voxy

Facade for voice generation: speech synthesis in any voice, on any service, plus cloning and designing voices.

To install: pip install voxy (add 'voxy[elevenlabs]', 'voxy[aix]' or 'voxy[fal]' for those services)

import voxy

voxy.text_to_speech("Hello!", voice="ada").save(
    "hi.mp3"
)  # a named voice from the library
voxy.text_to_speech(
    "A quick draft.", voice="Daniel", backend="say"
)  # free local voice (macOS)
voxy.list_voices()  # our named voices
voxy.list_voices("elevenlabs")  # a service's voices

Backends: elevenlabs (cloning, voice design, TTS), say (macOS, offline), aix (OpenAI voices and other LiteLLM providers), fal (fal.ai models via falaw), voicestudio (local text-described voices, voxy[voicestudio]), csm (local Sesame CSM-1B). Register more with voxy.register_speech_model(name, factory).

Command line: python -m voxy voices [--backend X] and python -m voxy speak "text" --voice ada -o out.mp3.

Designing a voice from a description

model = voxy.get_speech_model("elevenlabs")
line = (
    "Once upon a time, in a village at the edge of a very old forest, there lived a "
    "baker who could hear the bread singing as it rose."
)  # 100-1000 characters
previews = model.design_voice_previews("a warm, slow, elderly storyteller", text=line)
for i, p in enumerate(previews):
    p.save(f"preview_{i}.mp3")  # listen, then pick one
voxy.design_from_description(
    "storyteller", "a warm, slow, elderly storyteller", preview=previews[1]
)

Features

• Voice cloning from audio samples
• High-quality speech synthesis
• Flexible input formats (file paths, bytes, streams, tensors)
• Audio cleanup utilities
• Automatic audio transcription (using Whisper)
• Plugin architecture for different speech models

Installation

Prerequisites

• Python 3.10+
• PyTorch and TorchAudio
• CUDA-compatible GPU (recommended)
• FFmpeg for audio processing

Install at least one model

Example: the CSM Model

The intention is to make voxy into a plugin-enabled facade, where you can chose your own engine (for voice cloning, voice synthesis, etc.). But for now, we just support, what seems to be the best open-source model out there (at the time of writing this): Sesame AI Lab's CSM model. It's just that, well, they did an amazing job at the model, but a terrible one (so far) for the python interface -- which is what inspired me to develop voxy in the first place.

Follow the instructions in the CSM repository to install the CSM model and its dependencies.

Try the csm spaces where you can try various functionalities of csm.

Quick Start

Basic Usage

from voxy import create_speech_model

# Create a speech model
model = create_speech_model(model_type="csm")

# Generate speech with default voice
audio = model.generate_speech(
    text="Hello, this is a test of the CSM speech model.", output_path="output.wav"
)

Voice Cloning

from voxy import create_speech_model

# Create a speech model
model = create_speech_model(model_type="csm")

# Clone a voice from an audio file
voice_profile = model.clone_voice(
    audio_input="sample_voice.wav",
    transcript="This is a sample of my voice for cloning purposes.",
)

# Generate speech with the cloned voice
audio = model.generate_speech(
    text="This is my cloned voice speaking. Isn't it amazing?",
    voice_profile=voice_profile,
    output_path="cloned_voice.wav",
)

Automatic Transcription

from voxy import create_speech_model

# Create a speech model
model = create_speech_model(model_type="csm")

# Clone a voice with automatic transcription
voice_profile = model.clone_voice(
    audio_input="sample_voice.wav",
    # No transcript provided, will use automatic transcription
)

# Generate speech with the cloned voice
audio = model.generate_speech(
    text="This voice was cloned using automatic transcription.",
    voice_profile=voice_profile,
    output_path="auto_transcribed_voice.wav",
)

Flexible Input Formats

The module supports various input formats:

# From file path
voice_profile1 = model.clone_voice(
    audio_input="sample_voice.wav", transcript="Text transcript."
)

# From bytes
with open("sample_voice.wav", "rb") as f:
    audio_bytes = f.read()
voice_profile2 = model.clone_voice(
    audio_input=audio_bytes, transcript="Text transcript."
)

# From file object
with open("sample_voice.wav", "rb") as f:
    voice_profile3 = model.clone_voice(audio_input=f, transcript="Text transcript.")

