derp now handles both listening and speaking, so audition no longer needs cross-bot lookup or dual-play through merlin. Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
618 lines
20 KiB
Python
618 lines
20 KiB
Python
"""Plugin: voice STT/TTS for Mumble channels.
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Listens for voice audio via pymumble's sound callback, buffers PCM per
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user, transcribes via Whisper STT on silence, and provides TTS playback
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via Piper. Commands: !listen, !say.
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"""
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from __future__ import annotations
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import asyncio
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import io
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import json
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import logging
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import math
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import struct
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import threading
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import time
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import urllib.request
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import wave
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from derp.http import urlopen as _urlopen
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from derp.plugin import command
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log = logging.getLogger(__name__)
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# -- Constants ---------------------------------------------------------------
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_SAMPLE_RATE = 48000
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_CHANNELS = 1
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_SAMPLE_WIDTH = 2 # s16le = 2 bytes per sample
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_SILENCE_GAP = 1.5 # seconds of silence before flushing
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_MIN_DURATION = 0.5 # discard utterances shorter than this
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_MAX_DURATION = 30.0 # cap buffer at this many seconds
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_MIN_BYTES = int(_MIN_DURATION * _SAMPLE_RATE * _SAMPLE_WIDTH)
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_MAX_BYTES = int(_MAX_DURATION * _SAMPLE_RATE * _SAMPLE_WIDTH)
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_FLUSH_INTERVAL = 0.5 # flush monitor poll interval
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_MAX_SAY_LEN = 500 # max characters for !say
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_WHISPER_URL = "http://192.168.129.9:8080/inference"
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_PIPER_URL = "http://192.168.129.9:5100/"
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# -- Per-bot state -----------------------------------------------------------
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def _ps(bot):
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"""Per-bot plugin runtime state."""
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cfg = getattr(bot, "config", {}).get("voice", {})
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trigger = cfg.get("trigger", "")
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# Bias Whisper toward the trigger word unless explicitly configured
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default_prompt = f"{trigger.capitalize()}, " if trigger else ""
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return bot._pstate.setdefault("voice", {
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"listen": False,
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"trigger": trigger,
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"buffers": {}, # {username: bytearray}
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"last_ts": {}, # {username: float monotonic}
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"flush_task": None,
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"lock": threading.Lock(),
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"silence_gap": cfg.get("silence_gap", _SILENCE_GAP),
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"whisper_url": cfg.get("whisper_url", _WHISPER_URL),
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"piper_url": cfg.get("piper_url", _PIPER_URL),
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"voice": cfg.get("voice", ""),
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"length_scale": cfg.get("length_scale", 1.0),
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"noise_scale": cfg.get("noise_scale", 0.667),
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"noise_w": cfg.get("noise_w", 0.8),
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"fx": cfg.get("fx", ""),
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"initial_prompt": cfg.get("initial_prompt", default_prompt),
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"_listener_registered": False,
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})
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# -- Helpers -----------------------------------------------------------------
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def _is_mumble(bot) -> bool:
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"""Check if bot supports voice streaming."""
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return hasattr(bot, "stream_audio")
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def _pcm_to_wav(pcm: bytes) -> bytes:
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"""Wrap raw s16le 48kHz mono PCM in a WAV container."""
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buf = io.BytesIO()
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with wave.open(buf, "wb") as wf:
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wf.setnchannels(_CHANNELS)
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wf.setsampwidth(_SAMPLE_WIDTH)
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wf.setframerate(_SAMPLE_RATE)
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wf.writeframes(pcm)
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return buf.getvalue()
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# -- Acknowledge tone --------------------------------------------------------
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_ACK_FREQ = (880, 1320) # A5 -> E6 ascending
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_ACK_NOTE_DUR = 0.15 # seconds per note
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_ACK_AMP = 12000 # gentle amplitude
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_ACK_FRAME = 960 # 20ms at 48kHz, matches Mumble native
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async def _ack_tone(bot) -> None:
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"""Play a short two-tone ascending chime via pymumble sound_output."""
