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@@ -58,3 +58,7 @@ saved_model/**/* filter=lfs diff=lfs merge=lfs -text
 # Video files - compressed
 *.mp4 filter=lfs diff=lfs merge=lfs -text
 *.webm filter=lfs diff=lfs merge=lfs -text
+VibeVoice-finetuning/wandb/run-20260218_142500-puguclmi/run-puguclmi.wandb filter=lfs diff=lfs merge=lfs -text
+VibeVoice-finetuning/wandb/run-20260218_143617-09tsct60/run-09tsct60.wandb filter=lfs diff=lfs merge=lfs -text
+VibeVoice-finetuning/wandb/run-20260218_144236-a0h99ykt/run-a0h99ykt.wandb filter=lfs diff=lfs merge=lfs -text
+VibeVoice-finetuning/wandb/run-20260219_165256-e29g8fsh/run-e29g8fsh.wandb filter=lfs diff=lfs merge=lfs -text

VibeVoice-finetuning/LICENSE

@@ -0,0 +1,21 @@
+MIT License
+
+Copyright (c) 2025 Resemble AI
+
+Permission is hereby granted, free of charge, to any person obtaining a copy
+of this software and associated documentation files (the "Software"), to deal
+in the Software without restriction, including without limitation the rights
+to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+copies of the Software, and to permit persons to whom the Software is
+furnished to do so, subject to the following conditions:
+
+The above copyright notice and this permission notice shall be included in all
+copies or substantial portions of the Software.
+
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+SOFTWARE.

VibeVoice-finetuning/README.md

@@ -0,0 +1,762 @@
+
+  
+
+# Unofficial WIP Finetuning repo for VibeVoice
+
+  
+
+# Hardware requirements
+
+  
+
+To train a VibeVoice 1.5B LoRa, a machine with at least 16gb VRAM is recommended.
+
+To train a VibeVoice 7B LoRa, a machine with at least 48gb VRAM is recommended.
+
+Keep in mind longer audios increase VRAM requirements
+
+  
+
+# Installation
+
+It is recommended to install this in a fresh environment. Specifically, the Dockerized environment `runpod/pytorch:2.8.0-py3.11-cuda12.8.1-cudnn-devel-ubuntu22.04` has been tested to work.
+
+  
+
+Transformers version 4.51.3 is known to work, while other versions have errors related to Qwen2 architecture.
+
+  
+```
+git clone https://github.com/voicepowered-ai/VibeVoice-finetuning
+
+pip install -e .
+
+pip uninstall -y transformers && pip install transformers==4.51.3
+
+(OPTIONAL) wandb login
+
+(OPTIONAL) export HF_HOME=/workspace/hf_models
+```
+
+  
+
+# Usage
+
+  
+
+## VibeVoice 1.5B / 7B (LoRA) fine-tuning
+
+  
+
+  
+
+We put some code together for training VibeVoice (7B) with LoRA. This uses the vendored VibeVoice model/processor and trains with a dual loss: masked CE on text tokens plus diffusion MSE on acoustic latents.
+
+  
+
+  
+
+Requirements:
+
+  
+
+- Download a compatible VibeVoice 7B or 1.5b checkpoint (config + weights) and its processor files (preprocessor_config.json) or run straight from HF model.
+
+- A 24khz audio dataset with audio files (target audio), text prompts (transcriptions) and optionally voice prompts (reference audio)
+
+  
+
+  
+  
+
+### Training with Hugging Face Dataset
+
+  
+```
+python -m src.finetune_vibevoice_lora \
+
+--model_name_or_path aoi-ot/VibeVoice-Large \
+
+--processor_name_or_path src/vibevoice/processor \
+
+--dataset_name your/dataset \
+
+--text_column_name text \
+
+--audio_column_name audio \
+
+--voice_prompts_column_name voice_prompts \
+
+--output_dir outputTrain3 \
+
+--per_device_train_batch_size 8 \
+
+--gradient_accumulation_steps 16 \
+
+--learning_rate 2.5e-5 \
+
+--num_train_epochs 5 \
+
+--logging_steps 10 \
+
+--save_steps 100 \
+
+--eval_steps 100 \
+
+--report_to wandb \
+
+--remove_unused_columns False \
+
+--bf16 True \
+
+--do_train \
+
+--gradient_clipping \
+
+--gradient_checkpointing False \
+
+--ddpm_batch_mul 4 \
+
+--diffusion_loss_weight 1.4 \
+
+--train_diffusion_head True \
+
+--ce_loss_weight 0.04 \
+
+--voice_prompt_drop_rate 0.2 \
+
+--lora_target_modules q_proj,k_proj,v_proj,o_proj,gate_proj,up_proj,down_proj \
+
+--lr_scheduler_type cosine \
+
+--warmup_ratio 0.03 \
+
+--max_grad_norm 0.8
+```
+  
+
+----------
+
+  
+
+### Training with Local JSONL Dataset
+
+  
+```
+python -m src.finetune_vibevoice_lora \
+
+--model_name_or_path aoi-ot/VibeVoice-Large \
+
+--processor_name_or_path src/vibevoice/processor \
+
+--train_jsonl prompts.jsonl \
+
+--text_column_name text \
+
+--audio_column_name audio \
+
+--output_dir outputTrain3 \
+
+--per_device_train_batch_size 8 \
+
+--gradient_accumulation_steps 16 \
+
+--learning_rate 2.5e-5 \
+
+--num_train_epochs 5 \
+
+--logging_steps 10 \
+
+--save_steps 100 \
+
+--report_to wandb \
+
+--remove_unused_columns False \
+
+--bf16 True \
+
+--do_train \
+
+--gradient_clipping \
+
+--gradient_checkpointing False \
+
+--ddpm_batch_mul 4 \
+
+--diffusion_loss_weight 1.4 \
+
+--train_diffusion_head True \
+
+--ce_loss_weight 0.04 \
+
+--voice_prompt_drop_rate 0.2 \
+
+--lora_target_modules q_proj,k_proj,v_proj,o_proj,gate_proj,up_proj,down_proj \
+
+--lr_scheduler_type cosine \
+
+--warmup_ratio 0.03 \
+
+--max_grad_norm 0.8
+```
+
+
+### JSONL format:
+
+  
+
+You can provide an optional `voice_prompts` key. If it is omitted, a voice prompt will be automatically generated from the target audio.
+
+  
+
+**Example without a pre-defined voice prompt (will be auto-generated):**
+
+`{"text": "Speaker 0: Speaker0 transcription.", "audio": "/workspace/wavs/segment_000000.wav"}`
+
+  
+
+**Example with a pre-defined voice prompt:**
+
+`{"text": "Speaker 0: Speaker0 transcription.", "audio": "/workspace/wavs/segment_000000.wav", "voice_prompts": "/path/to/a/different/prompt.wav"}`
+
+  
+
+**Example with multiple speakers and voice prompts:**
+
+`{"text": "Speaker 0: How is the project coming along?\nSpeaker 1: It's going well, we should be finished by Friday.", "audio": "/data/conversations/convo_01.wav", "voice_prompts": ["/data/prompts/alice_voice_prompt.wav", "/data/prompts/bob_voice_prompt.wav"]}`
+
+  
+  
+  
+
+# Notes:
+
+  
+
+- Audio is assumed to be 24 kHz; input audio will be loaded/resampled to 24 kHz.
+
+  
+
+- If you pass raw NumPy arrays or torch Tensors as audio (without sampling rate metadata), the collator assumes they are already 24 kHz. To trigger resampling, provide dicts like {"array": <np.ndarray>, "sampling_rate": <int>} or file paths.
+
+  
+
+- Tokenizers (acoustic/semantic) are frozen by default. LoRA is applied to the LLM (Qwen) and optionally to the diffusion head.
+
+  
+
+- The collator builds interleaved sequences with speech placeholders and computes the required masks for diffusion loss.
+
+- If a voice_prompts column is not provided in your dataset for a given sample, a voice prompt is **automatically generated** by taking a random clip from the target audio. This fallback ensures the model's voice cloning ability is maintained. You can override this behavior by providing your own voice prompts.
+
+- Said voice_prompts are randomly dropped during training to improve generalization. Drop rates of 0.2 and 0.25 have been tested with satisfactory results.
+
+  
+
+- The model learns to emit a closing `[speech_end]` token after target placeholders.
+
+  
+
+- For multi‑speaker prompts, ensure `voice_prompts` list order matches `Speaker 0/1/...` tags in your text.
+
+  
+
+- LoRA adapters are saved under `output_dir/lora` after training.
+
+  
+
+  
+
+# Acknowledgements
+
+  
+
+- [VibeVoice](https://github.com/microsoft/VibeVoice)
+
+  
+
+- [chatterbox-finetuning](https://github.com/stlohrey/chatterbox-finetuning)
+
+  
+  
+
+## Training Script Arguments
+
+  
+
+Comprehensive list of all the command-line arguments available for the fine-tuning script.
+
+  
+
+### Model & Architecture Arguments
+
+Controls the base model, its configuration, and which components are trained.
+
+  
+
+*  `--model_name_or_path`
+
+*  **What it does:** Specifies the path to the pretrained VibeVoice base model. This can be a local directory or a Hugging Face Hub repository ID.
+
+*  **Required:** Yes.
+
+*  **Example:**
+
+```bash
+
+--model_name_or_path aoi-ot/VibeVoice-Large
+
+```
+
+  
+
+*  `--processor_name_or_path`
+
+*  **What it does:** Specifies the path to the VibeVoice processor configuration. If not provided, it defaults to the `model_name_or_path`.
+
+*  **Example:**
+
+```bash
+
+--processor_name_or_path src/vibevoice/processor
+
+```
+
+  
+
+*  `--train_diffusion_head`
+
+*  **What it does:** A boolean flag to enable **full fine-tuning** of the diffusion prediction head. When enabled, all parameters of the diffusion head become trainable.
+
+*  **Example:**
+
+```bash
+
+--train_diffusion_head True
+
+```
+
+  
+
+*  `--train_connectors`
+
+*  **What it does:** A boolean flag to enable training of the acoustic and semantic connectors, which bridge different parts of the model.
+
+*  **Example:**
+
+```bash
+
+--train_connectors True
+
+```
+
+  
+
+*  `--lora_target_modules`
+
+*  **What it does:** A comma-separated string of module names within the language model to apply LoRA adapters to. This is the primary way to enable LoRA for the text-processing part of the model.
+
+*  **Example:**
+
+```bash
+
+--lora_target_modules q_proj,k_proj,v_proj,o_proj,gate_proj,up_proj,down_proj
+
+```
+
+  
+
+*  `--lora_r`
+
+*  **What it does:** The rank (`r`) of the LoRA decomposition. A smaller number means fewer trainable parameters.
+
+*  **Default:**  `8`
+
+*  **Example:**
+
+```bash
+
+--lora_r 16
+
+```
+
+  
+
+*  `--lora_alpha`
+
+*  **What it does:** The scaling factor for the LoRA weights. A common practice is to set `lora_alpha` to be four times the value of `lora_r`.
+
+*  **Default:**  `32`
+
+*  **Example:**
+
+```bash
+
+--lora_alpha 64
+
+```
+
+*  `--lora_wrap_diffusion_head`
+
+*  **What it does:** An **alternative** to `--train_diffusion_head`. If `True`, it applies LoRA adapters to the diffusion head instead of fine-tuning it fully, enabling more parameter-efficient training of the head. Must only use `--train_diffusion_head` or `--lora_wrap_diffusion_head`
+
+*  **Default:**  `False`
+
+  
+  
+  
+
+*  `--layers_to_freeze`
+
+*  **What it does:** Comma-separated indices of diffusion head layers to freeze (e.g., '0,1,5,7,8'). [Diffusion head layer indices](https://github.com/voicepowered-ai/VibeVoice-finetuning/blob/main/diff_head_layers.txt)
+
+*  **Default:**  `None`
+
+### Data & Processing Arguments
+
+Defines the dataset to be used, its structure, and how it should be processed.
+
+  
+
+*  `--train_jsonl`
+
+*  **What it does:** Path to your local training data file in JSONL (JSON Lines) format. Each line should be a JSON object with keys for text and audio path.
+
+*  **Example:**
+
+```bash
+
+--train_jsonl prompts.jsonl
+
+```
+
+  
+
+*  `--validation_jsonl`
+
+*  **What it does:** Optional path to a local validation data file in JSONL format.
+
+*  **Example:**
+
+```bash
+
+--validation_jsonl validation_prompts.jsonl
+
+```
+
+  
+
+*  `--text_column_name`
+
+*  **What it does:** The name of the key in your JSONL file that contains the text transcription/prompt.
+
+*  **Default:**  `text`
+
+*  **Example:**
+
+```bash
+
+--text_column_name "prompt"
+
+```
+
+  
+
+*  `--audio_column_name`
+
+*  **What it does:** The name of the key in your JSONL file that contains the path to the audio file.
+
+*  **Default:**  `audio`
+
+*  **Example:**
+
+```bash
+
+--audio_column_name "file_path"
+
+```
+
+  
+
+*  `--voice_prompt_drop_rate`
+
+*  **What it does:** The probability (from 0.0 to 1.0) of randomly dropping the conditioning voice prompt during training. This acts as a regularizer.
+
+*  **Default:**  `0.0`
+
+*  **Example:**
+
+```bash
+
+--voice_prompt_drop_rate 0.2
+
+```
+
+  
+
+### Core Training Arguments
+
+Standard Hugging Face `TrainingArguments` that control the training loop, optimizer, and saving.
+
+  
+
+*  `--output_dir`
+
+*  **What it does:** The directory where model checkpoints and final outputs will be saved.
+
+*  **Required:** Yes.
+
+*  **Example:**
+
+```bash
+
+--output_dir output_model
+
+```
+
+  
+
+*  `--per_device_train_batch_size`
+
+*  **What it does:** The number of training examples processed per GPU in a single step.
+
+*  **Example:**
+
+```bash
+
+--per_device_train_batch_size 8
+
+```
+
+  
+
+*  `--gradient_accumulation_steps`
+
+*  **What it does:** The number of forward passes to accumulate gradients for before performing an optimizer step. This effectively increases the batch size without using more VRAM.
+
+*  **Example:**
+
+```bash
+
+--gradient_accumulation_steps 16
+
+```
+
+  
+
+*  `--learning_rate`
+
+*  **What it does:** The initial learning rate for the optimizer.
+
+*  **Example:**
+
+```bash
+
+--learning_rate 2.5e-5
+
+```
+
+  
+
+*  `--num_train_epochs`
+
+*  **What it does:** The total number of times to iterate over the entire training dataset.
+
+*  **Example:**
+
+```bash
+
+--num_train_epochs 5
+
+```
+
+  
+
+*  `--logging_steps`
+
+*  **What it does:** How often (in steps) to log training metrics like loss.
+
+*  **Example:**
+
+```bash
+
+--logging_steps 10
+
+```
+
+  
+
+*  `--save_steps`
+
+*  **What it does:** How often (in steps) to save a model checkpoint.
+
+*  **Example:**
+
+```bash
+
+--save_steps 100
+
+```
+
+  
+
+*  `--report_to`
+
+*  **What it does:** The integration to report logs to. Can be `wandb`, `tensorboard`, or `none`.
+
+*  **Example:**
+
+```bash
+
+--report_to wandb
+
+```
+
+  
+
+*  `--remove_unused_columns`
+
+*  **What it does:** Whether to remove columns from the dataset not used by the model's `forward` method. **This must be set to `False`** for this script to work correctly.
+
+*  **Example:**
+
+```bash
+
+--remove_unused_columns False
+
+```
+
+  
+
+*  `--bf16`
+
+*  **What it does:** Enables mixed-precision training using `bfloat16`. This speeds up training and reduces memory usage on compatible GPUs (NVIDIA Ampere series and newer).
+
+*  **Example:**
+
+```bash
+
+--bf16 True
+
+```
+
+  
+
+*  `--gradient_checkpointing`
+
+*  **What it does:** A memory-saving technique that trades compute for memory. Useful for training large models on limited VRAM.
+
+*  **Example:**
+
+```bash
+
+--gradient_checkpointing True
+
+```
+
+  
+
+*  `--lr_scheduler_type`
+
+*  **What it does:** The type of learning rate schedule to use (e.g., `linear`, `cosine`, `constant`).
+
+*  **Example:**
+
+```bash
+
+--lr_scheduler_type cosine
+
+```
+
+  
+
+*  `--warmup_ratio`
+
+*  **What it does:** The proportion of total training steps used for a linear warmup from 0 to the initial learning rate.
+
+*  **Example:**
+
+```bash
+
+--warmup_ratio 0.03
+
+```
+
+  
+
+### Custom VibeVoice Training Arguments
+
+Special arguments to control VibeVoice-specific training behaviors.
+
+  
+
+*  `--gradient_clipping`
+
+*  **What it does:** A custom boolean flag that acts as the master switch for gradient clipping. If you include this flag, the value from `--max_grad_norm` will be used to prevent exploding gradients.
+
+*  **Example:**
+
+```bash
+
+--gradient_clipping
+
+```
+
+*  `--max_grad_norm`
+
+*  **What it does:** The maximum value for gradient clipping. Only active if `--gradient_clipping` is also used.
+
+*  **Default:**  `1.0`
+
+*  **Example:**
+
+```bash
+
+--max_grad_norm 0.8
+
+```
+
+  
+
+*  `--diffusion_loss_weight`
+
+*  **What it does:** A float that scales the importance of the diffusion loss (for speech generation quality) in the total loss calculation.
+
+*  **Example:**
+
+```bash
+
+--diffusion_loss_weight 1.4
+
+```
+
+  
+
+*  `--ce_loss_weight`
+
+*  **What it does:** A float that scales the importance of the Cross-Entropy loss (for text prediction accuracy) in the total loss calculation.
+
+*  **Example:**
+
+```bash
+
+--ce_loss_weight 0.04
+
+```
+
+  
+
+*  `--ddpm_batch_mul`
+
+*  **What it does:** An integer multiplier for the batch size used specifically within the diffusion process.
+
+*  **Example:**
+
+```bash
+
+--ddpm_batch_mul 4
+
+
+```
+
+

VibeVoice-finetuning/checkpoint-1800/lora/acoustic_connector/pytorch_model.bin

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VibeVoice-finetuning/checkpoint-1800/lora/config.json

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+{
+  "architectures": [
+    "Qwen2Model"
+  ],
+  "attention_dropout": 0.0,
+  "hidden_act": "silu",
+  "hidden_size": 1536,
+  "initializer_range": 0.02,
+  "intermediate_size": 8960,
+  "max_position_embeddings": 65536,
+  "max_window_layers": 28,
+  "model_type": "qwen2",
+  "num_attention_heads": 12,
+  "num_hidden_layers": 28,
+  "num_key_value_heads": 2,
+  "rms_norm_eps": 1e-06,
+  "rope_scaling": null,
+  "rope_theta": 1000000.0,
+  "sliding_window": null,
+  "tie_word_embeddings": true,
+  "torch_dtype": "float16",
+  "transformers_version": "4.51.3",
+  "use_cache": true,
+  "use_sliding_window": false,
+  "vocab_size": 151936
+}

VibeVoice-finetuning/checkpoint-1800/lora/diffusion_head/config.json

@@ -0,0 +1,20 @@
+{
+  "architectures": [
+    "VibeVoiceDiffusionHead"
+  ],
+  "ddpm_batch_mul": 4,
+  "ddpm_beta_schedule": "cosine",
+  "ddpm_num_inference_steps": 20,
+  "ddpm_num_steps": 1000,
+  "diffusion_type": "ddpm",
+  "head_ffn_ratio": 3.0,
+  "head_layers": 4,
+  "hidden_size": 1536,
+  "latent_size": 64,
+  "model_type": "vibevoice_diffusion_head",
+  "prediction_type": "v_prediction",
+  "rms_norm_eps": 1e-05,
+  "speech_vae_dim": 64,
+  "torch_dtype": "float32",
+  "transformers_version": "4.51.3"
+}

