Upload folder using huggingface_hub
Browse files- README.md +49 -18
- __pycache__/pipeline.cpython-310.pyc +0 -0
- config.json +1 -1
- pipeline.py +73 -12
- requirements.txt +8 -0
- test.py +12 -0
README.md
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@@ -17,6 +17,8 @@ A regression probe trained on top of Whisper-large-v3 encoder features for estim
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**Score scale:** 1.0 (most severe dysarthria) to 7.0 (typical speech)
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## Model Description
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This model uses a three-stage training pipeline:
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|---|---|---|
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| `proposed_L_coarse_tau0.1` | Proposed (L_coarse) | 0.1 |
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| `proposed_L_coarse_tau1.0` | Proposed (L_coarse) | 1.0 |
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| `proposed_L_coarse_tau50.0` | Proposed (L_coarse) | 50.0 |
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| `proposed_L_coarse_tau100.0` | Proposed (L_coarse) | 100.0 |
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| `proposed_L_cont_tau0.1` | Proposed (L_cont) | 0.1 |
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| `proposed_L_dis_tau1.0` | Proposed (L_dis) | 1.0 |
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| `rank-n-contrast_tau100.0` | Rank-N-Contrast | 100.0 |
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| `simclr_tau0.1` | SimCLR | 0.1 |
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## Usage
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### With the custom pipeline
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from huggingface_hub import snapshot_download
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# Download the model
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model_dir = snapshot_download("jaesungbae/
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# Load pipeline (defaults to
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from pipeline import PreTrainedPipeline
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pipe = PreTrainedPipeline(model_dir)
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# Run inference
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result = pipe("/path/to/audio.wav")
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print(result)
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# {"severity_score": 4.25, "raw_score": 4.2483, "model_name": "
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```
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### Select a specific checkpoint
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result = pipe("/path/to/audio.wav", model_name="proposed_L_dis_tau1.0")
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```
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### List available checkpoints
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```python
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```bash
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python inference.py \
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--wav /path/to/audio.wav \
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--checkpoint ./checkpoints/stage3/
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```
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## Requirements
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- Python 3.10+
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- PyTorch + torchaudio
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- transformers >= 4.40.0
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- safetensors >= 0.4.0
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- Silero VAD (loaded via `torch.hub` at runtime)
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## Runtime Dependencies
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This model loads **openai/whisper-large-v3** (~6GB) and **Silero VAD** at initialization time. Ensure sufficient memory is available.
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## Citation
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```bibtex
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**Score scale:** 1.0 (most severe dysarthria) to 7.0 (typical speech)
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**GitHub:** [JaesungBae/DA-DSQA](https://github.com/JaesungBae/DA-DSQA)
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## Model Description
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This model uses a three-stage training pipeline:
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|---|---|---|
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| `proposed_L_coarse_tau0.1` | Proposed (L_coarse) | 0.1 |
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| `proposed_L_coarse_tau1.0` | Proposed (L_coarse) | 1.0 |
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| `proposed_L_coarse_tau10.0` | Proposed (L_coarse) | 10.0 |
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| `proposed_L_coarse_tau50.0` | Proposed (L_coarse) | 50.0 |
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| **`proposed_L_coarse_tau100.0`** (default) | Proposed (L_coarse) | 100.0 |
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| `proposed_L_cont_tau0.1` | Proposed (L_cont) | 0.1 |
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| `proposed_L_dis_tau1.0` | Proposed (L_dis) | 1.0 |
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| `rank-n-contrast_tau100.0` | Rank-N-Contrast | 100.0 |
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| `simclr_tau0.1` | SimCLR | 0.1 |
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## Setup
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### 1. Create conda environment
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```bash
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conda create -n da-dsqa python=3.10 -y
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conda activate da-dsqa
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```
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### 2. Install PyTorch with CUDA
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```bash
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conda install pytorch torchaudio -c pytorch -y
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```
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> For a GPU build with a specific CUDA version, see [pytorch.org](https://pytorch.org/get-started/locally/) for the appropriate command.
