``` ├── .github/ ├── workflows/ ├── ci.yaml ├── .gitignore ├── .python-version ├── LICENSE ├── README.md ├── app.py ├── cli.py ├── dia/ ├── __init__.py ├── audio.py ├── config.py ├── layers.py ├── model.py ├── state.py ├── static/ ├── images/ ├── banner.png ├── docker/ ├── Dockerfile.cpu ├── Dockerfile.gpu ├── example/ ├── simple.py ├── voice_clone.py ├── example_prompt.mp3 ├── pyproject.toml ├── uv.lock ``` ## /.github/workflows/ci.yaml ```yaml path="/.github/workflows/ci.yaml" name: Continuous Integration on: pull_request: branches: - main jobs: lint_and_format: runs-on: ubuntu-latest name: Lint and Format steps: - uses: actions/checkout@v4 - uses: astral-sh/ruff-action@v3 with: version: latest - name: Check Lint using Ruff run: ruff check - name: Check Format using Ruff run: ruff format --check --diff ``` ## /.gitignore ```gitignore path="/.gitignore" # Python-generated files __pycache__/ *.py[oc] build/ dist/ wheels/ *.egg-info # Virtual environments .venv .gradio **/*.pth **/*.mp3 !example_prompt.mp3 **/*.txt .ruff_cache .ipynb_checkpoints config.json ``` ## /.python-version ```python-version path="/.python-version" 3.10 ``` ## /LICENSE ``` path="/LICENSE" Apache License Version 2.0, January 2004 http://www.apache.org/licenses/ TERMS AND CONDITIONS FOR USE, REPRODUCTION, AND DISTRIBUTION 1. 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Static Badge LICENSE

Dataset on HuggingFace Space on HuggingFace

Dia is a 1.6B parameter text to speech model created by Nari Labs. Dia **directly generates highly realistic dialogue from a transcript**. You can condition the output on audio, enabling emotion and tone control. The model can also produce nonverbal communications like laughter, coughing, clearing throat, etc. To accelerate research, we are providing access to pretrained model checkpoints and inference code. The model weights are hosted on [Hugging Face](https://huggingface.co/nari-labs/Dia-1.6B). The model only supports English generation at the moment. We also provide a [demo page](https://yummy-fir-7a4.notion.site/dia) comparing our model to [ElevenLabs Studio](https://elevenlabs.io/studio) and [Sesame CSM-1B](https://github.com/SesameAILabs/csm). - (Update) We have a ZeroGPU Space running! Try it now [here](https://huggingface.co/spaces/nari-labs/Dia-1.6B). Thanks to the HF team for the support :) - Join our [discord server](https://discord.gg/yBrqQ9Dd) for community support and access to new features. - Play with a larger version of Dia: generate fun conversations, remix content, and share with friends. 🔮 Join the [waitlist](https://tally.so/r/meokbo) for early access. ## ⚡️ Quickstart ### Install via pip ```bash # Install directly from GitHub pip install git+https://github.com/nari-labs/dia.git ``` ### Run the Gradio UI This will open a Gradio UI that you can work on. ```bash git clone https://github.com/nari-labs/dia.git cd dia && uv run app.py ``` or if you do not have `uv` pre-installed: ```bash git clone https://github.com/nari-labs/dia.git cd dia python -m venv .venv source .venv/bin/activate pip install -e . python app.py ``` Note that the model was not fine-tuned on a specific voice. Hence, you will get different voices every time you run the model. You can keep speaker consistency by either adding an audio prompt (a guide coming VERY soon - try it with the second example on Gradio for now), or fixing the seed. ## Features - Generate dialogue via `[S1]` and `[S2]` tag - Generate non-verbal like `(laughs)`, `(coughs)`, etc. - Below verbal tags will be recognized, but might result in unexpected output. - `(laughs), (clears throat), (sighs), (gasps), (coughs), (singing), (sings), (mumbles), (beep), (groans), (sniffs), (claps), (screams), (inhales), (exhales), (applause), (burps), (humming), (sneezes), (chuckle), (whistles)` - Voice cloning. See [`example/voice_clone.py`](example/voice_clone.py) for more information. - In the Hugging Face space, you can upload the audio you want to clone and place its transcript before your script. Make sure the transcript follows the required format. The model will then output only the content of your script. ## ⚙️ Usage ### As a Python Library ```python from dia.model import Dia model = Dia.from_pretrained("nari-labs/Dia-1.6B", compute_dtype="float16") text = "[S1] Dia is an open weights text to dialogue model. [S2] You get full control over scripts and voices. [S1] Wow. Amazing. (laughs) [S2] Try it now on Git hub or Hugging Face." output = model.generate(text, use_torch_compile=True, verbose=True) model.save_audio("simple.mp3", output) ``` A pypi package and a working CLI tool will be available soon. ## 💻 Hardware and Inference Speed Dia has been tested on only GPUs (pytorch 2.0+, CUDA 12.6). CPU support is to be added soon. The initial run will take longer as the Descript Audio Codec also needs to be downloaded. These are the speed we benchmarked in RTX 4090. | precision | realtime factor w/ compile | realtime factor w/o compile | VRAM | |:-:|:-:|:-:|:-:| | `bfloat16` | x2.1 | x1.5 | ~10GB | | `float16` | x2.2 | x1.3 | ~10GB | | `float32` | x1 | x0.9 | ~13GB | We will be adding a quantized version in the future. If you don't have hardware available or if you want to play with bigger versions of our models, join the waitlist [here](https://tally.so/r/meokbo). ## 🪪 License This project is licensed under the Apache License 2.0 - see the [LICENSE](LICENSE) file for details. ## ⚠️ Disclaimer This project offers a high-fidelity speech generation model intended for research and educational use. The following uses are **strictly forbidden**: - **Identity Misuse**: Do not produce audio resembling real individuals without permission. - **Deceptive Content**: Do not use this model to generate misleading content (e.g. fake news) - **Illegal or Malicious Use**: Do not use this model for activities that are illegal or intended to cause harm. By using this model, you agree to uphold relevant legal standards and ethical responsibilities. We **are not responsible** for any misuse and firmly oppose any unethical usage of this technology. ## 🔭 TODO / Future Work - Docker support for ARM architecture and MacOS. - Optimize inference speed. - Add quantization for memory efficiency. ## 🤝 Contributing We are a tiny team of 1 full-time and 1 part-time research-engineers. We are extra-welcome to any contributions! Join our [Discord Server](https://discord.gg/yBrqQ9Dd) for discussions. ## 🤗 Acknowledgements - We thank the [Google TPU Research Cloud program](https://sites.research.google/trc/about/) for providing computation resources. - Our work was heavily inspired by [SoundStorm](https://arxiv.org/abs/2305.09636), [Parakeet](https://jordandarefsky.com/blog/2024/parakeet/), and [Descript Audio Codec](https://github.com/descriptinc/descript-audio-codec). - Hugging Face for providing the ZeroGPU Grant. - "Nari" is a pure Korean word for lily. - We thank Jason Y. for providing help with data filtering. ## ⭐ Star History Star History Chart ## /app.py ```py path="/app.py" import argparse import tempfile import time from pathlib import Path from typing import Optional, Tuple import gradio as gr import numpy as np import soundfile as sf import torch from dia.model import Dia # --- Global Setup --- parser = argparse.ArgumentParser(description="Gradio interface for Nari TTS") parser.add_argument("--device", type=str, default=None, help="Force device (e.g., 'cuda', 'mps', 'cpu')") parser.add_argument("--share", action="store_true", help="Enable Gradio sharing") args = parser.parse_args() # Determine device if args.device: device = torch.device(args.device) elif torch.cuda.is_available(): device = torch.device("cuda") # Simplified MPS check for broader compatibility elif hasattr(torch.backends, "mps") and torch.backends.mps.is_available(): # Basic check is usually sufficient, detailed check can be problematic device = torch.device("mps") else: device = torch.device("cpu") print(f"Using device: {device}") # Load Nari model and config print("Loading Nari model...") try: # Use the function from inference.py model = Dia.from_pretrained("nari-labs/Dia-1.6B", compute_dtype="float16", device=device) except Exception as e: print(f"Error loading Nari model: {e}") raise def run_inference( text_input: str, audio_prompt_input: Optional[Tuple[int, np.ndarray]], max_new_tokens: int, cfg_scale: float, temperature: float, top_p: float, cfg_filter_top_k: int, speed_factor: float, ): """ Runs Nari inference using the globally loaded model and provided inputs. Uses temporary files for text and audio prompt compatibility with inference.generate. """ global model, device # Access global model, config, device if not text_input or text_input.isspace(): raise gr.Error("Text input cannot be empty.") temp_txt_file_path = None temp_audio_prompt_path = None output_audio = (44100, np.zeros(1, dtype=np.float32)) try: prompt_path_for_generate = None if audio_prompt_input is not None: sr, audio_data = audio_prompt_input # Check if audio_data is valid if audio_data is None or audio_data.size == 0 or audio_data.max() == 0: # Check for silence/empty gr.Warning("Audio prompt seems empty or silent, ignoring prompt.") else: # Save prompt audio to a temporary WAV file with tempfile.NamedTemporaryFile(mode="wb", suffix=".wav", delete=False) as f_audio: temp_audio_prompt_path = f_audio.name # Store path for cleanup # Basic audio preprocessing for consistency # Convert to float32 in [-1, 1] range if integer type if np.issubdtype(audio_data.dtype, np.integer): max_val = np.iinfo(audio_data.dtype).max audio_data = audio_data.astype(np.float32) / max_val elif not np.issubdtype(audio_data.dtype, np.floating): gr.Warning(f"Unsupported audio prompt dtype {audio_data.dtype}, attempting