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VibeVoice-finetuning/checkpoint-1800/scaler.pt

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VibeVoice-finetuning/checkpoint-1800/scheduler.pt

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VibeVoice-finetuning/diff_head_layers.txt

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+[0] noisy_images_proj.weight (shape: (3584, 64), trainable: True)
+[1] cond_proj.weight (shape: (3584, 3584), trainable: True)
+[2] t_embedder.mlp.0.weight (shape: (3584, 256), trainable: True)
+[3] t_embedder.mlp.2.weight (shape: (3584, 3584), trainable: True)
+[4] layers.0.ffn.gate_proj.weight (shape: (10752, 3584), trainable: True)
+[5] layers.0.ffn.up_proj.weight (shape: (10752, 3584), trainable: True)
+[6] layers.0.ffn.down_proj.weight (shape: (3584, 10752), trainable: True)
+[7] layers.0.norm.weight (shape: (3584,), trainable: True)
+[8] layers.0.adaLN_modulation.1.weight (shape: (10752, 3584), trainable: True)
+[9] layers.1.ffn.gate_proj.weight (shape: (10752, 3584), trainable: True)
+[10] layers.1.ffn.up_proj.weight (shape: (10752, 3584), trainable: True)
+[11] layers.1.ffn.down_proj.weight (shape: (3584, 10752), trainable: True)
+[12] layers.1.norm.weight (shape: (3584,), trainable: True)
+[13] layers.1.adaLN_modulation.1.weight (shape: (10752, 3584), trainable: True)
+[14] layers.2.ffn.gate_proj.weight (shape: (10752, 3584), trainable: True)
+[15] layers.2.ffn.up_proj.weight (shape: (10752, 3584), trainable: True)
+[16] layers.2.ffn.down_proj.weight (shape: (3584, 10752), trainable: True)
+[17] layers.2.norm.weight (shape: (3584,), trainable: True)
+[18] layers.2.adaLN_modulation.1.weight (shape: (10752, 3584), trainable: True)
+[19] layers.3.ffn.gate_proj.weight (shape: (10752, 3584), trainable: True)
+[20] layers.3.ffn.up_proj.weight (shape: (10752, 3584), trainable: True)
+[21] layers.3.ffn.down_proj.weight (shape: (3584, 10752), trainable: True)
+[22] layers.3.norm.weight (shape: (3584,), trainable: True)
+[23] layers.3.adaLN_modulation.1.weight (shape: (10752, 3584), trainable: True)
+[24] final_layer.linear.weight (shape: (64, 3584), trainable: True)
+[25] final_layer.adaLN_modulation.1.weight (shape: (7168, 3584), trainable: True)

VibeVoice-finetuning/lora/acoustic_connector/pytorch_model.bin

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VibeVoice-finetuning/lora/config.json

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+{
+  "architectures": [
+    "Qwen2Model"
+  ],
+  "attention_dropout": 0.0,
+  "hidden_act": "silu",
+  "hidden_size": 1536,
+  "initializer_range": 0.02,
+  "intermediate_size": 8960,
+  "max_position_embeddings": 65536,
+  "max_window_layers": 28,
+  "model_type": "qwen2",
+  "num_attention_heads": 12,
+  "num_hidden_layers": 28,
+  "num_key_value_heads": 2,
+  "rms_norm_eps": 1e-06,
+  "rope_scaling": null,
+  "rope_theta": 1000000.0,
+  "sliding_window": null,
+  "tie_word_embeddings": true,
+  "torch_dtype": "float16",
+  "transformers_version": "4.51.3",
+  "use_cache": true,
+  "use_sliding_window": false,
+  "vocab_size": 151936
+}

VibeVoice-finetuning/lora/diffusion_head/config.json

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+{
+  "architectures": [
+    "VibeVoiceDiffusionHead"
+  ],
+  "ddpm_batch_mul": 4,
+  "ddpm_beta_schedule": "cosine",
+  "ddpm_num_inference_steps": 20,
+  "ddpm_num_steps": 1000,
+  "diffusion_type": "ddpm",
+  "head_ffn_ratio": 3.0,
+  "head_layers": 4,
+  "hidden_size": 1536,
+  "latent_size": 64,
+  "model_type": "vibevoice_diffusion_head",
+  "prediction_type": "v_prediction",
+  "rms_norm_eps": 1e-05,
+  "speech_vae_dim": 64,
+  "torch_dtype": "float32",
+  "transformers_version": "4.51.3"
+}

VibeVoice-finetuning/lora/diffusion_head/diffusion_head_full.bin

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VibeVoice-finetuning/lora/semantic_connector/pytorch_model.bin

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+size 5123867

VibeVoice-finetuning/preprocessed/.gitattributes

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+*.7z filter=lfs diff=lfs merge=lfs -text
+*.arrow filter=lfs diff=lfs merge=lfs -text
+*.avro filter=lfs diff=lfs merge=lfs -text
+*.bin filter=lfs diff=lfs merge=lfs -text
+*.bz2 filter=lfs diff=lfs merge=lfs -text
+*.ckpt filter=lfs diff=lfs merge=lfs -text
+*.ftz filter=lfs diff=lfs merge=lfs -text
+*.gz filter=lfs diff=lfs merge=lfs -text
+*.h5 filter=lfs diff=lfs merge=lfs -text
+*.joblib filter=lfs diff=lfs merge=lfs -text
+*.lfs.* filter=lfs diff=lfs merge=lfs -text
+*.lz4 filter=lfs diff=lfs merge=lfs -text
+*.mds filter=lfs diff=lfs merge=lfs -text
+*.mlmodel filter=lfs diff=lfs merge=lfs -text
+*.model filter=lfs diff=lfs merge=lfs -text
+*.msgpack filter=lfs diff=lfs merge=lfs -text
+*.npy filter=lfs diff=lfs merge=lfs -text
+*.npz filter=lfs diff=lfs merge=lfs -text
+*.onnx filter=lfs diff=lfs merge=lfs -text
+*.ot filter=lfs diff=lfs merge=lfs -text
+*.parquet filter=lfs diff=lfs merge=lfs -text
+*.pb filter=lfs diff=lfs merge=lfs -text
+*.pickle filter=lfs diff=lfs merge=lfs -text
+*.pkl filter=lfs diff=lfs merge=lfs -text
+*.pt filter=lfs diff=lfs merge=lfs -text
+*.pth filter=lfs diff=lfs merge=lfs -text
+*.rar filter=lfs diff=lfs merge=lfs -text
+*.safetensors filter=lfs diff=lfs merge=lfs -text
+saved_model/**/* filter=lfs diff=lfs merge=lfs -text
+*.tar.* filter=lfs diff=lfs merge=lfs -text
+*.tar filter=lfs diff=lfs merge=lfs -text
+*.tflite filter=lfs diff=lfs merge=lfs -text
+*.tgz filter=lfs diff=lfs merge=lfs -text
+*.wasm filter=lfs diff=lfs merge=lfs -text
+*.xz filter=lfs diff=lfs merge=lfs -text
+*.zip filter=lfs diff=lfs merge=lfs -text
+*.zst filter=lfs diff=lfs merge=lfs -text
+*tfevents* filter=lfs diff=lfs merge=lfs -text
+# Audio files - uncompressed
+*.pcm filter=lfs diff=lfs merge=lfs -text
+*.sam filter=lfs diff=lfs merge=lfs -text
+*.raw filter=lfs diff=lfs merge=lfs -text
+# Audio files - compressed
+*.aac filter=lfs diff=lfs merge=lfs -text
+*.flac filter=lfs diff=lfs merge=lfs -text
+*.mp3 filter=lfs diff=lfs merge=lfs -text
+*.ogg filter=lfs diff=lfs merge=lfs -text
+*.wav filter=lfs diff=lfs merge=lfs -text
+# Image files - uncompressed
+*.bmp filter=lfs diff=lfs merge=lfs -text
+*.gif filter=lfs diff=lfs merge=lfs -text
+*.png filter=lfs diff=lfs merge=lfs -text
+*.tiff filter=lfs diff=lfs merge=lfs -text
+# Image files - compressed
+*.jpg filter=lfs diff=lfs merge=lfs -text
+*.jpeg filter=lfs diff=lfs merge=lfs -text
+*.webp filter=lfs diff=lfs merge=lfs -text
+# Video files - compressed
+*.mp4 filter=lfs diff=lfs merge=lfs -text
+*.webm filter=lfs diff=lfs merge=lfs -text

VibeVoice-finetuning/preprocessed/preprocessed_batches.pt

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VibeVoice-finetuning/pyproject.toml

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+[project]
+name = "vibevoice-finetuning"
+version = "0.1.0"
+description = "Open Source finetuning code for VibeVoice"
+readme = "README.md"
+requires-python = ">=3.8"
+license = {file = "LICENSE"}
+authors = [
+    {name = "jpgallegoarvpb", email = "juanpablo.gallego@voicepowered.ai"}
+]
+dependencies = [
+    "numpy~=1.26.0",
+    "resampy==0.4.3",
+    "librosa==0.11.0",
+    "s3tokenizer",
+    "torch",
+    "torchaudio",
+    "transformers",
+    "datasets>=2.18.0",
+    "diffusers==0.29.0",
+    "resemble-perth==1.0.1",
+    "omegaconf==2.3.0",
+    "conformer==0.3.2",
+    "safetensors==0.5.3",
+    "peft>=0.11.0",
+    "tensorboard>=2.12",
+    "wandb"
+]
+
+[build-system]
+requires = ["setuptools>=61.0"]
+build-backend = "setuptools.build_meta"
+
+[tool.setuptools.packages.find]
+where = ["src"]

