Update eduport_tts_mal.py
Browse files- eduport_tts_mal.py +81 -44
eduport_tts_mal.py
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@@ -113,6 +113,7 @@ decoder_config = GPT2Config(vocab_size=len(tokenizer))
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decoder_config.add_cross_attention=True
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decoder = GPT2LMHeadModel(config=decoder_config)
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class SpeechRecognitionModel(torch.nn.Module):
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def __init__(self, encoder, decoder):
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super().__init__()
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@@ -120,57 +121,93 @@ class SpeechRecognitionModel(torch.nn.Module):
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self.decoder = decoder
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def forward(self, audio_input, text_input):
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encoder_output = self.encoder(audio_input).last_hidden_state
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for
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optimizer.zero_grad()
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audio_input = audio_input.squeeze(1).to(device)
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text_input = text_input.to(device)
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# Validation step
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model.eval()
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val_loss = 0
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with torch.no_grad():
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for audio_input, text_input in val_loader:
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audio_input = audio_input.to(device)
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text_input = text_input.to(device)
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output = model(audio_input, text_input)
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loss = torch.nn.CrossEntropyLoss()(output.logits.view(-1, output.logits.size(-1)), text_input.view(-1))
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val_loss += loss.item()
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#
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decoder_config.add_cross_attention=True
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decoder = GPT2LMHeadModel(config=decoder_config)
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# Model Architecture with Improved FP16 Support
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class SpeechRecognitionModel(torch.nn.Module):
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def __init__(self, encoder, decoder):
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super().__init__()
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self.decoder = decoder
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def forward(self, audio_input, text_input):
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# Extract encoder hidden states
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encoder_output = self.encoder(audio_input).last_hidden_state
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# Create an attention mask for the encoder output
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encoder_attention_mask = torch.ones(
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encoder_output.shape[:2],
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dtype=torch.long,
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device=encoder_output.device
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)
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# Forward pass through the decoder with cross-attention
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outputs = self.decoder(
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input_ids=text_input,
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encoder_hidden_states=encoder_output,
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encoder_attention_mask=encoder_attention_mask
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)
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return outputs
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# Training Loop with Improved Mixed Precision
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def train_model(num_epochs=10):
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# Prepare the models
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# Use float32 for most of the model, let autocast handle precision
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encoder = Wav2Vec2Model.from_pretrained('facebook/wav2vec2-base-960h')
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# Modify the decoder configuration
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decoder_config = GPT2Config(
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vocab_size=len(tokenizer),
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add_cross_attention=True
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)
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decoder = GPT2LMHeadModel(config=decoder_config)
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# Initialize the model
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model = SpeechRecognitionModel(encoder, decoder)
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# Move to GPU
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device = torch.device('cuda' if torch.cuda.is_available() else 'cpu')
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model = model.to(device)
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# Optimizer and learning rate scheduler
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optimizer = torch.optim.Adam(model.parameters(), lr=1e-4)
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scheduler = torch.optim.lr_scheduler.ReduceLROnPlateau(optimizer, 'min')
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# Gradient scaler for mixed precision training
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scaler = GradScaler()
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# Training loop
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for epoch in range(num_epochs):
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model.train()
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train_loss = 0
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for audio_input, text_input in train_loader:
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optimizer.zero_grad()
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# Move tensors to device
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audio_input = audio_input.squeeze(1).to(device)
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text_input = text_input.to(device)
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# Use autocast for mixed precision training
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with autocast(dtype=torch.float16):
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# Forward pass
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output = model(audio_input, text_input)
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# Compute loss
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loss = torch.nn.CrossEntropyLoss()(output.logits.view(-1, output.logits.size(-1)), text_input.view(-1))
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# Scaled loss for mixed precision
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scaler.scale(loss).backward()
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scaler.step(optimizer)
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scaler.update()
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train_loss += loss.item()
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# Validation step
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model.eval()
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val_loss = 0
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with torch.no_grad(), autocast(dtype=torch.float16):
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for audio_input, text_input in val_loader:
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audio_input = audio_input.to(device)
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text_input = text_input.to(device)
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output = model(audio_input, text_input)
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loss = torch.nn.CrossEntropyLoss()(output.logits.view(-1, output.logits.size(-1)), text_input.view(-1))
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val_loss += loss.item()
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# Update scheduler
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scheduler.step(val_loss)
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print(f'Epoch {epoch}: Train Loss: {train_loss / len(train_loader)}, Val Loss: {val_loss / len(val_loader)}')
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# Run the training
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train_model()
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