while i + n <= samples.size:
window = samples[i:i + n]
if self._step(window):
fired = True
# break — flush the rest on next call so we get one fire per word
i += n
break
i += n
self._sample_cursor += n
# Keep whatever didn't fit in a full window for next call.
self._carry = samples[i:].copy()
return fired
def reset(self) -> None:
"""Drop all state — call when resuming from a long pause."""
self._state = self.STATE_SILENCE
self._silence_run = 0
self._carry = np.zeros(0, dtype=np.int16)
self._burst_samples = []
def get_last_burst(self) -> Optional[np.ndarray]:
"""
Return the int16 PCM samples of the most-recent wake-triggering
burst, or None if no wake has fired yet. Used by marcus_voice to
verify the triggering word was actually 'Sanad' before proceeding.
"""
return self._last_burst_audio
# ── internal ──────────────────────────────────────────────────
def _step(self, window: np.ndarray) -> bool:
rms = float(np.sqrt(np.mean(window.astype(np.float64) ** 2)))
# Effective threshold = max(config floor, adaptive baseline * mult)
eff = self.cfg.speech_threshold
if self._baseline > 0:
eff = max(eff, self._baseline * self.cfg.adaptive_mult)
is_speech = rms > eff
# Track peak for diag. Log periodically so you can *see* what the
# detector is hearing — invaluable when "not hearing me" happens.
if rms > self._peak_since_diag:
self._peak_since_diag = rms
now = time.time()
if self.cfg.diag_log_sec > 0 and (now - self._last_diag) >= self.cfg.diag_log_sec:
if self._log is not None:
self._log.info(
"wake: peak=%.0f baseline=%.0f eff_threshold=%.0f state=%s",
self._peak_since_diag, self._baseline, eff, self._state,
)
self._peak_since_diag = 0.0
self._last_diag = now
if now < self._cooldown_until:
return False # silent during cooldown
if self._state == self.STATE_SILENCE:
# Learn the noise floor ONLY in silence — so speech bursts
# don't pull the baseline up and lock us out of wake.
if not is_speech:
self._baseline_buf.append(rms)
if len(self._baseline_buf) > self.cfg.adaptive_window_n:
self._baseline_buf.pop(0)
if self._baseline_buf:
self._baseline = sum(self._baseline_buf) / len(self._baseline_buf)
if is_speech:
self._state = self.STATE_SPEAKING
self._speech_start = self._sample_cursor
self._silence_run = 0
# Begin capturing the burst audio for later Whisper verify.
self._burst_samples = [window.copy()]
return False
# STATE_SPEAKING
# Accumulate every window (speech OR silence inside the burst)
# so we capture the full word + trailing quiet.
self._burst_samples.append(window.copy())
if is_speech:
self._silence_run = 0
# Abort if the burst is longer than a single word — user is
# just talking, not addressing the robot.
if self._sample_cursor - self._speech_start > self._max_speech:
self._state = self.STATE_SILENCE
self._burst_samples = []
return False
# Silent window inside SPEAKING — accumulate.
self._silence_run += window.size
if self._silence_run >= self._post_silence:
speech_len = (self._sample_cursor - self._silence_run) - self._speech_start
self._state = self.STATE_SILENCE
self._silence_run = 0
if self._min_speech <= speech_len <= self._max_speech:
# Snapshot burst audio for the caller's Whisper verify.
self._last_burst_audio = (
np.concatenate(self._burst_samples)
if self._burst_samples else None
)
self._burst_samples = []
self._cooldown_until = now + self.cfg.cooldown_s
return True
return False
# ── standalone test ─────────────────────────────────────────────
if __name__ == "__main__":
import os
import sys
_HERE = os.path.dirname(os.path.abspath(__file__))
sys.path.insert(0, os.path.dirname(_HERE))
from Voice.builtin_mic import BuiltinMic
print("WakeDetector standalone test — say 'Sanad' a few times.")
print("(Ctrl-C to quit)\n")
det = WakeDetector()
mic = BuiltinMic()
mic.start()
try:
while True:
chunk = mic.read_chunk(1024)
if det.process(chunk):
print(f" [WAKE] (t={time.strftime('%H:%M:%S')})")
except KeyboardInterrupt:
pass
finally:
mic.stop()