omnivoice

name

omnivoice

description

Local TTS, voice cloning, voice design, and video dubbing via the VoiceStudio MCP server (open-source ElevenLabs alternative; nothing leaves the machine, runs on MPS/CUDA/CPU). Use when: (1) generating speech from text in any of 646 languages, (2) cloning a voice from a 3-second reference clip, (3) designing a voice by gender/age/accent/pitch/style, (4) dubbing a video into another language, (5) listing voice profiles or personality presets, (6) producing narration where privacy, cost, or absent API keys matter, (7) non-English narration where Edge TTS/kokoro fall short, (8) batch audio for blog posts or content pipelines. Triggers: 'omnivoice', 'voice clone', 'clone this voice', 'tts', 'narrate', 'generate speech', 'voice synthesis', 'dub video', 'voice design', 'local tts', 'multilingual voice', 'narrate this post', 'elevenlabs alternative'.

VoiceStudio

The canonical cross-agent package lives at skills/omnivoice/SKILL.md. This Claude-specific package retains the MCP lifecycle helpers and references.

Overview

Generate audio locally via the VoiceStudio MCP server. Tools: generate_speech, list_voices, list_personalities, list_languages, check_health. Resources: voice://{id}, history://recent.

Prerequisites — Backend Must Be Running

The MCP tools all hit $OMNIVOICE_API_URL (default http://localhost:3900). If the backend is down, every tool returns a connection error. Install + boot:

git clone https://github.com/debpalash/VoiceStudio.git "$OMNIVOICE_HOME"
cd "$OMNIVOICE_HOME"
uv sync
VIRTUAL_ENV="$(pwd)/.venv" uv pip install 'mcp[cli]'

Then:

scripts/check-health.sh        # exit 0 if up
scripts/start-backend.sh       # boot in background (MPS/CUDA auto-detected)

First synthesis call lazy-downloads the k2-fsa/OmniVoice model (~2.4 GB) from HuggingFace — cached on subsequent boots.

Task Index — Pick the Right Tool

Task	Tool	Notes
Verify backend is up	check_health	Returns `{"status":"ok","device":"mps
Text → audio with a saved voice	generate_speech(text, profile_id)	Returns base64 WAV. profile_id="demo0001" is the bundled demo voice
Text → audio without a clone (voice design)	generate_speech(text, instruct="…")	Omit profile_id; pass an instruct like "warm middle-aged female narrator, calm pace"
Multilingual narration	generate_speech(text, language="es")	Any ISO 639 code or "Auto"
List existing voices	list_voices	Returns id, name, type, personality
List personality presets	list_personalities	Returns narrator / casual / news-anchor / etc. with their instruct strings
List supported languages	list_languages	646 total; returns 20 popular + the full count

For non-trivial decisions (which engine to use, when to pick VoiceStudio over kokoro / Edge TTS / ElevenLabs), see references/engines-comparison.md.

For MCP wiring details, backend lifecycle, troubleshooting, and a clean teardown, see references/mcp-setup.md.

Common Workflows

1. One-shot narration with the demo voice

# As called through the MCP client (your agent will do this for you):
result = generate_speech(
    text="Hello — this is VoiceStudio generating speech locally.",
    profile_id="demo0001",
    language="English",
    steps=16,                   # 8 = fast/draft · 16 = balanced · 32 = quality
)
# result is JSON with audio_id, generation_time_s, audio_duration_s, format, wav_base64

Benchmark: 4.2 s of audio in ~24 s server-side on Apple Silicon MPS at 16 diffusion steps.

2. Save the WAV to disk and play

Tool returns base64 PCM WAV (16-bit, mono, 24 kHz). Decode + write:

import base64, json
payload = json.loads(result_text)            # parse JSON the tool returns
open("out.wav","wb").write(base64.b64decode(payload["wav_base64"]))

On macOS: afplay out.wav. Convert to MP3 with ffmpeg -i out.wav -codec:a libmp3lame -b:a 128k out.mp3.

3. Voice clone — end-to-end recipe

Cloning needs a 3-10 second reference clip the model will use as a speaker embedding. The MCP server does NOT expose profile creation — it only reads existing profiles. Two paths to create one:

Path A — bundled helper (macOS, recommended for fresh clones):

scripts/record-reference.sh ~/Downloads/my-ref.wav 12 1
# args: output_path raw_duration_sec mic_index
# Default mic_index=1 (MacBook built-in); list devices via:
#   ffmpeg -f avfoundation -list_devices true -i ""

The script gives audible countdown + start/stop cues via macOS say + /System/Library/Sounds/Ping.aiff so the user knows when to speak (terminal stdout is buffered — text "speak now" prompts arrive too late). It records a longer raw window, then trims to ~10 seconds of speech via silenceremove + atrim, plays back for verification, and prints the next-step curl command.