conversion.") # Attempt conversion, might fail for complex types try: audio_data = audio_data.astype(np.float32) except Exception as conv_e: raise gr.Error(f"Failed to convert audio prompt to float32: {conv_e}") # Ensure mono (average channels if stereo) if audio_data.ndim > 1: if audio_data.shape[0] == 2: # Assume (2, N) audio_data = np.mean(audio_data, axis=0) elif audio_data.shape[1] == 2: # Assume (N, 2) audio_data = np.mean(audio_data, axis=1) else: gr.Warning( f"Audio prompt has unexpected shape {audio_data.shape}, taking first channel/axis." ) audio_data = ( audio_data[0] if audio_data.shape[0] < audio_data.shape[1] else audio_data[:, 0] ) audio_data = np.ascontiguousarray(audio_data) # Ensure contiguous after slicing/mean # Write using soundfile try: sf.write( temp_audio_prompt_path, audio_data, sr, subtype="FLOAT" ) # Explicitly use FLOAT subtype prompt_path_for_generate = temp_audio_prompt_path print(f"Created temporary audio prompt file: {temp_audio_prompt_path} (orig sr: {sr})") except Exception as write_e: print(f"Error writing temporary audio file: {write_e}") raise gr.Error(f"Failed to save audio prompt: {write_e}") # 3. Run Generation start_time = time.time() # Use torch.inference_mode() context manager for the generation call with torch.inference_mode(): output_audio_np = model.generate( text_input, max_tokens=max_new_tokens, cfg_scale=cfg_scale, temperature=temperature, top_p=top_p, cfg_filter_top_k=cfg_filter_top_k, # Pass the value here use_torch_compile=False, # Keep False for Gradio stability audio_prompt=prompt_path_for_generate, ) end_time = time.time() print(f"Generation finished in {end_time - start_time:.2f} seconds.") # 4. Convert Codes to Audio if output_audio_np is not None: # Get sample rate from the loaded DAC model output_sr = 44100 # --- Slow down audio --- original_len = len(output_audio_np) # Ensure speed_factor is positive and not excessively small/large to avoid issues speed_factor = max(0.1, min(speed_factor, 5.0)) target_len = int(original_len / speed_factor) # Target length based on speed_factor if target_len != original_len and target_len > 0: # Only interpolate if length changes and is valid x_original = np.arange(original_len) x_resampled = np.linspace(0, original_len - 1, target_len) resampled_audio_np = np.interp(x_resampled, x_original, output_audio_np) output_audio = ( output_sr, resampled_audio_np.astype(np.float32), ) # Use resampled audio print(f"Resampled audio from {original_len} to {target_len} samples for {speed_factor:.2f}x speed.") else: output_audio = ( output_sr, output_audio_np, ) # Keep original if calculation fails or no change print(f"Skipping audio speed adjustment (factor: {speed_factor:.2f}).") # --- End slowdown --- print(f"Audio conversion successful. Final shape: {output_audio[1].shape}, Sample Rate: {output_sr}") # Explicitly convert to int16 to prevent Gradio warning if output_audio[1].dtype == np.float32 or output_audio[1].dtype == np.float64: audio_for_gradio = np.clip(output_audio[1], -1.0, 1.0) audio_for_gradio = (audio_for_gradio * 32767).astype(np.int16) output_audio = (output_sr, audio_for_gradio) print("Converted audio to int16 for Gradio output.") else: print("\nGeneration finished, but no valid tokens were produced.") # Return default silence gr.Warning("Generation produced no output.") except Exception as e: print(f"Error during inference: {e}") import traceback traceback.print_exc() # Re-raise as Gradio error to display nicely in the UI raise gr.Error(f"Inference failed: {e}") finally: # 5. Cleanup Temporary Files defensively if temp_txt_file_path and Path(temp_txt_file_path).exists(): try: Path(temp_txt_file_path).unlink() print(f"Deleted temporary text file: {temp_txt_file_path}") except OSError as e: print(f"Warning: Error deleting temporary text file {temp_txt_file_path}: {e}") if temp_audio_prompt_path and Path(temp_audio_prompt_path).exists(): try: Path(temp_audio_prompt_path).unlink() print(f"Deleted temporary audio prompt file: {temp_audio_prompt_path}") except OSError as e: print(f"Warning: Error deleting temporary audio prompt file {temp_audio_prompt_path}: {e}") return output_audio # --- Create Gradio Interface --- css = """ #col-container {max-width: 90%; margin-left: auto; margin-right: auto;} """ # Attempt to load default text from example.txt default_text = "[S1] Dia is an open weights text to dialogue model. \n[S2] You get full control over scripts and voices. \n[S1] Wow. Amazing. (laughs) \n[S2] Try it now on Git hub or Hugging Face." example_txt_path = Path("./example.txt") if example_txt_path.exists(): try: default_text = example_txt_path.read_text(encoding="utf-8").strip() if not default_text: # Handle empty example file default_text = "Example text file was empty." except Exception as e: print(f"Warning: Could not read example.txt: {e}") # Build Gradio UI with gr.Blocks(css=css) as demo: gr.Markdown("# Nari Text-to-Speech Synthesis") with gr.Row(equal_height=False): with gr.Column(scale=1): text_input = gr.Textbox( label="Input Text", placeholder="Enter text here...", value=default_text, lines=5, # Increased lines ) audio_prompt_input = gr.Audio( label="Audio Prompt (Optional)", show_label=True, sources=["upload", "microphone"], type="numpy", ) with gr.Accordion("Generation Parameters", open=False): max_new_tokens = gr.Slider( label="Max New Tokens (Audio Length)", minimum=860, maximum=3072, value=model.config.data.audio_length, # Use config default if available, else fallback step=50, info="Controls the maximum length of the generated audio (more tokens = longer audio).", ) cfg_scale = gr.Slider( label="CFG Scale (Guidance Strength)", minimum=1.0, maximum=5.0, value=3.0, # Default from inference.py step=0.1, info="Higher values increase adherence to the text prompt.", ) temperature = gr.Slider( label="Temperature (Randomness)", minimum=1.0, maximum=1.5, value=1.3, # Default from inference.py step=0.05, info="Lower values make the output more deterministic, higher values increase randomness.", ) top_p = gr.Slider( label="Top P (Nucleus Sampling)", minimum=0.80, maximum=1.0, value=0.95, # Default from inference.py step=0.01, info="Filters vocabulary to the most likely tokens cumulatively reaching probability P.", ) cfg_filter_top_k = gr.Slider( label="CFG Filter Top K", minimum=15, maximum=50, value=30, step=1, info="Top k filter for CFG guidance.", ) speed_factor_slider = gr.Slider( label="Speed Factor", minimum=0.8, maximum=1.0, value=0.94, step=0.02, info="Adjusts the speed of the generated audio (1.0 = original speed).", ) run_button = gr.Button("Generate Audio", variant="primary") with gr.Column(scale=1): audio_output = gr.Audio( label="Generated Audio", type="numpy", autoplay=False, ) # Link button click to function run_button.click( fn=run_inference, inputs=[ text_input, audio_prompt_input, max_new_tokens, cfg_scale, temperature, top_p, cfg_filter_top_k, speed_factor_slider, ], outputs=[audio_output], # Add status_output here if using it api_name="generate_audio", ) # Add examples (ensure the prompt path is correct or remove it if example file doesn't exist) example_prompt_path = "./example_prompt.mp3" # Adjust if needed examples_list = [ [ "[S1] Oh fire! Oh my goodness! What's the procedure? What to we do people? The smoke could be coming through an air duct! \n[S2] Oh my god! Okay.. it's happening. Everybody stay calm! \n[S1] What's the procedure... \n[S2] Everybody stay fucking calm!!!... Everybody fucking calm down!!!!! \n[S1] No! No! If you touch the handle, if its hot there might be a fire down the hallway! ", None, 3072, 3.0, 1.3, 0.95, 35, 0.94, ], [ "[S1] Open weights text to dialogue model. \n[S2] You get full control over scripts and voices. \n[S1] I'm biased, but I think we clearly won. \n[S2] Hard to disagree. (laughs) \n[S1] Thanks for listening to this demo. \n[S2] Try it now on Git hub and Hugging Face. \n[S1] If you liked our model, please give us a star and share to your friends. \n[S2] This was Nari Labs.", example_prompt_path if Path(example_prompt_path).exists() else None, 3072, 3.0, 1.3, 0.95, 35, 0.94, ], ] if examples_list: gr.Examples( examples=examples_list, inputs=[ text_input, audio_prompt_input, max_new_tokens, cfg_scale, temperature, top_p, cfg_filter_top_k, speed_factor_slider, ], outputs=[audio_output], fn=run_inference, cache_examples=False, label="Examples (Click to Run)", ) else: gr.Markdown("_(No examples configured or example prompt file missing)_") # --- Launch the App --- if __name__ == "__main__": print("Launching Gradio interface...") # set `GRADIO_SERVER_NAME`, `GRADIO_SERVER_PORT` env vars to override default values # use `GRADIO_SERVER_NAME=0.0.0.0` for Docker demo.launch(share=args.share) ``` ## /cli.py ```py path="/cli.py" import argparse import os import random import numpy as np import soundfile as sf import torch from dia.model import Dia def set_seed(seed: int): """Sets the random seed for reproducibility.""" random.seed(seed) np.random.seed(seed) torch.manual_seed(seed) if torch.cuda.is_available(): torch.cuda.manual_seed(seed) torch.cuda.manual_seed_all(seed) # Ensure deterministic behavior for cuDNN (if used) torch.backends.cudnn.deterministic = True torch.backends.cudnn.benchmark = False def main(): parser = argparse.ArgumentParser(description="Generate audio using the Dia model.") parser.add_argument("text", type=str, help="Input text for speech generation.") parser.add_argument( "--output", type=str, required=True, help="Path to save the generated audio file (e.g., output.wav)." ) parser.add_argument( "--repo-id", type=str, default="nari-labs/Dia-1.6B", help="Hugging Face repository ID (e.g., nari-labs/Dia-1.6B).", ) parser.add_argument( "--local-paths", action="store_true", help="Load model from local config and