VibeVoice-finetuning/src/data_vibevoice.py

@@ -0,0 +1,453 @@
+import math
+from dataclasses import dataclass
+from typing import Any, Dict, List, Optional, Sequence, Tuple, Union
+
+import numpy as np
+import torch
+import warnings
+import random
+
+try:
+    import librosa  # type: ignore
+except Exception:  # pragma: no cover
+    librosa = None  # Fallback: user must install librosa when using local audio paths
+
+try:
+    import resampy  # type: ignore
+except Exception:  # pragma: no cover
+    resampy = None
+
+
+def _resample_if_needed(wav: np.ndarray, orig_sr: int, target_sr: int) -> np.ndarray:
+    if orig_sr == target_sr:
+        return wav.astype(np.float32, copy=False)
+    if resampy is not None:
+        return resampy.resample(wav.astype(np.float32), orig_sr, target_sr)
+    if librosa is not None:
+        return librosa.resample(y=wav.astype(np.float32), orig_sr=orig_sr, target_sr=target_sr)
+    warnings.warn(
+        "No resampler available; treating audio as target_sr without resampling. Install resampy or librosa.",
+        RuntimeWarning,
+    )
+    return wav.astype(np.float32, copy=False)
+
+
+# Lightweight HF-style dataset wrapper (optional). Trainer can also pass raw HF datasets directly.
+class VibeVoiceDataset:
+    def __init__(
+        self,
+        dataset: Any,
+        text_column: str = "text",
+        audio_column: str = "audio",
+        voice_prompts_column: Optional[str] = "voice_prompts",
+    ) -> None:
+        self.dataset = dataset
+        self.text_column = text_column
+        self.audio_column = audio_column
+        self.voice_prompts_column = voice_prompts_column
+
+    def __len__(self) -> int:
+        return len(self.dataset)
+
+    def __getitem__(self, idx: int) -> Dict[str, Any]:
+        item = self.dataset[idx]
+        data: Dict[str, Any] = {}
+        data["text"] = item[self.text_column]
+        data["audio"] = item[self.audio_column]
+
+        user_provided_prompt = None
+        if self.voice_prompts_column and self.voice_prompts_column in item:
+            user_provided_prompt = item[self.voice_prompts_column]
+
+        if user_provided_prompt:
+            # A prompt was provided in the dataset, so we use it.
+            if not isinstance(user_provided_prompt, list):
+                data["voice_prompts"] = [user_provided_prompt]
+            else:
+                data["voice_prompts"] = user_provided_prompt
+        else:
+            # FALLBACK: No prompt provided, so we auto-generate one from the target audio.
+            try:
+                target_sr = 24000
+                wav_array = _load_audio_to_24k(item[self.audio_column], target_sr=target_sr)
+                audio_len_seconds = len(wav_array) / target_sr
+
+                min_len_sec = min(5.0, audio_len_seconds / 4.0)
+                max_len_sec = min(15.0, audio_len_seconds / 2.0)
+                
+                if min_len_sec > max_len_sec:
+                    min_len_sec = max_len_sec
+                max_len_sec = min(max_len_sec, audio_len_seconds)
+
+                if max_len_sec > 0.1:
+                    prompt_len_sec = random.uniform(min_len_sec, max_len_sec)
+                    prompt_len_samples = int(prompt_len_sec * target_sr)
+
+                    max_start_sample = len(wav_array) - prompt_len_samples
+                    start_sample = random.randint(0, max_start_sample)
+                    
+                    prompt_crop = wav_array[start_sample : start_sample + prompt_len_samples]
+                    
+                    data["voice_prompts"] = [prompt_crop]
+                else:
+                    data["voice_prompts"] = None
+
+            except Exception as e:
+                warnings.warn(f"Could not create voice prompt for item {idx}: {e}")
+                data["voice_prompts"] = None            
+        return data
+
+
+def _apply_silence_with_crossfade(
+    wav: np.ndarray,
+    *,
+    sample_rate: int,
+    pre_silence_sec: float = 0.25,
+    pre_crossfade_sec: float = 0.25,
+    post_crossfade_sec: float = 0.25,
+    post_silence_sec: float = 0.75,
+) -> np.ndarray:
+    """Pad audio with leading/trailing silence and apply crossfades.
+
+    Structure: [pre_silence][pre_crossfade][audio_body][post_crossfade][post_silence]
+    Crossfades blend the audio with silence linearly to avoid hard edges.
+    """
+
+    wav = np.asarray(wav, dtype=np.float32).reshape(-1)
+
+    start_sil_samples = int(round(pre_silence_sec * sample_rate))
+    end_sil_samples = int(round(post_silence_sec * sample_rate))
+    pre_crossfade_samples = int(round(pre_crossfade_sec * sample_rate))
+    post_crossfade_samples = int(round(post_crossfade_sec * sample_rate))
+
+    total_len = wav.shape[0]
+    if total_len == 0:
+        pieces: List[np.ndarray] = []
+        if start_sil_samples > 0:
+            pieces.append(np.zeros(start_sil_samples, dtype=np.float32))
+        if end_sil_samples > 0:
+            pieces.append(np.zeros(end_sil_samples, dtype=np.float32))
+        return np.concatenate(pieces) if pieces else wav
+
+    start_len = min(pre_crossfade_samples, total_len)
+    remaining_after_start = max(total_len - start_len, 0)
+    end_len = min(post_crossfade_samples, remaining_after_start)
+    middle_end_idx = total_len - end_len
+
+    start_segment = wav[:start_len]
+    middle_segment = wav[start_len:middle_end_idx]
+    end_segment = wav[middle_end_idx:]
+
+    def _linear_fade(num_samples: int, start: float, end: float) -> np.ndarray:
+        if num_samples <= 0:
+            return np.zeros((0,), dtype=np.float32)
+        return np.linspace(start, end, num_samples, endpoint=True, dtype=np.float32)
+
+    start_crossfade = start_segment * _linear_fade(start_len, 0.0, 1.0)
+    end_crossfade = end_segment * _linear_fade(end_segment.shape[0], 1.0, 0.0)
+
+    pieces: List[np.ndarray] = []
+    if start_sil_samples > 0:
+        pieces.append(np.zeros(start_sil_samples, dtype=np.float32))
+    if start_crossfade.size > 0:
+        pieces.append(start_crossfade.astype(np.float32, copy=False))
+    if middle_segment.size > 0:
+        pieces.append(middle_segment.astype(np.float32, copy=False))
+    if end_crossfade.size > 0:
+        pieces.append(end_crossfade.astype(np.float32, copy=False))
+    if end_sil_samples > 0:
+        pieces.append(np.zeros(end_sil_samples, dtype=np.float32))
+
+    return np.concatenate(pieces)
+
+
+def _load_audio_to_24k(
+    audio: Union[str, np.ndarray, torch.Tensor, Dict[str, Any]],
+    *,
+    target_sr: int = 24000,
+    augment_with_silence: bool = False,
+) -> np.ndarray:
+    if isinstance(audio, np.ndarray):
+        wav_out = audio.astype(np.float32)
+    elif isinstance(audio, torch.Tensor):
+        wav_out = audio.detach().cpu().float().numpy()
+    elif isinstance(audio, str):
+        if librosa is None:
+            raise RuntimeError("librosa is required to load audio file paths. Please pip install librosa.")
+        wav, sr = librosa.load(audio, sr=None, mono=True)
+        wav_out = _resample_if_needed(wav, int(sr), target_sr)
+    elif isinstance(audio, dict) and "array" in audio and "sampling_rate" in audio:
+        arr = np.asarray(audio["array"], dtype=np.float32)
+        sr = int(audio["sampling_rate"])
+        wav_out = _resample_if_needed(arr, sr, target_sr)
+    else:
+        raise ValueError(f"Unsupported audio type: {type(audio)}")
+
+    wav_out = np.asarray(wav_out, dtype=np.float32)
+
+    if augment_with_silence:
+        wav_out = _apply_silence_with_crossfade(wav_out, sample_rate=target_sr)
+
+    return wav_out
+
+
+@dataclass
+class VibeVoiceCollator:
+    processor: Any  # VibeVoiceProcessor
+    max_length: Optional[int] = None
+    speech_compress_ratio: int = 3200
+    semantic_vae_dim: int = 128
+    compute_semantics: bool = False
+    debug_checks: bool = False
+
+    text_field: str = "text"
+    audio_field: str = "audio"
+    voice_prompts_field: str = "voice_prompts"
+    voice_prompt_drop_rate: float = 0.0
+
+    def __call__(self, features: Sequence[Dict[str, Any]]) -> Dict[str, Any]:
+        batch_size = len(features)
+
+        sample_input_ids: List[List[int]] = []
+        sample_attention_masks: List[List[int]] = []
+        sample_acoustic_input_masks: List[List[bool]] = []
+        sample_acoustic_loss_masks: List[List[bool]] = []
+
+        all_speech_waveforms: List[np.ndarray] = []
+        all_speech_latent_lengths: List[int] = []
+        per_segment_is_target: List[bool] = []
+
+        for ex in features:
+            text: str = ex.get(self.text_field, "")
+            voice_prompts: Optional[List[Union[str, np.ndarray, torch.Tensor]]] = ex.get(self.voice_prompts_field)
+            target_audio: Union[str, np.ndarray, torch.Tensor, Dict[str, Any]] = ex.get(self.audio_field)
+
+            # Clamp drop rate for safety
+            _drop_rate = self.voice_prompt_drop_rate
+            if _drop_rate < 0.0:
+                _drop_rate = 0.0
+            elif _drop_rate > 1.0:
+                _drop_rate = 1.0
+
+            proc = self.processor(
+                text=[text],
+                voice_samples=[voice_prompts] if voice_prompts is not None and random.random() >= _drop_rate else None,
+                padding=False,
+                truncation=False,
+                max_length=self.max_length,
+                return_tensors="pt",
+            )
+
+            ids = proc["input_ids"][0].tolist()
+            attn = proc.get("attention_mask", torch.ones_like(proc["input_ids"]))[0].tolist()
+            speech_input_mask = proc.get("speech_input_mask")
+            if speech_input_mask is None:
+                speech_input_mask = torch.zeros_like(proc["input_ids"], dtype=torch.bool)
+            speech_input_mask_list = speech_input_mask[0].tolist()
+
+            wav_target = _load_audio_to_24k(target_audio, target_sr=24000, augment_with_silence=True)
+            # Prefer exact frame count from acoustic tokenizer if available; fallback to compress ratio
+            target_latent_len = None
+            try:
+                acoustic_tok = getattr(self.processor, "acoustic_tokenizer", None)
+                if acoustic_tok is not None and hasattr(acoustic_tok, "encode"):
+                    enc_out = acoustic_tok.encode(wav_target)
+                    # Normalize various possible return formats to get time dimension
+                    T = None
+                    try:
+                        # Direct array-like with shape (T, D) or (T,)
+                        if hasattr(enc_out, "shape") and len(getattr(enc_out, "shape", [])) >= 1:
+                            T = int(enc_out.shape[0])
+                        else:
+                            # Nested lists/tuples or ModelOutput-like
+                            cand = enc_out
+                            # Drill down a couple of levels safely
+                            for _ in range(2):
+                                if isinstance(cand, (list, tuple)) and len(cand) > 0:
+                                    cand = cand[0]
+                            if hasattr(cand, "shape") and len(getattr(cand, "shape", [])) >= 1:
+                                T = int(cand.shape[0])
+                    except Exception:
+                        T = None
+                    if T is not None and T > 0:
+                        target_latent_len = T
+            except Exception:
+                target_latent_len = None
+            if target_latent_len is None:
+                target_latent_len = max(1, int(math.ceil(len(wav_target) / float(self.speech_compress_ratio))))
+
+            speech_diff_id = self.processor.tokenizer.speech_diffusion_id
+            target_placeholders = [speech_diff_id] * target_latent_len
+
+            ids_extended = ids + target_placeholders
+            attn_extended = attn + [1] * target_latent_len
+
+            acoustic_input_mask = speech_input_mask_list + [True] * target_latent_len
+            acoustic_loss_mask = ([False] * len(speech_input_mask_list)) + [True] * target_latent_len
+
+            speech_end_id = self.processor.tokenizer.speech_end_id
+            ids_extended.append(speech_end_id)
+            attn_extended.append(1)
+            acoustic_input_mask.append(False)
+            acoustic_loss_mask.append(False)
+
+            # Ensure text decoding sees an explicit end-of-sequence token after speech output.
+            eos_token_id = getattr(self.processor.tokenizer, "eos_id", None)
+            if eos_token_id is None:
+                eos_token_id = getattr(self.processor.tokenizer, "eos_token_id", None)
+            if eos_token_id is not None and eos_token_id >= 0:
+                ids_extended.append(eos_token_id)
+                attn_extended.append(1)
+                acoustic_input_mask.append(False)
+                acoustic_loss_mask.append(False)
+
+            if self.max_length is not None and len(ids_extended) > self.max_length:
+                cut = len(ids_extended) - int(self.max_length)
+                leading_non_acoustic = 0
+                for v in acoustic_input_mask:
+                    if v:
+                        break
+                    leading_non_acoustic += 1
+                if cut > leading_non_acoustic:
+                    raise ValueError(
+                        f"--max_length={self.max_length} would truncate into acoustic tokens. "
+                        f"Needed cut={cut}, but only {leading_non_acoustic} leading non-acoustic tokens available. "
+                        "Increase max_length or shorten text/voice-prompt preamble."
+                    )
+                ids_extended = ids_extended[cut:]
+                attn_extended = attn_extended[cut:]
+                acoustic_input_mask = acoustic_input_mask[cut:]
+                acoustic_loss_mask = acoustic_loss_mask[cut:]
+
+            sample_input_ids.append(ids_extended)
+            sample_attention_masks.append(attn_extended)
+            sample_acoustic_input_masks.append(acoustic_input_mask)
+            sample_acoustic_loss_masks.append(acoustic_loss_mask)
+
+            voice_speeches = []
+            voice_latent_lengths = []
+            if proc.get("speech_tensors") is not None:
+                voice_np = proc["speech_tensors"].cpu().numpy()
+                voice_masks = proc["speech_masks"].cpu().numpy().astype(bool)
+                for seg_idx in range(voice_np.shape[0]):
+                    voice_speeches.append(voice_np[seg_idx])
+                    voice_latent_lengths.append(int(voice_masks[seg_idx].sum()))
+
+            all_speech_waveforms.extend(voice_speeches)
+            all_speech_latent_lengths.extend(voice_latent_lengths)
+            per_segment_is_target.extend([False] * len(voice_speeches))
+
+            all_speech_waveforms.append(wav_target)
+            all_speech_latent_lengths.append(target_latent_len)
+            per_segment_is_target.append(True)
+
+        max_seq_len = max(len(x) for x in sample_input_ids)
+        padded_input_ids = []
+        padded_attention_masks = []
+        padded_acoustic_input_masks = []
+        padded_acoustic_loss_masks = []
+        tok = self.processor.tokenizer
+        pad_token_id = getattr(tok, "pad_token_id", None)
+        if pad_token_id is None or pad_token_id < 0:
+            pad_token_id = getattr(tok, "eos_token_id", None)
+            if pad_token_id is None or pad_token_id < 0:
+                raise ValueError(
+                    "Tokenizer has no pad_token_id or eos_token_id; please set one or pass a valid pad id."
+                )
+        for ids, attn, ain_mask, aloss_mask in zip(
+            sample_input_ids, sample_attention_masks, sample_acoustic_input_masks, sample_acoustic_loss_masks
+        ):
+            pad_len = max_seq_len - len(ids)
+            padded_input_ids.append(ids + [pad_token_id] * pad_len)
+            padded_attention_masks.append(attn + [0] * pad_len)
+            padded_acoustic_input_masks.append(ain_mask + [False] * pad_len)
+            padded_acoustic_loss_masks.append(aloss_mask + [False] * pad_len)
+
+        input_ids_tensor = torch.tensor(padded_input_ids, dtype=torch.long)
+        attention_mask_tensor = torch.tensor(padded_attention_masks, dtype=torch.long)
+        acoustic_input_mask_tensor = torch.tensor(padded_acoustic_input_masks, dtype=torch.bool)
+        acoustic_loss_mask_tensor = torch.tensor(padded_acoustic_loss_masks, dtype=torch.bool)
+
+        if all_speech_waveforms:
+            max_wave_len = max(w.shape[0] for w in all_speech_waveforms)
+            padded_speeches = np.zeros((len(all_speech_waveforms), max_wave_len), dtype=np.float32)
+            for i, w in enumerate(all_speech_waveforms):
+                L = w.shape[0]
+                padded_speeches[i, :L] = w
+
+            max_latent_len = max(all_speech_latent_lengths) if all_speech_latent_lengths else 1
+            speech_masks_np = np.zeros((len(all_speech_waveforms), max_latent_len), dtype=np.bool_)
+            for i, L_lat in enumerate(all_speech_latent_lengths):
+                speech_masks_np[i, :L_lat] = True
+
+            speech_tensors_tensor = torch.tensor(padded_speeches, dtype=torch.float32)
+            speech_masks_tensor = torch.tensor(speech_masks_np, dtype=torch.bool)
+
+            speeches_loss_input_np = np.zeros_like(speech_masks_np, dtype=np.bool_)
+            for i, is_target in enumerate(per_segment_is_target):
+                if is_target:
+                    speeches_loss_input_np[i] = speech_masks_np[i]
+            speeches_loss_input_tensor = torch.tensor(speeches_loss_input_np, dtype=torch.bool)
+
+            # Semantic features
+            if self.compute_semantics and hasattr(self.processor, "semantic_tokenizer") and self.processor.semantic_tokenizer is not None:
+                sem_feats: List[np.ndarray] = []
+                for w in all_speech_waveforms:
+                    try:
+                        # Expect [T, D]  where T ≈ ceil(len(w)/compress_ratio)
+                        sem = self.processor.semantic_tokenizer.encode(w)
+                        sem = np.asarray(sem, dtype=np.float32)
+                    except Exception:
+                        sem = np.zeros((0, self.semantic_vae_dim), dtype=np.float32)
+                    if sem.ndim != 2:
+                        raise RuntimeError(f"Semantic tokenizer returned unexpected shape {sem.shape}. Expect [T, D].")
+                    L = sem.shape[0]
+                    D = sem.shape[1]
+                    if D != self.semantic_vae_dim:
+                        if D < self.semantic_vae_dim:
+                            pad_d = np.zeros((L, self.semantic_vae_dim - D), dtype=np.float32)
+                            sem = np.concatenate([sem, pad_d], axis=1)
+                        else:
+                            sem = sem[:, : self.semantic_vae_dim]
+                    if L < max_latent_len:
+                        pad = np.zeros((max_latent_len - L, self.semantic_vae_dim), dtype=np.float32)
+                        sem = np.concatenate([sem, pad], axis=0)
+                    elif L > max_latent_len:
+                        sem = sem[:max_latent_len]
+                    sem_feats.append(sem.astype(np.float32))
+                speech_semantic_tensors = torch.tensor(np.stack(sem_feats, axis=0), dtype=torch.float32)
+            else:
+                # Semantic tokenizer unavailable - use zero features as fallback
+                # این امکان برای preprocess مفید است که semantic_tokenizer بارگذاری نشود
+                if not self.compute_semantics:
+                    # preprocess mode - صفر features
+                    sem_feats = [np.zeros((max_latent_len, self.semantic_vae_dim), dtype=np.float32) 
+                                for _ in all_speech_waveforms]
+                    speech_semantic_tensors = torch.tensor(np.stack(sem_feats, axis=0), dtype=torch.float32)
+                else:
+                    # training mode - اگر compute_semantics = True باشد اما tokenizer نباشد، خطا
+                    raise RuntimeError(
+                        "Semantic features are required but could not be computed. "
+                        "Ensure processor.semantic_tokenizer is available or set compute_semantics=False for preprocessing."
+                    )
+        else:
+            speech_tensors_tensor = None
+            speech_masks_tensor = None
+            speeches_loss_input_tensor = None
+            speech_semantic_tensors = None  # No segments in batch
+
+        if self.debug_checks:
+            assert (input_ids_tensor >= 0).all(), "input_ids contains negative indices"
+            if speech_tensors_tensor is not None:
+                assert speech_tensors_tensor.dim() == 2, "Expected speech_tensors 2D [segments, samples]"
+
+        return {
+            "input_ids": input_ids_tensor,
+            "attention_mask": attention_mask_tensor,
+            "speech_tensors": speech_tensors_tensor,
+            "speech_masks": speech_masks_tensor,
+            "speech_semantic_tensors": speech_semantic_tensors,
+            "acoustic_input_mask": acoustic_input_mask_tensor,
+            "acoustic_loss_mask": acoustic_loss_mask_tensor,
+            "speeches_loss_input": speeches_loss_input_tensor,
+        }

