✨ | Added support for .mp3 and .flac loading...
This commit is contained in:
39
data.py
39
data.py
@ -5,41 +5,42 @@ import torchaudio
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import os
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import random
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import torchaudio.transforms as T
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import tqdm
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import AudioUtils
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class AudioDataset(Dataset):
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audio_sample_rates = [11025]
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MAX_LENGTH = 44100 # Define your desired maximum length here
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def __init__(self, input_dir, device):
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self.input_files = [os.path.join(root, f) for root, _, files in os.walk(input_dir) for f in files if f.endswith('.wav')]
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self.device = device
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input_files = [os.path.join(root, f) for root, _, files in os.walk(input_dir) for f in files if f.endswith('.wav') or f.endswith('.mp3') or f.endswith('.flac')]
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def __len__(self):
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return len(self.input_files)
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def __getitem__(self, idx):
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# Load high-quality audio
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high_quality_audio, original_sample_rate = torchaudio.load(self.input_files[idx], normalize=True)
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# Change to mono
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high_quality_audio = AudioUtils.stereo_tensor_to_mono(high_quality_audio)
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data = []
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for audio_clip in tqdm.tqdm(input_files, desc=f"Processing {len(input_files)} audio file(s)"):
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audio, original_sample_rate = torchaudio.load(audio_clip, normalize=True)
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audio = AudioUtils.stereo_tensor_to_mono(audio)
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# Generate low-quality audio with random downsampling
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mangled_sample_rate = random.choice(self.audio_sample_rates)
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resample_transform_low = torchaudio.transforms.Resample(original_sample_rate, mangled_sample_rate)
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resample_transform_high = torchaudio.transforms.Resample(mangled_sample_rate, original_sample_rate)
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low_quality_audio = resample_transform_low(high_quality_audio)
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low_quality_audio = resample_transform_high(low_quality_audio)
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low_audio = resample_transform_low(audio)
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low_audio = resample_transform_high(low_audio)
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splitted_high_quality_audio = AudioUtils.split_audio(high_quality_audio, 128)
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splitted_high_quality_audio = AudioUtils.split_audio(audio, 128)
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splitted_high_quality_audio[-1] = AudioUtils.pad_tensor(splitted_high_quality_audio[-1], 128)
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splitted_high_quality_audio = [tensor.to(self.device) for tensor in splitted_high_quality_audio]
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splitted_low_quality_audio = AudioUtils.split_audio(low_quality_audio, 128)
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splitted_low_quality_audio = AudioUtils.split_audio(low_audio, 128)
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splitted_low_quality_audio[-1] = AudioUtils.pad_tensor(splitted_low_quality_audio[-1], 128)
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splitted_low_quality_audio = [tensor.to(self.device) for tensor in splitted_low_quality_audio]
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return (splitted_high_quality_audio, original_sample_rate), (splitted_low_quality_audio, mangled_sample_rate)
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for high_quality_sample, low_quality_sample in zip(splitted_high_quality_audio, splitted_low_quality_audio):
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data.append(((high_quality_sample, low_quality_sample), (original_sample_rate, mangled_sample_rate)))
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self.audio_data = data
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def __len__(self):
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return len(self.audio_data)
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def __getitem__(self, idx):
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return self.audio_data[idx]
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@ -2,19 +2,21 @@ import json
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filepath = "my_data.json"
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def write_data(filepath, data):
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def write_data(filepath, data, debug=False):
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try:
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with open(filepath, 'w') as f:
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json.dump(data, f, indent=4) # Use indent for pretty formatting
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if debug:
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print(f"Data written to '{filepath}'")
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except Exception as e:
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print(f"Error writing to file: {e}")
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def read_data(filepath):
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def read_data(filepath, debug=False):
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try:
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with open(filepath, 'r') as f:
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data = json.load(f)
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if debug:
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print(f"Data read from '{filepath}'")
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return data
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except FileNotFoundError:
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45
training.py
45
training.py
@ -76,7 +76,7 @@ os.makedirs(audio_output_dir, exist_ok=True)
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# ========= SINGLE =========
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train_data_loader = DataLoader(dataset, batch_size=1, shuffle=True)
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train_data_loader = DataLoader(dataset, batch_size=256, shuffle=True)
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# ========= MODELS =========
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@ -122,27 +122,28 @@ def start_training():
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times_correct = 0