checkpoint files." ) parser.add_argument( "--config", type=str, help="Path to local config.json file (required if --local-paths is set)." ) parser.add_argument( "--checkpoint", type=str, help="Path to local model checkpoint .pth file (required if --local-paths is set)." ) parser.add_argument( "--audio-prompt", type=str, default=None, help="Path to an optional audio prompt WAV file for voice cloning." ) gen_group = parser.add_argument_group("Generation Parameters") gen_group.add_argument( "--max-tokens", type=int, default=None, help="Maximum number of audio tokens to generate (defaults to config value).", ) gen_group.add_argument( "--cfg-scale", type=float, default=3.0, help="Classifier-Free Guidance scale (default: 3.0)." ) gen_group.add_argument( "--temperature", type=float, default=1.3, help="Sampling temperature (higher is more random, default: 0.7)." ) gen_group.add_argument("--top-p", type=float, default=0.95, help="Nucleus sampling probability (default: 0.95).") infra_group = parser.add_argument_group("Infrastructure") infra_group.add_argument("--seed", type=int, default=None, help="Random seed for reproducibility.") infra_group.add_argument( "--device", type=str, default="cuda" if torch.cuda.is_available() else "cpu", help="Device to run inference on (e.g., 'cuda', 'cpu', default: auto).", ) args = parser.parse_args() # Validation for local paths if args.local_paths: if not args.config: parser.error("--config is required when --local-paths is set.") if not args.checkpoint: parser.error("--checkpoint is required when --local-paths is set.") if not os.path.exists(args.config): parser.error(f"Config file not found: {args.config}") if not os.path.exists(args.checkpoint): parser.error(f"Checkpoint file not found: {args.checkpoint}") # Set seed if provided if args.seed is not None: set_seed(args.seed) print(f"Using random seed: {args.seed}") # Determine device device = torch.device(args.device) print(f"Using device: {device}") # Load model print("Loading model...") if args.local_paths: print(f"Loading from local paths: config='{args.config}', checkpoint='{args.checkpoint}'") try: model = Dia.from_local(args.config, args.checkpoint, device=device) except Exception as e: print(f"Error loading local model: {e}") exit(1) else: print(f"Loading from Hugging Face Hub: repo_id='{args.repo_id}'") try: model = Dia.from_pretrained(args.repo_id, device=device) except Exception as e: print(f"Error loading model from Hub: {e}") exit(1) print("Model loaded.") # Generate audio print("Generating audio...") try: sample_rate = 44100 # Default assumption output_audio = model.generate( text=args.text, audio_prompt=args.audio_prompt, max_tokens=args.max_tokens, cfg_scale=args.cfg_scale, temperature=args.temperature, top_p=args.top_p, ) print("Audio generation complete.") print(f"Saving audio to {args.output}...") os.makedirs(os.path.dirname(args.output) or ".", exist_ok=True) sf.write(args.output, output_audio, sample_rate) print(f"Audio successfully saved to {args.output}") except Exception as e: print(f"Error during audio generation or saving: {e}") exit(1) if __name__ == "__main__": main() ``` ## /dia/__init__.py ```py path="/dia/__init__.py" from .model import Dia __all__ = [ "Dia", ] ``` ## /dia/audio.py ```py path="/dia/audio.py" import typing as tp import torch def build_delay_indices(B: int, T: int, C: int, delay_pattern: tp.List[int]) -> tp.Tuple[torch.Tensor, torch.Tensor]: """ Precompute (t_idx_BxTxC, indices_BTCx3) so that out[t, c] = in[t - delay[c], c]. Negative t_idx => BOS; t_idx >= T => PAD. """ delay_arr = torch.tensor(delay_pattern, dtype=torch.int32) t_idx_BxT = torch.broadcast_to( torch.arange(T, dtype=torch.int32)[None, :], [B, T], ) t_idx_BxTx1 = t_idx_BxT[..., None] t_idx_BxTxC = t_idx_BxTx1 - delay_arr.view(1, 1, C) b_idx_BxTxC = torch.broadcast_to( torch.arange(B, dtype=torch.int32).view(B, 1, 1), [B, T, C], ) c_idx_BxTxC = torch.broadcast_to( torch.arange(C, dtype=torch.int32).view(1, 1, C), [B, T, C], ) # We must clamp time indices to [0..T-1] so gather_nd equivalent won't fail t_clamped_BxTxC = torch.clamp(t_idx_BxTxC, 0, T - 1) indices_BTCx3 = torch.stack( [ b_idx_BxTxC.reshape(-1), t_clamped_BxTxC.reshape(-1), c_idx_BxTxC.reshape(-1), ], dim=1, ).long() # Ensure indices are long type for indexing return t_idx_BxTxC, indices_BTCx3 def apply_audio_delay( audio_BxTxC: torch.Tensor, pad_value: int, bos_value: int, precomp: tp.Tuple[torch.Tensor, torch.Tensor], ) -> torch.Tensor: """ Applies the delay pattern to batched audio tokens using precomputed indices, inserting BOS where t_idx < 0 and PAD where t_idx >= T. Args: audio_BxTxC: [B, T, C] int16 audio tokens (or int32/float) pad_value: the padding token bos_value: the BOS token precomp: (t_idx_BxTxC, indices_BTCx3) from build_delay_indices Returns: result_BxTxC: [B, T, C] delayed audio tokens """ device = audio_BxTxC.device # Get device from input tensor t_idx_BxTxC, indices_BTCx3 = precomp t_idx_BxTxC = t_idx_BxTxC.to(device) # Move precomputed indices to device indices_BTCx3 = indices_BTCx3.to(device) # Equivalent of tf.gather_nd using advanced indexing # Ensure indices are long type if not already (build_delay_indices should handle this) gathered_flat = audio_BxTxC[indices_BTCx3[:, 0], indices_BTCx3[:, 1], indices_BTCx3[:, 2]] gathered_BxTxC = gathered_flat.view(audio_BxTxC.shape) # Create masks on the correct device mask_bos = t_idx_BxTxC < 0 # => place bos_value mask_pad = t_idx_BxTxC >= audio_BxTxC.shape[1] # => place pad_value # Create scalar tensors on the correct device bos_tensor = torch.tensor(bos_value, dtype=audio_BxTxC.dtype, device=device) pad_tensor = torch.tensor(pad_value, dtype=audio_BxTxC.dtype, device=device) # If mask_bos, BOS; else if mask_pad, PAD; else original gather # All tensors should now be on the same device result_BxTxC = torch.where(mask_bos, bos_tensor, torch.where(mask_pad, pad_tensor, gathered_BxTxC)) return result_BxTxC def build_revert_indices(B: int, T: int, C: int, delay_pattern: tp.List[int]) -> tp.Tuple[torch.Tensor, torch.Tensor]: """ Precompute indices for the revert operation using PyTorch. Returns: A tuple (t_idx_BxTxC, indices_BTCx3) where: - t_idx_BxTxC is a tensor of shape [B, T, C] computed as time indices plus the delay. - indices_BTCx3 is a tensor of shape [B*T*C, 3] used for gathering, computed from: batch indices, clamped time indices, and channel indices. """ # Use default device unless specified otherwise; assumes inputs might define device later device = None # Or determine dynamically if needed, e.g., from a model parameter delay_arr = torch.tensor(delay_pattern, dtype=torch.int32, device=device) t_idx_BT1 = torch.broadcast_to(torch.arange(T, device=device).unsqueeze(0), [B, T]) t_idx_BT1 = t_idx_BT1.unsqueeze(-1) t_idx_BxTxC = torch.minimum( t_idx_BT1 + delay_arr.view(1, 1, C), torch.tensor(T - 1, device=device), ) b_idx_BxTxC = torch.broadcast_to(torch.arange(B, device=device).view(B, 1, 1), [B, T, C]) c_idx_BxTxC = torch.broadcast_to(torch.arange(C, device=device).view(1, 1, C), [B, T, C]) indices_BTCx3 = torch.stack( [ b_idx_BxTxC.reshape(-1), t_idx_BxTxC.reshape(-1), c_idx_BxTxC.reshape(-1), ], axis=1, ).long() # Ensure indices are long type return t_idx_BxTxC, indices_BTCx3 def revert_audio_delay( audio_BxTxC: torch.Tensor, pad_value: int, precomp: tp.Tuple[torch.Tensor, torch.Tensor], T: int, ) -> torch.Tensor: """ Reverts a delay pattern from batched audio tokens using precomputed indices (PyTorch version). Args: audio_BxTxC: Input delayed audio tensor pad_value: Padding value for out-of-bounds indices precomp: Precomputed revert indices tuple containing: - t_idx_BxTxC: Time offset indices tensor - indices_BTCx3: Gather indices tensor for original audio T: Original sequence length before padding Returns: Reverted audio tensor with same shape as input """ t_idx_BxTxC, indices_BTCx3 = precomp device = audio_BxTxC.device # Get device from input tensor # Move precomputed indices to the same device as audio_BxTxC if they aren't already t_idx_BxTxC = t_idx_BxTxC.to(device) indices_BTCx3 = indices_BTCx3.to(device) # Using PyTorch advanced indexing (equivalent to tf.gather_nd or np equivalent) gathered_flat = audio_BxTxC[indices_BTCx3[:, 0], indices_BTCx3[:, 1], indices_BTCx3[:, 2]] gathered_BxTxC = gathered_flat.view(audio_BxTxC.size()) # Use .size() for robust reshaping # Create pad_tensor on the correct device pad_tensor = torch.tensor(pad_value, dtype=audio_BxTxC.dtype, device=device) # Create T tensor on the correct device for comparison T_tensor = torch.tensor(T, device=device) result_BxTxC = torch.where(t_idx_BxTxC >= T_tensor, pad_tensor, gathered_BxTxC) # Changed np.where to torch.where return result_BxTxC @torch.no_grad() @torch.inference_mode() def decode( model, audio_codes, ): """ Decodes the given frames into an output audio waveform """ if len(audio_codes) != 1: raise ValueError(f"Expected one frame, got {len(audio_codes)}") try: audio_values = model.quantizer.from_codes(audio_codes) audio_values = model.decode(audio_values[0]) return audio_values except Exception as e: print(f"Error in decode method: {str(e)}") raise ``` ## /dia/config.py ```py path="/dia/config.py" """Configuration management module for the Dia model. This module provides comprehensive configuration management for the Dia model, utilizing Pydantic for validation. It defines configurations for data processing, model architecture (encoder and decoder), and training settings. Key components: - DataConfig: Parameters for data loading and preprocessing. - EncoderConfig: Architecture details for the encoder module. - DecoderConfig: Architecture details for the decoder module. - ModelConfig: Combined model architecture settings. - TrainingConfig: Training hyperparameters and