VibeVoice-finetuning/src/finetune_vibevoice_lora.py

@@ -0,0 +1,902 @@
+# train_vibevoice_lora.py
+import logging
+import os
+from dataclasses import dataclass, field
+from typing import Any, Dict, List, Optional, Tuple
+
+import torch
+import torch.nn as nn
+import torch.nn.functional as F
+from datasets import load_dataset, DatasetDict, VerificationMode
+
+from transformers import (
+    HfArgumentParser,
+    Trainer,
+    set_seed,
+    TrainerCallback,
+)
+from transformers import TrainingArguments as HfTrainingArguments
+
+from peft import LoraConfig, get_peft_model, TaskType
+
+from vibevoice.modular.modeling_vibevoice import VibeVoiceForConditionalGeneration
+from vibevoice.modular.configuration_vibevoice import VibeVoiceConfig
+from vibevoice.processor.vibevoice_processor import VibeVoiceProcessor
+
+from data_vibevoice import VibeVoiceDataset, VibeVoiceCollator
+
+logger = logging.getLogger(__name__)
+
+# ================== SAMPLE CALLBACK UTILS ==================
+
+import copy
+import torch
+from transformers import TrainerCallback
+
+class EmaCallback(TrainerCallback):
+    def __init__(self, attr_path="model.prediction_head", decay=0.999, device="cpu"):
+        """
+        attr_path: where the head lives under self.model (Trainer wraps your VibeVoiceForConditionalGeneration)
+        decay:     EMA decay (0.999 ~ stable, 0.9999 ~ very smooth, slower to adapt)
+        """
+        self.attr_path = attr_path
+        self.decay = float(decay)
+        self.device = torch.device(device)
+        self.shadow = None
+        self._orig = None  # store non-EMA weights when we swap
+
+    def _get_module(self, model):
+        # Resolve dotted path like "model.prediction_head"
+        mod = model
+        for name in self.attr_path.split('.'):
+            mod = getattr(mod, name)
+        return mod
+
+    def on_train_begin(self, args, state, control, model=None, **kwargs):
+        head = self._get_module(model)
+        self.shadow = {k: p.detach().to(self.device).clone()
+                       for k, p in head.state_dict().items()}
+
+    def on_step_end(self, args, state, control, model=None, **kwargs):
+        if self.shadow is None: return
+        head = self._get_module(model)
+        with torch.no_grad():
+            for k, v in head.state_dict().items():
+                self.shadow[k].mul_(self.decay).add_(v.detach().to(self.device), alpha=(1.0 - self.decay))
+
+    # ---- Swap helpers ----
+    def _swap_in_ema(self, model):
+        head = self._get_module(model)
+        self._orig = copy.deepcopy(head.state_dict())
+        head.load_state_dict(self.shadow, strict=False)
+
+    def _swap_back(self, model):
+        if self._orig is None: return
+        head = self._get_module(model)
+        head.load_state_dict(self._orig, strict=False)
+        self._orig = None
+
+    def on_evaluate(self, args, state, control, model=None, **kwargs):
+        # use EMA during eval
+        self._swap_in_ema(model)
+
+    def on_evaluate_end(self, args, state, control, model=None, **kwargs):
+        self._swap_back(model)
+
+    def on_save(self, args, state, control, model=None, **kwargs):
+        # temporarily swap to EMA, let Trainer save, then swap back
+        self._swap_in_ema(model)
+
+    def on_save_end(self, args, state, control, model=None, **kwargs):
+        self._swap_back(model)
+
+    def on_train_end(self, args, state, control, model=None, **kwargs):
+        # final checkpoint: persist EMA
+        self._swap_in_ema(model)
+
+
+@dataclass
+class ModelArguments:
+    model_name_or_path: Optional[str] = field(
+        default=None, metadata={"help": "Path to VibeVoice base model with config.json"}
+    )
+    processor_name_or_path: Optional[str] = field(
+        default=None, metadata={"help": "Path to processor dir (preprocessor_config.json). Defaults to model path."}
+    )
+    cache_dir: Optional[str] = field(default=None)
+    freeze_acoustic_tokenizer: bool = field(default=True)
+    freeze_semantic_tokenizer: bool = field(default=True)
+    lora_r: int = field(default=8)
+    lora_alpha: int = field(default=32)
+    lora_dropout: float = field(default=0.05)
+    lora_target_modules: str = field(
+        default="q_proj,k_proj,v_proj,o_proj,gate_proj,up_proj,down_proj",
+        metadata={"help": "Comma-separated list of target module names in the LLM blocks"},
+    )
+    lora_wrap_diffusion_head: bool = field(default=False, metadata={"help": "Wrap diffusion head with PEFT LoRA"})
+    train_diffusion_head: bool = field(default=False, metadata={"help": "Train diffusion prediction head (full fine-tune)"})
+    train_connectors: bool = field(default=False, metadata={"help": "Train acoustic/semantic connectors (full fine-tune)"})
+    layers_to_freeze: Optional[str] = field(
+        default=None, 
+        metadata={"help": "Comma-separated indices of diffusion head layers to freeze (e.g., '0,1,5,7,8')."}
+    )
+
+@dataclass
+class DataArguments:
+    dataset_name: Optional[str] = field(default=None, metadata={"help": "HF dataset name or 'json' with --train_jsonl for local files"})
+    dataset_config_name: Optional[str] = field(default=None)
+    train_split_name: str = field(default="train")
+    eval_split_name: Optional[str] = field(default="validation")
+    text_column_name: str = field(default="text")
+    audio_column_name: str = field(default="audio")
+    voice_prompts_column_name: Optional[str] = field(default="voice_prompts")
+    eval_split_size: float = field(default=0.0)
+    ignore_verifications: bool = field(default=False)
+    max_length: Optional[int] = field(default=None)
+    train_jsonl: Optional[str] = field(default=None, metadata={"help": "Path to local train JSONL with {text, audio, [voice_prompts]}"})
+    validation_jsonl: Optional[str] = field(default=None, metadata={"help": "Optional path to local validation JSONL"})
+    voice_prompt_drop_rate: float = field(
+        default=0.0,
+        metadata={"help": "Probability to drop conditioning voice prompt during training (0.0 keep always, 1.0 drop always)."},
+    )
+
+@dataclass
+class CustomTrainingArguments(HfTrainingArguments):
+    ddpm_batch_mul: int = field(default=1)
+    ce_loss_weight: float = field(default=1.0)
+    diffusion_loss_weight: float = field(default=1.0)
+    debug_ce_details: bool = field(default=False)
+    debug_ce_topk: int = field(default=5)
+    debug_ce_max_examples: int = field(default=1)
+    debug_ce_every_n_steps: int = field(default=200)
+    gradient_clipping: bool = field(
+        default=False,
+        metadata={"help": "Enable gradient clipping using max_grad_norm (set via --max_grad_norm, default 1.0). When False, disables clipping by forcing max_grad_norm=0.0."},
+    )
+    debug_save: bool = field(
+        default=False,
+        metadata={"help": "If set, saves model components BEFORE training starts, into output_dir/debug_initial."},
+    )
+
+def build_lora_config(args: ModelArguments) -> LoraConfig:
+    target_modules = [s.strip() for s in args.lora_target_modules.split(",") if s.strip()]
+    return LoraConfig(
+        r=args.lora_r,
+        lora_alpha=args.lora_alpha,
+        lora_dropout=args.lora_dropout,
+        bias="none",
+        task_type=TaskType.CAUSAL_LM,
+        target_modules=target_modules,
+    )
+
+def build_head_lora_config(args: ModelArguments) -> LoraConfig:
+    target_modules = ["noisy_images_proj","cond_proj","gate_proj","up_proj","down_proj","linear"]
+    return LoraConfig(
+        r=args.lora_r,
+        lora_alpha=args.lora_alpha,
+        lora_dropout=args.lora_dropout,
+        bias="none",
+        task_type=TaskType.FEATURE_EXTRACTION,
+        target_modules=target_modules,
+    )
+
+def mask_for_ce(labels: torch.Tensor, attention_mask: torch.Tensor, acoustic_input_mask: torch.Tensor, pad_id: int = -100) -> torch.Tensor:
+    shifted = labels[:, 1:].contiguous()
+    base_mask = attention_mask[:, 1:].contiguous().eq(1) if (attention_mask is not None and attention_mask.numel() > 0) else torch.ones_like(shifted, dtype=torch.bool)
+    label_is_acoustic = acoustic_input_mask[:, 1:].contiguous()
+    final_mask = base_mask & (~label_is_acoustic)
+    out = shifted.clone()
+    out[~final_mask] = pad_id
+    return out
+
+def _patch_acoustic_encode_for_legacy_indexing(model_obj, logger_):
+    try:
+        acoustic = getattr(getattr(model_obj, "model", model_obj), "acoustic_tokenizer", None)
+        if acoustic is None or not hasattr(acoustic, "encode"):
+            logger_.warning("No acoustic_tokenizer.encode() found to patch.")
+            return
+        base_encode = acoustic.encode
+        def encode_wrapped(*args, **kwargs):
+            out = base_encode(*args, **kwargs)
+            try:
+                _ = out[0][0]
+                return out
+            except Exception:
+                pass
+            if isinstance(out, dict):
+                for k in ("frames", "codes", "tokens", "latents", "hidden_states"):
+                    if k in out:
+                        return [[out[k]]]
+                if len(out) > 0:
+                    return [[next(iter(out.values()))]]
+            for attr in ("frames", "codes", "tokens", "latents", "hidden_states"):
+                if hasattr(out, attr):
+                    return [[getattr(out, attr)]]
+            try:
+                if isinstance(out, torch.Tensor):
+                    return [[out]]
+            except Exception:
+                pass
+            return [[out]]
+        acoustic.encode = encode_wrapped
+        logger_.info("Patched acoustic_tokenizer.encode() to return [[...]] for legacy indexing.")
+    except Exception as e:
+        logger_.warning(f"Failed to patch acoustic_tokenizer.encode(): {e}")
+
+def main() -> None:
+    parser = HfArgumentParser((ModelArguments, DataArguments, CustomTrainingArguments))
+    model_args, data_args, training_args = parser.parse_args_into_dataclasses()
+
+    logging.basicConfig(
+        format="%(asctime)s - %(levelname)s - %(name)s - %(message)s",
+        datefmt="%m/%d/%Y %H:%M:%S",
+        level=logging.INFO if training_args.local_rank in [-1, 0] else logging.WARN,
+    )
+    logger.info("Training/evaluation parameters %s", training_args)
+    set_seed(training_args.seed)
+
+    # Configure gradient clipping
+    if not getattr(training_args, "gradient_clipping", False):
+        if hasattr(training_args, "max_grad_norm"):
+            training_args.max_grad_norm = 0.0
+            logger.info("Gradient clipping disabled (set max_grad_norm=0.0). Use --gradient_clipping to enable.")
+    else:
+        if (not hasattr(training_args, "max_grad_norm")) or training_args.max_grad_norm is None or training_args.max_grad_norm <= 0:
+            training_args.max_grad_norm = 1.0
+        logger.info(f"Gradient clipping enabled: max_grad_norm={training_args.max_grad_norm}")
+
+    # Load processor
+    processor_path = model_args.processor_name_or_path or model_args.model_name_or_path
+    if processor_path is None:
+        raise ValueError("--model_name_or_path (or --processor_name_or_path) must be provided")
+    processor: VibeVoiceProcessor = VibeVoiceProcessor.from_pretrained(processor_path)
+
+    # Required special tokens
+    tok = processor.tokenizer
+    for required in ["speech_start_id", "speech_diffusion_id", "speech_end_id"]:
+        if not hasattr(tok, required) or getattr(tok, required) is None:
+            raise RuntimeError(f"Tokenizer missing required special id: {required}")
+
+    # Load model
+    if model_args.model_name_or_path is None:
+        raise ValueError("--model_name_or_path is required to load VibeVoice base model")
+    dtype = torch.float32
+    if training_args.bf16:
+        dtype = torch.bfloat16
+    elif getattr(training_args, "fp16", False):
+        dtype = torch.float16
+    model = VibeVoiceForConditionalGeneration.from_pretrained(
+        model_args.model_name_or_path,
+        torch_dtype=dtype,
+    )
+    _patch_acoustic_encode_for_legacy_indexing(model, logger)
+    processor.semantic_tokenizer = getattr(model.model, "semantic_tokenizer", None)
+
+    # Diagnostics: LM head tie
+    try:
+        in_emb_mod = model.get_input_embeddings()
+        out_emb_mod = model.get_output_embeddings()
+        in_w = getattr(in_emb_mod, "weight", None)
+        out_w = getattr(out_emb_mod, "weight", None)
+        shared_ptr = bool(in_w is not None and out_w is not None and in_w.data_ptr() == out_w.data_ptr())
+        values_equal = False
+        if in_w is not None and out_w is not None and in_w.shape == out_w.shape:
+            try:
+                values_equal = bool(torch.allclose(in_w, out_w))
+            except Exception:
+                values_equal = False
+        try:
+            tie_cfg = getattr(getattr(model.config, "decoder_config", model.config), "tie_word_embeddings", None)
+        except Exception:
+            tie_cfg = getattr(model.config, "tie_word_embeddings", None)
+        logger.info(f"LM head diagnostics -> shared_params={shared_ptr}, values_equal={values_equal}, tie_word_embeddings={tie_cfg}")
+        if out_w is not None:
+            logger.info(f"LM head requires_grad before freeze: {bool(out_w.requires_grad)}")
+    except Exception as e:
+        logger.warning(f"LM head tie diagnostics failed: {e}")
+
+    # Hard-tie LM head
+    try:
+        emb_module = model.get_input_embeddings()
+        head_module = model.get_output_embeddings()
+        if hasattr(emb_module, "weight") and hasattr(head_module, "weight"):
+            if emb_module.weight.shape == head_module.weight.shape and emb_module.weight.data_ptr() != head_module.weight.data_ptr():
+                with torch.no_grad():
+                    head_module.weight = emb_module.weight
+                logger.info("Force-tied LM head weight to input embeddings (pointer share).")
+    except Exception as e:
+        logger.warning(f"Force-tie of LM head failed: {e}")
+
+    # Validate special IDs (info logs only)
+    try:
+        special_names = ["speech_start_id", "speech_diffusion_id", "speech_end_id"]
+        try:
+            vocab_size = int(getattr(model.config.decoder_config, "vocab_size", 0))
+        except Exception:
+            vocab_size = 0
+        in_emb_mod = model.get_input_embeddings()
+        out_emb_mod = model.get_output_embeddings()
+        in_w = getattr(in_emb_mod, "weight", None)
+        out_w = getattr(out_emb_mod, "weight", None)
+        for name in special_names:
+            val = getattr(tok, name, None)
+            exists = (val is not None)
+            in_range = (exists and isinstance(val, int) and 0 <= val < vocab_size)
+            equal_row = None
+            if in_range and in_w is not None and out_w is not None and in_w.shape == out_w.shape and in_w.size(0) > val:
+                try:
+                    equal_row = bool(torch.allclose(in_w[val], out_w[val]))
+                except Exception:
+                    equal_row = False
+            decoded_str = None
+            if exists and isinstance(val, int):
+                try:
+                    decoded_str = tok.decode([val])
+                except Exception:
+                    try:
+                        decoded_str = tok.convert_ids_to_tokens(val)
+                    except Exception:
+                        decoded_str = "<decode_failed>"
+            logger.info(f"Special token check -> {name}={val}, decoded='{decoded_str}', exists={exists}, in_vocab_range={in_range}, emb_vs_head_row_equal={equal_row}")
+    except Exception as e:
+        logger.warning(f"Special token ID/row validation failed: {e}")
+
+    # Quick tokenizer diagnostics (optional)
+    try:
+        logger.info("=== TOKENIZER DIAGNOSTICS ===")
+        logger.info(f"Tokenizer class: {type(tok).__name__}")
+        logger.info(f"Tokenizer vocab_size: {tok.vocab_size}")
+        # tiny CE smoke test
+        with torch.no_grad():
+            simple_text = "The cat sat on the mat."
+            simple_ids = torch.tensor([tok.encode(simple_text, add_special_tokens=True)], device=model.device)
+            simple_mask = torch.ones_like(simple_ids)
+            x = model.get_input_embeddings()(simple_ids)
+            outputs = model.model(inputs_embeds=x, attention_mask=simple_mask, return_dict=True)
+            logits = model.lm_head(outputs.last_hidden_state)
+            shift_logits = logits[:, :-1, :].contiguous()
+            shift_labels = simple_ids[:, 1:].contiguous()
+            ce_loss = F.cross_entropy(shift_logits.view(-1, shift_logits.size(-1)), shift_labels.view(-1), reduction='mean')
+            logger.info(f"Simple text CE loss: {ce_loss.item():.4f}")
+    except Exception as e:
+        logger.warning(f"Tokenizer diagnostics failed: {e}")
+
+    # Disable cache during training
+    if hasattr(model.config, "use_cache") and training_args.do_train:
+        model.config.use_cache = False
+
+    # Freeze tokenizers
+    if model_args.freeze_acoustic_tokenizer and hasattr(model.model, "acoustic_tokenizer"):
+        for p in model.model.acoustic_tokenizer.parameters():
+            p.requires_grad = False
+    if model_args.freeze_semantic_tokenizer and hasattr(model.model, "semantic_tokenizer"):
+        for p in model.model.semantic_tokenizer.parameters():
+            p.requires_grad = False
+
+    # LoRA wrap LLM (optional)
+    lora_cfg = build_lora_config(model_args)
+    tm_lower = [s.strip().lower() for s in model_args.lora_target_modules.split(",") if s.strip()]
+    skip_lm_lora = (len(tm_lower) == 0) or all(t in ("none", "off", "disable", "disabled") for t in tm_lower)
+    if not skip_lm_lora:
+        model.model.language_model = get_peft_model(model.model.language_model, lora_cfg)
+    else:
+        logger.info("Skipping LLM LoRA wrapping (lora_target_modules indicates none).")
+
+    try:
+        model.tie_weights()
+    except Exception:
+        pass
+
+    # Freeze all then enable trainable subsets
+    for _, p in model.named_parameters():
+        p.requires_grad = False
+
+    try:
+        for n, p in model.model.language_model.named_parameters():
+            if "lora_A" in n or "lora_B" in n:
+                p.requires_grad = True
+    except Exception:
+        logger.warning("Could not re-enable LoRA params on language_model.")
+
+    # Diffusion head LoRA wrapping (optional)
+    if getattr(model_args, "lora_wrap_diffusion_head", False) and hasattr(model.model, "prediction_head"):
+        class _HeadForwardShim(nn.Module):
+            def __init__(self, base: nn.Module): super().__init__(); self.base = base
+            def forward(self, *args, **kwargs):
+                if len(args) >= 3:
+                    noisy_images, timesteps, condition = args[:3]
+                else:
+                    noisy_images = kwargs.get("noisy_images")
+                    timesteps = kwargs.get("timesteps")
+                    condition = kwargs.get("condition")
+                return self.base(noisy_images, timesteps, condition)
+        try:
+            shim = _HeadForwardShim(model.model.prediction_head)
+            model.model.prediction_head = get_peft_model(shim, build_head_lora_config(model_args))
+            for n, p in model.model.prediction_head.named_parameters():
+                if "lora_A" in n or "lora_B" in n:
+                    p.requires_grad = True
+        except Exception as e:
+            logger.warning(f"Could not LoRA-wrap diffusion head: {e}")
+
+    # Train full diffusion head (optional)
+    if getattr(model_args, "train_diffusion_head", False) and hasattr(model.model, "prediction_head"):
+        for p in model.model.prediction_head.parameters():
+            p.requires_grad = True
+
+    # Freeze diffusion head layers (optional)
+    if model_args.layers_to_freeze is not None and hasattr(model.model, "prediction_head"):
+        head_params = list(model.model.prediction_head.named_parameters())
+        try:
+            indices_to_freeze = {int(x.strip()) for x in model_args.layers_to_freeze.split(',') if x.strip()}
+            frozen_count = 0
+            for i, (name, param) in enumerate(head_params):
+                if i in indices_to_freeze:
+                    param.requires_grad = False
+                    frozen_count += 1
+                    logger.info(f"Froze layer [{i}]: {name}")
+            logger.info(f"Successfully froze {frozen_count} parameter groups in the diffusion head.")
+        except Exception as e:
+            logger.error(f"Could not parse --layers_to_freeze: {e}")
+            raise
+    
+    # Connectors
+    if getattr(model_args, "train_connectors", False):
+        if hasattr(model.model, "acoustic_connector"):
+            for p in model.model.acoustic_connector.parameters():
+                p.requires_grad = True
+        if hasattr(model.model, "semantic_connector"):
+            for p in model.model.semantic_connector.parameters():
+                p.requires_grad = True
+    else:
+        if hasattr(model.model, "acoustic_connector"):
+            for p in model.model.acoustic_connector.parameters():
+                p.requires_grad = False
+        if hasattr(model.model, "semantic_connector"):
+            for p in model.model.semantic_connector.parameters():
+                p.requires_grad = False
+
+    # Freeze embedding + head
+    try:
+        emb = model.get_input_embeddings()
+        if hasattr(emb, "weight"):
+            emb.weight.requires_grad_(False)
+        head = model.get_output_embeddings()
+        if head is not None and hasattr(head, "weight"):
+            head.weight.requires_grad_(False)
+    except Exception:
+        pass
+
+    # Diagnostics
+    def _sum_params(named_iter):
+        return sum(p.numel() for _, p in named_iter if p.requires_grad)
+    try:
+        lm_lora = _sum_params(model.model.language_model.named_parameters()) if hasattr(model.model, "language_model") else 0
+        pred_head_train = _sum_params(model.model.prediction_head.named_parameters()) if hasattr(model.model, "prediction_head") else 0
+        ac_conn_train = _sum_params(model.model.acoustic_connector.named_parameters()) if hasattr(model.model, "acoustic_connector") else 0