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# ========= TRAINING =========
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for high_quality_data, low_quality_data in tqdm.tqdm(train_data_loader, desc=f"Training epoch {generator_epoch+1}/{generator_epochs}, Current epoch {epoch+1}"):
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for training_data in tqdm.tqdm(train_data_loader, desc=f"Training epoch {generator_epoch+1}/{generator_epochs}, Current epoch {epoch+1}"):
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## Data structure:
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# [[float..., float..., float...], sample_rate]
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# [[[float..., float..., float...], [float..., float..., float...]], [original_sample_rate, mangled_sample_rate]]
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# ========= LABELS =========
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good_quality_data = training_data[0][0].to(device)
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bad_quality_data = training_data[0][1].to(device)
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original_sample_rate = training_data[1][0]
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mangled_sample_rate = training_data[1][1]
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batch_size = high_quality_data[0][0].size(0)
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batch_size = good_quality_data.size(0)
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real_labels = torch.ones(batch_size, 1).to(device)
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fake_labels = torch.zeros(batch_size, 1).to(device)
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high_quality_audio = high_quality_data
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low_quality_audio = low_quality_data
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high_quality_audio = (good_quality_data, original_sample_rate)
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low_quality_audio = (bad_quality_data, mangled_sample_rate)
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ai_enhanced_outputs = []
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for high_quality_sample, low_quality_sample in tqdm.tqdm(zip(high_quality_data[0], low_quality_data[0]), desc=f"Processing audio clip.. Length: {len(high_quality_data[0])}"):
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# ========= DISCRIMINATOR =========
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discriminator.train()
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d_loss = discriminator_train(
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high_quality_sample,
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low_quality_sample,
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good_quality_data,
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bad_quality_data,
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real_labels,
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fake_labels,
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discriminator,
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@ -154,8 +155,8 @@ def start_training():
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# ========= GENERATOR =========
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generator.train()
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generator_output, combined_loss, adversarial_loss, mel_l1_tensor, log_stft_l1_tensor, mfcc_l_tensor = generator_train(
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low_quality_sample,
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high_quality_sample,
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bad_quality_data,
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good_quality_data,
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real_labels,
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generator,
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discriminator,
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@ -167,25 +168,23 @@ def start_training():
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mfcc_transform
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)
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ai_enhanced_outputs.append(generator_output)
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if debug:
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print(f"D_LOSS: {d_loss.item():.4f}, COMBINED_LOSS: {combined_loss.item():.4f}, ADVERSARIAL_LOSS: {adversarial_loss.item():.4f}, MEL_L1_LOSS: {mel_l1_tensor.item():.4f}, LOG_STFT_L1_LOSS: {log_stft_l1_tensor.item():.4f}, MFCC_LOSS: {mfcc_l_tensor.item():.4f}")
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scheduler_d.step(d_loss.detach())
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scheduler_g.step(adversarial_loss.detach())
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# ========= SAVE LATEST AUDIO =========
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high_quality_audio = (torch.cat(high_quality_data[0]), high_quality_data[1])
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low_quality_audio = (torch.cat(low_quality_data[0]), low_quality_data[1])
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ai_enhanced_audio = (torch.cat(ai_enhanced_outputs), high_quality_data[1])
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high_quality_audio = (good_quality_data, original_sample_rate)
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low_quality_audio = (bad_quality_data, original_sample_rate)
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ai_enhanced_audio = (generator_output, original_sample_rate)
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new_epoch = generator_epoch+epoch
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if generator_epoch % 25 == 0:
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print(f"Saved epoch {new_epoch}!")
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torchaudio.save(f"{audio_output_dir}/epoch-{new_epoch}-audio-crap.wav", low_quality_audio[0].cpu().detach(), high_quality_audio[1]) # <-- Because audio clip was resampled in data.py from original to crap and to original again.
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torchaudio.save(f"{audio_output_dir}/epoch-{new_epoch}-audio-ai.wav", ai_enhanced_audio[0].cpu().detach(), ai_enhanced_audio[1])
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torchaudio.save(f"{audio_output_dir}/epoch-{new_epoch}-audio-orig.wav", high_quality_audio[0].cpu().detach(), high_quality_audio[1])
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# if generator_epoch % 25 == 0:
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# print(f"Saved epoch {new_epoch}!")
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# torchaudio.save(f"{audio_output_dir}/epoch-{new_epoch}-audio-orig.wav", high_quality_audio[0][-1].cpu().detach(), high_quality_audio[1][-1])
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# torchaudio.save(f"{audio_output_dir}/epoch-{new_epoch}-audio-crap.wav", low_quality_audio[0][-1].cpu().detach(), high_quality_audio[1][-1]) # <-- Because audio clip was resampled in data.py from original to crap and to original again.
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# torchaudio.save(f"{audio_output_dir}/epoch-{new_epoch}-audio-ai.wav", ai_enhanced_audio[0][-1].cpu().detach(), high_quality_audio[1][-1])
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#if debug:
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# print(generator.state_dict().keys())
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