settings. - DiaConfig: Master configuration combining all components. """ import os from typing import Annotated from pydantic import BaseModel, BeforeValidator, Field class DataConfig(BaseModel, frozen=True): """Configuration for data loading and preprocessing. Attributes: text_length: Maximum length of text sequences (must be multiple of 128). audio_length: Maximum length of audio sequences (must be multiple of 128). channels: Number of audio channels. text_pad_value: Value used for padding text sequences. audio_eos_value: Value representing the end of audio sequences. audio_bos_value: Value representing the beginning of audio sequences. audio_pad_value: Value used for padding audio sequences. delay_pattern: List of delay values for each audio channel. """ text_length: Annotated[int, BeforeValidator(lambda x: (x + 127) // 128 * 128)] = Field(gt=0, multiple_of=128) audio_length: Annotated[int, BeforeValidator(lambda x: (x + 127) // 128 * 128)] = Field(gt=0, multiple_of=128) channels: int = Field(default=9, gt=0, multiple_of=1) text_pad_value: int = Field(default=0) audio_eos_value: int = Field(default=1024) audio_pad_value: int = Field(default=1025) audio_bos_value: int = Field(default=1026) delay_pattern: list[Annotated[int, Field(ge=0)]] = Field(default_factory=lambda: [0, 8, 9, 10, 11, 12, 13, 14, 15]) def __hash__(self) -> int: """Generate a hash based on all fields of the config.""" return hash( ( self.text_length, self.audio_length, self.channels, self.text_pad_value, self.audio_pad_value, self.audio_bos_value, self.audio_eos_value, tuple(self.delay_pattern), ) ) class EncoderConfig(BaseModel, frozen=True): """Configuration for the encoder component of the Dia model. Attributes: n_layer: Number of transformer layers. n_embd: Embedding dimension. n_hidden: Hidden dimension size in the MLP layers. n_head: Number of attention heads. head_dim: Dimension per attention head. """ n_layer: int = Field(gt=0) n_embd: int = Field(gt=0) n_hidden: int = Field(gt=0) n_head: int = Field(gt=0) head_dim: int = Field(gt=0) class DecoderConfig(BaseModel, frozen=True): """Configuration for the decoder component of the Dia model. Attributes: n_layer: Number of transformer layers. n_embd: Embedding dimension. n_hidden: Hidden dimension size in the MLP layers. gqa_query_heads: Number of query heads for grouped-query self-attention. kv_heads: Number of key/value heads for grouped-query self-attention. gqa_head_dim: Dimension per query head for grouped-query self-attention. cross_query_heads: Number of query heads for cross-attention. cross_head_dim: Dimension per cross-attention head. """ n_layer: int = Field(gt=0) n_embd: int = Field(gt=0) n_hidden: int = Field(gt=0) gqa_query_heads: int = Field(gt=0) kv_heads: int = Field(gt=0) gqa_head_dim: int = Field(gt=0) cross_query_heads: int = Field(gt=0) cross_head_dim: int = Field(gt=0) class ModelConfig(BaseModel, frozen=True): """Main configuration container for the Dia model architecture. Attributes: encoder: Configuration for the encoder component. decoder: Configuration for the decoder component. src_vocab_size: Size of the source (text) vocabulary. tgt_vocab_size: Size of the target (audio code) vocabulary. dropout: Dropout probability applied within the model. normalization_layer_epsilon: Epsilon value for normalization layers (e.g., LayerNorm). weight_dtype: Data type for model weights (e.g., "float32", "bfloat16"). rope_min_timescale: Minimum timescale for Rotary Positional Embeddings (RoPE). rope_max_timescale: Maximum timescale for Rotary Positional Embeddings (RoPE). """ encoder: EncoderConfig decoder: DecoderConfig src_vocab_size: int = Field(default=128, gt=0) tgt_vocab_size: int = Field(default=1028, gt=0) dropout: float = Field(default=0.0, ge=0.0, lt=1.0) normalization_layer_epsilon: float = Field(default=1.0e-5, ge=0.0) weight_dtype: str = Field(default="float32", description="Weight precision") rope_min_timescale: int = Field(default=1, description="Timescale For global Attention") rope_max_timescale: int = Field(default=10_000, description="Timescale For global Attention") class TrainingConfig(BaseModel, frozen=True): pass class DiaConfig(BaseModel, frozen=True): """Master configuration for the Dia model. Combines all sub-configurations into a single validated object. Attributes: version: Configuration version string. model: Model architecture configuration. training: Training process configuration (precision settings). data: Data loading and processing configuration. """ version: str = Field(default="1.0") model: ModelConfig # TODO: remove training. this is just for backward compatibility training: TrainingConfig | None = Field(default=None) data: DataConfig def save(self, path: str) -> None: """Save the current configuration instance to a JSON file. Ensures the parent directory exists and the file has a .json extension. Args: path: The target file path to save the configuration. Raises: ValueError: If the path is not a file with a .json extension. """ os.makedirs(os.path.dirname(path), exist_ok=True) config_json = self.model_dump_json(indent=2) with open(path, "w") as f: f.write(config_json) @classmethod def load(cls, path: str) -> "DiaConfig | None": """Load and validate a Dia configuration from a JSON file. Args: path: The path to the configuration file. Returns: A validated DiaConfig instance if the file exists and is valid, otherwise None if the file is not found. Raises: ValueError: If the path does not point to an existing .json file. pydantic.ValidationError: If the JSON content fails validation against the DiaConfig schema. """ try: with open(path, "r") as f: content = f.read() return cls.model_validate_json(content) except FileNotFoundError: return None ``` ## /dia/layers.py ```py path="/dia/layers.py" import torch import torch.nn as nn import torch.nn.functional as F from huggingface_hub import PyTorchModelHubMixin from torch import Tensor from torch.nn import RMSNorm from .config import DiaConfig from .state import DecoderInferenceState, EncoderInferenceState, KVCache def _normalize_axes(axes: tuple[int, ...], ndim: int) -> tuple[int, ...]: return tuple(ax if ax >= 0 else ndim + ax for ax in axes) class DenseGeneral(nn.Module): """ PyTorch equivalent of flax.linen.DenseGeneral with shapes defined at init. Stores weights (`kernel`) in the same layout as Jax and uses torch.tensordot for the generalized matrix multiplication. Weight/bias shapes are calculated and parameters created during initialization based on config. `load_weights` validates shapes and copies data. Attributes: axis (Tuple[int, ...]): Input axis or axes to contract. in_shapes (Tuple[int, ...]): Sizes of the input dimensions specified by `axis`. out_features (Tuple[int, ...]): Shape of the output features (non-contracted dims). use_bias (bool): Whether to add a bias term. weight (nn.Parameter): The kernel parameter. bias (Optional[nn.Parameter]): The bias parameter (if use_bias=True). """ def __init__( self, in_shapes: tuple[int, ...], out_features: tuple[int, ...], axis: tuple[int, ...] = (-1,), weight_dtype: torch.dtype | None = None, device: torch.device | None = None, ): super().__init__() self.in_shapes = in_shapes self.out_features = out_features self.axis = axis self.kernel_shape = self.in_shapes + self.out_features factory_kwargs = {"device": device, "dtype": weight_dtype} self.weight = nn.Parameter(torch.empty(self.kernel_shape, **factory_kwargs)) def forward(self, inputs: Tensor) -> Tensor: norm_axis = _normalize_axes(self.axis, inputs.ndim) kernel_contract_axes = tuple(range(len(norm_axis))) output = torch.tensordot( inputs.to(self.weight.dtype), self.weight, dims=(norm_axis, kernel_contract_axes), ).to(inputs.dtype) return output class MlpBlock(nn.Module): """MLP block using DenseGeneral.""" def __init__(self, embed_dim: int, intermediate_dim: int, compute_dtype: torch.dtype): super().__init__() self.dtype = compute_dtype self.wi_fused = DenseGeneral( in_shapes=(embed_dim,), out_features=(2, intermediate_dim), axis=(-1,), weight_dtype=compute_dtype, ) self.wo = DenseGeneral( in_shapes=(intermediate_dim,), out_features=(embed_dim,), axis=(-1,), weight_dtype=compute_dtype, ) def forward(self, x: torch.Tensor) -> torch.Tensor: """Forward pass.""" fused_x = self.wi_fused(x) gate = fused_x[..., 0, :] up = fused_x[..., 1, :] hidden = torch.mul(F.silu(gate), up).to(self.dtype) output = self.wo(hidden) return output class RotaryEmbedding(nn.Module): """Rotary Position Embedding (RoPE) implementation in PyTorch.""" def __init__( self, embedding_dims: int, min_timescale: int = 1, max_timescale: int = 10000, dtype: torch.dtype = torch.float32, ): super().__init__() if embedding_dims % 2 != 0: raise ValueError("Embedding dim must be even for RoPE.") self.embedding_dims = embedding_dims self.min_timescale = min_timescale self.max_timescale = max_timescale self.compute_dtype = dtype half_embedding_dim = embedding_dims // 2 fraction = (2.0 * torch.arange(0, half_embedding_dim)) / embedding_dims timescale = (self.min_timescale * (self.max_timescale / self.min_timescale) ** fraction).to(torch.float32) self.register_buffer("timescale", timescale, persistent=False) def forward(self, inputs: torch.Tensor, position: torch.Tensor): """Applies RoPE.""" position = position.unsqueeze(-1).unsqueeze(-1) sinusoid_inp = position / self.timescale sin = torch.sin(sinusoid_inp) cos = torch.cos(sinusoid_inp) first_half, second_half = torch.chunk(inputs.to(torch.float32), 2, dim=-1) first_part = first_half * cos - second_half * sin second_part = second_half * cos + first_half * sin return torch.cat((first_part.to(self.compute_dtype), second_part.to(self.compute_dtype)), dim=-1) class Attention(nn.Module): """Attention using DenseGeneral.""" def __init__( self, config: DiaConfig, q_embed_dim: int, kv_embed_dim: int, num_query_heads: int, num_kv_heads: int, head_dim: int, compute_dtype: torch.dtype, is_cross_attn: bool = False, out_embed_dim: int | None = None, ): super().