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mu = getattr(bot, "_mumble", None)
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if mu is None:
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return
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so = mu.sound_output
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if so is None:
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return
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# Unmute if self-muted (stream_audio handles re-mute later)
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if getattr(bot, "_self_mute", False):
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if bot._mute_task and not bot._mute_task.done():
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bot._mute_task.cancel()
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bot._mute_task = None
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try:
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mu.users.myself.unmute()
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except Exception:
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pass
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frames_per_note = int(_ACK_NOTE_DUR / 0.02) # 0.02s per frame
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for freq in _ACK_FREQ:
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for i in range(frames_per_note):
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samples = []
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for j in range(_ACK_FRAME):
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t = (i * _ACK_FRAME + j) / _SAMPLE_RATE
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samples.append(int(_ACK_AMP * math.sin(2 * math.pi * freq * t)))
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pcm = struct.pack(f"<{_ACK_FRAME}h", *samples)
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so.add_sound(pcm)
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while so.get_buffer_size() > 0.5:
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await asyncio.sleep(0.02)
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# Wait for tone to finish
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while so.get_buffer_size() > 0:
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await asyncio.sleep(0.05)
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# -- STT: Sound listener (pymumble thread) ----------------------------------
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def _on_voice(bot, user, sound_chunk):
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"""Buffer incoming voice PCM per user. Runs on pymumble thread."""
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ps = _ps(bot)
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if not ps["listen"] and not ps["trigger"]:
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return
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try:
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name = user["name"]
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except (KeyError, TypeError):
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name = None
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if not name or name == bot.nick:
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return
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pcm = sound_chunk.pcm
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if not pcm:
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return
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with ps["lock"]:
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if name not in ps["buffers"]:
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ps["buffers"][name] = bytearray()
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buf = ps["buffers"][name]
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buf.extend(pcm)
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if len(buf) > _MAX_BYTES:
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ps["buffers"][name] = bytearray(buf[-_MAX_BYTES:])
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ps["last_ts"][name] = time.monotonic()
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# -- STT: Whisper transcription ---------------------------------------------
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def _transcribe(ps, pcm: bytes) -> str:
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"""POST PCM (as WAV) to Whisper and return transcribed text. Blocking."""
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wav_data = _pcm_to_wav(pcm)
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boundary = "----derp_voice_boundary"
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body = (
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f"--{boundary}\r\n"
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f'Content-Disposition: form-data; name="file"; filename="audio.wav"\r\n'
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f"Content-Type: audio/wav\r\n\r\n"
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).encode() + wav_data + (
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f"\r\n--{boundary}\r\n"
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f'Content-Disposition: form-data; name="response_format"\r\n\r\n'
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f"json"
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).encode()
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# Bias Whisper toward the trigger word when configured
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prompt = ps.get("initial_prompt", "")
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if prompt:
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body += (
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f"\r\n--{boundary}\r\n"
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f'Content-Disposition: form-data; name="initial_prompt"\r\n\r\n'
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f"{prompt}"
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).encode()
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body += f"\r\n--{boundary}--\r\n".encode()
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req = urllib.request.Request(ps["whisper_url"], data=body, method="POST")
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req.add_header("Content-Type", f"multipart/form-data; boundary={boundary}")
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resp = _urlopen(req, timeout=30, proxy=False)
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data = json.loads(resp.read())
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resp.close()
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return data.get("text", "").strip()
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# -- STT: Flush monitor (asyncio background task) ---------------------------
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async def _flush_monitor(bot):
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"""Poll for silence gaps and transcribe completed utterances."""
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ps = _ps(bot)
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loop = asyncio.get_running_loop()
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try:
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while ps["listen"] or ps["trigger"]:
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await asyncio.sleep(_FLUSH_INTERVAL)
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now = time.monotonic()
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to_flush: list[tuple[str, bytes]] = []
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with ps["lock"]:
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for name in list(ps["last_ts"]):
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elapsed = now - ps["last_ts"][name]
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if elapsed >= ps["silence_gap"] and name in ps["buffers"]:
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pcm = bytes(ps["buffers"].pop(name))
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del ps["last_ts"][name]
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to_flush.append((name, pcm))
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for name, pcm in to_flush:
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if len(pcm) < _MIN_BYTES:
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continue
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try:
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text = await loop.run_in_executor(
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None, _transcribe, ps, pcm,
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)
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except Exception:
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log.exception("voice: transcription failed for %s", name)
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continue
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if not text or text.strip("., ") == "":
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continue
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trigger = ps["trigger"]
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if trigger and text.lower().startswith(trigger.lower()):
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remainder = text[len(trigger):].strip().lstrip(",.;:!?")
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if remainder:
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log.info("voice: trigger from %s: %s", name, remainder)
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bot._spawn(
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_tts_play(bot, remainder), name="voice-tts",
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)
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continue
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if ps["listen"]:
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log.info("voice: %s said: %s", name, text)
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await bot.action("0", f"heard {name} say: {text}")
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except asyncio.CancelledError:
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pass
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except Exception:
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log.exception("voice: flush monitor error")
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# -- TTS: Piper fetch + playback --------------------------------------------
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def _fetch_tts(piper_url: str, text: str) -> str | None:
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"""POST text to Piper TTS and save the WAV response. Blocking."""
