8aab13d12d
The counter was incremented in start/stop/cancel but never read anywhere in the codebase. The race condition it was meant to guard against is practically impossible with the persistent stream design.
753 lines
28 KiB
Python
753 lines
28 KiB
Python
"""Voice Mode -- Push-to-talk audio recording and playback for the CLI.
|
|
|
|
Provides audio capture via sounddevice, WAV encoding via stdlib wave,
|
|
STT dispatch via tools.transcription_tools, and TTS playback via
|
|
sounddevice or system audio players.
|
|
|
|
Dependencies (optional):
|
|
pip install sounddevice numpy
|
|
or: pip install hermes-agent[voice]
|
|
"""
|
|
|
|
import logging
|
|
import os
|
|
import platform
|
|
import re
|
|
import shutil
|
|
import subprocess
|
|
import tempfile
|
|
import threading
|
|
import time
|
|
import wave
|
|
from pathlib import Path
|
|
from typing import Any, Dict, List, Optional
|
|
|
|
logger = logging.getLogger(__name__)
|
|
|
|
# ---------------------------------------------------------------------------
|
|
# Lazy audio imports -- never imported at module level to avoid crashing
|
|
# in headless environments (SSH, Docker, WSL, no PortAudio).
|
|
# ---------------------------------------------------------------------------
|
|
|
|
def _import_audio():
|
|
"""Lazy-import sounddevice and numpy. Returns (sd, np).
|
|
|
|
Raises ImportError or OSError if the libraries are not available
|
|
(e.g. PortAudio missing on headless servers).
|
|
"""
|
|
import sounddevice as sd
|
|
import numpy as np
|
|
return sd, np
|
|
|
|
|
|
def _audio_available() -> bool:
|
|
"""Return True if audio libraries can be imported."""
|
|
try:
|
|
_import_audio()
|
|
return True
|
|
except (ImportError, OSError):
|
|
return False
|
|
|
|
|
|
def detect_audio_environment() -> dict:
|
|
"""Detect if the current environment supports audio I/O.
|
|
|
|
Returns dict with 'available' (bool) and 'warnings' (list of strings).
|
|
"""
|
|
warnings = []
|
|
|
|
# SSH detection
|
|
if any(os.environ.get(v) for v in ('SSH_CLIENT', 'SSH_TTY', 'SSH_CONNECTION')):
|
|
warnings.append("Running over SSH -- no audio devices available")
|
|
|
|
# Docker detection
|
|
if os.path.exists('/.dockerenv'):
|
|
warnings.append("Running inside Docker container -- no audio devices")
|
|
|
|
# WSL detection
|
|
try:
|
|
with open('/proc/version', 'r') as f:
|
|
if 'microsoft' in f.read().lower():
|
|
warnings.append("Running in WSL -- audio requires PulseAudio bridge to Windows")
|
|
except (FileNotFoundError, PermissionError, OSError):
|
|
pass
|
|
|
|
# Check audio libraries
|
|
try:
|
|
sd, _ = _import_audio()
|
|
try:
|
|
devices = sd.query_devices()
|
|
if not devices:
|
|
warnings.append("No audio input/output devices detected")
|
|
except Exception:
|
|
warnings.append("Audio subsystem error (PortAudio cannot query devices)")
|
|
except (ImportError, OSError):
|
|
warnings.append("Audio libraries not installed (pip install sounddevice numpy)")
|
|
|
|
return {
|
|
"available": len(warnings) == 0,
|
|
"warnings": warnings,
|
|
}
|
|
|
|
# ---------------------------------------------------------------------------
|
|
# Recording parameters
|
|
# ---------------------------------------------------------------------------
|
|
SAMPLE_RATE = 16000 # Whisper native rate
|
|
CHANNELS = 1 # Mono
|
|
DTYPE = "int16" # 16-bit PCM
|
|
SAMPLE_WIDTH = 2 # bytes per sample (int16)
|
|
MAX_RECORDING_SECONDS = 120 # Safety cap
|
|
|
|
# Silence detection defaults
|
|
SILENCE_RMS_THRESHOLD = 200 # RMS below this = silence (int16 range 0-32767)
|
|
SILENCE_DURATION_SECONDS = 3.0 # Seconds of continuous silence before auto-stop
|
|
|
|
# Temp directory for voice recordings
|
|
_TEMP_DIR = os.path.join(tempfile.gettempdir(), "hermes_voice")
|
|
|
|
|
|
# ============================================================================
|
|
# Audio cues (beep tones)
|
|
# ============================================================================
|
|
def play_beep(frequency: int = 880, duration: float = 0.12, count: int = 1) -> None:
|
|
"""Play a short beep tone using numpy + sounddevice.
