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--- |
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license: mit |
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language: |
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- en |
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pipeline_tag: text-to-speech |
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tags: |
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- text-to-speech |
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- speech |
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- speech-generation |
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- voice-cloning |
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library_name: Chatterbox |
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base_model: |
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- ResembleAI/chatterbox |
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--- |
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<img width="800" alt="cb-big2" src="https://github.com/user-attachments/assets/bd8c5f03-e91d-4ee5-b680-57355da204d1" /> |
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<h1 style="font-size: 32px">Chatterbox TTS</h1> |
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<div style="display: flex; align-items: center; gap: 12px"> |
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<a href="https://resemble-ai.github.io/chatterbox_demopage/"> |
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<img src="https://img.shields.io/badge/listen-demo_samples-blue" alt="Listen to Demo Samples" /> |
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</a> |
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<a href="https://huggingface.co/spaces/ResembleAI/Chatterbox"> |
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<img src="https://huggingface.co/datasets/huggingface/badges/resolve/main/open-in-hf-spaces-sm.svg" alt="Open in HF Spaces" /> |
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</a> |
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<a href="https://podonos.com/resembleai/chatterbox"> |
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<img src="https://static-public.podonos.com/badges/insight-on-pdns-sm-dark.svg" alt="Insight on Podos" /> |
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</a> |
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</div> |
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<div style="display: flex; align-items: center; gap: 8px;"> |
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<img width="100" alt="resemble-logo-horizontal" src="https://github.com/user-attachments/assets/35cf756b-3506-4943-9c72-c05ddfa4e525" /> |
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</div> |
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**Chatterbox** [Resemble AI's](https://resemble.ai) production-grade open source TTS model. Chatterbox supports **English** out of the box. Licensed under MIT, Chatterbox has been benchmarked against leading closed-source systems like ElevenLabs, and is consistently preferred in side-by-side evaluations. |
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Whether you're working on memes, videos, games, or AI agents, Chatterbox brings your content to life. It's also the first open source TTS model to support **emotion exaggeration control**, a powerful feature that makes your voices stand out. |
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Chatterbox is provided in an exported ONNX format, enabling fast and portable inference with ONNX Runtime across platforms. |
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# Key Details |
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- SoTA zeroshot English TTS |
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- 0.5B Llama backbone |
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- Unique exaggeration/intensity control |
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- Ultra-stable with alignment-informed inference |
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- Trained on 0.5M hours of cleaned data |
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- Watermarked outputs (optional) |
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- Easy voice conversion script using onnxruntime |
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- [Outperforms ElevenLabs](https://podonos.com/resembleai/chatterbox) |
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# Tips |
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- **General Use (TTS and Voice Agents):** |
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- The default settings (`exaggeration=0.5`, `cfg=0.5`) work well for most prompts. |
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- **Expressive or Dramatic Speech:** |
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- Try increase `exaggeration` to around `0.7` or higher. |
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- Higher `exaggeration` tends to speed up speech; |
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# Usage |
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[Link to GitHub ONNX Export and Inference script](https://github.com/VladOS95-cyber/onnx_conversion_scripts/tree/main/chatterbox) |
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```python |
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# !pip install --upgrade onnxruntime==1.22.1 huggingface_hub==0.34.4 transformers==4.46.3 numpy==2.2.6 tqdm==4.67.1 librosa==0.11.0 soundfile==0.13.1 resemble-perth==1.0.1 |
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import onnxruntime |
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from huggingface_hub import hf_hub_download |
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from transformers import AutoTokenizer |
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import numpy as np |
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from tqdm import tqdm |
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import librosa |
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import soundfile as sf |
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S3GEN_SR = 24000 |
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START_SPEECH_TOKEN = 6561 |
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STOP_SPEECH_TOKEN = 6562 |
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class RepetitionPenaltyLogitsProcessor: |
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def __init__(self, penalty: float): |
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if not isinstance(penalty, float) or not (penalty > 0): |