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import tempfile
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try:
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payload = json.dumps({"text": text}).encode()
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req = urllib.request.Request(
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piper_url, data=payload, method="POST",
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)
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req.add_header("Content-Type", "application/json")
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resp = _urlopen(req, timeout=30, proxy=False)
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data = resp.read()
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resp.close()
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if not data:
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return None
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tmp = tempfile.NamedTemporaryFile(
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suffix=".wav", prefix="derp_tts_", delete=False,
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)
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tmp.write(data)
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tmp.close()
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return tmp.name
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except Exception:
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log.exception("voice: TTS fetch failed")
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return None
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async def _tts_play(bot, text: str):
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"""Fetch TTS audio and play it via stream_audio.
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Uses the configured voice profile (voice, fx, piper params) when set,
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otherwise falls back to Piper's default voice.
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"""
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from pathlib import Path
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ps = _ps(bot)
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loop = asyncio.get_running_loop()
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if ps["voice"] or ps["fx"]:
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wav_path = await loop.run_in_executor(
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None, lambda: _fetch_tts_voice(
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ps["piper_url"], text,
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voice=ps["voice"],
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length_scale=ps["length_scale"],
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noise_scale=ps["noise_scale"],
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noise_w=ps["noise_w"],
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fx=ps["fx"],
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),
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)
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else:
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wav_path = await loop.run_in_executor(
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None, _fetch_tts, ps["piper_url"], text,
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)
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if wav_path is None:
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return
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try:
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# Signal music plugin to duck, wait for it to take effect
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bot.registry._tts_active = True
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await asyncio.sleep(1.5)
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await _ack_tone(bot)
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done = asyncio.Event()
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await bot.stream_audio(str(wav_path), volume=1.0, on_done=done)
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await done.wait()
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finally:
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bot.registry._tts_active = False
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Path(wav_path).unlink(missing_ok=True)
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# -- Listener lifecycle -----------------------------------------------------
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def _ensure_listener(bot):
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"""Register the sound listener callback (idempotent)."""
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ps = _ps(bot)
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if ps["_listener_registered"]:
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return
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if not hasattr(bot, "_sound_listeners"):
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return
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bot._sound_listeners.append(lambda user, chunk: _on_voice(bot, user, chunk))
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ps["_listener_registered"] = True
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log.info("voice: registered sound listener")
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def _ensure_flush_task(bot):
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"""Start the flush monitor if not running."""
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ps = _ps(bot)
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task = ps.get("flush_task")
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if task and not task.done():
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return
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ps["flush_task"] = bot._spawn(
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_flush_monitor(bot), name="voice-flush-monitor",
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)
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def _stop_flush_task(bot):
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"""Cancel the flush monitor."""
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ps = _ps(bot)
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task = ps.get("flush_task")
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if task and not task.done():
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task.cancel()
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ps["flush_task"] = None
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# -- Commands ----------------------------------------------------------------
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@command("listen", help="Voice: !listen [on|off] -- toggle STT", tier="admin")
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async def cmd_listen(bot, message):
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"""Toggle voice-to-text transcription."""
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if not _is_mumble(bot):
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await bot.reply(message, "Voice is Mumble-only")
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return
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ps = _ps(bot)
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parts = message.text.split()
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if len(parts) < 2:
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state = "on" if ps["listen"] else "off"
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trigger = ps["trigger"]
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info = f"Listen: {state}"
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if trigger:
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info += f" | Trigger: {trigger}"
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await bot.reply(message, info)
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return
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sub = parts[1].lower()
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if sub == "on":
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ps["listen"] = True
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_ensure_listener(bot)
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_ensure_flush_task(bot)
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await bot.reply(message, "Listening for voice")
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elif sub == "off":
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ps["listen"] = False
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if not ps["trigger"]:
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with ps["lock"]:
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ps["buffers"].clear()
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ps["last_ts"].clear()
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_stop_flush_task(bot)
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await bot.reply(message, "Stopped listening")
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else:
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await bot.reply(message, "Usage: !listen [on|off]")
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@command("say", help="Voice: !say <text> -- text-to-speech")
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async def cmd_say(bot, message):
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"""Speak text aloud via Piper TTS."""
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if not _is_mumble(bot):
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await bot.reply(message, "Voice is Mumble-only")
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return
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parts = message.text.split(None, 1)
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if len(parts) < 2:
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await bot.reply(message, "Usage: !say <text>")
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return
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text = parts[1].strip()
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if len(text) > _MAX_SAY_LEN:
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await bot.reply(message, f"Text too long (max {_MAX_SAY_LEN} chars)")
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return
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bot._spawn(_tts_play(bot, text), name="voice-tts")
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def _split_fx(fx: str) -> tuple[list[str], str]:
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"""Split FX chain into rubberband CLI args and ffmpeg filter string.
