--- license: mit language: - en base_model: - ResembleAI/chatterbox-turbo pipeline_tag: text-to-speech tags: - text-to-speech - speech - speech-generation - voice-cloning ---  # Chatterbox TTS
### ⚡ Model Zoo
Choose the right model for your application.
| Model | Size | Languages | Key Features | Best For | 🤗 | Examples |
|:----------------------------------------------------------------------------------------------------------------| :--- | :--- |:--------------------------------------------------------|:---------------------------------------------|:--------------------------------------------------------------------------| :--- |
| **Chatterbox-Turbo** | **350M** | **English** | Paralinguistic Tags (`[laugh]`), Lower Compute and VRAM | Zero-shot voice agents, Production | [Demo](https://huggingface.co/spaces/ResembleAI/chatterbox-turbo-demo) | [Listen](https://resemble-ai.github.io/chatterbox_turbo_demopage/) |
| Chatterbox-Multilingual [(Language list)](#supported-languages) | 500M | 23+ | Zero-shot cloning, Multiple Languages | Global applications, Localization | [Demo](https://huggingface.co/spaces/ResembleAI/Chatterbox-Multilingual-TTS) | [Listen](https://resemble-ai.github.io/chatterbox_demopage/) |
| Chatterbox [(Tips and Tricks)](#original-chatterbox-tips) | 500M | English | CFG & Exaggeration tuning | General zero-shot TTS with creative controls | [Demo](https://huggingface.co/spaces/ResembleAI/Chatterbox) | [Listen](https://resemble-ai.github.io/chatterbox_demopage/) |
## Usage
##### Chatterbox-Turbo
```python
import onnxruntime
from transformers import AutoTokenizer
from huggingface_hub import hf_hub_download
import numpy as np
from tqdm import trange
import librosa
import soundfile as sf
MODEL_ID = "ResembleAI/chatterbox-turbo-ONNX"
SAMPLE_RATE = 24000
START_SPEECH_TOKEN = 6561
STOP_SPEECH_TOKEN = 6562
SILENCE_TOKEN = 4299
NUM_KV_HEADS = 16
HEAD_DIM = 64
class RepetitionPenaltyLogitsProcessor:
def __init__(self, penalty: float):
if not isinstance(penalty, float) or not (penalty > 0):
raise ValueError(f"`penalty` must be a strictly positive float, but is {penalty}")
self.penalty = penalty
def __call__(self, input_ids: np.ndarray, scores: np.ndarray) -> np.ndarray:
score = np.take_along_axis(scores, input_ids, axis=1)
score = np.where(score < 0, score * self.penalty, score / self.penalty)
scores_processed = scores.copy()
np.put_along_axis(scores_processed, input_ids, score, axis=1)
return scores_processed
def download_model(name: str, dtype: str = "fp32") -> str:
filename = f"{name}{'' if dtype == 'fp32' else '_quantized' if dtype == 'q8' else f'_{dtype}'}.onnx"
graph = hf_hub_download(MODEL_ID, subfolder="onnx", filename=filename) # Download graph
hf_hub_download(MODEL_ID, subfolder="onnx", filename=f"{filename}_data") # Download weights
return graph
# Download models
## dtype options: fp32, fp16, q8, q4, q4f16
conditional_decoder_path = download_model("conditional_decoder", dtype="fp32")
speech_encoder_path = download_model("speech_encoder", dtype="fp32")
embed_tokens_path = download_model("embed_tokens", dtype="fp32")
language_model_path = download_model("language_model", dtype="fp32")
# Create ONNX sessions
speech_encoder_session = onnxruntime.InferenceSession(speech_encoder_path)
embed_tokens_session = onnxruntime.InferenceSession(embed_tokens_path)
language_model_session = onnxruntime.InferenceSession(language_model_path)
cond_decoder_session = onnxruntime.InferenceSession(conditional_decoder_path)
# Generation parameters
text = "Oh, that's hilarious! [chuckle] Um anyway, how are you doing today?"
