Audio Classification
Transformers
ONNX
Safetensors
English
dualturn_endpointing
feature-extraction
turn-taking
endpointing
end-of-turn
voice-activity-detection
voice-agents
conversation
speech
audio
mimi
dualturn
real-time
custom_code
Instructions to use anyreach-ai/dualturn-endpointing with libraries, inference providers, notebooks, and local apps. Follow these links to get started.
- Libraries
- Transformers
How to use anyreach-ai/dualturn-endpointing with Transformers:
# Use a pipeline as a high-level helper from transformers import pipeline pipe = pipeline("audio-classification", model="anyreach-ai/dualturn-endpointing", trust_remote_code=True)# Load model directly from transformers import AutoModel model = AutoModel.from_pretrained("anyreach-ai/dualturn-endpointing", trust_remote_code=True, device_map="auto") - Notebooks
- Google Colab
- Kaggle
Upload src/dualturn/config/base_config.py with huggingface_hub
Browse files
src/dualturn/config/base_config.py
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| 1 |
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"""
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| 2 |
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Configuration management for turn-taking model training.
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| 3 |
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| 4 |
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Provides dataclass-based config with validation and YAML loading.
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| 5 |
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"""
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from dataclasses import dataclass, field, asdict
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| 8 |
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from pathlib import Path
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| 9 |
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from typing import Literal, Optional, List
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import yaml
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| 11 |
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@dataclass
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class ModelConfig:
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"""Model architecture hyperparameters."""
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# Qwen backbone
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qwen_model_name: str = "Qwen/Qwen2.5-0.5B"
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| 19 |
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# LoRA config
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lora_r: int = 16
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lora_alpha: int = 32
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| 22 |
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lora_dropout: float = 0.05
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| 23 |
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lora_target_modules: List[str] = field(
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| 24 |
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default_factory=lambda: ["q_proj", "k_proj", "v_proj", "o_proj", "gate_proj", "up_proj", "down_proj"]
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| 25 |
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)
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# Audio embeddings
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num_codebooks: int = 8
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codebook_size: int = 2049
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hidden_dim: int = 896
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# Depth predictor
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depth_dim: int = 512
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# Backbone type: "qwen" (default LLM) or "lstm" (lightweight baseline)
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backbone_type: str = "qwen"
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# LSTM backbone config (only used when backbone_type="lstm")
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lstm_hidden_dim: int = 512
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| 40 |
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lstm_num_layers: int = 2
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| 41 |
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lstm_dropout: float = 0.1
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| 42 |
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lstm_bidirectional: bool = True
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| 43 |
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| 44 |
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# Transformer backbone config (only used when backbone_type="transformer")
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transformer_num_layers: int = 10
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transformer_num_heads: int = 14
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transformer_ff_dim: int = 3584
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transformer_dropout: float = 0.1
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| 49 |
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# Layer probing: None = last layer (default), "weighted" = learned weighted avg,
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| 51 |
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# int = specific layer index (0=embedding, 1-24=transformer layers)
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probe_layer: Optional[str] = None
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# Random init: if True, initialize Qwen with random weights instead of pretrained
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random_init: bool = False
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# Input mode: "discrete" (codebook embeddings) or "continuous" (Mimi encoder features)
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input_mode: str = "discrete"
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mimi_feat_dim: int = 512
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| 61 |
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# AudioAdapter (dual-task only)
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| 62 |
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use_audio_adapter: bool = False
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audio_adapter_heads: int = 8
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audio_adapter_ff_dim: int = 2048
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| 65 |
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audio_adapter_dropout: float = 0.1
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| 66 |
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| 67 |
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# Prediction head architecture: "linear" (legacy) or "mlp" (2-layer MLP for sparse tasks)
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head_type: str = "linear"
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| 69 |
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head_hidden_dim: int = 256
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| 70 |
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head_dropout: float = 0.1
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# Per-task learned layer attention (ELMo-style, per task)
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| 73 |
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per_task_layer_attention: bool = False
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| 74 |
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| 75 |
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# Per-codebook output heads in depth predictor (vs shared MLP)
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| 76 |
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per_codebook_heads: bool = False
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| 77 |
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| 78 |
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# Two-stream backbone (autoresearch best_model): siamese per-channel + cross-channel block.