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### 3. Install remaining dependencies
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```bash
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pip install -r requirements.txt
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```
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> **Note:** [Silero VAD](https://github.com/snakers4/silero-vad) is loaded automatically at runtime via `torch.hub` — no separate installation needed.
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### Runtime Dependencies
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This model loads **openai/whisper-large-v3** (~6GB) and **Silero VAD** at initialization time. Ensure sufficient memory is available.
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## Usage
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### With the custom pipeline
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from huggingface_hub import snapshot_download
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# Download the model
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model_dir = snapshot_download("jaesungbae/da-dsqa")
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# Load pipeline (defaults to proposed_L_coarse_tau100.0)
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from pipeline import PreTrainedPipeline
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pipe = PreTrainedPipeline(model_dir)
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# Run inference
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result = pipe("/path/to/audio.wav")
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print(result)
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# {"severity_score": 4.25, "raw_score": 4.2483, "model_name": "proposed_L_coarse_tau100.0"}
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```
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### Select a specific checkpoint
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result = pipe("/path/to/audio.wav", model_name="proposed_L_dis_tau1.0")
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```
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### Batch inference
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```python
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results = pipe.batch_inference([
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"/path/to/audio1.wav",
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"/path/to/audio2.wav",
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"/path/to/audio3.wav",
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])
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for r in results:
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print(f"{r['file']}: {r['severity_score']}")
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```
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### List available checkpoints
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```python
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```bash
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python inference.py \
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--wav /path/to/audio.wav \
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--checkpoint ./checkpoints/stage3/proposed_L_coarse_tau100.0/average
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```
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## Citation
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```bibtex
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__pycache__/pipeline.cpython-310.pyc
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Binary file (9.13 kB). View file
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config.json
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"num_classes": 1,
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"whisper_model_name": "openai/whisper-large-v3",
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"sampling_rate": 16000,
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"default_checkpoint": "
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"available_checkpoints": [
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"proposed_L_coarse_tau0.1",
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"proposed_L_coarse_tau1.0",
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"num_classes": 1,
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"whisper_model_name": "openai/whisper-large-v3",
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"sampling_rate": 16000,
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"default_checkpoint": "proposed_L_coarse_tau100.0",
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"available_checkpoints": [
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"proposed_L_coarse_tau0.1",
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"proposed_L_coarse_tau1.0",
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pipeline.py
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import torch
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import torch.nn as nn
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import torchaudio
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SAMPLING_RATE = 16000
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WHISPER_MODEL_NAME = "openai/whisper-large-v3"
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WHISPER_HIDDEN_DIM = 1280
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DEFAULT_CHECKPOINT = "
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class WhisperFeatureProbeV2(nn.Module):
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"""Return list of available checkpoint names."""
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return list(self.available_checkpoints)
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def __call__(self, inputs, model_name: str = None):
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"""
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Run severity estimation on audio input.
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if model_name is not None:
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self.switch_model(model_name)
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-
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if isinstance(inputs, str):
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wav, sr = torchaudio.load(inputs)
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elif isinstance(inputs, bytes):
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wav, sr = torchaudio.load(io.BytesIO(inputs))
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else:
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wav, sr = torchaudio.load(io.BytesIO(inputs))
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if sr != SAMPLING_RATE:
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wav = torchaudio.functional.resample(wav, sr, SAMPLING_RATE)
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wav = wav.squeeze()
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# VAD
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wav = _apply_vad(wav, self.vad_model, self.get_speech_timestamps)
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"raw_score": round(score, 4),
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"model_name": self.current_model_name,
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}
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import torch
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import torch.nn as nn
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import soundfile as sf
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import torchaudio
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SAMPLING_RATE = 16000
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WHISPER_MODEL_NAME = "openai/whisper-large-v3"
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WHISPER_HIDDEN_DIM = 1280
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DEFAULT_CHECKPOINT = "proposed_L_coarse_tau100.0"
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class WhisperFeatureProbeV2(nn.Module):
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"""Return list of available checkpoint names."""