# From tensor
import torch
import torchaudio

audio_tensor, sample_rate = torchaudio.load("sample_voice.wav")
voice_profile4 = model.clone_voice(
    audio_input=audio_tensor, transcript="Text transcript."
)

ElevenLabs backend

Install with pip install 'voxy[elevenlabs]' and set ELEVEN_API_KEY or ELEVENLABS_API_KEY (checked in that order), or pass api_key=.

from voxy import create_speech_model

model = create_speech_model("elevenlabs")

# Instant voice clone: one sample or several (paths, bytes, file-likes, tensors), at most 25 files.
# ElevenLabs recommends 1-2 minutes of clean single-speaker audio in total (at most ~3).
profile = model.clone_voice(["clip1.wav", "clip2.m4a"], name="my-voice")
print(profile.segment)  # the ElevenLabs voice_id

audio = model.generate_speech(
    "Hello from my cloned voice.", profile, output_path="hello.wav"
)

# Later sessions: rebuild the profile from the id, no re-upload.
profile = model.voice_profile("<voice_id>")

generate_speech returns a tensor for pcm_*/wav_* output formats (default pcm_24000); synthesize_bytes returns the raw payload in any format (e.g. mp3_44100_128). delete_voice removes a clone from the account. ElevenLabs requires that you have the right, and the voice owner's consent, to clone a voice.

Live tests are opt-in: VOXY_LIVE_ELEVENLABS=1 runs a short synthesis; adding VOXY_LIVE_ELEVENLABS_SAMPLE=<audio you may clone> also clones, speaks and deletes.

Local voice cloning (no service, CPU works)

Five local models clone a voice from one clean clip and its transcript: voxcpm (VoxCPM2), qwen3tts (Qwen3-TTS), chatterbox, f5tts (F5-TTS) and xtts (XTTS-v2). Each runs in its own Python environment, built on first use with uv under $VOXY_DATA_DIR/envs/ (weights under $VOXY_DATA_DIR/models/), so their clashing torch/transformers pins never touch yours. Requirements: uv and ffmpeg on the PATH.

import voxy
from voxy.local_clone import reference_profile

model = voxy.create_speech_model("voxcpm")  # or qwen3tts, chatterbox, f5tts, xtts
voice = reference_profile("ref.wav", "Exactly what the clip says.")
model.synthesize(chapter_text, voice, cfg_value=2.0).save("chapter1.wav")
model.close()  # free the model's memory

Long text is split into paragraphs and sentence-packed chunks, each spoken separately and joined with controlled pauses (gap_ms=, paragraph_gap_ms=). Model parameters are keyword arguments of synthesize (see voxy.local_clone.LOCAL_CLONE_BACKENDS[name].params); model options (a variant, a fine-tuned checkpoint) are keyword arguments of create_speech_model. Threads are capped at $VOXY_NUM_THREADS (default 8). Licences differ: F5-TTS and XTTS-v2 weights are non-commercial (XTTS needs accept_licence=True); VoxCPM2, Qwen3-TTS and Chatterbox are Apache-2.0/MIT.

Languages: VoxCPM2 (30 languages), Qwen3-TTS (10, pass language="French" etc.), XTTS-v2 (17, language="fr") and Chatterbox with variant="multilingual" (23, language="fr") speak beyond English; Chatterbox's original and turbo variants and the F5-TTS base model are English (F5 also Chinese).

Which backend? docs/benchmarks.md compares the cloning backends on published benchmarks (French and English accuracy, voice likeness rated by people, arenas, licence, price, watermark), with tables, charts and a link to every source. The data is voxy/data/benchmarks.json; python -m voxy benchmarks docs/benchmarks.md regenerates the tables and charts after you edit it.

model = voxy.create_speech_model("chatterbox", variant="multilingual")
speech = model.synthesize("Bonjour à tous.", voice, language="fr")

python -m voxy setup voxcpm qwen3tts      # build environments ahead of time
python -m voxy prep ada tape1.m4a tape2.mp3 --language en
python -m voxy diagnose spec.json --out renders/first-listen --publish

Reference preparation (voxy.refprep.prepare_reference): converts any recording to mono WAV, denoises (denoise= afftdn, anlmdn, or deepfilternet, which lowers real room noise far more but builds its own torch environment on first use) and normalises loudness (ffmpeg), transcribes locally with Whisper (scribed's faster-whisper backend) with word timings, cuts clips of 3-12 s at sentence boundaries, and scores each clip (signal-to-noise, clipping, level, ASR confidence, speaking rate). Clips and manifest.jsonl go to the voice's samples store; best_references(manifest) picks reference clips and voxy.finetune.export_dataset writes training sets (LJSpeech or F5-TTS layout).