Path B — manual:

# 1. Record (mono, 24 kHz native — matches model's internal rate)
ffmpeg -f avfoundation -i ":1" -t 12 -ac 1 -ar 24000 raw.wav

# 2. Trim leading silence + take first 10 sec of speech
ffmpeg -i raw.wav \
  -af "silenceremove=start_periods=1:start_silence=0.05:start_threshold=-40dB,atrim=end=10" \
  -ac 1 -ar 24000 ref.wav

# 3. Verify
ffmpeg -i ref.wav -af volumedetect -f null - 2>&1 | grep volume   # max should be > -20 dB
afplay ref.wav

POST to /profiles (multipart/form-data — required fields: name, ref_audio):

curl -X POST http://127.0.0.1:3900/profiles \
  -F "name=carlos-clone" \
  -F "ref_audio=@ref.wav" \
  -F "ref_text=The exact text spoken in the clip" \
  -F "language=English" \
  | python3 -m json.tool
# returns { "id": "abc12345", "name": "carlos-clone" }

Once created, pass profile_id to generate_speech (via MCP) or directly via POST /generate. Profiles persist in SQLite + reference-audio files at ~/Library/Application Support/OmniVoice/voices/<id>.<ext> (the backend preserves the uploaded extension — .wav if you uploaded a WAV, .mp3 if MP3, etc.). State persists across backend restarts.

Reference clip tips that materially affect quality:

Factor	Why it matters
Single speaker	Mixed speakers blur the embedding
Clean speech, no music/noise	Model embeds the noise too
Natural prosody (avoid pangrams)	Diffusion samples replicate prosody, not just timbre
3-10 sec is the sweet spot	< 3 s lacks information; > 10 s adds compute without quality gain
Match ref_text to what's spoken	Improves alignment, especially on noisy refs
language correct	Wrong language → cross-lingual transfer artifacts
Loudness peak ≥ -15 dB	Quiet refs work but normalize poorly

4. Voice design (no reference clip)

Skip profile_id; provide an instruct string describing the desired voice:

generate_speech(
    text="Welcome to the future of agentic systems.",
    instruct="warm middle-aged female narrator, calm authoritative pace, documentary style",
)

Get pre-made instructs via list_personalities and copy the one matching the brief (narrator, casual, news-anchor, etc.).

5. Video dubbing (web UI only)

The MCP server does not expose the dubbing endpoint. The full transcribe → translate → re-voice → mux pipeline lives behind the desktop UI (bun run desktop in $OMNIVOICE_HOME) and the /dub/* REST routes. When the user asks to dub a video, point them to the UI; surface this skill only for the synthesis primitives above.

When NOT to use VoiceStudio

• Fast English-only narration on weak hardware → kokoro-tts is ~10× smaller and 2× realtime on CPU (see references/engines-comparison.md)
• Lowest-friction one-off TTS → Edge TTS needs no install or backend
• Highest possible quality regardless of cost → ElevenLabs still wins on English narration polish; VoiceStudio ties or wins on multilingual + cloning
• Real-time streaming dictation → use the VoiceStudio desktop widget (⌘+⇧+Space), not the MCP server

Resources

• references/engines-comparison.md — Decision tree across VoiceStudio / kokoro / Voicebox / Edge TTS / ElevenLabs / cloud APIs
• references/mcp-setup.md — MCP wiring, backend lifecycle, env vars, troubleshooting
• scripts/check-health.sh — curl /health, exit 0/1
• scripts/start-backend.sh — Start uvicorn on 127.0.0.1:3900 with health probe
• scripts/stop-backend.sh — Clean shutdown via kill -TERM on the bound PID
• scripts/record-reference.sh — macOS-only: record + trim + verify a reference clip for cloning, with audible cues (say + system beeps) that bypass terminal output buffering

Backend Swagger / OpenAPI: http://127.0.0.1:3900/docs (when backend is up).

Upstream: github.com/debpalash/VoiceStudio. The app uses AGPL-3.0-only; optional engines and downloaded models retain their own licenses. See LICENSE-NOTICE.md in the repository.