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raise ValueError(f"`penalty` must be a strictly positive float, but is {penalty}") |
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self.penalty = penalty |
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def __call__(self, input_ids: np.ndarray, scores: np.ndarray) -> np.ndarray: |
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score = np.take_along_axis(scores, input_ids, axis=1) |
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score = np.where(score < 0, score * self.penalty, score / self.penalty) |
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scores_processed = scores.copy() |
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np.put_along_axis(scores_processed, input_ids, score, axis=1) |
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return scores_processed |
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def run_inference( |
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text="The Lord of the Rings is the greatest work of literature.", |
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target_voice_path=None, |
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max_new_tokens = 256, |
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exaggeration=0.5, |
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output_dir="converted", |
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output_file_name="output.wav", |
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apply_watermark=True, |
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): |
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model_id = "onnx-community/chatterbox-onnx" |
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if not target_voice_path: |
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target_voice_path = hf_hub_download(repo_id=model_id, filename="default_voice.wav", local_dir=output_dir) |
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## Load model |
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speech_encoder_path = hf_hub_download(repo_id=model_id, filename="speech_encoder.onnx", local_dir=output_dir, subfolder='onnx') |
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hf_hub_download(repo_id=model_id, filename="speech_encoder.onnx_data", local_dir=output_dir, subfolder='onnx') |
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embed_tokens_path = hf_hub_download(repo_id=model_id, filename="embed_tokens.onnx", local_dir=output_dir, subfolder='onnx') |
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hf_hub_download(repo_id=model_id, filename="embed_tokens.onnx_data", local_dir=output_dir, subfolder='onnx') |
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conditional_decoder_path = hf_hub_download(repo_id=model_id, filename="conditional_decoder.onnx", local_dir=output_dir, subfolder='onnx') |
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hf_hub_download(repo_id=model_id, filename="conditional_decoder.onnx_data", local_dir=output_dir, subfolder='onnx') |
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language_model_path = hf_hub_download(repo_id=model_id, filename="language_model.onnx", local_dir=output_dir, subfolder='onnx') |
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hf_hub_download(repo_id=model_id, filename="language_model.onnx_data", local_dir=output_dir, subfolder='onnx') |
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# # Start inferense sessions |
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speech_encoder_session = onnxruntime.InferenceSession(speech_encoder_path) |
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embed_tokens_session = onnxruntime.InferenceSession(embed_tokens_path) |
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llama_with_past_session = onnxruntime.InferenceSession(language_model_path) |
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cond_decoder_session = onnxruntime.InferenceSession(conditional_decoder_path) |
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def execute_text_to_audio_inference(text): |
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print("Start inference script...") |
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audio_values, _ = librosa.load(target_voice_path, sr=S3GEN_SR) |
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audio_values = audio_values[np.newaxis, :].astype(np.float32) |
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## Prepare input |
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tokenizer = AutoTokenizer.from_pretrained(model_id) |
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input_ids = tokenizer(text, return_tensors="np")["input_ids"].astype(np.int64) |
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position_ids = np.where( |
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input_ids >= START_SPEECH_TOKEN, |
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0, |
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np.arange(input_ids.shape[1])[np.newaxis, :] - 1 |
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) |
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ort_embed_tokens_inputs = { |
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"input_ids": input_ids, |
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"position_ids": position_ids, |
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"exaggeration": np.array([exaggeration], dtype=np.float32) |
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} |
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## Instantiate the logits processors. |
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repetition_penalty = 1.2 |
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repetition_penalty_processor = RepetitionPenaltyLogitsProcessor(penalty=repetition_penalty) |
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num_hidden_layers = 30 |
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num_key_value_heads = 16 |
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head_dim = 64 |
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generate_tokens = np.array([[START_SPEECH_TOKEN]], dtype=np.long) |
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# ---- Generation Loop using kv_cache ---- |
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for i in tqdm(range(max_new_tokens), desc="Sampling", dynamic_ncols=True): |
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inputs_embeds = embed_tokens_session.run(None, ort_embed_tokens_inputs)[0] |
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if i == 0: |