+        se_conn_train = _sum_params(model.model.semantic_connector.named_parameters()) if hasattr(model.model, "semantic_connector") else 0
+        total_trainable = sum(p.numel() for p in model.parameters() if p.requires_grad)
+        logger.info(f"Trainable by block -> LLM-LoRA: {lm_lora:,} | diff_head: {pred_head_train:,} | ac_conn: {ac_conn_train:,} | se_conn: {se_conn_train:,}")
+        logger.info("TOTAL trainable: %s", f"{total_trainable:,}")
+    except Exception:
+        pass
+
+    # Datasets
+    verification_mode = VerificationMode.NO_CHECKS if data_args.ignore_verifications else VerificationMode.BASIC_CHECKS
+    if data_args.train_jsonl is not None:
+        data_files: Dict[str, str] = {"train": data_args.train_jsonl}
+        if data_args.validation_jsonl is not None:
+            data_files["validation"] = data_args.validation_jsonl
+        raw = load_dataset("json", data_files=data_files, verification_mode=verification_mode, cache_dir=model_args.cache_dir)
+    else:
+        if data_args.dataset_name is None:
+            raise ValueError("Provide --dataset_name (HF datasets) or use --train_jsonl/--validation_jsonl for local files.")
+        raw = load_dataset(
+            data_args.dataset_name,
+            data_args.dataset_config_name,
+            verification_mode=verification_mode,
+            cache_dir=model_args.cache_dir,
+        )
+    train_ds = raw[data_args.train_split_name]
+    eval_ds = None
+    if training_args.do_eval:
+        if data_args.eval_split_name and data_args.eval_split_name in raw:
+            eval_ds = raw[data_args.eval_split_name]
+        elif data_args.eval_split_size and data_args.eval_split_size > 0 and len(train_ds) > 1:
+            split = train_ds.train_test_split(test_size=data_args.eval_split_size, seed=training_args.seed)
+            train_ds, eval_ds = split["train"], split["test"]
+
+    train_dataset = VibeVoiceDataset(
+        train_ds,
+        text_column=data_args.text_column_name,
+        audio_column=data_args.audio_column_name,
+        voice_prompts_column=data_args.voice_prompts_column_name,
+    )
+    eval_dataset = None
+    if eval_ds is not None:
+        eval_dataset = VibeVoiceDataset(
+            eval_ds,
+            text_column=data_args.text_column_name,
+            audio_column=data_args.audio_column_name,
+            voice_prompts_column=data_args.voice_prompts_column_name,
+        )
+
+    # Ratios/dims from processor+model
+    speech_compress_ratio = getattr(processor, "speech_tok_compress_ratio", 3200)
+    semantic_dim = getattr(model.config, "semantic_vae_dim", None)
+    if semantic_dim is None:
+        try:
+            semantic_dim = int(getattr(model.config.semantic_tokenizer_config, "vae_dim", 128))
+        except Exception:
+            semantic_dim = 128
+
+    compute_semantics_flag = hasattr(processor, "semantic_tokenizer") and processor.semantic_tokenizer is not None
+
+    data_collator = VibeVoiceCollator(
+        processor=processor,
+        max_length=data_args.max_length,
+        speech_compress_ratio=speech_compress_ratio,
+        semantic_vae_dim=semantic_dim,
+        compute_semantics=compute_semantics_flag,
+        debug_checks=False,
+        voice_prompt_drop_rate=data_args.voice_prompt_drop_rate,
+    )
+
+    class LoRADebugCallback(TrainerCallback):
+        def __init__(self, log_every_n_steps: int = 50):
+            self.log_every_n_steps = max(1, int(log_every_n_steps))
+            self.prev_param_norms: Dict[str, float] = {}
+            self.lora_param_names: List[str] = []
+
+        def on_train_begin(self, args, state, control, model=None, **kwargs):
+            try:
+                if model is None:
+                    return
+                named: Dict[str, torch.nn.Parameter] = dict(model.named_parameters())
+                self.lora_param_names = [n for n in named.keys() if ("lora_A" in n or "lora_B" in n)]
+                for n in self.lora_param_names:
+                    p = named[n]
+                    self.prev_param_norms[n] = float(p.data.norm().item())
+                total = len(self.lora_param_names)
+                req_grad = sum(1 for n in self.lora_param_names if named[n].requires_grad)
+                num_A = sum(1 for n in self.lora_param_names if "lora_A" in n)
+                num_B = sum(1 for n in self.lora_param_names if "lora_B" in n)
+                zero_B = sum(1 for n in self.lora_param_names if ("lora_B" in n and float(named[n].data.norm().item()) == 0.0))
+                logger.info(f"LoRA debug: found {total} LoRA params (A={num_A}, B={num_B}); trainable={req_grad}. Initial lora_B_zero={zero_B}.")
+                if total == 0:
+                    logger.warning("LoRA debug: No LoRA parameters found. Check lora_target_modules.")
+                if req_grad != total:
+                    logger.warning("LoRA debug: Some LoRA params are frozen. They should be trainable.")
+            except Exception as e:
+                logger.warning(f"LoRA debug (on_train_begin) failed: {e}")
+
+        def on_step_end(self, args, state, control, model=None, **kwargs):
+            try:
+                if model is None or len(self.lora_param_names) == 0:
+                    return
+                step = int(getattr(state, "global_step", 0) or 0)
+                if step % self.log_every_n_steps != 0 and step != 1:
+                    return
+                named: Dict[str, torch.nn.Parameter] = dict(model.named_parameters())
+                changed_A = 0
+                changed_B = 0
+                zero_B = 0
+                eps = 1e-12
+                for n in self.lora_param_names:
+                    p = named.get(n, None)
+                    if p is None:
+                        continue
+                    prev = self.prev_param_norms.get(n, 0.0)
+                    curr = float(p.data.norm().item())
+                    if "lora_A" in n and abs(curr - prev) > eps:
+                        changed_A += 1
+                    if "lora_B" in n:
+                        if abs(curr - prev) > eps:
+                            changed_B += 1
+                        if curr == 0.0:
+                            zero_B += 1
+                    self.prev_param_norms[n] = curr
+                total_A = sum(1 for n in self.lora_param_names if "lora_A" in n)
+                total_B = sum(1 for n in self.lora_param_names if "lora_B" in n)
+                logger.info(f"LoRA debug step {step}: changed A {changed_A}/{total_A}, changed B {changed_B}/{total_B}, lora_B_zero_now={zero_B}.")
+            except Exception as e:
+                logger.warning(f"LoRA debug (on_step_end) failed: {e}")
+
+    class VibeVoiceTrainer(Trainer):
+        def compute_loss(self, model: VibeVoiceForConditionalGeneration, inputs: Dict[str, Any], return_outputs=False, num_items_in_batch: Optional[int] = None):
+            labels = inputs.get("input_ids")
+            attention_mask = inputs.get("attention_mask")
+            acoustic_input_mask = inputs.get("acoustic_input_mask")
+
+            # Ensure semantic tensors exist and have correct dtype/device
+            sem = inputs.get("speech_semantic_tensors", None)
+            try:
+                target_dtype = next(model.model.semantic_connector.parameters()).dtype
+            except Exception:
+                target_dtype = model.get_input_embeddings().weight.dtype
+
+            if sem is None:
+                sm = inputs.get("speech_masks")
+                if sm is not None:
+                    zeros = torch.zeros(
+                        sm.size(0), sm.size(1),
+                        getattr(model.config, "semantic_vae_dim", 128),
+                        dtype=target_dtype,
+                        device=sm.device,
+                    )
+                    inputs["speech_semantic_tensors"] = zeros
+            else:
+                if isinstance(sem, torch.Tensor):
+                    inputs["speech_semantic_tensors"] = sem.to(dtype=target_dtype)
+
+            outputs = model(
+                input_ids=inputs.get("input_ids"),
+                attention_mask=attention_mask,
+                speech_tensors=inputs.get("speech_tensors"),
+                speech_masks=inputs.get("speech_masks"),
+                speech_semantic_tensors=inputs.get("speech_semantic_tensors"),
+                acoustic_input_mask=acoustic_input_mask,
+                acoustic_loss_mask=inputs.get("acoustic_loss_mask"),
+                speeches_loss_input=inputs.get("speeches_loss_input"),
+                ddpm_batch_mul=training_args.ddpm_batch_mul,
+            )
+
+            # Invariants: token/latent selection equality across views (warn, don't assert)
+            try:
+                al_mask = inputs.get("acoustic_loss_mask")
+                sp_masks = inputs.get("speech_masks")
+                sp_loss_sel = inputs.get("speeches_loss_input")
+                num_tok_total = int(acoustic_input_mask.sum().item()) if acoustic_input_mask is not None else 0
+                num_tok_loss = int(al_mask.sum().item()) if al_mask is not None else 0
+                num_lat_total = int(sp_masks.sum().item()) if sp_masks is not None else 0
+                num_lat_loss = int(((sp_loss_sel & sp_masks).sum().item())) if (sp_loss_sel is not None and sp_masks is not None) else 0
+                self.log({
+                    "debug/num_tok_total": float(num_tok_total),
+                    "debug/num_tok_loss": float(num_tok_loss),
+                    "debug/num_lat_total": float(num_lat_total),
+                    "debug/num_lat_loss": float(num_lat_loss),
+                })
+                if sp_loss_sel is not None and sp_masks is not None and al_mask is not None:
+                    if num_tok_loss != num_lat_loss:
+                        logger.warning(f"Loss selection mismatch: acoustic_loss_mask={num_tok_loss} vs speeches_loss_input={num_lat_loss}")
+            except Exception:
+                pass
+
+            # CE Loss
+            logits = outputs.logits
+            ce_labels = mask_for_ce(labels, attention_mask, acoustic_input_mask, pad_id=-100)
+            shift_logits = logits[:, :-1, :].contiguous()
+            loss_fct = nn.CrossEntropyLoss(ignore_index=-100)
+            ce_loss = loss_fct(shift_logits.view(-1, shift_logits.size(-1)), ce_labels.view(-1))
+
+            # Optional CE diagnostics
+            try:
+                self._debug_ce(shift_logits, ce_labels, attention_mask, acoustic_input_mask)
+            except Exception as e:
+                logger.warning(f"Failed invoking CE debug: {e}")
+
+            # Diffusion loss
+            diffusion_loss = outputs.diffusion_loss if outputs.diffusion_loss is not None else torch.tensor(0.0, device=ce_loss.device)
+            total = training_args.ce_loss_weight * ce_loss + training_args.diffusion_loss_weight * diffusion_loss
+
+            # Logs
+            try:
+                prefix = "train" if model.training else "eval"
+                self.log({
+                    f"{prefix}/ce_loss": ce_loss.detach().item(),
+                    f"{prefix}/diffusion_loss": diffusion_loss.detach().item() if isinstance(diffusion_loss, torch.Tensor) else float(diffusion_loss),
+                })
+                if hasattr(self, "optimizer") and self.optimizer is not None and len(self.optimizer.param_groups) > 0:
+                    lr_val = self.optimizer.param_groups[0].get("lr", None)
+                    if lr_val is not None:
+                        self.log({"train/learning_rate_real": float(lr_val)})
+            except Exception:
+                pass
+
+            return (total, outputs) if return_outputs else total
+
+        def _debug_ce(self, shift_logits: torch.Tensor, ce_labels: torch.Tensor, attention_mask: Optional[torch.Tensor], acoustic_input_mask: Optional[torch.Tensor]):
+            try:
+                if not getattr(training_args, "debug_ce_details", False):
+                    return
+                step = int(getattr(self.state, "global_step", 0) or 0)
+                every_n = max(1, int(getattr(training_args, "debug_ce_every_n_steps", 200) or 200))
+                if not (step <= 1 or (step % every_n == 0)):
+                    return
+
+                with torch.no_grad():
+                    vocab = shift_logits.size(-1)
+                    per_token_loss = F.cross_entropy(
+                        shift_logits.view(-1, vocab),
+                        ce_labels.view(-1),
+                        reduction="none",
+                        ignore_index=-100,
+                    ).view_as(ce_labels)
+
+                    valid_mask = ce_labels.ne(-100)
+                    num_valid = int(valid_mask.sum().item())
+                    avg_loss = float((per_token_loss[valid_mask].mean().item())) if num_valid > 0 else float("nan")
+
+                    per_ex_avgs = []
+                    max_examples = max(1, int(getattr(training_args, "debug_ce_max_examples", 1) or 1))
+                    B = ce_labels.size(0)
+                    for b in range(min(B, max_examples)):
+                        vb = valid_mask[b]
+                        if int(vb.sum().item()) > 0:
+                            per_ex_avgs.append(float(per_token_loss[b][vb].mean().item()))
+                        else:
+                            per_ex_avgs.append(float("nan"))
+                    logger.info(f"CE debug: tokens_in_loss={num_valid}, avg_loss={avg_loss:.4f}, per_example_avgs={[round(x,4) if x==x else None for x in per_ex_avgs]}")
+            except Exception as e:
+                logger.warning(f"CE detailed debug failed: {e}")
+
+        # --------- CRITICAL SAVE OVERRIDES: also dump FULL head/connectors for inference ---------
+  
+
+        def _save(self, output_dir: Optional[str] = None, state_dict=None) -> None:
+            try:
+                target_dir = output_dir or self.args.output_dir
+                lora_out = os.path.join(target_dir, "lora")
+                os.makedirs(lora_out, exist_ok=True)
+    
+                # --- LLM PEFT adapters (if LoRA-wrapped) ---
+                language_model = getattr(self.model.model, "language_model", None)
+                if hasattr(language_model, "save_pretrained"):
+                    language_model.save_pretrained(lora_out)
+    
+                # --- Diffusion head PEFT adapters (if LoRA-wrapped) ---
+                pred_head = getattr(self.model.model, "prediction_head", None)
+                if hasattr(pred_head, "save_pretrained"):
+                    ph_dir = os.path.join(lora_out, "diffusion_head")
+                    os.makedirs(ph_dir, exist_ok=True)
+                    pred_head.save_pretrained(ph_dir)
+    
+                # --- ALWAYS save FULL diffusion head state_dict for fallback ---
+                if pred_head is not None and hasattr(pred_head, "state_dict"):
+                    sd = pred_head.state_dict()
+                    torch.save(sd, os.path.join(lora_out, "diffusion_head_full.bin"))
+                    ph_dir = os.path.join(lora_out, "diffusion_head")
+                    os.makedirs(ph_dir, exist_ok=True)
+                    torch.save(sd, os.path.join(ph_dir, "diffusion_head_full.bin"))
+    
+                # --- Connectors (plain state_dicts) ---
+                ac = getattr(self.model.model, "acoustic_connector", None)
+                if ac is not None:
+                    ac_dir = os.path.join(lora_out, "acoustic_connector")
+                    os.makedirs(ac_dir, exist_ok=True)
+                    torch.save(ac.state_dict(), os.path.join(ac_dir, "pytorch_model.bin"))
+    
+                se = getattr(self.model.model, "semantic_connector", None)
+                if se is not None:
+                    se_dir = os.path.join(lora_out, "semantic_connector")
+                    os.makedirs(se_dir, exist_ok=True)
+                    torch.save(se.state_dict(), os.path.join(se_dir, "pytorch_model.bin"))
+    
+            except Exception as e:
+                logger.warning(f"Failed to save LoRA assets: {e}")
+
+
+    # ------------- Build the Trainer -------------
+
+    # Resolve which adapters to apply in samples
+
+    ema_cb = EmaCallback(attr_path="model.prediction_head", decay=0.999, device="cpu")
+
+    trainer = VibeVoiceTrainer(
+        model=model,
+        args=training_args,
+        train_dataset=train_dataset,
+        eval_dataset=eval_dataset,
+        data_collator=data_collator,
+        callbacks=[ema_cb, LoRADebugCallback(log_every_n_steps=(int(getattr(training_args, "logging_steps", 50) or 50)))],
+    )
+
+    # Optional debug pre-training save
+    if getattr(training_args, "debug_save", False):
+        try:
+            debug_dir = os.path.join(training_args.output_dir, "debug_initial")
+            lora_out = os.path.join(debug_dir, "lora")
+            os.makedirs(lora_out, exist_ok=True)
+            logger.info(f"[debug_save] Saving initial (pre-training) model components to: {debug_dir}")
+            # language model adapters / base
+            try:
+                if hasattr(model.model.language_model, "save_pretrained"):
+                    model.model.language_model.save_pretrained(lora_out)
+            except Exception as e_lm:
+                logger.warning(f"[debug_save] Failed to save language_model: {e_lm}")
+            # diffusion head
+            try:
+                if hasattr(model.model, "prediction_head") and hasattr(model.model.prediction_head, "save_pretrained"):
+                    model.model.prediction_head.save_pretrained(os.path.join(lora_out, "diffusion_head"))
+            except Exception as e_head:
+                logger.warning(f"[debug_save] Failed to save prediction_head: {e_head}")
+            # NEW: full diffusion head state_dict as fallback
+            try:
+                ph = getattr(model.model, "prediction_head", None)
+                if ph is not None and hasattr(ph, "state_dict"):
+                    sd = ph.state_dict()
+                    torch.save(sd, os.path.join(lora_out, "diffusion_head_full.bin"))
+                    os.makedirs(os.path.join(lora_out, "diffusion_head"), exist_ok=True)
+                    torch.save(sd, os.path.join(lora_out, "diffusion_head", "diffusion_head_full.bin"))
+            except Exception as e:
+                logger.warning(f"[debug_save] Failed to save FULL diffusion head: {e}")
+            # connectors
+            try:
+                ac_conn = getattr(model.model, "acoustic_connector", None)
+                if ac_conn is not None:
+                    ac_dir = os.path.join(lora_out, "acoustic_connector")
+                    os.makedirs(ac_dir, exist_ok=True)
+                    torch.save(ac_conn.state_dict(), os.path.join(ac_dir, "pytorch_model.bin"))
+            except Exception as e_ac:
+                logger.warning(f"[debug_save] Failed to save acoustic_connector: {e_ac}")
+            try:
+                se_conn = getattr(model.model, "semantic_connector", None)
+                if se_conn is not None:
+                    se_dir = os.path.join(lora_out, "semantic_connector")
+                    os.makedirs(se_dir, exist_ok=True)
+                    torch.save(se_conn.state_dict(), os.path.join(se_dir, "pytorch_model.bin"))
+            except Exception as e_se:
+                logger.warning(f"[debug_save] Failed to save semantic_connector: {e_se}")
+        except Exception as e:
+            logger.warning(f"[debug_save] Unexpected failure saving initial components: {e}")
+
+    if getattr(training_args, "gradient_checkpointing", False):
+        try:
+            model.gradient_checkpointing_enable()
+        except Exception:
+            logger.warning("Failed to enable gradient checkpointing on the model.")
+
+    if training_args.do_train:
+        trainer.train(resume_from_checkpoint=training_args.resume_from_checkpoint)
+    
+        lora_out = os.path.join(training_args.output_dir, "lora")
+        os.makedirs(lora_out, exist_ok=True)
+    
+        # LLM PEFT (if any)
+        lm = getattr(model.model, "language_model", None)
+        if hasattr(lm, "save_pretrained"):
+            lm.save_pretrained(lora_out)
+    
+        # Diffusion head PEFT (if any)
+        ph = getattr(model.model, "prediction_head", None)
+        if hasattr(ph, "save_pretrained"):
+            ph_dir = os.path.join(lora_out, "diffusion_head")
+            os.makedirs(ph_dir, exist_ok=True)
+            ph.save_pretrained(ph_dir)
+    
+        # ALWAYS: full diffusion head state_dict fallback
+        try:
+            if ph is not None and hasattr(ph, "state_dict"):
+                sd = ph.state_dict()
+                torch.save(sd, os.path.join(lora_out, "diffusion_head_full.bin"))
+                ph_dir = os.path.join(lora_out, "diffusion_head")
+                os.makedirs(ph_dir, exist_ok=True)
+                torch.save(sd, os.path.join(ph_dir, "diffusion_head_full.bin"))
+        except Exception as e:
+            logger.warning(f"Failed to save FULL diffusion head at end: {e}")
+    
+        # Connectors (if trained)
+        try:
+            ac = getattr(model.model, "acoustic_connector", None)
+            if ac is not None:
+                ac_dir = os.path.join(lora_out, "acoustic_connector")
+                os.makedirs(ac_dir, exist_ok=True)
+                torch.save(ac.state_dict(), os.path.join(ac_dir, "pytorch_model.bin"))
+        except Exception as e:
+            logger.warning(f"Failed to save acoustic_connector: {e}")
+    
+        try:
+            se = getattr(model.model, "semantic_connector", None)
+            if se is not None:
+                se_dir = os.path.join(lora_out, "semantic_connector")
+                os.makedirs(se_dir, exist_ok=True)
+                torch.save(se.state_dict(), os.path.join(se_dir, "pytorch_model.bin"))
+        except Exception as e:
+            logger.warning(f"Failed to save semantic_connector: {e}")
+
+    if training_args.do_eval and eval_dataset is not None:
+        trainer.evaluate()
+
+
+if __name__ == "__main__":
+    main()