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Alpine's ffmpeg lacks librubberband, so pitch shifting is handled by
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the ``rubberband`` CLI tool and remaining filters by ffmpeg.
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"""
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import math
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parts = fx.split(",")
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rb_args: list[str] = []
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ff_parts: list[str] = []
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for part in parts:
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if part.startswith("rubberband="):
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opts: dict[str, str] = {}
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for kv in part[len("rubberband="):].split(":"):
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k, _, v = kv.partition("=")
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opts[k] = v
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if "pitch" in opts:
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semitones = 12 * math.log2(float(opts["pitch"]))
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rb_args += ["--pitch", f"{semitones:.2f}"]
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if opts.get("formant") == "1":
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rb_args.append("--formant")
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else:
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ff_parts.append(part)
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return rb_args, ",".join(ff_parts)
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def _fetch_tts_voice(piper_url: str, text: str, *, voice: str = "",
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speaker_id: int = 0, length_scale: float = 1.0,
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noise_scale: float = 0.667, noise_w: float = 0.8,
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fx: str = "") -> str | None:
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"""Fetch TTS with explicit voice params and optional FX. Blocking.
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Pitch shifting uses the ``rubberband`` CLI (Alpine ffmpeg has no
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librubberband); remaining audio filters go through ffmpeg.
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"""
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import os
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import subprocess
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import tempfile
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payload = {"text": text}
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if voice:
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payload["voice"] = voice
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if speaker_id:
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payload["speaker_id"] = speaker_id
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payload["length_scale"] = length_scale
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payload["noise_scale"] = noise_scale
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payload["noise_w"] = noise_w
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data = json.dumps(payload).encode()
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req = urllib.request.Request(piper_url, data=data, method="POST")
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req.add_header("Content-Type", "application/json")
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resp = _urlopen(req, timeout=30, proxy=False)
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wav_data = resp.read()
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resp.close()
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if not wav_data:
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return None
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tmp = tempfile.NamedTemporaryFile(suffix=".wav", prefix="derp_aud_", delete=False)
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tmp.write(wav_data)
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tmp.close()
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if not fx:
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return tmp.name
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rb_args, ff_filters = _split_fx(fx)
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current = tmp.name
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# Pitch shift via rubberband CLI
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if rb_args:
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rb_out = tempfile.NamedTemporaryFile(
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suffix=".wav", prefix="derp_aud_", delete=False,
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)
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rb_out.close()
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r = subprocess.run(
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["rubberband"] + rb_args + [current, rb_out.name],
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capture_output=True, timeout=15,
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)
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os.unlink(current)
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if r.returncode != 0:
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log.warning("voice: rubberband failed: %s", r.stderr[:200])
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os.unlink(rb_out.name)
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return None
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current = rb_out.name
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# Remaining filters via ffmpeg
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if ff_filters:
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ff_out = tempfile.NamedTemporaryFile(
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suffix=".wav", prefix="derp_aud_", delete=False,
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)
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ff_out.close()
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r = subprocess.run(
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["ffmpeg", "-y", "-i", current, "-af", ff_filters, ff_out.name],
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capture_output=True, timeout=15,
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)
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os.unlink(current)
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if r.returncode != 0:
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log.warning("voice: ffmpeg failed: %s", r.stderr[:200])
|
|
os.unlink(ff_out.name)
|
|
return None
|
|
current = ff_out.name
|
|
|
|
return current
|
|
|
|
|
|
@command("audition", help="Voice: !audition -- play voice samples", tier="admin")
|
|
async def cmd_audition(bot, message):
|
|
"""Play voice samples through Mumble for comparison."""
|
|
if not _is_mumble(bot):
|
|
return
|
|
|
|
ps = _ps(bot)
|
|
piper_url = ps["piper_url"]
|
|
phrase = "The sorcerer has arrived. I have seen things beyond your understanding."