__init__() self.num_query_heads = num_query_heads self.num_kv_heads = num_kv_heads self.head_dim = head_dim self.is_cross_attn = is_cross_attn self.output_dim = out_embed_dim if out_embed_dim is not None else q_embed_dim self.projected_query_dim = num_query_heads * head_dim if num_query_heads % num_kv_heads != 0: raise ValueError(f"num_query_heads ({num_query_heads}) must be divisible by num_kv_heads ({num_kv_heads})") self.num_gqa_groups = num_query_heads // num_kv_heads # --- Projection Layers using DenseGeneral --- self.q_proj = DenseGeneral( in_shapes=(q_embed_dim,), out_features=(num_query_heads, head_dim), axis=(-1,), weight_dtype=compute_dtype, ) self.k_proj = DenseGeneral( in_shapes=(kv_embed_dim,), out_features=(num_kv_heads, head_dim), axis=(-1,), weight_dtype=compute_dtype, ) self.v_proj = DenseGeneral( in_shapes=(kv_embed_dim,), out_features=(num_kv_heads, head_dim), axis=(-1,), weight_dtype=compute_dtype, ) self.o_proj = DenseGeneral( in_shapes=(num_query_heads, head_dim), out_features=(self.output_dim,), axis=(-2, -1), weight_dtype=compute_dtype, ) # --- Rotary Embedding --- self.rotary_emb = RotaryEmbedding( embedding_dims=self.head_dim, min_timescale=config.model.rope_min_timescale, max_timescale=config.model.rope_max_timescale, dtype=compute_dtype, ) def forward( self, Xq: torch.Tensor, # (B, T, D) T = 1 in AR generation Xkv: torch.Tensor, # (B, S, E) S = 1 in AR generation q_positions: torch.Tensor, # (B, T) kv_positions: torch.Tensor | None = None, # (B, S) attn_mask: torch.Tensor | None = None, # None in Decoder Self Attention, Valid mask in Others cache: KVCache | None = None, # None in Encoder, KVCache in Decoder prefill: bool = False, is_causal: bool = False, ) -> tuple[torch.Tensor, tuple[torch.Tensor, torch.Tensor] | None]: """ Performs attention calculation with optional KV caching. Args: Xq: Query tensor (B, T, D). T=1 during single-step decoding. Xkv: Key/Value source tensor (B, S, E). S=1 during single-step decoding for self-attn. q_positions: Positions for queries (B, T). kv_positions: Positions for keys/values (B, S). If None, uses q_positions. attn_mask: Attention mask. cache: KVCache. prefill: If True, use prefill mode. Returns: A tuple containing: - output: The attention output tensor (B, T, output_dim). - present_kv: The K/V state to be cached for the next step ((B, N, S_new, H), (B, N, S_new, H)). For self-attn, S_new = S_past + S. For cross-attn, S_new = S_kv. """ if kv_positions is None: kv_positions = q_positions original_dtype = Xq.dtype Xq_BxTxNxH = self.q_proj(Xq) Xq_BxTxNxH = self.rotary_emb(Xq_BxTxNxH, position=q_positions) Xq_BxNxTxH = Xq_BxTxNxH.transpose(1, 2) attn_k: torch.Tensor | None = None attn_v: torch.Tensor | None = None if self.is_cross_attn: attn_k, attn_v = cache.k, cache.v else: Xk_BxSxKxH = self.k_proj(Xkv) # (B, S, K, H) Xv_BxSxKxH = self.v_proj(Xkv) # (B, S, K, H) Xk_BxSxKxH = self.rotary_emb(Xk_BxSxKxH, position=kv_positions) # (B, S, K, H) Xk_BxKxSxH = Xk_BxSxKxH.transpose(1, 2) # (B, K, S, H) Xv_BxKxSxH = Xv_BxSxKxH.transpose(1, 2) # (B, K, S, H) if cache is None: attn_k = Xk_BxKxSxH attn_v = Xv_BxKxSxH else: if prefill: attn_k, attn_v = Xk_BxKxSxH, Xv_BxKxSxH cache.prefill(attn_k, attn_v) else: attn_k, attn_v = cache.update(Xk_BxKxSxH, Xv_BxKxSxH) attn_output = F.scaled_dot_product_attention( Xq_BxNxTxH, attn_k, attn_v, attn_mask=attn_mask, scale=1.0, enable_gqa=self.num_gqa_groups > 1, is_causal=is_causal, ) attn_output = attn_output.transpose(1, 2).contiguous() # (B, T, N, H) output = self.o_proj(attn_output) return output.to(original_dtype) class EncoderLayer(nn.Module): """Transformer Encoder Layer using DenseGeneral.""" def __init__(self, config: DiaConfig, compute_dtype: torch.dtype): super().__init__() self.config = config model_config = config.model enc_config = config.model.encoder embed_dim = enc_config.n_embd self.compute_dtype = compute_dtype self.pre_sa_norm = RMSNorm( embed_dim, eps=model_config.normalization_layer_epsilon, dtype=torch.float32, ) self.self_attention = Attention( config, q_embed_dim=embed_dim, kv_embed_dim=embed_dim, num_query_heads=enc_config.n_head, num_kv_heads=enc_config.n_head, head_dim=enc_config.head_dim, compute_dtype=compute_dtype, is_cross_attn=False, out_embed_dim=embed_dim, ) self.post_sa_norm = RMSNorm( embed_dim, eps=model_config.normalization_layer_epsilon, dtype=torch.float32, ) self.mlp = MlpBlock(embed_dim=embed_dim, intermediate_dim=enc_config.n_hidden, compute_dtype=compute_dtype) def forward( self, x: torch.Tensor, state: EncoderInferenceState, ) -> torch.Tensor: residual = x x_norm = self.pre_sa_norm(x).to(self.compute_dtype) sa_out = self.self_attention( Xq=x_norm, Xkv=x_norm, q_positions=state.positions, kv_positions=state.positions, attn_mask=state.attn_mask, ) x = residual + sa_out residual = x x_norm = self.post_sa_norm(x).to(self.compute_dtype) mlp_out = self.mlp(x_norm) x = residual + mlp_out return x class Encoder(nn.Module): """Transformer Encoder Stack using DenseGeneral.""" def __init__(self, config: DiaConfig, compute_dtype: torch.dtype): super().__init__() self.config = config model_config = config.model enc_config = config.model.encoder self.compute_dtype = compute_dtype self.embedding = nn.Embedding( model_config.src_vocab_size, enc_config.n_embd, dtype=compute_dtype, ) self.layers = nn.ModuleList([EncoderLayer(config, compute_dtype) for _ in range(enc_config.n_layer)]) self.norm = RMSNorm( enc_config.n_embd, eps=model_config.normalization_layer_epsilon, dtype=torch.float32, ) def forward( self, x_ids: torch.Tensor, state: EncoderInferenceState, ) -> torch.Tensor: x = self.embedding(x_ids) for layer in self.layers: x = layer(x, state) x = self.norm(x).to(self.compute_dtype) return x class DecoderLayer(nn.Module): """Transformer Decoder Layer using DenseGeneral.""" def __init__(self, config: DiaConfig, compute_dtype: torch.dtype): super().__init__() self.config = config model_config = config.model dec_config = config.model.decoder enc_config = config.model.encoder dec_embed_dim = dec_config.n_embd enc_embed_dim = enc_config.n_embd self.compute_dtype = compute_dtype # Norms self.pre_sa_norm = RMSNorm( dec_embed_dim, eps=model_config.normalization_layer_epsilon, dtype=torch.float32, ) self.pre_ca_norm = RMSNorm( dec_embed_dim, eps=model_config.normalization_layer_epsilon, dtype=torch.float32, ) self.pre_mlp_norm = RMSNorm( dec_embed_dim, eps=model_config.normalization_layer_epsilon, dtype=torch.float32, ) # Self-Attention (GQA) with Causal Masking self.self_attention = Attention( config, q_embed_dim=dec_embed_dim, kv_embed_dim=dec_embed_dim, num_query_heads=dec_config.gqa_query_heads, num_kv_heads=dec_config.kv_heads, head_dim=dec_config.gqa_head_dim, compute_dtype=compute_dtype, is_cross_attn=False, out_embed_dim=dec_embed_dim, ) # Cross-Attention (MHA) self.cross_attention = Attention( config=config, q_embed_dim=dec_embed_dim, kv_embed_dim=enc_embed_dim, # Note kv_embed_dim num_query_heads=dec_config.cross_query_heads, num_kv_heads=dec_config.cross_query_heads, head_dim=dec_config.cross_head_dim, compute_dtype=compute_dtype, is_cross_attn=True, out_embed_dim=dec_embed_dim, ) # MLP self.mlp = MlpBlock( embed_dim=dec_embed_dim, intermediate_dim=dec_config.n_hidden, compute_dtype=compute_dtype, ) def forward( self, x: torch.Tensor, state: DecoderInferenceState, self_attn_cache: KVCache | None = None, cross_attn_cache: KVCache | None = None, prefill: bool = False, ) -> torch.Tensor: residual = x x_norm = self.pre_sa_norm(x).to(self.compute_dtype) sa_out = self.self_attention( Xq=x_norm, # (2, 1, D) Xkv=x_norm, # (2, 1, D) q_positions=state.dec_positions, # (2, 1) kv_positions=state.dec_positions, # (2, 1) attn_mask=None, cache=self_attn_cache, prefill=prefill, is_causal=prefill, ) x = residual + sa_out residual = x x_norm = self.pre_ca_norm(x).to(self.compute_dtype) ca_out = self.cross_attention( Xq=x_norm, Xkv=state.enc_out, q_positions=state.dec_positions, kv_positions=state.enc_positions, attn_mask=state.dec_cross_attn_mask, cache=cross_attn_cache, ) x = residual + ca_out residual = x x_norm = self.pre_mlp_norm(x).to(self.compute_dtype) mlp_out = self.mlp(x_norm) x = residual + mlp_out return x class Decoder(nn.Module): """Transformer Decoder Stack using DenseGeneral.""" def __init__(self, config: DiaConfig, compute_dtype: torch.dtype): super().