target_voice_path = "path/to/voice.wav"
output_file_name = "output.wav"
max_new_tokens = 1024
repetition_penalty = 1.2
apply_watermark = False
# Prepare audio input
audio_values, _ = librosa.load(target_voice_path, sr=SAMPLE_RATE)
audio_values = audio_values[np.newaxis, :].astype(np.float32)
# Prepare text input
tokenizer = AutoTokenizer.from_pretrained(MODEL_ID)
input_ids = tokenizer(text, return_tensors="np")["input_ids"].astype(np.int64)
# Generation loop
repetition_penalty_processor = RepetitionPenaltyLogitsProcessor(penalty=repetition_penalty)
generate_tokens = np.array([[START_SPEECH_TOKEN]], dtype=np.int64)
for i in trange(max_new_tokens, desc="Sampling", dynamic_ncols=True):
inputs_embeds = embed_tokens_session.run(None, {"input_ids": input_ids})[0]
if i == 0:
ort_speech_encoder_input = {"audio_values": audio_values}
cond_emb, prompt_token, speaker_embeddings, speaker_features = speech_encoder_session.run(None, ort_speech_encoder_input)
inputs_embeds = np.concatenate((cond_emb, inputs_embeds), axis=1)
# Initialize cache and LLM inputs
batch_size, seq_len, _ = inputs_embeds.shape
past_key_values = {
i.name: np.zeros([batch_size, NUM_KV_HEADS, 0, HEAD_DIM], dtype=np.float16 if i.type == 'tensor(float16)' else np.float32)
for i in language_model_session.get_inputs()
if "past_key_values" in i.name
}
attention_mask = np.ones((batch_size, seq_len), dtype=np.int64)
position_ids = np.arange(seq_len, dtype=np.int64).reshape(1, -1).repeat(batch_size, axis=0)
logits, *present_key_values = language_model_session.run(None, dict(
inputs_embeds=inputs_embeds,
attention_mask=attention_mask,
position_ids=position_ids,
**past_key_values,
))
logits = logits[:, -1, :]
next_token_logits = repetition_penalty_processor(generate_tokens, logits)
input_ids = np.argmax(next_token_logits, axis=-1, keepdims=True).astype(np.int64)
generate_tokens = np.concatenate((generate_tokens, input_ids), axis=-1)
if (input_ids.flatten() == STOP_SPEECH_TOKEN).all():
break
# Update values for next generation loop
attention_mask = np.concatenate([attention_mask, np.ones((batch_size, 1), dtype=np.int64)], axis=1)
position_ids = position_ids[:, -1:] + 1
for j, key in enumerate(past_key_values):
past_key_values[key] = present_key_values[j]
# Decode audio
speech_tokens = generate_tokens[:, 1:-1]
silence_tokens = np.full((speech_tokens.shape[0], 3), SILENCE_TOKEN, dtype=np.int64) # Add silence at the end
speech_tokens = np.concatenate([prompt_token, speech_tokens, silence_tokens], axis=1)
wav = cond_decoder_session.run(None, dict(
speech_tokens=speech_tokens,
speaker_embeddings=speaker_embeddings,
speaker_features=speaker_features,
))[0].squeeze(axis=0)
# Optional: Apply watermark
if apply_watermark:
import perth
watermarker = perth.PerthImplicitWatermarker()
wav = watermarker.apply_watermark(wav, sample_rate=SAMPLE_RATE)
sf.write(output_file_name, wav, SAMPLE_RATE)
```
## Built-in PerTh Watermarking for Responsible AI
Every audio file generated by Chatterbox includes [Resemble AI's Perth (Perceptual Threshold) Watermarker](https://github.com/resemble-ai/perth) - imperceptible neural watermarks that survive MP3 compression, audio editing, and common manipulations while maintaining nearly 100% detection accuracy.
## Watermark extraction
You can look for the watermark using the following script.
```python
import perth
import librosa
AUDIO_PATH = "YOUR_FILE.wav"
# Load the watermarked audio
watermarked_audio, sr = librosa.load(AUDIO_PATH, sr=None)
# Initialize watermarker (same as used for embedding)
watermarker = perth.PerthImplicitWatermarker()
# Extract watermark
watermark = watermarker.get_watermark(watermarked_audio, sample_rate=sr)
print(f"Extracted watermark: {watermark}")
# Output: 0.0 (no watermark) or 1.0 (watermarked)
```
## Official Discord
👋 Join us on [Discord](https://discord.gg/rJq9cRJBJ6) and let's build something awesome together!
## Acknowledgements
- [Cosyvoice](https://github.com/FunAudioLLM/CosyVoice)
- [Real-Time-Voice-Cloning](https://github.com/CorentinJ/Real-Time-Voice-Cloning)
- [HiFT-GAN](https://github.com/yl4579/HiFTNet)
- [Llama 3](https://github.com/meta-llama/llama3)
- [S3Tokenizer](https://github.com/xingchensong/S3Tokenizer)
## Citation
If you find this model useful, please consider citing.
```
@misc{chatterboxtts2025,
author = {{Resemble AI}},
title = {{Chatterbox-TTS}},
year = {2025},
howpublished = {\url{https://github.com/resemble-ai/chatterbox}},
note = {GitHub repository}
}
```
## Disclaimer
Don't use this model to do bad things. Prompts are sourced from freely available data on the internet.