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ort_speech_encoder_input = { |
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"audio_values": audio_values, |
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} |
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cond_emb, prompt_token, ref_x_vector, prompt_feat = speech_encoder_session.run(None, ort_speech_encoder_input) |
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inputs_embeds = np.concatenate((cond_emb, inputs_embeds), axis=1) |
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## Prepare llm inputs |
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batch_size, seq_len, _ = inputs_embeds.shape |
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past_key_values = { |
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f"past_key_values.{layer}.{kv}": np.zeros([batch_size, num_key_value_heads, 0, head_dim], dtype=np.float32) |
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for layer in range(num_hidden_layers) |
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for kv in ("key", "value") |
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} |
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attention_mask = np.ones((batch_size, seq_len), dtype=np.int64) |
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logits, *present_key_values = llama_with_past_session.run(None, dict( |
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inputs_embeds=inputs_embeds, |
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attention_mask=attention_mask, |
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**past_key_values, |
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)) |
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logits = logits[:, -1, :] |
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next_token_logits = repetition_penalty_processor(generate_tokens, logits) |
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next_token = np.argmax(next_token_logits, axis=-1, keepdims=True).astype(np.int64) |
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generate_tokens = np.concatenate((generate_tokens, next_token), axis=-1) |
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if (next_token.flatten() == STOP_SPEECH_TOKEN).all(): |
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break |
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# Get embedding for the new token. |
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position_ids = np.full( |
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(input_ids.shape[0], 1), |
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i + 1, |
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dtype=np.int64, |
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) |
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ort_embed_tokens_inputs["input_ids"] = next_token |
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ort_embed_tokens_inputs["position_ids"] = position_ids |
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## Update values for next generation loop |
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attention_mask = np.concatenate([attention_mask, np.ones((batch_size, 1), dtype=np.int64)], axis=1) |
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for j, key in enumerate(past_key_values): |
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past_key_values[key] = present_key_values[j] |
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speech_tokens = generate_tokens[:, 1:-1] |
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speech_tokens = np.concatenate([prompt_token, speech_tokens], axis=1) |
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return speech_tokens, ref_x_vector, prompt_feat |
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speech_tokens, speaker_embeddings, speaker_features = execute_text_to_audio_inference(text) |
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cond_incoder_input = { |
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"speech_tokens": speech_tokens, |
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"speaker_embeddings": speaker_embeddings, |
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"speaker_features": speaker_features, |
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} |
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wav = cond_decoder_session.run(None, cond_incoder_input)[0] |
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wav = np.squeeze(wav, axis=0) |
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# Optional: Apply watermark |
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if apply_watermark: |
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import perth |
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watermarker = perth.PerthImplicitWatermarker() |
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wav = watermarker.apply_watermark(wav, sample_rate=S3GEN_SR) |
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sf.write(output_file_name, wav, S3GEN_SR) |
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print(f"{output_file_name} was successfully saved") |
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if __name__ == "__main__": |
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run_inference( |
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text="Ezreal and Jinx teamed up with Ahri, Yasuo, and Teemo to take down the enemy's Nexus in an epic late-game pentakill.", |
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exaggeration=0.5, |
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output_file_name="output.wav", |
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apply_watermark=False, |
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) |
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``` |
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# Acknowledgements |
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- [Xenova](https://huggingface.co/Xenova) |
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- [Vladislav Bronzov](https://github.com/VladOS95-cyber) |
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- [Resemble AI](https://github.com/resemble-ai/chatterbox) |
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# Built-in PerTh Watermarking for Responsible AI |
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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. |
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# Disclaimer |
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Don't use this model to do bad things. Prompts are sourced from freely available data on the internet. |