|
|
|
|
# FX building blocks
|
|
_deep = "rubberband=pitch=0.87:formant=1"
|
|
_bass = "bass=g=6:f=110:w=0.6"
|
|
_bass_heavy = "equalizer=f=80:t=h:w=150:g=8"
|
|
_echo_subtle = "aecho=0.8:0.6:25|40:0.25|0.15"
|
|
_echo_chamber = "aecho=0.8:0.88:60:0.35"
|
|
_echo_cave = "aecho=0.8:0.7:40|70|100:0.3|0.2|0.1"
|
|
|
|
samples = [
|
|
# -- Base voices (no FX) for reference
|
|
("ryan-high raw", "en_US-ryan-high", 0, ""),
|
|
("lessac-high raw", "en_US-lessac-high", 0, ""),
|
|
# -- Deep pitch only
|
|
("ryan deep", "en_US-ryan-high", 0,
|
|
_deep),
|
|
("lessac deep", "en_US-lessac-high", 0,
|
|
_deep),
|
|
# -- Deep + bass boost
|
|
("ryan deep+bass", "en_US-ryan-high", 0,
|
|
f"{_deep},{_bass}"),
|
|
("lessac deep+bass", "en_US-lessac-high", 0,
|
|
f"{_deep},{_bass}"),
|
|
# -- Deep + heavy bass
|
|
("ryan deep+heavy bass", "en_US-ryan-high", 0,
|
|
f"{_deep},{_bass_heavy}"),
|
|
# -- Deep + bass + subtle echo
|
|
("ryan deep+bass+echo", "en_US-ryan-high", 0,
|
|
f"{_deep},{_bass},{_echo_subtle}"),
|
|
("lessac deep+bass+echo", "en_US-lessac-high", 0,
|
|
f"{_deep},{_bass},{_echo_subtle}"),
|
|
# -- Deep + bass + chamber reverb
|
|
("ryan deep+bass+chamber", "en_US-ryan-high", 0,
|
|
f"{_deep},{_bass},{_echo_chamber}"),
|
|
("lessac deep+bass+chamber", "en_US-lessac-high", 0,
|
|
f"{_deep},{_bass},{_echo_chamber}"),
|
|
# -- Deep + heavy bass + cave reverb
|
|
("ryan deep+heavybass+cave", "en_US-ryan-high", 0,
|
|
f"{_deep},{_bass_heavy},{_echo_cave}"),
|
|
# -- Libritts best candidates with full sorcerer chain
|
|
("libritts #20 deep+bass+echo", "en_US-libritts_r-medium", 20,
|
|
f"{_deep},{_bass},{_echo_subtle}"),
|
|
("libritts #22 deep+bass+echo", "en_US-libritts_r-medium", 22,
|
|
f"{_deep},{_bass},{_echo_subtle}"),
|
|
("libritts #79 deep+bass+chamber", "en_US-libritts_r-medium", 79,
|
|
f"{_deep},{_bass},{_echo_chamber}"),
|
|
]
|
|
|
|
await bot.reply(message, f"Auditioning {len(samples)} voice samples...")
|
|
loop = asyncio.get_running_loop()
|
|
from pathlib import Path
|
|
|
|
for i, (label, voice, sid, fx) in enumerate(samples, 1):
|
|
await bot.send("0", f"[{i}/{len(samples)}] {label}")
|
|
await asyncio.sleep(1)
|
|
sample_wav = await loop.run_in_executor(
|
|
None, lambda v=voice, s=sid, f=fx: _fetch_tts_voice(
|
|
piper_url, phrase, voice=v, speaker_id=s,
|
|
length_scale=1.15, noise_scale=0.4, noise_w=0.5, fx=f,
|
|
),
|
|
)
|
|
if sample_wav is None:
|
|
await bot.send("0", " (failed)")
|
|
continue
|
|
try:
|
|
done = asyncio.Event()
|
|
await bot.stream_audio(sample_wav, volume=1.0, on_done=done)
|
|
await done.wait()
|
|
finally:
|
|
Path(sample_wav).unlink(missing_ok=True)
|
|
await asyncio.sleep(2)
|
|
|
|
await bot.send("0", "Audition complete.")
|
|
|
|
|
|
# -- Plugin lifecycle --------------------------------------------------------
|
|
|
|
|
|
async def on_connected(bot) -> None:
|
|
"""Re-register listener after reconnect; play TTS greeting on first connect."""
|
|
if not _is_mumble(bot):
|
|
return
|
|
ps = _ps(bot)
|
|
if ps["listen"] or ps["trigger"]:
|
|
_ensure_listener(bot)
|
|
_ensure_flush_task(bot)
|
|
|
|
# Greet via TTS on first connection only
|
|
greet = getattr(bot, "config", {}).get("mumble", {}).get("greet")
|
|
if greet and not ps.get("_greeted"):
|
|
ps["_greeted"] = True
|
|
ready = getattr(bot, "_is_audio_ready", None)
|
|
if ready:
|
|
for _ in range(20):
|
|
if ready():
|
|
break
|
|
await asyncio.sleep(0.5)
|
|
bot._spawn(_tts_play(bot, greet), name="voice-greet")
|