|
|
|
|
Args:
|
|
frequency: Tone frequency in Hz (default 880 = A5).
|
|
duration: Duration of each beep in seconds.
|
|
count: Number of beeps to play (with short gap between).
|
|
"""
|
|
try:
|
|
sd, np = _import_audio()
|
|
except (ImportError, OSError):
|
|
return
|
|
try:
|
|
gap = 0.06 # seconds between beeps
|
|
samples_per_beep = int(SAMPLE_RATE * duration)
|
|
samples_per_gap = int(SAMPLE_RATE * gap)
|
|
|
|
parts = []
|
|
for i in range(count):
|
|
t = np.linspace(0, duration, samples_per_beep, endpoint=False)
|
|
# Apply fade in/out to avoid click artifacts
|
|
tone = np.sin(2 * np.pi * frequency * t)
|
|
fade_len = min(int(SAMPLE_RATE * 0.01), samples_per_beep // 4)
|
|
tone[:fade_len] *= np.linspace(0, 1, fade_len)
|
|
tone[-fade_len:] *= np.linspace(1, 0, fade_len)
|
|
parts.append((tone * 0.3 * 32767).astype(np.int16))
|
|
if i < count - 1:
|
|
parts.append(np.zeros(samples_per_gap, dtype=np.int16))
|
|
|
|
audio = np.concatenate(parts)
|
|
sd.play(audio, samplerate=SAMPLE_RATE)
|
|
# sd.wait() calls Event.wait() without timeout — hangs forever if the
|
|
# audio device stalls. Poll with a 2s ceiling and force-stop.
|
|
deadline = time.monotonic() + 2.0
|
|
while sd.get_stream() and sd.get_stream().active and time.monotonic() < deadline:
|
|
time.sleep(0.01)
|
|
sd.stop()
|
|
except Exception as e:
|
|
logger.debug("Beep playback failed: %s", e)
|
|
|
|
|
|
# ============================================================================
|
|
# AudioRecorder
|
|
# ============================================================================
|
|
class AudioRecorder:
|
|
"""Thread-safe audio recorder using sounddevice.InputStream.
|
|
|
|
Usage::
|
|
|
|
recorder = AudioRecorder()
|
|
recorder.start(on_silence_stop=my_callback)
|
|
# ... user speaks ...
|
|
wav_path = recorder.stop() # returns path to WAV file
|
|
# or
|
|
recorder.cancel() # discard without saving
|
|
|
|
If ``on_silence_stop`` is provided, recording automatically stops when
|
|
the user is silent for ``silence_duration`` seconds and calls the callback.
|
|
"""
|
|
|
|
def __init__(self) -> None:
|
|
self._lock = threading.Lock()
|
|
self._stream: Any = None
|
|
self._frames: List[Any] = []
|
|
self._recording = False
|
|
self._start_time: float = 0.0
|
|
# Silence detection state
|
|
self._has_spoken = False
|
|
self._speech_start: float = 0.0 # When speech attempt began
|
|
self._dip_start: float = 0.0 # When current below-threshold dip began
|
|
self._min_speech_duration: float = 0.3 # Seconds of speech needed to confirm
|
|
self._max_dip_tolerance: float = 0.3 # Max dip duration before resetting speech
|
|
self._silence_start: float = 0.0
|
|
self._resume_start: float = 0.0 # Tracks sustained speech after silence starts
|
|
self._resume_dip_start: float = 0.0 # Dip tolerance tracker for resume detection
|
|
self._on_silence_stop = None
|
|
self._silence_threshold: int = SILENCE_RMS_THRESHOLD
|
|
self._silence_duration: float = SILENCE_DURATION_SECONDS
|
|
self._max_wait: float = 15.0 # Max seconds to wait for speech before auto-stop
|
|
# Peak RMS seen during recording (for speech presence check in stop())
|
|
self._peak_rms: int = 0
|
|
# Live audio level (read by UI for visual feedback)
|
|
self._current_rms: int = 0
|
|
|
|
# -- public properties ---------------------------------------------------
|
|
|
|
@property
|
|
def is_recording(self) -> bool:
|
|
return self._recording
|
|
|
|
@property
|
|
def elapsed_seconds(self) -> float:
|
|
if not self._recording:
|
|
return 0.0
|
|
return time.monotonic() - self._start_time
|
|
|
|
@property
|
|
def current_rms(self) -> int:
|
|
"""Current audio input RMS level (0-32767). Updated each audio chunk."""