VibeVoice-finetuning/src/finetune_vibevoice_lora0.py

@@ -0,0 +1,984 @@
+# train_vibevoice_lora.py
+import os
+os.environ["CUDA_VISIBLE_DEVICES"] = "0"
+os.environ["TOKENIZERS_PARALLELISM"] = "false"
+
+import logging
+import os
+from dataclasses import dataclass, field
+from typing import Any, Dict, List, Optional, Tuple
+
+import torch
+import torch.nn as nn
+import torch.nn.functional as F
+from datasets import load_dataset, DatasetDict, VerificationMode
+
+from transformers import (
+    HfArgumentParser,
+    Trainer,
+    set_seed,
+    TrainerCallback,
+)
+from transformers import TrainingArguments as HfTrainingArguments
+
+from peft import LoraConfig, get_peft_model, TaskType
+
+from vibevoice.modular.modeling_vibevoice import VibeVoiceForConditionalGeneration
+from vibevoice.modular.configuration_vibevoice import VibeVoiceConfig
+from vibevoice.processor.vibevoice_processor import VibeVoiceProcessor
+
+from data_vibevoice import VibeVoiceDataset, VibeVoiceCollator
+
+logger = logging.getLogger(__name__)
+
+# ================== SAMPLE CALLBACK UTILS ==================
+
+import copy
+import torch
+from transformers import TrainerCallback
+
+class EmaCallback(TrainerCallback):
+    def __init__(self, attr_path="model.prediction_head", decay=0.999, device="cuda"):
+        """
+        attr_path: where the head lives under self.model (Trainer wraps your VibeVoiceForConditionalGeneration)
+        decay:     EMA decay (0.999 ~ stable, 0.9999 ~ very smooth, slower to adapt)
+        """
+        self.attr_path = attr_path
+        self.decay = float(decay)
+        self.device = torch.device(device)
+        self.shadow = None
+        self._orig = None  # store non-EMA weights when we swap
+
+    def _get_module(self, model):
+        # Resolve dotted path like "model.prediction_head"
+        mod = model
+        for name in self.attr_path.split('.'):
+            mod = getattr(mod, name)
+        return mod
+
+    def on_train_begin(self, args, state, control, model=None, **kwargs):
+        head = self._get_module(model)
+        self.shadow = {k: p.detach().to(self.device).clone()
+                       for k, p in head.state_dict().items()}
+
+    def on_step_end(self, args, state, control, model=None, **kwargs):
+        if self.shadow is None: return
+        head = self._get_module(model)
+        with torch.no_grad():
+            for k, v in head.state_dict().items():
+                self.shadow[k].mul_(self.decay).add_(v.detach().to(self.device), alpha=(1.0 - self.decay))
+
+    # ---- Swap helpers ----
+    def _swap_in_ema(self, model):
+        head = self._get_module(model)
+        self._orig = copy.deepcopy(head.state_dict())
+        head.load_state_dict(self.shadow, strict=False)
+
+    def _swap_back(self, model):
+        if self._orig is None: return
+        head = self._get_module(model)
+        head.load_state_dict(self._orig, strict=False)
+        self._orig = None
+
+    def on_evaluate(self, args, state, control, model=None, **kwargs):
+        # use EMA during eval
+        self._swap_in_ema(model)
+
+    def on_evaluate_end(self, args, state, control, model=None, **kwargs):
+        self._swap_back(model)
+
+    def on_save(self, args, state, control, model=None, **kwargs):
+        # temporarily swap to EMA, let Trainer save, then swap back
+        self._swap_in_ema(model)
+
+    def on_save_end(self, args, state, control, model=None, **kwargs):
+        self._swap_back(model)
+
+    def on_train_end(self, args, state, control, model=None, **kwargs):
+        # final checkpoint: persist EMA
+        self._swap_in_ema(model)
+
+
+@dataclass
+class ModelArguments:
+    model_name_or_path: Optional[str] = field(
+        default=None, metadata={"help": "Path to VibeVoice base model with config.json"}
+    )
+    processor_name_or_path: Optional[str] = field(
+        default=None, metadata={"help": "Path to processor dir (preprocessor_config.json). Defaults to model path."}
+    )
+    cache_dir: Optional[str] = field(default=None)
+    freeze_acoustic_tokenizer: bool = field(default=True)
+    freeze_semantic_tokenizer: bool = field(default=True)
+    lora_r: int = field(default=8)
+    lora_alpha: int = field(default=32)
+    lora_dropout: float = field(default=0.05)
+    lora_target_modules: str = field(
+        default="q_proj,k_proj,v_proj,o_proj,gate_proj,up_proj,down_proj",
+        metadata={"help": "Comma-separated list of target module names in the LLM blocks"},
+    )
+    lora_wrap_diffusion_head: bool = field(default=False, metadata={"help": "Wrap diffusion head with PEFT LoRA"})
+    train_diffusion_head: bool = field(default=False, metadata={"help": "Train diffusion prediction head (full fine-tune)"})
+    train_connectors: bool = field(default=False, metadata={"help": "Train acoustic/semantic connectors (full fine-tune)"})
+    layers_to_freeze: Optional[str] = field(
+        default=None, 
+        metadata={"help": "Comma-separated indices of diffusion head layers to freeze (e.g., '0,1,5,7,8')."}
+    )
+
+@dataclass
+class DataArguments:
+    dataset_name: Optional[str] = field(default=None, metadata={"help": "HF dataset name or 'json' with --train_jsonl for local files"})
+    dataset_config_name: Optional[str] = field(default=None)
+    train_split_name: str = field(default="train")
+    eval_split_name: Optional[str] = field(default="validation")
+    text_column_name: str = field(default="text")
+    audio_column_name: str = field(default="audio")
+    voice_prompts_column_name: Optional[str] = field(default="voice_prompts")
+    eval_split_size: float = field(default=0.0)
+    ignore_verifications: bool = field(default=False)
+    max_length: Optional[int] = field(default=None)
+    train_jsonl: Optional[str] = field(default=None, metadata={"help": "Path to local train JSONL with {text, audio, [voice_prompts]}"})
+    validation_jsonl: Optional[str] = field(default=None, metadata={"help": "Optional path to local validation JSONL"})
+    voice_prompt_drop_rate: float = field(
+        default=0.0,
+        metadata={"help": "Probability to drop conditioning voice prompt during training (0.0 keep always, 1.0 drop always)."},
+    )
+
+@dataclass
+class CustomTrainingArguments(HfTrainingArguments):
+    ddpm_batch_mul: int = field(default=1)
+    ce_loss_weight: float = field(default=1.0)
+    diffusion_loss_weight: float = field(default=1.0)
+    debug_ce_details: bool = field(default=False)
+    debug_ce_topk: int = field(default=5)
+    debug_ce_max_examples: int = field(default=1)
+    debug_ce_every_n_steps: int = field(default=200)
+    gradient_clipping: bool = field(
+        default=False,
+        metadata={"help": "Enable gradient clipping using max_grad_norm (set via --max_grad_norm, default 1.0). When False, disables clipping by forcing max_grad_norm=0.0."},
+    )
+    debug_save: bool = field(
+        default=False,
+        metadata={"help": "If set, saves model components BEFORE training starts, into output_dir/debug_initial."},
+    )
+
+def build_lora_config(args: ModelArguments) -> LoraConfig:
+    target_modules = [s.strip() for s in args.lora_target_modules.split(",") if s.strip()]
+    return LoraConfig(
+        r=args.lora_r,
+        lora_alpha=args.lora_alpha,
+        lora_dropout=args.lora_dropout,
+        bias="none",
+        task_type=TaskType.CAUSAL_LM,
+        target_modules=target_modules,
+    )
+
+def build_head_lora_config(args: ModelArguments) -> LoraConfig:
+    target_modules = ["noisy_images_proj","cond_proj","gate_proj","up_proj","down_proj","linear"]
+    return LoraConfig(
+        r=args.lora_r,
+        lora_alpha=args.lora_alpha,
+        lora_dropout=args.lora_dropout,
+        bias="none",
+        task_type=TaskType.FEATURE_EXTRACTION,
+        target_modules=target_modules,
+    )
+
+def mask_for_ce(labels: torch.Tensor, attention_mask: torch.Tensor, acoustic_input_mask: torch.Tensor, pad_id: int = -100) -> torch.Tensor:
+    shifted = labels[:, 1:].contiguous()
+    base_mask = attention_mask[:, 1:].contiguous().eq(1) if (attention_mask is not None and attention_mask.numel() > 0) else torch.ones_like(shifted, dtype=torch.bool)
+    label_is_acoustic = acoustic_input_mask[:, 1:].contiguous()
+    final_mask = base_mask & (~label_is_acoustic)
+    out = shifted.clone()
+    out[~final_mask] = pad_id
+    return out
+
+def _patch_acoustic_encode_for_legacy_indexing(model_obj, logger_):
+    try:
+        acoustic = getattr(getattr(model_obj, "model", model_obj), "acoustic_tokenizer", None)
+        if acoustic is None or not hasattr(acoustic, "encode"):
+            logger_.warning("No acoustic_tokenizer.encode() found to patch.")
+            return
+        base_encode = acoustic.encode
+        def encode_wrapped(*args, **kwargs):
+            out = base_encode(*args, **kwargs)
+            try:
+                _ = out[0][0]
+                return out
+            except Exception:
+                pass
+            if isinstance(out, dict):
+                for k in ("frames", "codes", "tokens", "latents", "hidden_states"):
+                    if k in out:
+                        return [[out[k]]]
+                if len(out) > 0:
+                    return [[next(iter(out.values()))]]
+            for attr in ("frames", "codes", "tokens", "latents", "hidden_states"):
+                if hasattr(out, attr):
+                    return [[getattr(out, attr)]]
+            try:
+                if isinstance(out, torch.Tensor):
+                    return [[out]]
+            except Exception:
+                pass
+            return [[out]]
+        acoustic.encode = encode_wrapped
+        logger_.info("Patched acoustic_tokenizer.encode() to return [[...]] for legacy indexing.")
+    except Exception as e:
+        logger_.warning(f"Failed to patch acoustic_tokenizer.encode(): {e}")
+
+def main() -> None:
+    parser = HfArgumentParser((ModelArguments, DataArguments, CustomTrainingArguments))
+    model_args, data_args, training_args = parser.parse_args_into_dataclasses()
+
+    logging.basicConfig(
+        format="%(asctime)s - %(levelname)s - %(name)s - %(message)s",
+        datefmt="%m/%d/%Y %H:%M:%S",
+        level=logging.INFO if training_args.local_rank in [-1, 0] else logging.WARN,
+    )
+    logger.info("Training/evaluation parameters %s", training_args)
+    set_seed(training_args.seed)
+
+    # Configure gradient clipping
+    if not getattr(training_args, "gradient_clipping", False):
+        if hasattr(training_args, "max_grad_norm"):
+            training_args.max_grad_norm = 0.0
+            logger.info("Gradient clipping disabled (set max_grad_norm=0.0). Use --gradient_clipping to enable.")
+    else:
+        if (not hasattr(training_args, "max_grad_norm")) or training_args.max_grad_norm is None or training_args.max_grad_norm <= 0:
+            training_args.max_grad_norm = 1.0
+        logger.info(f"Gradient clipping enabled: max_grad_norm={training_args.max_grad_norm}")
+
+    # Load processor
+    processor_path = model_args.processor_name_or_path or model_args.model_name_or_path
+    if processor_path is None:
+        raise ValueError("--model_name_or_path (or --processor_name_or_path) must be provided")
+    processor: VibeVoiceProcessor = VibeVoiceProcessor.from_pretrained(processor_path)
+
+    # Required special tokens
+    tok = processor.tokenizer
+    for required in ["speech_start_id", "speech_diffusion_id", "speech_end_id"]:
+        if not hasattr(tok, required) or getattr(tok, required) is None:
+            raise RuntimeError(f"Tokenizer missing required special id: {required}")
+
+    # Load model
+    if model_args.model_name_or_path is None:
+        raise ValueError("--model_name_or_path is required to load VibeVoice base model")
+    dtype = torch.float32
+    if training_args.bf16:
+        dtype = torch.bfloat16
+    elif getattr(training_args, "fp16", False):
+        dtype = torch.float16
+    model = VibeVoiceForConditionalGeneration.from_pretrained(
+        model_args.model_name_or_path,
+        torch_dtype=dtype, device_map={"": 0},
+    )
+    _patch_acoustic_encode_for_legacy_indexing(model, logger)
+    processor.semantic_tokenizer = getattr(model.model, "semantic_tokenizer", None)
+
+    # Diagnostics: LM head tie
+    try:
+        in_emb_mod = model.get_input_embeddings()
+        out_emb_mod = model.get_output_embeddings()
+        in_w = getattr(in_emb_mod, "weight", None)
+        out_w = getattr(out_emb_mod, "weight", None)
+        shared_ptr = bool(in_w is not None and out_w is not None and in_w.data_ptr() == out_w.data_ptr())
+        values_equal = False
+        if in_w is not None and out_w is not None and in_w.shape == out_w.shape:
+            try:
+                values_equal = bool(torch.allclose(in_w, out_w))
+            except Exception:
+                values_equal = False
+        try:
+            tie_cfg = getattr(getattr(model.config, "decoder_config", model.config), "tie_word_embeddings", None)
+        except Exception:
+            tie_cfg = getattr(model.config, "tie_word_embeddings", None)
+        logger.info(f"LM head diagnostics -> shared_params={shared_ptr}, values_equal={values_equal}, tie_word_embeddings={tie_cfg}")
+        if out_w is not None:
+            logger.info(f"LM head requires_grad before freeze: {bool(out_w.requires_grad)}")
+    except Exception as e:
+        logger.warning(f"LM head tie diagnostics failed: {e}")
+
+    # Hard-tie LM head
+    try:
+        emb_module = model.get_input_embeddings()
+        head_module = model.get_output_embeddings()
+        if hasattr(emb_module, "weight") and hasattr(head_module, "weight"):
+            if emb_module.weight.shape == head_module.weight.shape and emb_module.weight.data_ptr() != head_module.weight.data_ptr():
+                with torch.no_grad():
+                    head_module.weight = emb_module.weight
+                logger.info("Force-tied LM head weight to input embeddings (pointer share).")
+    except Exception as e:
+        logger.warning(f"Force-tie of LM head failed: {e}")
+
+    # Validate special IDs (info logs only)
+    try:
+        special_names = ["speech_start_id", "speech_diffusion_id", "speech_end_id"]
+        try:
+            vocab_size = int(getattr(model.config.decoder_config, "vocab_size", 0))
+        except Exception:
+            vocab_size = 0
+        in_emb_mod = model.get_input_embeddings()
+        out_emb_mod = model.get_output_embeddings()
+        in_w = getattr(in_emb_mod, "weight", None)
+        out_w = getattr(out_emb_mod, "weight", None)
+        for name in special_names:
+            val = getattr(tok, name, None)
+            exists = (val is not None)
+            in_range = (exists and isinstance(val, int) and 0 <= val < vocab_size)
+            equal_row = None
+            if in_range and in_w is not None and out_w is not None and in_w.shape == out_w.shape and in_w.size(0) > val:
+                try:
+                    equal_row = bool(torch.allclose(in_w[val], out_w[val]))
+                except Exception:
+                    equal_row = False
+            decoded_str = None
+            if exists and isinstance(val, int):
+                try:
+                    decoded_str = tok.decode([val])
+                except Exception:
+                    try:
+                        decoded_str = tok.convert_ids_to_tokens(val)
+                    except Exception:
+                        decoded_str = "<decode_failed>"
+            logger.info(f"Special token check -> {name}={val}, decoded='{decoded_str}', exists={exists}, in_vocab_range={in_range}, emb_vs_head_row_equal={equal_row}")
+    except Exception as e:
+        logger.warning(f"Special token ID/row validation failed: {e}")
+
+    # Quick tokenizer diagnostics (optional)
+    try:
+        logger.info("=== TOKENIZER DIAGNOSTICS ===")
+        logger.info(f"Tokenizer class: {type(tok).__name__}")
+        logger.info(f"Tokenizer vocab_size: {tok.vocab_size}")
+        # tiny CE smoke test
+        with torch.no_grad():
+            simple_text = "The cat sat on the mat."
+            simple_ids = torch.tensor([tok.encode(simple_text, add_special_tokens=True)], device=model.device)
+            simple_mask = torch.ones_like(simple_ids)
+            x = model.get_input_embeddings()(simple_ids)
+            outputs = model.model(inputs_embeds=x, attention_mask=simple_mask, return_dict=True)
+            logits = model.lm_head(outputs.last_hidden_state)
+            shift_logits = logits[:, :-1, :].contiguous()
+            shift_labels = simple_ids[:, 1:].contiguous()
+            ce_loss = F.cross_entropy(shift_logits.view(-1, shift_logits.size(-1)), shift_labels.view(-1), reduction='mean')
+            logger.info(f"Simple text CE loss: {ce_loss.item():.4f}")
+    except Exception as e:
+        logger.warning(f"Tokenizer diagnostics failed: {e}")
+
+    # Disable cache during training
+    if hasattr(model.config, "use_cache") and training_args.do_train:
+        model.config.use_cache = False
+
+    # Freeze tokenizers
+    if model_args.freeze_acoustic_tokenizer and hasattr(model.model, "acoustic_tokenizer"):
+        for p in model.model.acoustic_tokenizer.parameters():
+            p.requires_grad = False
+    if model_args.freeze_semantic_tokenizer and hasattr(model.model, "semantic_tokenizer"):
+        for p in model.model.semantic_tokenizer.parameters():
+            p.requires_grad = False
+
+    # LoRA wrap LLM (optional)
+    lora_cfg = build_lora_config(model_args)
+    tm_lower = [s.strip().lower() for s in model_args.lora_target_modules.split(",") if s.strip()]
+    skip_lm_lora = (len(tm_lower) == 0) or all(t in ("none", "off", "disable", "disabled") for t in tm_lower)
+    if not skip_lm_lora:
+        model.model.language_model = get_peft_model(model.model.language_model, lora_cfg)
+    else:
+        logger.info("Skipping LLM LoRA wrapping (lora_target_modules indicates none).")
+
+    try:
+        model.tie_weights()
+    except Exception:
+        pass
+
+    # Freeze all then enable trainable subsets
+    for _, p in model.named_parameters():
+        p.requires_grad = False
+
+    try:
+        for n, p in model.model.language_model.named_parameters():
+            if "lora_A" in n or "lora_B" in n:
+                p.requires_grad = True
+    except Exception:
+        logger.warning("Could not re-enable LoRA params on language_model.")
+
+    # Diffusion head LoRA wrapping (optional)
+    if getattr(model_args, "lora_wrap_diffusion_head", False) and hasattr(model.model, "prediction_head"):
+        class _HeadForwardShim(nn.Module):
+            def __init__(self, base: nn.Module): super().__init__(); self.base = base
+            def forward(self, *args, **kwargs):
+                if len(args) >= 3:
+                    noisy_images, timesteps, condition = args[:3]
+                else:
+                    noisy_images = kwargs.get("noisy_images")
+                    timesteps = kwargs.get("timesteps")
+                    condition = kwargs.get("condition")
+                return self.base(noisy_images, timesteps, condition)
+        try:
+            shim = _HeadForwardShim(model.model.prediction_head)
+            model.model.prediction_head = get_peft_model(shim, build_head_lora_config(model_args))
+            for n, p in model.model.prediction_head.named_parameters():
+                if "lora_A" in n or "lora_B" in n:
+                    p.requires_grad = True
+        except Exception as e:
+            logger.warning(f"Could not LoRA-wrap diffusion head: {e}")
+
+    # Train full diffusion head (optional)
+    if getattr(model_args, "train_diffusion_head", False) and hasattr(model.model, "prediction_head"):
+        for p in model.model.prediction_head.parameters():
+            p.requires_grad = True
+
+    # Freeze diffusion head layers (optional)
+    if model_args.layers_to_freeze is not None and hasattr(model.model, "prediction_head"):
+        head_params = list(model.model.prediction_head.named_parameters())
+        try:
+            indices_to_freeze = {int(x.strip()) for x in model_args.layers_to_freeze.split(',') if x.strip()}
+            frozen_count = 0
+            for i, (name, param) in enumerate(head_params):
+                if i in indices_to_freeze:
+                    param.requires_grad = False
+                    frozen_count += 1
+                    logger.info(f"Froze layer [{i}]: {name}")
+            logger.info(f"Successfully froze {frozen_count} parameter groups in the diffusion head.")
+        except Exception as e:
+            logger.error(f"Could not parse --layers_to_freeze: {e}")
+            raise
+    
+    # Connectors
+    if getattr(model_args, "train_connectors", False):
+        if hasattr(model.model, "acoustic_connector"):
+            for p in model.model.acoustic_connector.parameters():
+                p.requires_grad = True
+        if hasattr(model.model, "semantic_connector"):
+            for p in model.model.semantic_connector.parameters():
+                p.requires_grad = True
+    else:
+        if hasattr(model.model, "acoustic_connector"):
+            for p in model.model.acoustic_connector.parameters():
+                p.requires_grad = False
+        if hasattr(model.model, "semantic_connector"):
+            for p in model.model.semantic_connector.parameters():
+                p.requires_grad = False
+
+    # Freeze embedding + head
+    try:
+        emb = model.get_input_embeddings()
+        if hasattr(emb, "weight"):
+            emb.weight.requires_grad_(False)
+        head = model.get_output_embeddings()
+        if head is not None and hasattr(head, "weight"):
+            head.weight.requires_grad_(False)
+    except Exception:
+        pass
+
+    # Diagnostics
+    def _sum_params(named_iter):
+        return sum(p.numel() for _, p in named_iter if p.requires_grad)
+    try:
+        lm_lora = _sum_params(model.model.language_model.named_parameters()) if hasattr(model.model, "language_model") else 0
+        pred_head_train = _sum_params(model.model.prediction_head.named_parameters()) if hasattr(model.model, "prediction_head") else 0
+        ac_conn_train = _sum_params(model.model.acoustic_connector.named_parameters()) if hasattr(model.model, "acoustic_connector") else 0
+        se_conn_train = _sum_params(model.model.semantic_connector.named_parameters()) if hasattr(model.model, "semantic_connector") else 0
+        total_trainable = sum(p.numel() for p in model.parameters() if p.requires_grad)
+        logger.info(f"Trainable by block -> LLM-LoRA: {lm_lora:,} | diff_head: {pred_head_train:,} | ac_conn: {ac_conn_train:,} | se_conn: {se_conn_train:,}")
+        logger.info("TOTAL trainable: %s", f"{total_trainable:,}")
+    except Exception:
+        pass
+
+    # Preprocessed data classes
+    class PreprocessedBatchDataset:
+        def __init__(self, preprocessed_file: str):
+            self.data = torch.load(preprocessed_file, map_location='cpu')
+            logger.info(f"Loaded {len(self.data)} preprocessed batches from {preprocessed_file}")
+        
+        def __len__(self):
+            return len(self.data)
+        
+        def __getitem__(self, idx):
+            batch = self.data[idx]
+            result = {}
+            for k, v in batch.items():
+                if isinstance(v, torch.Tensor):
+                    result[k] = v
+                else:
+                    result[k] = v
+            return result
+
+    class PreprocessedBatchSubset:
+        def __init__(self, dataset: 'PreprocessedBatchDataset', indices: List[int]):
+            self.dataset = dataset
+            self.indices = indices
+        
+        def __len__(self):
+            return len(self.indices)
+        
+        def __getitem__(self, idx):
+            actual_idx = self.indices[idx]
+            return self.dataset[actual_idx]
+
+    class PreprocessedBatchCollator:
+        def __call__(self, batch: List[Dict[str, torch.Tensor]]) -> Dict[str, torch.Tensor]:
+            if not batch:
+                return {}
