Create README.md
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README.md
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---
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license: mit
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language:
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- ar
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- da
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- de
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- el
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- en
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- es
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- fi
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- fr
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- he
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- hi
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- it
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- ja
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- ko
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- ms
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- nl
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- no
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- pl
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- pt
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- ru
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- sv
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- sw
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- tr
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- zh
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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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- multilingual-tts
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library_name: 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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<span style="font-style: italic;white-space: pre-wrap">Made with ❤️ by</span>
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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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# 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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[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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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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# Sampling rate of the inputs to S3TokenizerV2
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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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llm_position_ids = np.cumsum(attention_mask, axis=1, dtype=np.int64) - 1
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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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position_ids=llm_position_ids,
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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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llm_position_ids = llm_position_ids[:, -1:] + 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.
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