|
|
return self._current_rms
|
|
|
|
# -- public methods ------------------------------------------------------
|
|
|
|
def _ensure_stream(self) -> None:
|
|
"""Create the audio InputStream once and keep it alive.
|
|
|
|
The stream stays open for the lifetime of the recorder. Between
|
|
recordings the callback simply discards audio chunks (``_recording``
|
|
is ``False``). This avoids the CoreAudio bug where closing and
|
|
re-opening an ``InputStream`` hangs indefinitely on macOS.
|
|
"""
|
|
if self._stream is not None:
|
|
return # already alive
|
|
|
|
sd, np = _import_audio()
|
|
|
|
def _callback(indata, frames, time_info, status): # noqa: ARG001
|
|
if status:
|
|
logger.debug("sounddevice status: %s", status)
|
|
# When not recording the stream is idle — discard audio.
|
|
if not self._recording:
|
|
return
|
|
self._frames.append(indata.copy())
|
|
|
|
# Compute RMS for level display and silence detection
|
|
rms = int(np.sqrt(np.mean(indata.astype(np.float64) ** 2)))
|
|
self._current_rms = rms
|
|
if rms > self._peak_rms:
|
|
self._peak_rms = rms
|
|
|
|
# Silence detection
|
|
if self._on_silence_stop is not None:
|
|
now = time.monotonic()
|
|
elapsed = now - self._start_time
|
|
|
|
if rms > self._silence_threshold:
|
|
# Audio is above threshold -- this is speech (or noise).
|
|
self._dip_start = 0.0 # Reset dip tracker
|
|
if self._speech_start == 0.0:
|
|
self._speech_start = now
|
|
elif not self._has_spoken and now - self._speech_start >= self._min_speech_duration:
|
|
self._has_spoken = True
|
|
logger.debug("Speech confirmed (%.2fs above threshold)",
|
|
now - self._speech_start)
|
|
# After speech is confirmed, only reset silence timer if
|
|
# speech is sustained (>0.3s above threshold). Brief
|
|
# spikes from ambient noise should NOT reset the timer.
|
|
if not self._has_spoken:
|
|
self._silence_start = 0.0
|
|
else:
|
|
# Track resumed speech with dip tolerance.
|
|
# Brief dips below threshold are normal during speech,
|
|
# so we mirror the initial speech detection pattern:
|
|
# start tracking, tolerate short dips, confirm after 0.3s.
|
|
self._resume_dip_start = 0.0 # Above threshold — no dip
|
|
if self._resume_start == 0.0:
|
|
self._resume_start = now
|
|
elif now - self._resume_start >= self._min_speech_duration:
|
|
self._silence_start = 0.0
|
|
self._resume_start = 0.0
|
|
elif self._has_spoken:
|
|
# Below threshold after speech confirmed.
|
|
# Use dip tolerance before resetting resume tracker —
|
|
# natural speech has brief dips below threshold.
|
|
if self._resume_start > 0:
|
|
if self._resume_dip_start == 0.0:
|
|
self._resume_dip_start = now
|
|
elif now - self._resume_dip_start >= self._max_dip_tolerance:
|
|
# Sustained dip — user actually stopped speaking
|
|
self._resume_start = 0.0
|
|
self._resume_dip_start = 0.0
|
|
elif self._speech_start > 0:
|
|
# We were in a speech attempt but RMS dipped.
|
|
# Tolerate brief dips (micro-pauses between syllables).
|
|
if self._dip_start == 0.0:
|
|
self._dip_start = now
|
|
elif now - self._dip_start >= self._max_dip_tolerance:
|
|
# Dip lasted too long -- genuine silence, reset
|
|
logger.debug("Speech attempt reset (dip lasted %.2fs)",
|
|
now - self._dip_start)
|
|
self._speech_start = 0.0
|
|
self._dip_start = 0.0
|
|
|
|
# Fire silence callback when:
|
|
# 1. User spoke then went silent for silence_duration, OR
|
|
# 2. No speech detected at all for max_wait seconds
|
|
should_fire = False
|
|
if self._has_spoken and rms <= self._silence_threshold:
|
|
# User was speaking and now is silent
|
|
if self._silence_start == 0.0:
|
|
self._silence_start = now
|
|
elif now - self._silence_start >= self._silence_duration:
|
|
logger.info("Silence detected (%.1fs), auto-stopping",
|
|
self._silence_duration)
|
|
should_fire = True
|
|
elif not self._has_spoken and elapsed >= self._max_wait:
|
|
logger.info("No speech within %.0fs, auto-stopping",
|
|
self._max_wait)
|
|
should_fire = True
|
|
|
|
if should_fire:
|
|
cb = self._on_silence_stop
|
|
self._on_silence_stop = None # fire only once
|
|
if cb:
|
|
def _safe_cb():
|
|
try:
|
|
cb()
|
|
except Exception as e:
|
|
logger.error("Silence callback failed: %s", e, exc_info=True)
|
|
threading.Thread(target=_safe_cb, daemon=True).start()
|
|
|
|
# Create stream — may block on CoreAudio (first call only).
|
|
try:
|
|
stream = sd.InputStream(
|
|
samplerate=SAMPLE_RATE,
|
|
channels=CHANNELS,
|
|
dtype=DTYPE,
|
|
callback=_callback,
|
|
)
|
|
stream.start()
|
|
except Exception as e:
|
|
raise RuntimeError(
|
|
f"Failed to open audio input stream: {e}. "
|
|
"Check that a microphone is connected and accessible."
|
|
) from e
|
|
self._stream = stream
|
|
|
|
def start(self, on_silence_stop=None) -> None:
|
|
"""Start capturing audio from the default input device.
|
|
|
|
The underlying InputStream is created once and kept alive across
|
|
recordings. Subsequent calls simply reset detection state and
|
|
toggle frame collection via ``_recording``.
|
|
|
|
Args:
|
|
on_silence_stop: Optional callback invoked (in a daemon thread) when
|
|
silence is detected after speech. The callback receives no arguments.
|
|
Use this to auto-stop recording and trigger transcription.
|
|
|
|
Raises ``RuntimeError`` if sounddevice/numpy are not installed
|
|
or if a recording is already in progress.
|
|
"""
|
|
try:
|
|
_import_audio()
|
|
except (ImportError, OSError) as e:
|
|
raise RuntimeError(
|
|
"Voice mode requires sounddevice and numpy.\n"
|
|
"Install with: pip install sounddevice numpy\n"
|
|
"Or: pip install hermes-agent[voice]"
|
|
) from e
|
|
|
|
with self._lock:
|
|
if self._recording:
|
|
return # already recording
|
|
|
|
self._frames = []
|
|
self._start_time = time.monotonic()
|
|
self._has_spoken = False
|
|
self._speech_start = 0.0
|
|
self._dip_start = 0.0
|
|
self._silence_start = 0.0
|
|
self._resume_start = 0.0
|
|
self._resume_dip_start = 0.0
|
|
self._peak_rms = 0
|
|
self._current_rms = 0
|
|
self._on_silence_stop = on_silence_stop
|
|
|
|
# Ensure the persistent stream is alive (no-op after first call).
|
|
self._ensure_stream()
|
|
|
|
with self._lock:
|
|
self._recording = True
|
|
logger.info("Voice recording started (rate=%d, channels=%d)", SAMPLE_RATE, CHANNELS)
|
|
|
|
def _close_stream_with_timeout(self, timeout: float = 3.0) -> None:
|
|
"""Close the audio stream with a timeout to prevent CoreAudio hangs."""
|
|
if self._stream is None:
|
|
return
|
|
|
|
stream = self._stream
|
|
self._stream = None
|
|
|
|
def _do_close():
|
|
try:
|
|
stream.stop()
|
|
stream.close()
|
|
except Exception:
|
|
pass
|
|
|
|
t = threading.Thread(target=_do_close, daemon=True)
|
|
t.start()
|
|
t.join(timeout=timeout)
|
|
if t.is_alive():
|
|
logger.warning("Audio stream close timed out after %.1fs — forcing ahead", timeout)
|
|
|
|
def stop(self) -> Optional[str]:
|
|
"""Stop recording and write captured audio to a WAV file.
|
|
|
|
The underlying stream is kept alive for reuse — only frame
|
|
collection is stopped.
|
|
|
|
Returns:
|
|
Path to the WAV file, or ``None`` if no audio was captured.