+            result = {}
+            for key in batch[0].keys():
+                tensors = [b[key] for b in batch if b[key] is not None]
+                if tensors and isinstance(tensors[0], torch.Tensor):
+                    result[key] = torch.cat(tensors, dim=0)
+                else:
+                    result[key] = tensors[0] if tensors else None
+            return result
+
+    # Datasets
+    preprocessed_dir = os.path.join(training_args.output_dir, "preprocessed")
+    preprocessed_file = os.path.join(preprocessed_dir, "preprocessed_batches.pt")
+    
+    if os.path.exists(preprocessed_file):
+        logger.info(f"Loading preprocessed data from {preprocessed_file}")
+        preprocessed_data = PreprocessedBatchDataset(preprocessed_file)
+        
+        train_dataset = preprocessed_data
+        eval_dataset = None
+        
+        if training_args.do_eval and data_args.eval_split_size and data_args.eval_split_size > 0 and len(preprocessed_data) > 1:
+            num_eval = max(1, int(len(preprocessed_data) * data_args.eval_split_size))
+            num_train = len(preprocessed_data) - num_eval
+            indices = list(range(len(preprocessed_data)))
+            import random
+            random.Random(training_args.seed).shuffle(indices)
+            train_indices = indices[:num_train]
+            eval_indices = indices[num_train:]
+            train_dataset = PreprocessedBatchSubset(preprocessed_data, train_indices)
+            eval_dataset = PreprocessedBatchSubset(preprocessed_data, eval_indices)
+    else:
+        logger.info(f"Preprocessed data not found at {preprocessed_file}, loading from raw JSONL/HF datasets")
+        verification_mode = VerificationMode.NO_CHECKS if data_args.ignore_verifications else VerificationMode.BASIC_CHECKS
+        if data_args.train_jsonl is not None:
+            data_files: Dict[str, str] = {"train": data_args.train_jsonl}
+            if data_args.validation_jsonl is not None:
+                data_files["validation"] = data_args.validation_jsonl
+            raw = load_dataset("json", data_files=data_files, verification_mode=verification_mode, cache_dir=model_args.cache_dir)
+        else:
+            if data_args.dataset_name is None:
+                raise ValueError("Provide --dataset_name (HF datasets) or use --train_jsonl/--validation_jsonl for local files.")
+            raw = load_dataset(
+                data_args.dataset_name,
+                data_args.dataset_config_name,
+                verification_mode=verification_mode,
+                cache_dir=model_args.cache_dir,
+            )
+        train_ds = raw[data_args.train_split_name]
+        eval_ds = None
+        if training_args.do_eval:
+            if data_args.eval_split_name and data_args.eval_split_name in raw:
+                eval_ds = raw[data_args.eval_split_name]
+            elif data_args.eval_split_size and data_args.eval_split_size > 0 and len(train_ds) > 1:
+                split = train_ds.train_test_split(test_size=data_args.eval_split_size, seed=training_args.seed)
+                train_ds, eval_ds = split["train"], split["test"]
+
+        train_dataset = VibeVoiceDataset(
+            train_ds,
+            text_column=data_args.text_column_name,
+            audio_column=data_args.audio_column_name,
+            voice_prompts_column=data_args.voice_prompts_column_name,
+        )
+        eval_dataset = None
+        if eval_ds is not None:
+            eval_dataset = VibeVoiceDataset(
+                eval_ds,
+                text_column=data_args.text_column_name,
+                audio_column=data_args.audio_column_name,
+                voice_prompts_column=data_args.voice_prompts_column_name,
+            )
+
+    # Ratios/dims from processor+model
+    speech_compress_ratio = getattr(processor, "speech_tok_compress_ratio", 3200)
+    semantic_dim = getattr(model.config, "semantic_vae_dim", None)
+    if semantic_dim is None:
+        try:
+            semantic_dim = int(getattr(model.config.semantic_tokenizer_config, "vae_dim", 128))
+        except Exception:
+            semantic_dim = 128
+
+    compute_semantics_flag = hasattr(processor, "semantic_tokenizer") and processor.semantic_tokenizer is not None
+
+    if os.path.exists(preprocessed_file):
+        data_collator = PreprocessedBatchCollator()
+    else:
+        data_collator = VibeVoiceCollator(
+            processor=processor,
+            max_length=data_args.max_length,
+            speech_compress_ratio=speech_compress_ratio,
+            semantic_vae_dim=semantic_dim,
+            compute_semantics=compute_semantics_flag,
+            debug_checks=False,
+            voice_prompt_drop_rate=data_args.voice_prompt_drop_rate,
+        )
+
+    class LoRADebugCallback(TrainerCallback):
+        def __init__(self, log_every_n_steps: int = 50):
+            self.log_every_n_steps = max(1, int(log_every_n_steps))
+            self.prev_param_norms: Dict[str, float] = {}
+            self.lora_param_names: List[str] = []
+
+        def on_train_begin(self, args, state, control, model=None, **kwargs):
+            try:
+                if model is None:
+                    return
+                named: Dict[str, torch.nn.Parameter] = dict(model.named_parameters())
+                self.lora_param_names = [n for n in named.keys() if ("lora_A" in n or "lora_B" in n)]
+                for n in self.lora_param_names:
+                    p = named[n]
+                    self.prev_param_norms[n] = float(p.data.norm().item())
+                total = len(self.lora_param_names)
+                req_grad = sum(1 for n in self.lora_param_names if named[n].requires_grad)
+                num_A = sum(1 for n in self.lora_param_names if "lora_A" in n)
+                num_B = sum(1 for n in self.lora_param_names if "lora_B" in n)
+                zero_B = sum(1 for n in self.lora_param_names if ("lora_B" in n and float(named[n].data.norm().item()) == 0.0))
+                logger.info(f"LoRA debug: found {total} LoRA params (A={num_A}, B={num_B}); trainable={req_grad}. Initial lora_B_zero={zero_B}.")
+                if total == 0:
+                    logger.warning("LoRA debug: No LoRA parameters found. Check lora_target_modules.")
+                if req_grad != total:
+                    logger.warning("LoRA debug: Some LoRA params are frozen. They should be trainable.")
+            except Exception as e:
+                logger.warning(f"LoRA debug (on_train_begin) failed: {e}")
+
+        def on_step_end(self, args, state, control, model=None, **kwargs):
+            try:
+                if model is None or len(self.lora_param_names) == 0:
+                    return
+                step = int(getattr(state, "global_step", 0) or 0)
+                if step % self.log_every_n_steps != 0 and step != 1:
+                    return
+                named: Dict[str, torch.nn.Parameter] = dict(model.named_parameters())
+                changed_A = 0
+                changed_B = 0
+                zero_B = 0
+                eps = 1e-12
+                for n in self.lora_param_names:
+                    p = named.get(n, None)
+                    if p is None:
+                        continue
+                    prev = self.prev_param_norms.get(n, 0.0)
+                    curr = float(p.data.norm().item())
+                    if "lora_A" in n and abs(curr - prev) > eps:
+                        changed_A += 1
+                    if "lora_B" in n:
+                        if abs(curr - prev) > eps:
+                            changed_B += 1
+                        if curr == 0.0:
+                            zero_B += 1
+                    self.prev_param_norms[n] = curr
+                total_A = sum(1 for n in self.lora_param_names if "lora_A" in n)
+                total_B = sum(1 for n in self.lora_param_names if "lora_B" in n)
+                logger.info(f"LoRA debug step {step}: changed A {changed_A}/{total_A}, changed B {changed_B}/{total_B}, lora_B_zero_now={zero_B}.")
+            except Exception as e:
+                logger.warning(f"LoRA debug (on_step_end) failed: {e}")
+
+    class VibeVoiceTrainer(Trainer):
+        def compute_loss(self, model: VibeVoiceForConditionalGeneration, inputs: Dict[str, Any], return_outputs=False, num_items_in_batch: Optional[int] = None):
+            labels = inputs.get("input_ids")
+            attention_mask = inputs.get("attention_mask")
+            acoustic_input_mask = inputs.get("acoustic_input_mask")
+
+            # Ensure semantic tensors exist and have correct dtype/device
+            sem = inputs.get("speech_semantic_tensors", None)
+            try:
+                target_dtype = next(model.model.semantic_connector.parameters()).dtype
+            except Exception:
+                target_dtype = model.get_input_embeddings().weight.dtype
+
+            if sem is None:
+                sm = inputs.get("speech_masks")
+                if sm is not None:
+                    zeros = torch.zeros(
+                        sm.size(0), sm.size(1),
+                        getattr(model.config, "semantic_vae_dim", 128),
+                        dtype=target_dtype,
+                        device=sm.device,
+                    )
+                    inputs["speech_semantic_tensors"] = zeros
+            else:
+                if isinstance(sem, torch.Tensor):
+                    inputs["speech_semantic_tensors"] = sem.to(dtype=target_dtype)
+
+            outputs = model(
+                input_ids=inputs.get("input_ids"),
+                attention_mask=attention_mask,
+                speech_tensors=inputs.get("speech_tensors"),
+                speech_masks=inputs.get("speech_masks"),
+                speech_semantic_tensors=inputs.get("speech_semantic_tensors"),
+                acoustic_input_mask=acoustic_input_mask,
+                acoustic_loss_mask=inputs.get("acoustic_loss_mask"),
+                speeches_loss_input=inputs.get("speeches_loss_input"),
+                ddpm_batch_mul=training_args.ddpm_batch_mul,
+            )
+
+            # Invariants: token/latent selection equality across views (warn, don't assert)
+            try:
+                al_mask = inputs.get("acoustic_loss_mask")
+                sp_masks = inputs.get("speech_masks")
+                sp_loss_sel = inputs.get("speeches_loss_input")
+                num_tok_total = int(acoustic_input_mask.sum().item()) if acoustic_input_mask is not None else 0
+                num_tok_loss = int(al_mask.sum().item()) if al_mask is not None else 0
+                num_lat_total = int(sp_masks.sum().item()) if sp_masks is not None else 0
+                num_lat_loss = int(((sp_loss_sel & sp_masks).sum().item())) if (sp_loss_sel is not None and sp_masks is not None) else 0
+                self.log({
+                    "debug/num_tok_total": float(num_tok_total),
+                    "debug/num_tok_loss": float(num_tok_loss),
+                    "debug/num_lat_total": float(num_lat_total),
+                    "debug/num_lat_loss": float(num_lat_loss),
+                })
+                if sp_loss_sel is not None and sp_masks is not None and al_mask is not None:
+                    if num_tok_loss != num_lat_loss:
+                        logger.warning(f"Loss selection mismatch: acoustic_loss_mask={num_tok_loss} vs speeches_loss_input={num_lat_loss}")
+            except Exception:
+                pass
+
+            # CE Loss
+            logits = outputs.logits
+            ce_labels = mask_for_ce(labels, attention_mask, acoustic_input_mask, pad_id=-100)
+            shift_logits = logits[:, :-1, :].contiguous()
+            loss_fct = nn.CrossEntropyLoss(ignore_index=-100)
+            ce_loss = loss_fct(shift_logits.view(-1, shift_logits.size(-1)), ce_labels.view(-1))
+
+            # Optional CE diagnostics
+            try:
+                self._debug_ce(shift_logits, ce_labels, attention_mask, acoustic_input_mask)
+            except Exception as e:
+                logger.warning(f"Failed invoking CE debug: {e}")
+
+            # Diffusion loss
+            diffusion_loss = outputs.diffusion_loss if outputs.diffusion_loss is not None else torch.tensor(0.0, device=ce_loss.device)
+            total = training_args.ce_loss_weight * ce_loss + training_args.diffusion_loss_weight * diffusion_loss
+
+            # Logs
+            try:
+                prefix = "train" if model.training else "eval"
+                self.log({
+                    f"{prefix}/ce_loss": ce_loss.detach().item(),
+                    f"{prefix}/diffusion_loss": diffusion_loss.detach().item() if isinstance(diffusion_loss, torch.Tensor) else float(diffusion_loss),
+                })
+                if hasattr(self, "optimizer") and self.optimizer is not None and len(self.optimizer.param_groups) > 0:
+                    lr_val = self.optimizer.param_groups[0].get("lr", None)
+                    if lr_val is not None:
+                        self.log({"train/learning_rate_real": float(lr_val)})
+            except Exception:
+                pass
+
+            return (total, outputs) if return_outputs else total
+
+        def _debug_ce(self, shift_logits: torch.Tensor, ce_labels: torch.Tensor, attention_mask: Optional[torch.Tensor], acoustic_input_mask: Optional[torch.Tensor]):
+            try:
+                if not getattr(training_args, "debug_ce_details", False):
+                    return
+                step = int(getattr(self.state, "global_step", 0) or 0)
+                every_n = max(1, int(getattr(training_args, "debug_ce_every_n_steps", 200) or 200))
+                if not (step <= 1 or (step % every_n == 0)):
+                    return
+
+                with torch.no_grad():
+                    vocab = shift_logits.size(-1)
+                    per_token_loss = F.cross_entropy(
+                        shift_logits.view(-1, vocab),
+                        ce_labels.view(-1),
+                        reduction="none",
+                        ignore_index=-100,
+                    ).view_as(ce_labels)
+
+                    valid_mask = ce_labels.ne(-100)
+                    num_valid = int(valid_mask.sum().item())
+                    avg_loss = float((per_token_loss[valid_mask].mean().item())) if num_valid > 0 else float("nan")
+
+                    per_ex_avgs = []
+                    max_examples = max(1, int(getattr(training_args, "debug_ce_max_examples", 1) or 1))
+                    B = ce_labels.size(0)
+                    for b in range(min(B, max_examples)):
+                        vb = valid_mask[b]
+                        if int(vb.sum().item()) > 0:
+                            per_ex_avgs.append(float(per_token_loss[b][vb].mean().item()))
+                        else:
+                            per_ex_avgs.append(float("nan"))
+                    logger.info(f"CE debug: tokens_in_loss={num_valid}, avg_loss={avg_loss:.4f}, per_example_avgs={[round(x,4) if x==x else None for x in per_ex_avgs]}")
+            except Exception as e:
+                logger.warning(f"CE detailed debug failed: {e}")
+
+        # --------- CRITICAL SAVE OVERRIDES: also dump FULL head/connectors for inference ---------
+  
+
+        def _save(self, output_dir: Optional[str] = None, state_dict=None) -> None:
+            try:
+                target_dir = output_dir or self.args.output_dir
+                lora_out = os.path.join(target_dir, "lora")
+                os.makedirs(lora_out, exist_ok=True)
+    
+                # --- LLM PEFT adapters (if LoRA-wrapped) ---
+                language_model = getattr(self.model.model, "language_model", None)
+                if hasattr(language_model, "save_pretrained"):
+                    language_model.save_pretrained(lora_out)
+    
+                # --- Diffusion head PEFT adapters (if LoRA-wrapped) ---
+                pred_head = getattr(self.model.model, "prediction_head", None)
+                if hasattr(pred_head, "save_pretrained"):
+                    ph_dir = os.path.join(lora_out, "diffusion_head")
+                    os.makedirs(ph_dir, exist_ok=True)
+                    pred_head.save_pretrained(ph_dir)
+    
+                # --- ALWAYS save FULL diffusion head state_dict for fallback ---
+                if pred_head is not None and hasattr(pred_head, "state_dict"):
+                    sd = pred_head.state_dict()
+                    torch.save(sd, os.path.join(lora_out, "diffusion_head_full.bin"))
+                    ph_dir = os.path.join(lora_out, "diffusion_head")
+                    os.makedirs(ph_dir, exist_ok=True)
+                    torch.save(sd, os.path.join(ph_dir, "diffusion_head_full.bin"))
+    
+                # --- Connectors (plain state_dicts) ---
+                ac = getattr(self.model.model, "acoustic_connector", None)
+                if ac is not None:
+                    ac_dir = os.path.join(lora_out, "acoustic_connector")
+                    os.makedirs(ac_dir, exist_ok=True)
+                    torch.save(ac.state_dict(), os.path.join(ac_dir, "pytorch_model.bin"))
+    
+                se = getattr(self.model.model, "semantic_connector", None)
+                if se is not None:
+                    se_dir = os.path.join(lora_out, "semantic_connector")
+                    os.makedirs(se_dir, exist_ok=True)
+                    torch.save(se.state_dict(), os.path.join(se_dir, "pytorch_model.bin"))
+    
+            except Exception as e:
+                logger.warning(f"Failed to save LoRA assets: {e}")
+
+
+    # ------------- Build the Trainer -------------
+
+    # Resolve which adapters to apply in samples
+
+    ema_cb = EmaCallback(attr_path="model.prediction_head", decay=0.999, device="cuda")
+
+    # --- CRITICAL FIX: CAST TRAINABLE PARAMS TO FP32 ---
+    # This prevents 'ValueError: Attempting to unscale FP16 gradients'
+    if getattr(training_args, 'fp16', False) or getattr(training_args, 'bf16', False):
+        print('>>> INFO: Enforcing float32 for trainable parameters (LoRA/Head) to fix GradScaler.')
+        for name, param in model.named_parameters():
+            if param.requires_grad:
+                param.data = param.data.to(torch.float32)
+    # ---------------------------------------------------
+
+    trainer = VibeVoiceTrainer(
+        model=model,
+        args=training_args,
+        train_dataset=train_dataset,
+        eval_dataset=eval_dataset,
+        data_collator=data_collator,
+        callbacks=[ema_cb, LoRADebugCallback(log_every_n_steps=(int(getattr(training_args, "logging_steps", 50) or 50)))],
+    )
+
+    # Optional debug pre-training save
+    if getattr(training_args, "debug_save", False):
+        try:
+            debug_dir = os.path.join(training_args.output_dir, "debug_initial")
+            lora_out = os.path.join(debug_dir, "lora")
+            os.makedirs(lora_out, exist_ok=True)
+            logger.info(f"[debug_save] Saving initial (pre-training) model components to: {debug_dir}")
+            # language model adapters / base
+            try:
+                if hasattr(model.model.language_model, "save_pretrained"):
+                    model.model.language_model.save_pretrained(lora_out)
+            except Exception as e_lm:
+                logger.warning(f"[debug_save] Failed to save language_model: {e_lm}")
+            # diffusion head
+            try:
+                if hasattr(model.model, "prediction_head") and hasattr(model.model.prediction_head, "save_pretrained"):
+                    model.model.prediction_head.save_pretrained(os.path.join(lora_out, "diffusion_head"))
+            except Exception as e_head:
+                logger.warning(f"[debug_save] Failed to save prediction_head: {e_head}")
+            # NEW: full diffusion head state_dict as fallback
+            try:
+                ph = getattr(model.model, "prediction_head", None)
+                if ph is not None and hasattr(ph, "state_dict"):
+                    sd = ph.state_dict()
+                    torch.save(sd, os.path.join(lora_out, "diffusion_head_full.bin"))
+                    os.makedirs(os.path.join(lora_out, "diffusion_head"), exist_ok=True)
+                    torch.save(sd, os.path.join(lora_out, "diffusion_head", "diffusion_head_full.bin"))
+            except Exception as e:
+                logger.warning(f"[debug_save] Failed to save FULL diffusion head: {e}")
+            # connectors
+            try:
+                ac_conn = getattr(model.model, "acoustic_connector", None)
+                if ac_conn is not None:
+                    ac_dir = os.path.join(lora_out, "acoustic_connector")
+                    os.makedirs(ac_dir, exist_ok=True)
+                    torch.save(ac_conn.state_dict(), os.path.join(ac_dir, "pytorch_model.bin"))
+            except Exception as e_ac:
+                logger.warning(f"[debug_save] Failed to save acoustic_connector: {e_ac}")
+            try:
+                se_conn = getattr(model.model, "semantic_connector", None)
+                if se_conn is not None:
+                    se_dir = os.path.join(lora_out, "semantic_connector")
+                    os.makedirs(se_dir, exist_ok=True)
+                    torch.save(se_conn.state_dict(), os.path.join(se_dir, "pytorch_model.bin"))
+            except Exception as e_se:
+                logger.warning(f"[debug_save] Failed to save semantic_connector: {e_se}")
+        except Exception as e:
+            logger.warning(f"[debug_save] Unexpected failure saving initial components: {e}")
+
+    if getattr(training_args, "gradient_checkpointing", False):
+        try:
+            model.gradient_checkpointing_enable()
+        except Exception:
+            logger.warning("Failed to enable gradient checkpointing on the model.")
+
+    if training_args.do_train:
+        trainer.train(resume_from_checkpoint=training_args.resume_from_checkpoint)
+    
+        lora_out = os.path.join(training_args.output_dir, "lora")
+        os.makedirs(lora_out, exist_ok=True)
+    
+        # LLM PEFT (if any)
+        lm = getattr(model.model, "language_model", None)
+        if hasattr(lm, "save_pretrained"):
+            lm.save_pretrained(lora_out)
+    
+        # Diffusion head PEFT (if any)
+        ph = getattr(model.model, "prediction_head", None)
+        if hasattr(ph, "save_pretrained"):
+            ph_dir = os.path.join(lora_out, "diffusion_head")
+            os.makedirs(ph_dir, exist_ok=True)
+            ph.save_pretrained(ph_dir)
+    
+        # ALWAYS: full diffusion head state_dict fallback
+        try:
+            if ph is not None and hasattr(ph, "state_dict"):
+                sd = ph.state_dict()
+                torch.save(sd, os.path.join(lora_out, "diffusion_head_full.bin"))
+                ph_dir = os.path.join(lora_out, "diffusion_head")
+                os.makedirs(ph_dir, exist_ok=True)
+                torch.save(sd, os.path.join(ph_dir, "diffusion_head_full.bin"))
+        except Exception as e:
+            logger.warning(f"Failed to save FULL diffusion head at end: {e}")
+    
+        # Connectors (if trained)
+        try:
+            ac = getattr(model.model, "acoustic_connector", None)
+            if ac is not None:
+                ac_dir = os.path.join(lora_out, "acoustic_connector")
+                os.makedirs(ac_dir, exist_ok=True)
+                torch.save(ac.state_dict(), os.path.join(ac_dir, "pytorch_model.bin"))
+        except Exception as e:
+            logger.warning(f"Failed to save acoustic_connector: {e}")
+    
+        try:
+            se = getattr(model.model, "semantic_connector", None)
+            if se is not None:
+                se_dir = os.path.join(lora_out, "semantic_connector")
+                os.makedirs(se_dir, exist_ok=True)
+                torch.save(se.state_dict(), os.path.join(se_dir, "pytorch_model.bin"))
+        except Exception as e:
+            logger.warning(f"Failed to save semantic_connector: {e}")
+
+    if training_args.do_eval and eval_dataset is not None:
+        trainer.evaluate()
+
+
+if __name__ == "__main__":
+    main()