__init__() self.config = config model_config = config.model dec_config = config.model.decoder data_config = config.data self.num_channels = data_config.channels self.num_layers = dec_config.n_layer self.embeddings = nn.ModuleList( [ nn.Embedding(model_config.tgt_vocab_size, dec_config.n_embd, dtype=compute_dtype) for _ in range(self.num_channels) ] ) self.layers = nn.ModuleList( [DecoderLayer(config=config, compute_dtype=compute_dtype) for _ in range(self.num_layers)] ) self.norm = RMSNorm( dec_config.n_embd, eps=model_config.normalization_layer_epsilon, dtype=torch.float32, ) self.logits_dense = DenseGeneral( in_shapes=(dec_config.n_embd,), out_features=(self.num_channels, model_config.tgt_vocab_size), axis=(-1,), weight_dtype=compute_dtype, ) def precompute_cross_attn_cache( self, enc_out: torch.Tensor, # (B, S, E) enc_positions: torch.Tensor, # (B, S) ) -> list[KVCache]: """ Computes the Key and Value tensors for cross-attention for each layer from the encoder output. """ per_layer_kv_cache: list[KVCache] = [] for layer in self.layers: cross_attn_module = layer.cross_attention k_proj = cross_attn_module.k_proj(enc_out) v_proj = cross_attn_module.v_proj(enc_out) k_proj = cross_attn_module.rotary_emb(k_proj, position=enc_positions) k = k_proj.transpose(1, 2) v = v_proj.transpose(1, 2) per_layer_kv_cache.append(KVCache.from_kv(k, v)) return per_layer_kv_cache def decode_step( self, tgt_ids_Bx1xC: torch.Tensor, # [B, 1, C] state: DecoderInferenceState, ) -> torch.Tensor: """ Performs a single decoding step, managing KV caches layer by layer. Returns: A tuple containing: - logits_Bx1xCV: The final output logits for the current step (B, 1, C*V), cast to float32. """ x = None for i in range(self.num_channels): channel_tokens = tgt_ids_Bx1xC[..., i] channel_embed = self.embeddings[i](channel_tokens) x = channel_embed if x is None else x + channel_embed for i, layer in enumerate(self.layers): self_cache = state.self_attn_cache[i] cross_cache = state.cross_attn_cache[i] x = layer( x, # (2, 1, D) state, self_attn_cache=self_cache, cross_attn_cache=cross_cache, ) x = self.norm(x) logits_Bx1xCxV = self.logits_dense(x) return logits_Bx1xCxV.to(torch.float32) def forward(self, tgt_ids_BxTxC: torch.Tensor, state: DecoderInferenceState) -> torch.Tensor: """ Forward pass for the Decoder stack, managing KV caches. Args: tgt_ids_BxTxC: Target token IDs (B, T, C). encoder_out: Output from the encoder (B, S, E). tgt_positions: Positions for target sequence (B, T). src_positions: Positions for source sequence (B, S). self_attn_mask: Mask for self-attention. cross_attn_mask: Mask for cross-attention. past_key_values: List containing the self-attention KV cache for each layer from the previous decoding step. `len(past_key_values)` should equal `num_layers`. precomputed_cross_attn_kv: A single tuple containing the pre-computed K/V cache derived from `encoder_out`. This is passed identically to all layers. Returns: A tuple containing: - logits: The final output logits (B, T, C * V), cast to float32. - present_key_values: A list containing the updated self-attention KV cache for each layer for the *current* decoding step. """ _, _, num_channels_in = tgt_ids_BxTxC.shape assert num_channels_in == self.num_channels, "Input channels mismatch" # Embeddings x = None for i in range(self.num_channels): channel_tokens = tgt_ids_BxTxC[..., i] channel_embed = self.embeddings[i](channel_tokens) x = channel_embed if x is None else x + channel_embed for i, layer in enumerate(self.layers): self_cache = state.self_attn_cache[i] cross_cache = state.cross_attn_cache[i] x = layer(x, state, self_attn_cache=self_cache, cross_attn_cache=cross_cache, prefill=True) # Final Norm x = self.norm(x) logits_BxTxCxV = self.logits_dense(x) return logits_BxTxCxV.to(torch.float32) class DiaModel( nn.Module, PyTorchModelHubMixin, repo_url="https://github.com/nari-labs/dia", pipeline_tag="text-to-speech", license="apache-2.0", coders={ DiaConfig: ( lambda x: x.model_dump(), lambda data: DiaConfig.model_validate(data), ), }, ): """PyTorch Dia Model using DenseGeneral.""" def __init__(self, config: DiaConfig, compute_dtype: torch.dtype): super().__init__() self.config = config self.encoder = Encoder(config, compute_dtype) self.decoder = Decoder(config, compute_dtype) ``` ## /dia/model.py ```py path="/dia/model.py" import time from enum import Enum import dac import numpy as np import torch import torchaudio from .audio import apply_audio_delay, build_delay_indices, build_revert_indices, decode, revert_audio_delay from .config import DiaConfig from .layers import DiaModel from .state import DecoderInferenceState, DecoderOutput, EncoderInferenceState DEFAULT_SAMPLE_RATE = 44100 def _get_default_device(): if torch.cuda.is_available(): return torch.device("cuda") elif hasattr(torch.backends, "mps") and torch.backends.mps.is_available(): return torch.device("mps") return torch.device("cpu") def _sample_next_token( logits_BCxV: torch.Tensor, temperature: float, top_p: float, cfg_filter_top_k: int | None = None, ) -> torch.Tensor: if temperature == 0.0: return torch.argmax(logits_BCxV, dim=-1) logits_BCxV = logits_BCxV / temperature if cfg_filter_top_k is not None: _, top_k_indices_BCxV = torch.topk(logits_BCxV, k=cfg_filter_top_k, dim=-1) mask = torch.ones_like(logits_BCxV, dtype=torch.bool) mask.scatter_(dim=-1, index=top_k_indices_BCxV, value=False) logits_BCxV = logits_BCxV.masked_fill(mask, -torch.inf) if top_p < 1.0: probs_BCxV = torch.softmax(logits_BCxV, dim=-1) sorted_probs_BCxV, sorted_indices_BCxV = torch.sort(probs_BCxV, dim=-1, descending=True) cumulative_probs_BCxV = torch.cumsum(sorted_probs_BCxV, dim=-1) sorted_indices_to_remove_BCxV = cumulative_probs_BCxV > top_p sorted_indices_to_remove_BCxV[..., 1:] = sorted_indices_to_remove_BCxV[..., :-1].clone() sorted_indices_to_remove_BCxV[..., 0] = 0 indices_to_remove_BCxV = torch.zeros_like(sorted_indices_to_remove_BCxV) indices_to_remove_BCxV.scatter_(dim=-1, index=sorted_indices_BCxV, src=sorted_indices_to_remove_BCxV) logits_BCxV = logits_BCxV.masked_fill(indices_to_remove_BCxV, -torch.inf) final_probs_BCxV = torch.softmax(logits_BCxV, dim=-1) sampled_indices_BC = torch.multinomial(final_probs_BCxV, num_samples=1) sampled_indices_C = sampled_indices_BC.squeeze(-1) return sampled_indices_C class ComputeDtype(str, Enum): FLOAT32 = "float32" FLOAT16 = "float16" BFLOAT16 = "bfloat16" def to_dtype(self) -> torch.dtype: if self == ComputeDtype.FLOAT32: return torch.float32 elif self == ComputeDtype.FLOAT16: return torch.float16 elif self == ComputeDtype.BFLOAT16: return torch.bfloat16 else: raise ValueError(f"Unsupported compute dtype: {self}") class Dia: def __init__( self, config: DiaConfig, compute_dtype: str | ComputeDtype = ComputeDtype.FLOAT32, device: torch.device | None = None, ): """Initializes the Dia model. Args: config: The configuration object for the model. device: The device to load the model onto. If None, will automatically select the best available device. Raises: RuntimeError: If there is an error loading the DAC model. """ super().__init__() self.config = config self.device = device if device is not None else _get_default_device() if isinstance(compute_dtype, str): compute_dtype = ComputeDtype(compute_dtype) self.compute_dtype = compute_dtype.to_dtype() self.model = DiaModel(config, self.compute_dtype) self.dac_model = None @classmethod def from_local( cls, config_path: str, checkpoint_path: str, compute_dtype: str | ComputeDtype = ComputeDtype.FLOAT32, device: torch.device | None = None, ) -> "Dia": """Loads the Dia model from local configuration and checkpoint files. Args: config_path: Path to the configuration JSON file. checkpoint_path: Path to the model checkpoint (.pth) file. device: The device to load the model onto. If None, will automatically select the best available device. Returns: An instance of the Dia model loaded with weights and set to eval mode. Raises: FileNotFoundError: If the config or checkpoint file is not found. RuntimeError: If there is an error loading the checkpoint. """ config = DiaConfig.load(config_path) if config is None: raise FileNotFoundError(f"Config file not found at {config_path}") dia = cls(config, compute_dtype, device) try: state_dict = torch.load(checkpoint_path, map_location=dia.device) dia.model.load_state_dict(state_dict) except FileNotFoundError: raise FileNotFoundError(f"Checkpoint file not found at {checkpoint_path}") except Exception as e: raise RuntimeError(f"Error loading checkpoint from {checkpoint_path}") from e dia.model.to(dia.device) dia.model.eval() dia._load_dac_model() return dia @classmethod def from_pretrained( cls, model_name: str = "nari-labs/Dia-1.6B", compute_dtype: str | ComputeDtype = ComputeDtype.FLOAT32, device: torch.device | None = None, ) -> "Dia": """Loads the Dia model from a Hugging Face Hub repository. Downloads the configuration and checkpoint files from the specified repository ID and then loads the model. Args: model_name: The Hugging Face Hub repository ID (e.g., "nari-labs/Dia-1.6B"). compute_dtype: The computation dtype to use. device: The device to load the model onto. If None, will automatically select the best available device. Returns: An instance of the Dia model loaded with weights and set to eval mode. Raises: FileNotFoundError: If config or checkpoint download/loading fails. RuntimeError: If there is an error loading the checkpoint. """ if isinstance(compute_dtype, str): compute_dtype = ComputeDtype(compute_dtype) loaded_model = DiaModel.from_pretrained(model_name, compute_dtype=compute_dtype.to_dtype()) config = loaded_model.config dia = cls(config, compute_dtype, device) dia.model = loaded_model dia.model.to(dia.device) dia.model.eval() dia._load_dac_model() return dia def _load_dac_model(self): try: dac_model_path = dac.utils.download() dac_model = dac.DAC.load(dac_model_path).to(self.device) except Exception as e: raise RuntimeError("Failed to load DAC model") from e self.dac_model = dac_model def _prepare_text_input(self, text: str) -> torch.Tensor: """Encodes text prompt, pads, and creates attention mask and positions.""" text_pad_value = self.config.data.text_pad_value max_len = self.config.data.text_length byte_text = text.encode("utf-8") replaced_bytes = byte_text.replace(b"[S1]", b"\x01").replace(b"[S2]", b"\x02") text_tokens = list(replaced_bytes) current_len = len(text_tokens) padding_needed = max_len - current_len if padding_needed <= 0: text_tokens = text_tokens[:max_len] padded_text_np = np.array(text_tokens, dtype=np.uint8) else: padded_text_np = np.pad( text_tokens, (0, padding_needed), mode="constant", constant_values=text_pad_value, ).astype(np.uint8) src_tokens = torch.from_numpy(padded_text_np).to(torch.long).to(self.device).unsqueeze(0) # [1, S] return src_tokens def _prepare_audio_prompt(self, audio_prompt: torch.Tensor | None) -> tuple[torch.Tensor, int]: num_channels = self.config.data.channels audio_bos_value = self.config.data.audio_bos_value