|
|
"""
|
|
with self._lock:
|
|
if not self._recording:
|
|
return None
|
|
|
|
self._recording = False
|
|
self._current_rms = 0
|
|
# Stream stays alive — no close needed.
|
|
|
|
if not self._frames:
|
|
return None
|
|
|
|
# Concatenate frames and write WAV
|
|
_, np = _import_audio()
|
|
audio_data = np.concatenate(self._frames, axis=0)
|
|
self._frames = []
|
|
|
|
elapsed = time.monotonic() - self._start_time
|
|
logger.info("Voice recording stopped (%.1fs, %d samples)", elapsed, len(audio_data))
|
|
|
|
# Skip very short recordings (< 0.3s of audio)
|
|
min_samples = int(SAMPLE_RATE * 0.3)
|
|
if len(audio_data) < min_samples:
|
|
logger.debug("Recording too short (%d samples), discarding", len(audio_data))
|
|
return None
|
|
|
|
# Skip silent recordings using peak RMS (not overall average, which
|
|
# gets diluted by silence at the end of the recording).
|
|
if self._peak_rms < SILENCE_RMS_THRESHOLD:
|
|
logger.info("Recording too quiet (peak RMS=%d < %d), discarding",
|
|
self._peak_rms, SILENCE_RMS_THRESHOLD)
|
|
return None
|
|
|
|
return self._write_wav(audio_data)
|
|
|
|
def cancel(self) -> None:
|
|
"""Stop recording and discard all captured audio.
|
|
|
|
The underlying stream is kept alive for reuse.
|
|
"""
|
|
with self._lock:
|
|
self._recording = False
|
|
self._frames = []
|
|
self._on_silence_stop = None
|
|
self._current_rms = 0
|
|
logger.info("Voice recording cancelled")
|
|
|
|
def shutdown(self) -> None:
|
|
"""Release the audio stream. Call when voice mode is disabled."""
|
|
with self._lock:
|
|
self._recording = False
|
|
self._frames = []
|
|
self._on_silence_stop = None
|
|
self._close_stream_with_timeout()
|
|
logger.info("AudioRecorder shut down")
|
|
|
|
# -- private helpers -----------------------------------------------------
|
|
|
|
@staticmethod
|
|
def _write_wav(audio_data) -> str:
|
|
"""Write numpy int16 audio data to a WAV file.
|
|
|
|
Returns the file path.
|
|
"""
|
|
os.makedirs(_TEMP_DIR, exist_ok=True)
|
|
timestamp = time.strftime("%Y%m%d_%H%M%S")
|
|
wav_path = os.path.join(_TEMP_DIR, f"recording_{timestamp}.wav")
|
|
|
|
with wave.open(wav_path, "wb") as wf:
|
|
wf.setnchannels(CHANNELS)
|
|
wf.setsampwidth(SAMPLE_WIDTH)
|
|
wf.setframerate(SAMPLE_RATE)
|
|
wf.writeframes(audio_data.tobytes())
|
|
|
|
file_size = os.path.getsize(wav_path)
|
|
logger.info("WAV written: %s (%d bytes)", wav_path, file_size)
|
|
return wav_path
|
|
|
|
|
|
# ============================================================================
|
|
# Whisper hallucination filter
|
|
# ============================================================================
|
|
# Whisper commonly hallucinates these phrases on silent/near-silent audio.
|
|
WHISPER_HALLUCINATIONS = {
|
|
"thank you.",
|
|
"thank you",
|
|
"thanks for watching.",
|
|
"thanks for watching",
|
|
"subscribe to my channel.",
|
|
"subscribe to my channel",
|
|
"like and subscribe.",
|
|
"like and subscribe",
|
|
"please subscribe.",
|
|
"please subscribe",
|
|
"thank you for watching.",
|
|
"thank you for watching",
|
|
"bye.",
|
|
"bye",
|
|
"you",
|
|
"the end.",
|
|
"the end",
|
|
# Non-English hallucinations (common on silence)
|
|
"продолжение следует",
|
|
"продолжение следует...",
|
|
"sous-titres",
|
|
"sous-titres réalisés par la communauté d'amara.org",
|
|
"sottotitoli creati dalla comunità amara.org",
|
|
"untertitel von stephanie geiges",
|
|
"amara.org",
|
|
"www.mooji.org",
|
|
"ご視聴ありがとうございました",
|
|
}
|
|
|
|
# Regex patterns for repetitive hallucinations (e.g. "Thank you. Thank you. Thank you.")
|
|
_HALLUCINATION_REPEAT_RE = re.compile(
|
|
r'^(?:thank you|thanks|bye|you|ok|okay|the end|\.|\s|,|!)+$',
|
|
flags=re.IGNORECASE,
|
|
)
|
|
|
|
|
|
def is_whisper_hallucination(transcript: str) -> bool:
|
|
"""Check if a transcript is a known Whisper hallucination on silence."""