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| 79 |
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# Requires backbone_type="transformer" and input_mode="continuous".
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| 80 |
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two_stream_backbone: bool = False
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| 81 |
+
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| 82 |
+
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| 83 |
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@dataclass
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| 84 |
+
class DataConfig:
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| 85 |
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"""Data loading and augmentation hyperparameters."""
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| 86 |
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# Paths
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| 87 |
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processed_dir: str = "data/otospeech_processed_npy"
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| 88 |
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splits_path: str = "data/splits.json"
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| 89 |
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metadata_path: str = "data/otospeech_metadata.json"
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| 90 |
+
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| 91 |
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# Windowing
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| 92 |
+
window_frames: int = 125 # 10s at 12.5Hz
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| 93 |
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hop_frames_train: int = 25 # 2s hop for training (overlapping)
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| 94 |
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hop_frames_val: int = 125 # 10s hop for val/test (non-overlapping)
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+
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| 96 |
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# Augmentation
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| 97 |
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channel_swap_prob: float = 0.5 # Channel swap augmentation (train only)
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| 98 |
+
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| 99 |
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# Soft labels
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| 100 |
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soft_labels: bool = True
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sigma_before: float = 3.0 # frames (240ms at 12.5Hz)
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sigma_after: float = 1.0 # frames (80ms at 12.5Hz)
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| 104 |
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# Balanced sampling
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balanced_sampling: bool = True
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shift_oversample_ratio: float = 3.0 # Oversample windows with shifts
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dynamic_window_sampling: bool = False # Random window positions around shifts (Stage 3)
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# Event label widening: dilate sparse labels to N frames (1 = no widening)
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event_label_width: int = 1
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| 112 |
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# Dual-task data paths (SWB + Oto separate)
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swb_processed_dir: Optional[str] = None
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| 114 |
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swb_splits_path: Optional[str] = None
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| 115 |
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oto_processed_dir: Optional[str] = None
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| 116 |
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oto_splits_path: Optional[str] = None
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| 117 |
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oto_metadata_path: Optional[str] = None
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| 118 |
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max_text_tokens: int = 512
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| 119 |
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| 120 |
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# S2S dataset (production call recordings) — included as extra dataset alongside Oto/SWB
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s2s_processed_dir: Optional[str] = None
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s2s_splits_path: Optional[str] = None
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| 123 |
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| 124 |
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@dataclass
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class TrainingConfig:
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| 127 |
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"""Training loop hyperparameters."""
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| 128 |
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# Stage
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| 129 |
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stage: Literal["stage2", "stage2_dual_task", "stage3", "linear_probe", "full_finetune", "lstm_baseline"] = "stage2"
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| 130 |
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| 131 |
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# Optimization
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| 132 |
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learning_rate: float = 1e-4
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| 133 |
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weight_decay: float = 0.01
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| 134 |
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adam_beta1: float = 0.9
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| 135 |
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adam_beta2: float = 0.999
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| 136 |
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adam_eps: float = 1e-8
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| 137 |
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max_grad_norm: float = 1.0 # Gradient clipping
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| 138 |
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# Scheduler
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| 140 |
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warmup_steps: int = 1000
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| 141 |
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scheduler_type: str = "cosine" # "cosine", "linear", or "constant"
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| 142 |
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# Training loop
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| 144 |
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batch_size: int = 16
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| 145 |
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num_epochs: int = 5
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| 146 |
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max_steps: Optional[int] = None # If set, overrides num_epochs
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| 147 |
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gradient_accumulation_steps: int = 1
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| 148 |
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| 149 |
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# Loss weights -- context-aware tasks (v3)
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| 150 |
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weight_eot: float = 1.0 # End of Turn
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| 151 |
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weight_hold: float = 1.0 # Turn Hold / Pause
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| 152 |
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weight_bot: float = 1.0 # Beginning of Turn
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| 153 |
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weight_bc: float = 1.0 # Backchannel
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| 154 |
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weight_vad: float = 1.0 # Voice Activity Detection (BCE)
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| 155 |
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weight_codebook: float = 0.0 # Codebook prediction loss weight
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| 156 |
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| 157 |
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# Sparse event loss type: "focal" (recommended) or "wbce" (weighted BCE)
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| 158 |
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# Focal loss avoids gradient spikes from high pos_weight by down-weighting
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| 159 |
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# easy negatives adaptively. Use "wbce" only for backward compat.
