Delete modeling_gender.py
Browse files- modeling_gender.py +0 -177
modeling_gender.py
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import torch
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import torchaudio
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import numpy as np
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import pandas as pd
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import soundfile as sf
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from transformers import Pipeline
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from typing import Union, List, Tuple, Dict, Any
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from speechbrain.inference.speaker import EncoderClassifier
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class GenderClassificationPipeline(Pipeline):
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def __init__(self, svm_model, scaler, device="cpu"):
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"""
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Initialize the pipeline with SVM model and scaler.
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The ECAPA-TDNN model is loaded using SpeechBrain's EncoderClassifier.
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"""
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# Convert device string to torch.device
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self.device = torch.device(device)
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# Initialize the model with the proper device
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self.model = EncoderClassifier.from_hparams(
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source="speechbrain/spkrec-ecapa-voxceleb",
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run_opts={"device": str(self.device)} # SpeechBrain expects string
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)
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self.feature_names = [f"{i}_speechbrain_embedding" for i in range(192)]
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# Add feature_extractor for handling multiple files
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self.feature_extractor = lambda x: {"sampling_rate": 16000, "raw_speech": x}
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# Audio processing parameters
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self.target_sample_rate = 16000
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self.target_bitrate = 256000 # 256 kbps
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self.bits_per_sample = 16
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self.svm_model = svm_model
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self.scaler = scaler
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self.labels = ["female", "male"]
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# Required by Pipeline class
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self.framework = "pt"
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self._batch_size = 1
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self._num_workers = None
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self._preprocess_params, self._forward_params, self._postprocess_params = self._sanitize_parameters()
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self._framework = "pt"
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self.call_count = 0
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self.sequential = True
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# self.torch_dtype = None
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self.is_encoder_decoder = False
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def _sanitize_parameters(self, **kwargs) -> Tuple[Dict[str, Any], Dict[str, Any], Dict[str, Any]]:
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"""Sanitize parameters for preprocess, forward, and postprocess steps"""
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preprocess_kwargs = {}
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forward_kwargs = {}
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postprocess_kwargs = {}
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return preprocess_kwargs, forward_kwargs, postprocess_kwargs
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def _process_audio(self, waveform: torch.Tensor, sample_rate: int) -> torch.Tensor:
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"""Process audio to match target specifications"""
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# Convert to mono if needed
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if len(waveform.shape) > 1 and waveform.shape[0] > 1:
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waveform = torch.mean(waveform, dim=0, keepdim=True)
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# Resample to 16kHz if needed
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if sample_rate != self.target_sample_rate:
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resampler = torchaudio.transforms.Resample(sample_rate, self.target_sample_rate)
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waveform = resampler(waveform)
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# Normalize the audio to be between -1 and 1
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if waveform.abs().max() > 1:
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waveform = waveform / waveform.abs().max()
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# Convert to 16-bit precision
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waveform = (waveform * 32767).round() / 32767
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# Calculate target samples based on bitrate
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# bitrate = sample_rate * bits_per_sample * channels
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target_samples = int((self.target_bitrate * waveform.shape[1]) /
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(self.target_sample_rate * self.bits_per_sample))
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# Adjust number of samples if needed
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if waveform.shape[1] != target_samples:
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# Either truncate or pad with zeros
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if waveform.shape[1] > target_samples:
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waveform = waveform[:, :target_samples]
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else:
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padding = target_samples - waveform.shape[1]
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waveform = torch.nn.functional.pad(waveform, (0, padding))
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return waveform
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def preprocess(self, audio_input: Union[str, np.ndarray, torch.Tensor]) -> Dict[str, torch.Tensor]:
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"""Preprocess audio input"""
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if isinstance(audio_input, list):
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waveforms = []
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for audio_file in audio_input:
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if isinstance(audio_file, str):
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wave, sr = sf.read(audio_file)
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wave = torch.from_numpy(wave).float()
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if len(wave.shape) == 1:
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wave = wave.unsqueeze(0)
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else:
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wave = wave.T
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wave = self._process_audio(wave, sr)
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waveforms.append(wave)
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else:
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raise ValueError(f"Unsupported audio input type in list: {type(audio_file)}")
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# Stack all waveforms
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waveform = torch.stack(waveforms)
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return {"inputs": waveform.to(self.device)}
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# Handle single input
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if isinstance(audio_input, str):
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waveform, sample_rate = sf.read(audio_input)
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waveform = torch.from_numpy(waveform).float()
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if len(waveform.shape) == 1:
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waveform = waveform.unsqueeze(0)
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else:
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waveform = waveform.T
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elif isinstance(audio_input, np.ndarray):
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waveform = torch.from_numpy(audio_input).float()
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if len(waveform.shape) == 1:
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waveform = waveform.unsqueeze(0)
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sample_rate = self.target_sample_rate
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else:
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waveform = audio_input
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sample_rate = self.target_sample_rate
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waveform = self._process_audio(waveform, sample_rate)
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return {"inputs": waveform.to(self.device)}
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def _forward(self, model_inputs: Dict[str, torch.Tensor]) -> torch.Tensor:
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"""Extract embeddings using the model"""
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with torch.no_grad():
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embeddings = self.model.encode_batch(model_inputs["inputs"])
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return embeddings
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def postprocess(self, model_outputs: torch.Tensor) -> List[str]:
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"""Process model outputs to final predictions"""
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# Convert to numpy and reshape
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embeddings = model_outputs.cpu().numpy().ravel().reshape(1, -1)
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df_embeddings = pd.DataFrame(embeddings, columns=self.feature_names)
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# embeddings = np.squeeze(embeddings, axis=1)
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# if len(embeddings.shape) == 1:
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# embeddings = embeddings.reshape(1, -1)
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# Scale features
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scaled_features = self.scaler.transform(df_embeddings)
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# Get SVM predictions and probabilities
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predictions = self.svm_model.predict(scaled_features)
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# Format output
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results = [self.labels[p] for p in predictions]
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return results
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@classmethod
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def from_pretrained(cls, model_path: str, device="cpu"):
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"""Load all model components"""
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import joblib
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import json
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# Load configuration
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with open(f"{model_path}/config.json", "r") as f:
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config = json.load(f)
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# Load SVM and scaler
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svm_model = joblib.load(f"{model_path}/svm_model.joblib")
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scaler = joblib.load(f"{model_path}/scaler.joblib")
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# Create pipeline instance
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pipeline = cls(svm_model=svm_model, scaler=scaler, device=device)
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pipeline.labels = config["labels"]
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return pipeline
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