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return list(self.available_checkpoints)
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def _load_wav(self, inputs):
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"""Load and preprocess a single audio input to a 1D waveform tensor."""
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if isinstance(inputs, (bytes, bytearray)):
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data, sr = sf.read(io.BytesIO(inputs), dtype="float32")
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else:
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data, sr = sf.read(inputs, dtype="float32")
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wav = torch.from_numpy(data).float()
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if wav.dim() > 1:
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wav = wav.mean(dim=-1)
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if sr != SAMPLING_RATE:
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wav = torchaudio.functional.resample(wav, sr, SAMPLING_RATE)
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return wav
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def __call__(self, inputs, model_name: str = None):
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"""
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Run severity estimation on audio input.
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if model_name is not None:
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self.switch_model(model_name)
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wav = self._load_wav(inputs)
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# VAD
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wav = _apply_vad(wav, self.vad_model, self.get_speech_timestamps)
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"raw_score": round(score, 4),
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"model_name": self.current_model_name,
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}
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def batch_inference(self, input_list, model_name: str = None):
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"""
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Run severity estimation on a batch of audio files.
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Whisper processes one file at a time (due to variable-length VAD output),
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but the probe runs as a single padded batch for efficiency.
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Args:
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input_list: list of file paths (str) or raw audio bytes
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model_name: optionally override the checkpoint for this call
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Returns:
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list of dicts, each with "file", "severity_score", "raw_score",
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and "model_name"
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"""
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if model_name is not None:
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self.switch_model(model_name)
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# Extract features for each file
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all_features = []
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lengths = []
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for inputs in input_list:
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wav = self._load_wav(inputs)
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wav = _apply_vad(wav, self.vad_model, self.get_speech_timestamps)
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features = _extract_features(
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wav, self.whisper, self.processor, self.device
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)
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feat = features.squeeze(0) # (T, hidden_dim)
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all_features.append(feat)
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lengths.append(feat.shape[0])
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# Pad and batch
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max_len = max(lengths)
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hidden_dim = all_features[0].shape[1]
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batch_size = len(all_features)
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padded = torch.zeros(batch_size, max_len, hidden_dim, device=self.device)
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for i, feat in enumerate(all_features):
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padded[i, : lengths[i]] = feat
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lengths_tensor = torch.tensor(lengths, device=self.device)
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# Batched probe inference
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with torch.no_grad():
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output = self.probe(padded, lengths=lengths_tensor)
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scores = output.logits.squeeze(-1).cpu().tolist()
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results = []
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for i, inputs in enumerate(input_list):
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score = scores[i]
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results.append({
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"file": inputs if isinstance(inputs, str) else f"input_{i}",
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"severity_score": round(max(1.0, min(7.0, score)), 2),
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"raw_score": round(score, 4),
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"model_name": self.current_model_name,
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})
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return results
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requirements.txt
ADDED
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# Install PyTorch separately first (via conda or pip):
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# conda install pytorch torchaudio -c pytorch -y
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# See https://pytorch.org/get-started/locally/ for GPU builds.
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transformers>=4.40.0
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safetensors>=0.4.0
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huggingface_hub>=0.20.0
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soundfile>=0.12.0
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test.py
ADDED
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from huggingface_hub import snapshot_download
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# Download the model from HuggingFace
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model_dir = snapshot_download("jaesungbae/da-dsqa")
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# Load pipeline (defaults to proposed_L_coarse_tau100.0)
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from pipeline import PreTrainedPipeline
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pipe = PreTrainedPipeline(model_dir)
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# Run inference
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result = pipe("/projects/bedl/jbae4/workspace_2026/severity_level_classifier_release/sample_wavs/Naturalness/level_1/d1b9444a-2ed1-438e-fd68-08dcb5d1edd7_1071_8831.wav")
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print(result)
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