audio_pad_value = self.config.data.audio_pad_value delay_pattern = self.config.data.delay_pattern max_delay_pattern = max(delay_pattern) prefill = torch.full( (1, num_channels), fill_value=audio_bos_value, dtype=torch.int, device=self.device, ) prefill_step = 1 if audio_prompt is not None: prefill_step += audio_prompt.shape[0] prefill = torch.cat([prefill, audio_prompt], dim=0) delay_pad_tensor = torch.full( (max_delay_pattern, num_channels), fill_value=-1, dtype=torch.int, device=self.device ) prefill = torch.cat([prefill, delay_pad_tensor], dim=0) delay_precomp = build_delay_indices( B=1, T=prefill.shape[0], C=num_channels, delay_pattern=delay_pattern, ) prefill = apply_audio_delay( audio_BxTxC=prefill.unsqueeze(0), pad_value=audio_pad_value, bos_value=audio_bos_value, precomp=delay_precomp, ).squeeze(0) return prefill, prefill_step def _prepare_generation(self, text: str, audio_prompt: str | torch.Tensor | None, verbose: bool): enc_input_cond = self._prepare_text_input(text) enc_input_uncond = torch.zeros_like(enc_input_cond) enc_input = torch.cat([enc_input_uncond, enc_input_cond], dim=0) if isinstance(audio_prompt, str): audio_prompt = self.load_audio(audio_prompt) prefill, prefill_step = self._prepare_audio_prompt(audio_prompt) if verbose: print("generate: data loaded") enc_state = EncoderInferenceState.new(self.config, enc_input_cond) encoder_out = self.model.encoder(enc_input, enc_state) dec_cross_attn_cache = self.model.decoder.precompute_cross_attn_cache(encoder_out, enc_state.positions) dec_state = DecoderInferenceState.new( self.config, enc_state, encoder_out, dec_cross_attn_cache, self.compute_dtype ) dec_output = DecoderOutput.new(self.config, self.device) dec_output.prefill(prefill, prefill_step) dec_step = prefill_step - 1 if dec_step > 0: dec_state.prepare_step(0, dec_step) tokens_BxTxC = dec_output.get_tokens_at(0, dec_step).unsqueeze(0).expand(2, -1, -1) self.model.decoder.forward(tokens_BxTxC, dec_state) return dec_state, dec_output def _decoder_step( self, tokens_Bx1xC: torch.Tensor, dec_state: DecoderInferenceState, cfg_scale: float, temperature: float, top_p: float, cfg_filter_top_k: int, ) -> torch.Tensor: audio_eos_value = self.config.data.audio_eos_value logits_Bx1xCxV = self.model.decoder.decode_step(tokens_Bx1xC, dec_state) logits_last_BxCxV = logits_Bx1xCxV[:, -1, :, :] uncond_logits_CxV = logits_last_BxCxV[0, :, :] cond_logits_CxV = logits_last_BxCxV[1, :, :] logits_CxV = cond_logits_CxV + cfg_scale * (cond_logits_CxV - uncond_logits_CxV) logits_CxV[:, audio_eos_value + 1 :] = -torch.inf logits_CxV[1:, audio_eos_value:] = -torch.inf pred_C = _sample_next_token( logits_CxV.float(), temperature=temperature, top_p=top_p, cfg_filter_top_k=cfg_filter_top_k, ) return pred_C def _generate_output(self, generated_codes: torch.Tensor) -> np.ndarray: num_channels = self.config.data.channels seq_length = generated_codes.shape[0] delay_pattern = self.config.data.delay_pattern audio_pad_value = self.config.data.audio_pad_value max_delay_pattern = max(delay_pattern) revert_precomp = build_revert_indices( B=1, T=seq_length, C=num_channels, delay_pattern=delay_pattern, ) codebook = revert_audio_delay( audio_BxTxC=generated_codes.unsqueeze(0), pad_value=audio_pad_value, precomp=revert_precomp, T=seq_length, )[:, :-max_delay_pattern, :] min_valid_index = 0 max_valid_index = 1023 invalid_mask = (codebook < min_valid_index) | (codebook > max_valid_index) codebook[invalid_mask] = 0 audio = decode(self.dac_model, codebook.transpose(1, 2)) return audio.squeeze().cpu().numpy() def load_audio(self, audio_path: str) -> torch.Tensor: audio, sr = torchaudio.load(audio_path, channels_first=True) # C, T if sr != DEFAULT_SAMPLE_RATE: audio = torchaudio.functional.resample(audio, sr, DEFAULT_SAMPLE_RATE) audio = audio.to(self.device).unsqueeze(0) # 1, C, T audio_data = self.dac_model.preprocess(audio, DEFAULT_SAMPLE_RATE) _, encoded_frame, _, _, _ = self.dac_model.encode(audio_data) # 1, C, T return encoded_frame.squeeze(0).transpose(0, 1) def save_audio(self, path: str, audio: np.ndarray): import soundfile as sf sf.write(path, audio, DEFAULT_SAMPLE_RATE) @torch.inference_mode() def generate( self, text: str, max_tokens: int | None = None, cfg_scale: float = 3.0, temperature: float = 1.3, top_p: float = 0.95, use_torch_compile: bool = False, cfg_filter_top_k: int = 35, audio_prompt: str | torch.Tensor | None = None, audio_prompt_path: str | None = None, use_cfg_filter: bool | None = None, verbose: bool = False, ) -> np.ndarray: audio_eos_value = self.config.data.audio_eos_value audio_pad_value = self.config.data.audio_pad_value delay_pattern = self.config.data.delay_pattern max_tokens = self.config.data.audio_length if max_tokens is None else max_tokens max_delay_pattern = max(delay_pattern) self.model.eval() if audio_prompt_path: print("Warning: audio_prompt_path is deprecated. Use audio_prompt instead.") audio_prompt = audio_prompt_path if use_cfg_filter is not None: print("Warning: use_cfg_filter is deprecated.") if verbose: total_start_time = time.time() dec_state, dec_output = self._prepare_generation(text, audio_prompt, verbose) dec_step = dec_output.prefill_step - 1 bos_countdown = max_delay_pattern eos_detected = False eos_countdown = -1 if use_torch_compile: step_fn = torch.compile(self._decoder_step, mode="default") else: step_fn = self._decoder_step if verbose: print("generate: starting generation loop") if use_torch_compile: print("generate: by using use_torch_compile=True, the first step would take long") start_time = time.time() while dec_step < max_tokens: dec_state.prepare_step(dec_step) tokens_Bx1xC = dec_output.get_tokens_at(dec_step).unsqueeze(0).expand(2, -1, -1) pred_C = step_fn( tokens_Bx1xC, dec_state, cfg_scale, temperature, top_p, cfg_filter_top_k, ) if (not eos_detected and pred_C[0] == audio_eos_value) or dec_step == max_tokens - max_delay_pattern - 1: eos_detected = True eos_countdown = max_delay_pattern if eos_countdown > 0: step_after_eos = max_delay_pattern - eos_countdown for i, d in enumerate(delay_pattern): if step_after_eos == d: pred_C[i] = audio_eos_value elif step_after_eos > d: pred_C[i] = audio_pad_value eos_countdown -= 1 bos_countdown = max(0, bos_countdown - 1) dec_output.update_one(pred_C, dec_step + 1, bos_countdown > 0) if eos_countdown == 0: break dec_step += 1 if verbose and dec_step % 86 == 0: duration = time.time() - start_time print( f"generate step {dec_step}: speed={86 / duration:.3f} tokens/s, realtime factor={1 / duration:.3f}x" ) start_time = time.time() if dec_output.prefill_step >= dec_step + 1: print("Warning: Nothing generated") return None generated_codes = dec_output.generated_tokens[dec_output.prefill_step : dec_step + 1, :] if verbose: total_step = dec_step + 1 - dec_output.prefill_step total_duration = time.time() - total_start_time print(f"generate: total step={total_step}, total duration={total_duration:.3f}s") return self._generate_output(generated_codes) ``` ## /dia/state.py ```py path="/dia/state.py" from dataclasses import dataclass import torch from .config import DiaConfig def create_attn_mask( q_padding_mask_1d: torch.Tensor, k_padding_mask_1d: torch.Tensor, device: torch.device, is_causal: bool = False, ) -> torch.Tensor: """ Creates the attention mask (self or cross) mimicking JAX segment ID logic. """ B1, Tq = q_padding_mask_1d.shape B2, Tk = k_padding_mask_1d.shape assert B1 == B2, "Query and key batch dimensions must match" p_mask_q = q_padding_mask_1d.unsqueeze(2) # Shape [B, Tq, 1] p_mask_k = k_padding_mask_1d.unsqueeze(1) # Shape [B, 1, Tk] # Condition A: Non-padding query attends to non-padding key non_pad_attends_non_pad = p_mask_q & p_mask_k # Shape [B, Tq, Tk] # Condition B: Padding query attends to padding key pad_attends_pad = (~p_mask_q) & (~p_mask_k) # Shape [B, Tq, Tk] # Combine: True if padding status is compatible (both non-pad OR both pad) mask = non_pad_attends_non_pad | pad_attends_pad # Shape [B, Tq, Tk] if is_causal: assert Tq == Tk, "Causal mask requires query and key sequence lengths to be equal" causal_mask_2d = torch.tril(torch.ones((Tq, Tk), dtype=torch.bool, device=device)) # Shape [Tq, Tk] causal_mask = mask & causal_mask_2d # Shape [B, Tq, Tk] return causal_mask.unsqueeze(1) # Shape [B, 1, Tq, Tk] else: return mask.unsqueeze(1) # Shape [B, 1, Tq, Tk] @dataclass class EncoderInferenceState: """Parameters specifically for encoder inference.""" max_seq_len: int device: torch.device positions: torch.Tensor padding_mask: torch.Tensor attn_mask: torch.Tensor @classmethod def new(cls, config: DiaConfig, cond_src: torch.Tensor) -> "EncoderInferenceState": """Creates EtorchrInferenceParams from DiaConfig and a device.""" device = cond_src.device positions = ( torch.arange(config.data.text_length, dtype=torch.float32, device=device).unsqueeze(0).expand(2, -1) ) padding_mask = (cond_src != config.data.text_pad_value).to(device).expand(2, -1) attn_mask = create_attn_mask(padding_mask, padding_mask, device, is_causal=False) return cls( max_seq_len=config.data.text_length, device=device, positions=positions, padding_mask=padding_mask, attn_mask=attn_mask, ) class KVCache: def __init__( self, num_heads: int, max_len: int, head_dim: int, dtype: torch.dtype, device: torch.device, k: torch.Tensor | None = None, v: torch.Tensor | None = None, ): self.k = torch.zeros((2, num_heads, max_len, head_dim), dtype=dtype, device=device) if k is None else k self.v = torch.zeros((2, num_heads, max_len, head_dim), dtype=dtype, device=device) if v is None else v self.current_idx = torch.tensor(0) @classmethod def from_kv(cls, k: torch.Tensor, v: torch.Tensor) -> "KVCache": return cls( num_heads=k.shape[1], max_len=k.shape[2], head_dim=k.shape[3], dtype=k.dtype, device=k.device, k=k, v=v, ) def