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| 160 |
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sparse_loss_type: str = "focal"
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| 161 |
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| 162 |
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# Focal loss per-task alpha (positive class weight, in [0,1]).
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| 163 |
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# Higher alpha = more weight on positives. Combined with gamma, this is
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| 164 |
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# much gentler than wBCE pos_weight while achieving better gradient balance.
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| 165 |
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eot_alpha: float = 0.75
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| 166 |
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hold_alpha: float = 0.60
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| 167 |
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bot_alpha: float = 0.80
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| 168 |
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bc_alpha: float = 0.80
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| 169 |
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focal_gamma_sparse: float = 2.0
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| 170 |
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| 171 |
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# Per-task pos_weight for wBCE (only used when sparse_loss_type="wbce").
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| 172 |
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# Derived from combined dataset positive rates after 5-frame widening + VAD mask:
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| 173 |
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# EOT~2.7%, HOLD~5.6%, BOT~1.3%, BC~1.4%
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| 174 |
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eot_pos_weight: float = 37.0
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| 175 |
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hold_pos_weight: float = 17.0
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| 176 |
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bot_pos_weight: float = 75.0
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| 177 |
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bc_pos_weight: float = 73.0
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| 178 |
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sparse_pos_weight: float = 20.0 # legacy fallback if per-task not set in config
|
| 179 |
+
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| 180 |
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# Legacy weights (backward compat -- set to 0.0 for new training)
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| 181 |
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weight_shift: float = 0.0
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| 182 |
+
weight_end: float = 0.0
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| 183 |
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weight_start: float = 0.0
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| 184 |
+
focal_gamma: float = 2.0
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| 185 |
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focal_alpha: float = 0.75
|
| 186 |
+
|
| 187 |
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# Future VAD projection -- VAP-style binned voice activity prediction
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| 188 |
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weight_fvad: float = 0.0 # Future VAD projection loss weight (0 = disabled)
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| 189 |
+
fvad_bins: List[int] = field(default_factory=lambda: [3, 6, 12, 25])
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| 190 |
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# Bin edges in frames at 12.5Hz: [3,6,12,25] ->
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| 191 |
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# bin0: t+1..t+3 (80-240ms), bin1: t+4..t+6 (320-480ms)
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| 192 |
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# bin2: t+7..t+12 (560-960ms), bin3: t+13..t+25 (1.04-2.0s)
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| 193 |
+
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| 194 |
+
# Text loss (dual-task)
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| 195 |
+
weight_text: float = 0.0 # ASR text prediction loss weight (Stage 2 dual-task)
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| 196 |
+
mode: str = "codebook" # forward mode: "codebook", "dual_task", "shift"
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| 197 |
+
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| 198 |
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# Full finetune: merge LoRA into base and unfreeze all params
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| 199 |
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full_finetune: bool = False
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| 200 |
+
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| 201 |
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# Per-codebook weighting: [code0, ..., code7]. Default equal.
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| 202 |
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# Moshi-inspired: code0 carries prosody -> weight higher.