Diagnosis pages (voxy.diagnosis): render one text through a grid of models and parameters, then build one page with a player per variant, its parameters, generation time and real-time factor. Each variant has a stable ID (e.g. VX-7KQ2) derived from its configuration, not the text, so listeners can say "use VX-7KQ2" across pages and chapters; each listener can rate and annotate variants and copy their notes. A spec:

{"title": "First listen", "text_file": "excerpt.txt", "voice": "ada", "n_references": 1,
 "grid": [{"backend": "voxcpm", "params": {"cfg_value": [1.5, 2.0, 2.5]}},
          {"backend": "chatterbox", "params": {"exaggeration": [0.3, 0.6]}}]}

Cards are titled with readable names (the backend, then the settings that vary, e.g. "Chatterbox 0.3"). For a page you share with other listeners, optional spec keys help: references (named clips, e.g. a raw and a denoised one, each with a label), appendix (reference name → a note: those variants go in a collapsed section at the end), imported (clips rendered elsewhere, such as a cloud service, with their backend, label and audio_file), backend_notes, noindex, and notes_sink + page_url, which save every rating, note, name and clip play as they happen to an annoquest server ({"kind": "annoquest", "url": "/api/annoquest", "request": "<id>"}). The page says so, and annoquest-request.json is written next to it for registering on that server; add ?reader=<name> to the link you send each listener to tell them apart.

The voice library: clone once, reuse by name

voxy keeps voice data in stores (MutableMappings) under $VOXY_DATA_DIR (default ~/.local/share/voxy): voices/{name}.json records (with each backend's saved profile, e.g. the ElevenLabs voice_id), samples/{name}/ (audio a clone is made from), sources/{name}/ (raw media). Every library function takes its stores as arguments (voices=, samples=), so any MutableMapping (another folder, S3) can stand in.

from voxy import samples_store, clone_from_samples, load_voice, create_speech_model

samples = samples_store("ada")
samples["clip1.wav"] = open("clip1.wav", "rb").read()
clone_from_samples("ada")  # uploads the samples, saves the profile

model = create_speech_model("elevenlabs")
model.generate_speech("Hi!", load_voice("ada"), output_path="hi.wav")

Configuration

You can configure the default device by setting the DFLT_VOXY_DEVICE environment variable:

# Use CUDA
export DFLT_VOXY_DEVICE=cuda

# Use CPU
export DFLT_VOXY_DEVICE=cpu

# Use MPS (Apple Silicon)
export DFLT_VOXY_DEVICE=mps

Advanced Usage

Audio Cleanup

The module includes an audio cleanup function that normalizes volume and removes silence:

from voxy import cleanup_audio
import torchaudio

# Load audio
audio, sample_rate = torchaudio.load("noisy_audio.wav")

# Clean up audio
cleaned_audio = cleanup_audio(
    audio=audio,
    sample_rate=sample_rate,
    normalize=True,
    remove_silence=True,
    silence_threshold=0.02,
    min_silence_duration=0.2,
)

# Save cleaned audio
torchaudio.save("cleaned_audio.wav", cleaned_audio, sample_rate)

Disabling Audio Cleanup

You can disable audio cleanup when cloning a voice:

voice_profile = model.clone_voice(
    audio_input="sample_voice.wav",
    transcript="This is a sample of my voice.",
    cleanup_audio_fn=None,  # Disable audio cleanup
)

Custom Audio Cleanup

You can also provide your own audio cleanup function:

def my_custom_cleanup(audio, sample_rate, **kwargs):
    # Custom cleanup logic
    return processed_audio

voice_profile = model.