|
|
cleaned = transcript.strip().lower()
|
|
if not cleaned:
|
|
return True
|
|
# Exact match against known phrases
|
|
if cleaned.rstrip('.!') in WHISPER_HALLUCINATIONS or cleaned in WHISPER_HALLUCINATIONS:
|
|
return True
|
|
# Repetitive patterns (e.g. "Thank you. Thank you. Thank you. you")
|
|
if _HALLUCINATION_REPEAT_RE.match(cleaned):
|
|
return True
|
|
return False
|
|
|
|
|
|
# ============================================================================
|
|
# STT dispatch
|
|
# ============================================================================
|
|
def transcribe_recording(wav_path: str, model: Optional[str] = None) -> Dict[str, Any]:
|
|
"""Transcribe a WAV recording using the existing Whisper pipeline.
|
|
|
|
Delegates to ``tools.transcription_tools.transcribe_audio()``.
|
|
Filters out known Whisper hallucinations on silent audio.
|
|
|
|
Args:
|
|
wav_path: Path to the WAV file.
|
|
model: Whisper model name (default: from config or ``whisper-1``).
|
|
|
|
Returns:
|
|
Dict with ``success``, ``transcript``, and optionally ``error``.
|
|
"""
|
|
from tools.transcription_tools import transcribe_audio
|
|
|
|
result = transcribe_audio(wav_path, model=model)
|
|
|
|
# Filter out Whisper hallucinations (common on silent/near-silent audio)
|
|
if result.get("success") and is_whisper_hallucination(result.get("transcript", "")):
|
|
logger.info("Filtered Whisper hallucination: %r", result["transcript"])
|
|
return {"success": True, "transcript": "", "filtered": True}
|
|
|
|
return result
|
|
|
|
|
|
# ============================================================================
|
|
# Audio playback (interruptable)
|
|
# ============================================================================
|
|
|
|
# Global reference to the active playback process so it can be interrupted.
|
|
_active_playback: Optional[subprocess.Popen] = None
|
|
_playback_lock = threading.Lock()
|
|
|
|
|
|
def stop_playback() -> None:
|
|
"""Interrupt the currently playing audio (if any)."""
|
|
global _active_playback
|
|
with _playback_lock:
|
|
proc = _active_playback
|
|
_active_playback = None
|
|
if proc and proc.poll() is None:
|
|
try:
|
|
proc.terminate()
|
|
logger.info("Audio playback interrupted")
|
|
except Exception:
|
|
pass
|
|
# Also stop sounddevice playback if active
|
|
try:
|
|
sd, _ = _import_audio()
|
|
sd.stop()
|
|
except Exception:
|
|
pass
|
|
|
|
|
|
def play_audio_file(file_path: str) -> bool:
|
|
"""Play an audio file through the default output device.
|
|
|
|
Strategy:
|
|
1. WAV files via ``sounddevice.play()`` when available.
|
|
2. System commands: ``afplay`` (macOS), ``ffplay`` (cross-platform),
|
|
``aplay`` (Linux ALSA).
|
|
|
|
Playback can be interrupted by calling ``stop_playback()``.
|
|
|
|
Returns:
|
|
``True`` if playback succeeded, ``False`` otherwise.