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| 203 |
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codebook_weights: Optional[List[float]] = None
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| 204 |
+
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| 205 |
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# Future VAD auxiliary task (legacy, kept for old config compat)
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| 206 |
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vad_lookahead_frames: int = 3
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| 207 |
+
|
| 208 |
+
# Label mode
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| 209 |
+
label_mode: str = "start_end" # "shift" (legacy) or "start_end" (v2)
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| 210 |
+
|
| 211 |
+
# Validation & checkpointing
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| 212 |
+
eval_every_steps: int = 500
|
| 213 |
+
save_every_steps: int = 1000
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| 214 |
+
early_stopping_patience: int = 5 # Stop after N evals without improvement
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| 215 |
+
max_val_batches: Optional[int] = None # Limit val batches (None = full val set; useful for quick tests)
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| 216 |
+
save_generated_audio: bool = False # Save autoregressive audio samples each validation (Stage-1)
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| 217 |
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gen_context_frames: int = 125 # Context frames for generation (125 = 10s at 12.5Hz)
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| 218 |
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gen_gen_frames: int = 62 # Frames to generate (62 = ~5s at 12.5Hz)
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| 219 |
+
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| 220 |
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# System
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| 221 |
+
num_workers: int = 4
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| 222 |
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pin_memory: bool = True
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| 223 |
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mixed_precision: bool = False # bf16 training (A100 native support)
|
| 224 |
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seed: int = 42
|
| 225 |
+
|
| 226 |
+
# Logging
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| 227 |
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log_every_steps: int = 10
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| 228 |
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log_perplexity: bool = True # Log perplexity for codebook prediction (Stage 2)
|
| 229 |
+
wandb_project: Optional[str] = None # "turn-taking-interspeech"
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| 230 |
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wandb_run_name: Optional[str] = None
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| 231 |
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experiment_name: str = "default"
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| 232 |
+
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| 233 |
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# Paths
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| 234 |
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checkpoint_dir: str = "checkpoints"
|
| 235 |
+
log_dir: str = "logs"
|
| 236 |
+
stage2_checkpoint: Optional[str] = None # Path to Stage 2 checkpoint (for Stage 3)
|
| 237 |
+
|
| 238 |
+
|
| 239 |
+
@dataclass
|
| 240 |
+
class ExperimentConfig:
|
| 241 |
+
"""Complete experiment configuration."""
|
| 242 |
+
model: ModelConfig = field(default_factory=ModelConfig)
|
| 243 |
+
data: DataConfig = field(default_factory=DataConfig)
|
| 244 |
+
training: TrainingConfig = field(default_factory=TrainingConfig)
|
| 245 |
+
|
| 246 |
+
@classmethod
|
| 247 |
+
def from_yaml(cls, path: str) -> "ExperimentConfig":
|
| 248 |
+
"""Load config from YAML file."""
|
| 249 |
+
with open(path, 'r') as f:
|
| 250 |
+
data = yaml.safe_load(f)
|
| 251 |
+
|
| 252 |
+
config = cls(
|
| 253 |
+
model=ModelConfig(**data.get("model", {})),
|
| 254 |
+
data=DataConfig(**data.get("data", {})),
|
| 255 |
+
training=TrainingConfig(**data.get("training", {})),
|
| 256 |
+
)
|
| 257 |
+
return config
|
| 258 |
+
|
| 259 |
+
def to_yaml(self, path: str):
|
| 260 |
+
"""Save config to YAML file."""
|
| 261 |
+
data = {
|
| 262 |
+
"model": asdict(self.model),
|
| 263 |
+
"data": asdict(self.data),
|
| 264 |
+
"training": asdict(self.training),
|
| 265 |
+
}
|
| 266 |
+
with open(path, 'w') as f:
|
| 267 |
+
yaml.dump(data, f, default_flow_style=False, sort_keys=False)
|
| 268 |
+
|
| 269 |
+
def to_dict(self) -> dict:
|
| 270 |
+
"""Convert to dictionary."""
|
| 271 |
+
return {
|
| 272 |
+
"model": asdict(self.model),
|
| 273 |
+
"data": asdict(self.data),
|
| 274 |
+
"training": asdict(self.training),
|
| 275 |
+
}
|