update(self, k: torch.Tensor, v: torch.Tensor) -> tuple[torch.Tensor, torch.Tensor]: self.k[:, :, self.current_idx : self.current_idx + 1, :] = k self.v[:, :, self.current_idx : self.current_idx + 1, :] = v self.current_idx += 1 return self.k[:, :, : self.current_idx, :], self.v[:, :, : self.current_idx, :] def prefill(self, k: torch.Tensor, v: torch.Tensor) -> tuple[torch.Tensor, torch.Tensor]: prefill_len = k.shape[2] self.k[:, :, :prefill_len, :] = k self.v[:, :, :prefill_len, :] = v self.current_idx = prefill_len - 1 @dataclass class DecoderInferenceState: """Parameters specifically for decoder inference.""" device: torch.device dtype: torch.dtype enc_out: torch.Tensor enc_positions: torch.Tensor dec_positions: torch.Tensor dec_cross_attn_mask: torch.Tensor self_attn_cache: list[KVCache] cross_attn_cache: list[KVCache] @classmethod def new( cls, config: DiaConfig, enc_state: EncoderInferenceState, enc_out: torch.Tensor, dec_cross_attn_cache: list[KVCache], compute_dtype: torch.dtype, ) -> "DecoderInferenceState": """Creates DecoderInferenceParams from DiaConfig and a device.""" device = enc_out.device max_audio_len = config.data.audio_length dec_positions = torch.full((2, 1), fill_value=0, dtype=torch.long, device=device) tgt_padding_mask = torch.ones((2, 1), dtype=torch.bool, device=device) dec_cross_attn_mask = create_attn_mask(tgt_padding_mask, enc_state.padding_mask, device, is_causal=False) self_attn_cache = [ KVCache( config.model.decoder.kv_heads, max_audio_len, config.model.decoder.gqa_head_dim, compute_dtype, device, ) for _ in range(config.model.decoder.n_layer) ] return cls( device=device, dtype=compute_dtype, enc_out=enc_out, enc_positions=enc_state.positions, dec_positions=dec_positions, dec_cross_attn_mask=dec_cross_attn_mask, self_attn_cache=self_attn_cache, cross_attn_cache=dec_cross_attn_cache, ) def prepare_step(self, step_from: int, step_to: int | None = None) -> None: if step_to is None: step_to = step_from + 1 self.dec_positions = ( torch.arange(step_from, step_to, dtype=torch.float32, device=self.device).unsqueeze(0).expand(2, -1) ) @dataclass class DecoderOutput: generated_tokens: torch.Tensor prefill_step: int @classmethod def new(cls, config: DiaConfig, device: torch.device) -> "DecoderOutput": max_audio_len = config.data.audio_length return cls( generated_tokens=torch.full( (max_audio_len, config.data.channels), fill_value=-1, dtype=torch.int, device=device, ), prefill_step=0, ) def get_tokens_at(self, step_from: int, step_to: int | None = None) -> torch.Tensor: if step_to is None: step_to = step_from + 1 return self.generated_tokens[step_from:step_to, :] def update_one(self, dec_out: torch.Tensor, step: int, apply_mask: bool = False): if apply_mask: mask = self.generated_tokens[step : step + 1, :] == -1 self.generated_tokens[step : step + 1, :] = torch.where( mask, dec_out, self.generated_tokens[step : step + 1, :] ) else: self.generated_tokens[step : step + 1, :] = dec_out def prefill(self, dec_out: torch.Tensor, prefill_step: int): length = dec_out.shape[0] self.generated_tokens[0:length, :] = dec_out self.prefill_step = prefill_step ``` ## /dia/static/images/banner.png Binary file available at https://raw.githubusercontent.com/nari-labs/dia/refs/heads/main/dia/static/images/banner.png ## /docker/Dockerfile.cpu ```cpu path="/docker/Dockerfile.cpu" # Dockerfile.cpu - CPU-only deployment for DIA # -------------------------------------------------- # Build: docker build . -f docker/Dockerfile.cpu -t dia-cpu # Run: docker run --rm -p 7860:7860 dia-cpu FROM python:3.10-slim # Set non-interactive frontend ENV DEBIAN_FRONTEND=noninteractive # Install venv, and system dependencies RUN apt-get update && apt-get install -y \ python3-venv \ libsndfile1 \ ffmpeg \ curl \ && apt-get clean && rm -rf /var/lib/apt/lists/* # Create non-root user and set up directories RUN useradd -m -u 1001 appuser && \ mkdir -p /app/outputs /app && \ chown -R appuser:appuser /app USER appuser WORKDIR /app # Copy all code (including pyproject.toml) COPY --chown=appuser:appuser . . # Create and activate virtual environment RUN python3 -m venv /app/venv ENV PATH="/app/venv/bin:$PATH" # Install all project dependencies (CPU-only PyTorch) RUN pip install --upgrade pip && \ pip install torch torchaudio --index-url https://download.pytorch.org/whl/cpu && \ pip install --no-cache-dir -e .[dev] # Set environment variables ENV PYTHONUNBUFFERED=1 \ PYTHONPATH=/app # Expose Gradio default port ENV GRADIO_SERVER_NAME="0.0.0.0" EXPOSE 7860 # Entrypoint CMD ["python3", "app.py"] ``` ## /docker/Dockerfile.gpu ```gpu path="/docker/Dockerfile.gpu" # Dockerfile.gpu - GPU deployment for DIA # -------------------------------------------------- # Build: docker build . -f docker/Dockerfile.gpu -t dia-gpu # Run: docker run --rm --gpus all -p 7860:7860 dia-gpu # Requires NVIDIA Container Toolkit on host. FROM pytorch/pytorch:2.1.2-cuda12.1-cudnn8-runtime # Set non-interactive frontend ENV DEBIAN_FRONTEND=noninteractive # Install venv, and system dependencies RUN apt-get update && apt-get install -y \ python3-venv \ libsndfile1 \ ffmpeg \ curl \ && apt-get clean && rm -rf /var/lib/apt/lists/* # Create non-root user and set up directories RUN useradd -m -u 1001 appuser && \ mkdir -p /app/outputs /app && \ chown -R appuser:appuser /app USER appuser WORKDIR /app # Copy all code (including pyproject.toml) COPY --chown=appuser:appuser . . # Create and activate virtual environment RUN python3 -m venv /app/venv ENV PATH="/app/venv/bin:$PATH" # Install all project dependencies RUN pip install --upgrade pip && pip install --no-cache-dir . # Set environment variables ENV PYTHONUNBUFFERED=1 \ PYTHONPATH=/app \ USE_GPU=true \ LD_LIBRARY_PATH=/usr/local/cuda/lib64:/usr/local/cuda-12.1/lib64:${LD_LIBRARY_PATH} # Expose Gradio default port ENV GRADIO_SERVER_NAME="0.0.0.0" EXPOSE 7860 # Entrypoint CMD ["python3", "app.py"] ``` ## /example/simple.py ```py path="/example/simple.py" from dia.model import Dia model = Dia.from_pretrained("nari-labs/Dia-1.6B", compute_dtype="float16") text = "[S1] Dia is an open weights text to dialogue model. [S2] You get full control over scripts and voices. [S1] Wow. Amazing. (laughs) [S2] Try it now on Git hub or Hugging Face." output = model.generate(text, use_torch_compile=True, verbose=True) model.save_audio("simple.mp3", output) ``` ## /example/voice_clone.py ```py path="/example/voice_clone.py" from dia.model import Dia model = Dia.from_pretrained("nari-labs/Dia-1.6B", compute_dtype="float16") # You should put the transcript of the voice you want to clone # We will use the audio created by running simple.py as an example. # Note that you will be REQUIRED TO RUN simple.py for the script to work as-is. clone_from_text = "[S1] Dia is an open weights text to dialogue model. [S2] You get full control over scripts and voices. [S1] Wow. Amazing. (laughs) [S2] Try it now on Git hub or Hugging Face." clone_from_audio = "simple.mp3" # For your custom needs, replace above with below and add your audio file to this directory: # clone_from_text = "[S1] ... [S2] ... [S1] ... corresponding to your_audio_name.mp3" # clone_from_audio = "your_audio_name.mp3" # Text to generate text_to_generate = "[S1] Hello, how are you? [S2] I'm good, thank you. [S1] What's your name? [S2] My name is Dia. [S1] Nice to meet you. [S2] Nice to meet you too." # It will only return the audio from the text_to_generate output = model.generate( clone_from_text + text_to_generate, audio_prompt=clone_from_audio, use_torch_compile=True, verbose=True ) model.save_audio("voice_clone.mp3", output) ``` ## /example_prompt.mp3 Binary file available at https://raw.githubusercontent.com/nari-labs/dia/refs/heads/main/example_prompt.mp3 ## /pyproject.toml ```toml path="/pyproject.toml" [project] name = "nari-tts" version = "0.1.0" description = "Dia - A text-to-speech model for dialogue generation" readme = "README.md" requires-python = ">=3.10" license = {file = "LICENSE"} authors = [ {name = "Nari Labs", email = "contact@narilabs.ai"} ] dependencies = [ "descript-audio-codec>=1.0.0", "gradio>=5.25.2", "huggingface-hub>=0.30.2", "numpy>=2.2.4", "pydantic>=2.11.3", "safetensors>=0.5.3", "soundfile>=0.13.1", "torch==2.6.0", "torchaudio==2.6.0", "triton==3.2.0 ; sys_platform == 'linux'", "triton-windows==3.2.0.post18 ; sys_platform == 'win32'", ] [build-system] requires = ["hatchling"] build-backend = "hatchling.build" [project.urls] "Homepage" = "https://github.com/nari-labs/dia" "Bug Tracker" = "https://github.com/nari-labs/dia/issues" [tool.hatch.build.targets.wheel] packages = ["dia"] [tool.ruff] # Never enforce `E501` (line length violations). lint.ignore = ["C901", "E501", "E741", "W605"] lint.select = ["C", "E", "F", "I", "W"] line-length = 119 # Ignore import violations in all `__init__.py` files. [tool.ruff.lint.per-file-ignores] "__init__.py" = ["E402", "F401", "F403", "F811"] [tool.ruff.lint.isort] lines-after-imports = 2 [tool.uv.sources] torch = [ { index = "pytorch-cu126", marker = "sys_platform == 'linux' or sys_platform == 'win32'" }, ] torchaudio = [ { index = "pytorch-cu126", marker = "sys_platform == 'linux' or sys_platform == 'win32'" }, ] [[tool.uv.index]] name = "pytorch-cu126" url = "https://download.pytorch.org/whl/cu126" explicit = true [dependency-groups] dev = [ "ninja>=1.11.1.4", "packaging>=25.0", ] ``` The content has been capped at 50000 tokens, and files over NaN bytes have been omitted. The user could consider applying other filters to refine the result. The better and more specific the context, the better the LLM can follow instructions. If the context seems verbose, the user can refine the filter using uithub. Thank you for using https://uithub.com - Perfect LLM context for any GitHub repo.