|
|
"""
|
|
global _active_playback
|
|
|
|
if not os.path.isfile(file_path):
|
|
logger.warning("Audio file not found: %s", file_path)
|
|
return False
|
|
|
|
# Try sounddevice for WAV files
|
|
if file_path.endswith(".wav"):
|
|
try:
|
|
sd, np = _import_audio()
|
|
with wave.open(file_path, "rb") as wf:
|
|
frames = wf.readframes(wf.getnframes())
|
|
audio_data = np.frombuffer(frames, dtype=np.int16)
|
|
sample_rate = wf.getframerate()
|
|
|
|
sd.play(audio_data, samplerate=sample_rate)
|
|
sd.wait()
|
|
return True
|
|
except (ImportError, OSError):
|
|
pass # audio libs not available, fall through to system players
|
|
except Exception as e:
|
|
logger.debug("sounddevice playback failed: %s", e)
|
|
|
|
# Fall back to system audio players (using Popen for interruptability)
|
|
system = platform.system()
|
|
players = []
|
|
|
|
if system == "Darwin":
|
|
players.append(["afplay", file_path])
|
|
players.append(["ffplay", "-nodisp", "-autoexit", "-loglevel", "quiet", file_path])
|
|
if system == "Linux":
|
|
players.append(["aplay", "-q", file_path])
|
|
|
|
for cmd in players:
|
|
exe = shutil.which(cmd[0])
|
|
if exe:
|
|
try:
|
|
proc = subprocess.Popen(cmd, stdout=subprocess.DEVNULL, stderr=subprocess.DEVNULL)
|
|
with _playback_lock:
|
|
_active_playback = proc
|
|
proc.wait(timeout=300)
|
|
with _playback_lock:
|
|
_active_playback = None
|
|
return True
|
|
except Exception as e:
|
|
logger.debug("System player %s failed: %s", cmd[0], e)
|
|
with _playback_lock:
|
|
_active_playback = None
|
|
|
|
logger.warning("No audio player available for %s", file_path)
|
|
return False
|
|
|
|
|
|
# ============================================================================
|
|
# Requirements check
|
|
# ============================================================================
|
|
def check_voice_requirements() -> Dict[str, Any]:
|
|
"""Check if all voice mode requirements are met.
|
|
|
|
Returns:
|
|
Dict with ``available``, ``audio_available``, ``stt_key_set``,
|
|
``missing_packages``, and ``details``.
|
|
"""
|
|
openai_key = bool(os.getenv("VOICE_TOOLS_OPENAI_KEY"))
|
|
groq_key = bool(os.getenv("GROQ_API_KEY"))
|
|
stt_key_set = openai_key or groq_key
|
|
missing: List[str] = []
|
|
has_audio = _audio_available()
|
|
|
|
if not has_audio:
|
|
missing.extend(["sounddevice", "numpy"])
|
|
|
|
# Environment detection
|
|
env_check = detect_audio_environment()
|
|
|
|
available = has_audio and stt_key_set and env_check["available"]
|
|
details_parts = []
|
|
|
|
if has_audio:
|
|
details_parts.append("Audio capture: OK")
|
|
else:
|
|
details_parts.append("Audio capture: MISSING (pip install sounddevice numpy)")
|
|
|
|
if openai_key:
|
|
details_parts.append("STT API key: OK (OpenAI)")
|
|
elif groq_key:
|
|
details_parts.append("STT API key: OK (Groq)")
|
|
else:
|
|
details_parts.append("STT API key: MISSING (set GROQ_API_KEY or VOICE_TOOLS_OPENAI_KEY)")
|
|
|
|
for warning in env_check["warnings"]:
|
|
details_parts.append(f"Environment: {warning}")
|
|
|
|
return {
|
|
"available": available,
|
|
"audio_available": has_audio,
|
|
"stt_key_set": stt_key_set,
|
|
"missing_packages": missing,
|
|
"details": "\n".join(details_parts),
|
|
"environment": env_check,
|
|
}
|
|
|
|
|
|
# ============================================================================
|
|
# Temp file cleanup
|
|
# ============================================================================
|
|
def cleanup_temp_recordings(max_age_seconds: int = 3600) -> int:
|
|
"""Remove old temporary voice recording files.
|
|
|
|
Args:
|
|
max_age_seconds: Delete files older than this (default: 1 hour).
|
|
|
|
Returns:
|
|
Number of files deleted.
|
|
"""
|
|
if not os.path.isdir(_TEMP_DIR):
|
|
return 0
|
|
|
|
deleted = 0
|
|
now = time.time()
|
|
|
|
for entry in os.scandir(_TEMP_DIR):
|
|
if entry.is_file() and entry.name.startswith("recording_") and entry.name.endswith(".wav"):
|
|
try:
|
|
age = now - entry.stat().st_mtime
|
|
if age > max_age_seconds:
|
|
os.unlink(entry.path)
|
|
deleted += 1
|
|
except OSError:
|
|
pass
|
|
|
|
if deleted:
|
|
logger.debug("Cleaned up %d old voice recordings", deleted)
|
|
return deleted
|