VibeVoice-finetuning/src/vibevoice/configs/qwen2.5_1.5b_64k.json

@@ -0,0 +1,112 @@
+{
+  "_attn_implementation_autoset": true,
+  "acoustic_vae_dim": 64,
+  "acoustic_tokenizer_config": {
+    "causal": true,
+    "channels": 1,
+    "conv_bias": true,
+    "conv_norm": "none",
+    "corpus_normalize": 0.0,
+    "decoder_depths": null,
+    "decoder_n_filters": 32,
+    "decoder_ratios": [
+      8,
+      5,
+      5,
+      4,
+      2,
+      2
+    ],
+    "disable_last_norm": true,
+    "encoder_depths": "3-3-3-3-3-3-8",
+    "encoder_n_filters": 32,
+    "encoder_ratios": [
+      8,
+      5,
+      5,
+      4,
+      2,
+      2
+    ],
+    "fix_std": 0.5,
+    "layer_scale_init_value": 1e-06,
+    "layernorm": "RMSNorm",
+    "layernorm_elementwise_affine": true,
+    "layernorm_eps": 1e-05,
+    "mixer_layer": "depthwise_conv",
+    "model_type": "vibepod_acoustic_tokenizer",
+    "pad_mode": "constant",
+    "std_dist_type": "gaussian",
+    "vae_dim": 64,
+    "weight_init_value": 0.01
+  },
+  "decoder_config": {
+    "attention_dropout": 0.0,
+    "hidden_act": "silu",
+    "hidden_size": 1536,
+    "initializer_range": 0.02,
+    "intermediate_size": 8960,
+    "max_position_embeddings": 65536,
+    "max_window_layers": 28,
+    "model_type": "qwen2",
+    "num_attention_heads": 12,
+    "num_hidden_layers": 28,
+    "num_key_value_heads": 2,
+    "rms_norm_eps": 1e-06,
+    "rope_scaling": null,
+    "rope_theta": 1000000.0,
+    "sliding_window": null,
+    "tie_word_embeddings": true,
+    "torch_dtype": "bfloat16",
+    "use_cache": true,
+    "use_sliding_window": false,
+    "vocab_size": 151936
+  },
+  "diffusion_head_config": {
+    "ddpm_batch_mul": 4,
+    "ddpm_beta_schedule": "cosine",
+    "ddpm_num_inference_steps": 20,
+    "ddpm_num_steps": 1000,
+    "diffusion_type": "ddpm",
+    "head_ffn_ratio": 3.0,
+    "head_layers": 4,
+    "hidden_size": 1536,
+    "latent_size": 64,
+    "model_type": "vibepod_diffusion_head",
+    "prediction_type": "v_prediction",
+    "rms_norm_eps": 1e-05,
+    "speech_vae_dim": 64
+  },
+  "model_type": "vibepod",
+  "semantic_tokenizer_config": {
+    "causal": true,
+    "channels": 1,
+    "conv_bias": true,
+    "conv_norm": "none",
+    "corpus_normalize": 0.0,
+    "disable_last_norm": true,
+    "encoder_depths": "3-3-3-3-3-3-8",
+    "encoder_n_filters": 32,
+    "encoder_ratios": [
+      8,
+      5,
+      5,
+      4,
+      2,
+      2
+    ],
+    "fix_std": 0,
+    "layer_scale_init_value": 1e-06,
+    "layernorm": "RMSNorm",
+    "layernorm_elementwise_affine": true,
+    "layernorm_eps": 1e-05,
+    "mixer_layer": "depthwise_conv",
+    "model_type": "vibepod_semantic_tokenizer",
+    "pad_mode": "constant",
+    "std_dist_type": "none",
+    "vae_dim": 128,
+    "weight_init_value": 0.01
+  },
+  "semantic_vae_dim": 128,
+  "torch_dtype": "bfloat16"
+}

VibeVoice-finetuning/src/vibevoice/configs/qwen2.5_7b_32k.json

@@ -0,0 +1,113 @@
+{
+  "_attn_implementation_autoset": true,
+  "acoustic_vae_dim": 64,
+  "acoustic_tokenizer_config": {
+    "causal": true,
+    "channels": 1,
+    "conv_bias": true,
+    "conv_norm": "none",
+    "corpus_normalize": 0.0,
+    "decoder_depths": null,
+    "decoder_n_filters": 32,
+    "decoder_ratios": [
+      8,
+      5,
+      5,
+      4,
+      2,
+      2
+    ],
+    "disable_last_norm": true,
+    "encoder_depths": "3-3-3-3-3-3-8",
+    "encoder_n_filters": 32,
+    "encoder_ratios": [
+      8,
+      5,
+      5,
+      4,
+      2,
+      2
+    ],
+    "fix_std": 0.5,
+    "layer_scale_init_value": 1e-06,
+    "layernorm": "RMSNorm",
+    "layernorm_elementwise_affine": true,
+    "layernorm_eps": 1e-05,
+    "mixer_layer": "depthwise_conv",
+    "model_type": "vibepod_acoustic_tokenizer",
+    "pad_mode": "constant",
+    "std_dist_type": "gaussian",
+    "vae_dim": 64,
+    "weight_init_value": 0.01
+  },
+  "decoder_config": {
+    "attention_dropout": 0.0,
+    "hidden_act": "silu",
+    "hidden_size": 3584,
+    "initializer_range": 0.02,
+    "intermediate_size": 18944,
+    "max_position_embeddings": 32768,
+    "max_window_layers": 28,
+    "model_type": "qwen2",
+    "num_attention_heads": 28,
+    "num_hidden_layers": 28,
+    "num_key_value_heads": 4,
+    "rms_norm_eps": 1e-06,
+    "rope_theta": 1000000.0,
+    "sliding_window": null,
+    "tie_word_embeddings": false,
+    "torch_dtype": "bfloat16",
+    "transformers_version": "4.40.1",
+    "use_cache": true,
+    "use_mrope": false,
+    "use_sliding_window": false,
+    "vocab_size": 152064
+  },
+  "diffusion_head_config": {
+    "ddpm_batch_mul": 4,
+    "ddpm_beta_schedule": "cosine",
+    "ddpm_num_inference_steps": 20,
+    "ddpm_num_steps": 1000,
+    "diffusion_type": "ddpm",
+    "head_ffn_ratio": 3.0,
+    "head_layers": 4,
+    "hidden_size": 3584,
+    "latent_size": 64,
+    "model_type": "vibepod_diffusion_head",
+    "prediction_type": "v_prediction",
+    "rms_norm_eps": 1e-05,
+    "speech_vae_dim": 64
+  },
+  "model_type": "vibepod",
+  "semantic_tokenizer_config": {
+    "causal": true,
+    "channels": 1,
+    "conv_bias": true,
+    "conv_norm": "none",
+    "corpus_normalize": 0.0,
+    "disable_last_norm": true,
+    "encoder_depths": "3-3-3-3-3-3-8",
+    "encoder_n_filters": 32,
+    "encoder_ratios": [
+      8,
+      5,
+      5,
+      4,
+      2,
+      2
+    ],
+    "fix_std": 0,
+    "layer_scale_init_value": 1e-06,
+    "layernorm": "RMSNorm",
+    "layernorm_elementwise_affine": true,
+    "layernorm_eps": 1e-05,
+    "mixer_layer": "depthwise_conv",
+    "model_type": "vibepod_semantic_tokenizer",
+    "pad_mode": "constant",
+    "std_dist_type": "none",
+    "vae_dim": 128,
+    "weight_init_value": 0.01
+  },
+  "semantic_vae_dim": 128,
+  "torch_dtype": "bfloat16"
+}

VibeVoice-finetuning/src/vibevoice/modular/configuration_vibevoice.py

@@ -0,0 +1,248 @@
+""" VibeVoice_AcousticTokenizer model configuration"""
+
+from typing import Dict, List, Optional, Tuple
+
+from transformers.configuration_utils import PretrainedConfig 
+from transformers.utils import logging
+
+from transformers.models.qwen2.configuration_qwen2 import Qwen2Config
+
+logger = logging.get_logger(__name__)
+
+
+class VibeVoiceAcousticTokenizerConfig(PretrainedConfig):
+    model_type = "vibevoice_acoustic_tokenizer"
+
+    def __init__(
+        self,
+        channels: int = 1,
+        corpus_normalize: float = 0.0,
+        causal: bool = True,
+        vae_dim: int = 64,
+        fix_std: float = 0.5,
+        std_dist_type: str = 'gaussian',
+        # common 
+        mixer_layer: str = 'depthwise_conv',
+        conv_norm: str = 'none',
+        pad_mode: str = 'constant',
+        disable_last_norm: bool = True,
+        layernorm: str = 'RMSNorm',
+        layernorm_eps: float = 1e-5,
+        layernorm_elementwise_affine: bool = True,
+        conv_bias: bool = True,
+        layer_scale_init_value: float = 1e-6,
+        weight_init_value: float = 1e-2,
+        # encoder specific
+        encoder_n_filters: int = 32,
+        encoder_ratios: Optional[List[int]] = [8,5,5,4,2,2],
+        encoder_depths: str = "3-3-3-3-3-3-8",
+        # decoder specific
+        decoder_n_filters: int = 32,
+        decoder_ratios: Optional[List[int]] = None, # if None, same as encoder
+        decoder_depths: Optional[str] = None,
+        **kwargs
+    ):
+        super().__init__(**kwargs)
+        self.channels = channels
+        self.corpus_normalize = corpus_normalize
+        self.causal = causal
+        self.vae_dim = vae_dim
+        self.fix_std = fix_std
+        self.std_dist_type = std_dist_type
+        
+        # common parameters
+        self.conv_norm = conv_norm
+        self.pad_mode = pad_mode
+        self.layernorm_eps = layernorm_eps
+        self.disable_last_norm = disable_last_norm
+        self.layernorm = layernorm
+        self.layernorm_elementwise_affine = layernorm_elementwise_affine
+        self.conv_bias = conv_bias
+        self.layer_scale_init_value = layer_scale_init_value
+        self.weight_init_value = weight_init_value
+        self.mixer_layer = mixer_layer
+
+        # encoder specific parameters
+        self.encoder_n_filters = encoder_n_filters
+        self.encoder_ratios = encoder_ratios
+        self.encoder_depths = encoder_depths
+        
+        # decoder specific parameters
+        self.decoder_ratios = decoder_ratios if decoder_ratios is not None else encoder_ratios
+        self.decoder_n_filters = decoder_n_filters
+        self.decoder_depths = decoder_depths
+
+
+class VibeVoiceSemanticTokenizerConfig(PretrainedConfig):
+    model_type = "vibevoice_semantic_tokenizer"
+    
+    def __init__(
+        self,
+        channels: int = 1,
+        corpus_normalize: float = 0.0,
+        causal: bool = True,
+        vae_dim: int = 64,
+        fix_std: float = 0,
+        std_dist_type: str = 'none',
+        # common 
+        mixer_layer: str = 'depthwise_conv',
+        conv_norm: str = 'none',
+        pad_mode: str = 'constant',
+        disable_last_norm: bool = True,
+        layernorm: str = 'RMSNorm',
+        layernorm_eps: float = 1e-5,
+        layernorm_elementwise_affine: bool = True,
+        conv_bias: bool = True,
+        layer_scale_init_value: float = 1e-6,
+        weight_init_value: float = 1e-2,
+        # encoder specific
+        encoder_n_filters: int = 32,
+        encoder_ratios: Optional[List[int]] = [8,5,5,4,2,2],
+        encoder_depths: str = "3-3-3-3-3-3-8",
+        **kwargs
+    ):
+        super().__init__(**kwargs)
+        self.channels = channels
+        self.corpus_normalize = corpus_normalize
+        self.causal = causal
+        self.vae_dim = vae_dim
+        self.fix_std = fix_std
+        self.std_dist_type = std_dist_type
+        
+        # common parameters
+        self.conv_norm = conv_norm
+        self.pad_mode = pad_mode
+        self.layernorm_eps = layernorm_eps
+        self.disable_last_norm = disable_last_norm
+        self.layernorm = layernorm
+        self.layernorm_elementwise_affine = layernorm_elementwise_affine
+        self.conv_bias = conv_bias
+        self.layer_scale_init_value = layer_scale_init_value
+        self.weight_init_value = weight_init_value
+        self.mixer_layer = mixer_layer
+
+        # encoder specific parameters
+        self.encoder_n_filters = encoder_n_filters
+        self.encoder_ratios = encoder_ratios
+        self.encoder_depths = encoder_depths
+        
+
+class VibeVoiceDiffusionHeadConfig(PretrainedConfig):
+    model_type = "vibevoice_diffusion_head"
+
+    def __init__(
+        self,
+        hidden_size=768,
+        head_layers=4,
+        head_ffn_ratio=3.0,
+        rms_norm_eps=1e-5,
+        latent_size=64,
+        speech_vae_dim=None,
+        prediction_type="v_prediction",
+        diffusion_type="ddpm",
+        ddpm_num_steps=1000,
+        ddpm_num_inference_steps=20,
+        ddpm_beta_schedule="cosine",
+        ddpm_batch_mul=4,
+        **kwargs
+    ):
+        self.hidden_size = hidden_size
+        self.head_layers = head_layers
+        self.head_ffn_ratio = head_ffn_ratio
+        self.rms_norm_eps = rms_norm_eps
+        self.latent_size = latent_size
+        self.speech_vae_dim = speech_vae_dim
+        self.prediction_type = prediction_type
+        self.diffusion_type = diffusion_type
+        self.ddpm_num_steps = ddpm_num_steps
+        self.ddpm_num_inference_steps = ddpm_num_inference_steps
+        self.ddpm_beta_schedule = ddpm_beta_schedule
+        self.ddpm_batch_mul = ddpm_batch_mul
+        
+        super().__init__(**kwargs)
+
+class VibeVoiceConfig(PretrainedConfig):
+    model_type = "vibevoice"
+    is_composition = True
+    sub_configs = {
+        "acoustic_tokenizer_config": VibeVoiceAcousticTokenizerConfig, 
+        "semantic_tokenizer_config": VibeVoiceSemanticTokenizerConfig,
+        "decoder_config": Qwen2Config,
+        "diffusion_head_config": VibeVoiceDiffusionHeadConfig,
+    }
+    # keys_to_ignore_at_inference = ["past_key_values"]
+    # Default tensor parallel plan for base model `Qwen2`
+    base_model_tp_plan = {
+        "layers.*.self_attn.q_proj": "colwise",
+        "layers.*.self_attn.k_proj": "colwise",
+        "layers.*.self_attn.v_proj": "colwise",
+        "layers.*.self_attn.o_proj": "rowwise",
+        "layers.*.mlp.gate_proj": "colwise",
+        "layers.*.mlp.up_proj": "colwise",
+        "layers.*.mlp.down_proj": "rowwise",
+    }
+    
+    def __init__(
+        self,
+        acoustic_tokenizer_config=None,
+        semantic_tokenizer_config=None,
+        decoder_config=None,
+        diffusion_head_config=None,
+        **kwargs
+    ):
+
+        # kwargs["_attn_implementation"] = "flash_attention_2"
+        kwargs["_attn_implementation_autoset"] = False 
+
+        if acoustic_tokenizer_config is None:
+            self.acoustic_tokenizer_config = self.sub_configs["acoustic_tokenizer_config"]()
+        elif isinstance(acoustic_tokenizer_config, dict):
+            acoustic_tokenizer_config["model_type"] = "vibevoice_acoustic_tokenizer"
+            self.acoustic_tokenizer_config = self.sub_configs["acoustic_tokenizer_config"](**acoustic_tokenizer_config)
+        elif isinstance(acoustic_tokenizer_config, VibeVoiceAcousticTokenizerConfig):
+            # If an instance of the config class is provided
+            self.acoustic_tokenizer_config = acoustic_tokenizer_config
+
+        if semantic_tokenizer_config is None:
+            self.semantic_tokenizer_config = self.sub_configs["semantic_tokenizer_config"]()
+        elif isinstance(semantic_tokenizer_config, dict):
+            semantic_tokenizer_config["model_type"] = "vibevoice_semantic_tokenizer"
+            self.semantic_tokenizer_config = self.sub_configs["semantic_tokenizer_config"](**semantic_tokenizer_config)
+        elif isinstance(semantic_tokenizer_config, VibeVoiceSemanticTokenizerConfig):
+            # If an instance of the config class is provided
+            self.semantic_tokenizer_config = semantic_tokenizer_config
+
+        if decoder_config is None:
+            self.decoder_config = self.sub_configs["decoder_config"]()
+        elif isinstance(decoder_config, dict):
+            # If a dictionary is provided, instantiate the config class with it
+            # self.decoder_config = self.sub_configs["decoder_config"](**decoder_config)
+            if decoder_config.get("model_type", '') == "qwen2":
+                self.decoder_config = Qwen2Config(**decoder_config)
+            else:
+                raise ValueError(f"Unsupported decoder model type: {decoder_config.get('model_type', '')}")
+        elif isinstance(decoder_config, (Qwen2Config,)):
+            # If an instance of the config class is provided
+            self.decoder_config = decoder_config
+
+        if diffusion_head_config is None:
+            self.diffusion_head_config = self.sub_configs["diffusion_head_config"]()
+        elif isinstance(diffusion_head_config, dict):
+            diffusion_head_config["model_type"] = "vibevoice_diffusion_head"
+            self.diffusion_head_config = self.sub_configs["diffusion_head_config"](**diffusion_head_config)
+        elif isinstance(diffusion_head_config, VibeVoiceDiffusionHeadConfig):
+            # If an instance of the config class is provided
+            self.diffusion_head_config = diffusion_head_config
+
+        # other parameters
+        self.acoustic_vae_dim = getattr(self.acoustic_tokenizer_config, 'vae_dim', 64)
+        self.semantic_vae_dim = getattr(self.semantic_tokenizer_config, 'vae_dim', 128)
+
+        super().__init__(**kwargs)
+
+__all__ = [
+    "VibeVoiceAcousticTokenizerConfig", 
+    "VibeVoiceSemanticTokenizerConfig", 
+    "VibeVoiceDiffusionHeadConfig", 
+    "VibeVoiceConfig"
+]