Delete dataloader

dataloader/dataloader.py

@@ -1,498 +0,0 @@
-import numpy as np
-import math, os, csv
-import torchaudio
-import torch
-import torch.nn as nn
-import torch.utils.data as data
-import torch.distributed as dist
-import soundfile as sf
-from torch.utils.data import Dataset
-import torch.utils.data as data
-import os 
-import sys
-sys.path.append(os.path.dirname(__file__))
-
-from dataloader.misc import read_and_config_file
-import librosa
-import random
-EPS = 1e-6
-MAX_WAV_VALUE = 32768.0
-
-def audioread(path, sampling_rate):
-    """
-    Reads an audio file from the specified path, normalizes the audio, 
-    resamples it to the desired sampling rate (if necessary), and ensures it is single-channel.
-
-    Parameters:
-    path (str): The file path of the audio file to be read.
-    sampling_rate (int): The target sampling rate for the audio.
-
-    Returns:
-    numpy.ndarray: The processed audio data, normalized, resampled (if necessary),
-                   and converted to mono (if the input audio has multiple channels).
-    """
-    
-    # Read audio data and its sample rate from the file.
-    data, fs = sf.read(path)
-    
-    # Normalize the audio data.
-    data, scalar = audio_norm(data)
-    
-    # Resample the audio if the sample rate is different from the target sampling rate.
-    if fs != sampling_rate:
-        data = librosa.resample(data, orig_sr=fs, target_sr=sampling_rate)
-    
-    # Convert to mono by selecting the first channel if the audio has multiple channels.
-    if len(data.shape) > 1:
-        data = data[:, 0]
-    
-    # Return the processed audio data.
-    return data, scalar
-
-def audio_norm(x):
-    """
-    Normalizes the input audio signal to a target Root Mean Square (RMS) level, 
-    applying two stages of scaling. This ensures the audio signal is neither too quiet 
-    nor too loud, keeping its amplitude consistent.
-
-    Parameters:
-    x (numpy.ndarray): Input audio signal to be normalized.
-
-    Returns:
-    numpy.ndarray: Normalized audio signal.
-    """
-    
-    # Compute the root mean square (RMS) of the input audio signal.
-    rms = (x ** 2).mean() ** 0.5
-    
-    # Calculate the scalar to adjust the signal to the target level (-25 dB).
-    scalar = 10 ** (-25 / 20) / (rms + EPS)
-    
-    # Scale the input audio by the computed scalar.
-    x = x * scalar
-    
-    # Compute the power of the scaled audio signal.
-    pow_x = x ** 2
-    
-    # Calculate the average power of the audio signal.
-    avg_pow_x = pow_x.mean()
-    
-    # Compute RMS only for audio segments with higher-than-average power.
-    rmsx = pow_x[pow_x > avg_pow_x].mean() ** 0.5
-    
-    # Calculate another scalar to further normalize based on higher-power segments.
-    scalarx = 10 ** (-25 / 20) / (rmsx + EPS)
-    
-    # Apply the second scalar to the audio.
-    x = x * scalarx
-    
-    # Return the doubly normalized audio signal.
-    return x, 1/(scalar * scalarx + EPS)
-
-class DataReader(object):
-    """
-    A class for reading audio data from a list of files, normalizing it, 
-    and extracting features for further processing. It supports extracting 
-    features from each file, reshaping the data, and returning metadata 
-    like utterance ID and data length.
-
-    Parameters:
-    args: Arguments containing the input path and target sampling rate.
-
-    Attributes:
-    file_list (list): A list of audio file paths to process.
-    sampling_rate (int): The target sampling rate for audio files.
-    """
-
-    def __init__(self, args):
-        # Read and configure the file list from the input path provided in the arguments.
-        # The file list is decoded, if necessary.
-        self.file_list = read_and_config_file(args, args.input_path, decode=True)
-        
-        # Store the target sampling rate.
-        self.sampling_rate = args.sampling_rate
-
-        # Store the args file
-        self.args = args
-
-    def __len__(self):
-        """
-        Returns the number of audio files in the file list.
-
-        Returns:
-        int: Number of files to process.
-        """
-        return len(self.file_list)
-
-    def __getitem__(self, index):
-        """
-        Retrieves the features of the audio file at the given index.
-
-        Parameters:
-        index (int): Index of the file in the file list.
-
-        Returns:
-        tuple: Features (inputs, utterance ID, data length) for the selected audio file.
-        """
-        if self.args.task == 'target_speaker_extraction':
-            if self.args.network_reference.cue== 'lip':
-                return self.file_list[index]
-        return self.extract_feature(self.file_list[index])
-
-    def extract_feature(self, path):
-        """
-        Extracts features from the given audio file path.
-
-        Parameters:
-        path (str): The file path of the audio file.
-
-        Returns:
-        inputs (numpy.ndarray): Reshaped audio data for further processing.
-        utt_id (str): The unique identifier of the audio file, usually the filename.
-        length (int): The length of the original audio data.
-        """
-        # Extract the utterance ID from the file path (usually the filename).
-        utt_id = path.split('/')[-1]
-        
-        # Read and normalize the audio data, converting it to float32 for processing.
-        #data = audioread(path, self.sampling_rate).astype(np.float32)
-        data, scalar = audioread(path, self.sampling_rate)        
-        data = data.astype(np.float32)
-        
-        # Reshape the data to ensure it's in the format [1, data_length].
-        inputs = np.reshape(data, [1, data.shape[0]])
-        
-        # Return the reshaped audio data, utterance ID, and the length of the original data.
-        return inputs, utt_id, data.shape[0], scalar
-
-class Wave_Processor(object):
-    """
-    A class for processing audio data, specifically for reading input and label audio files,
-    segmenting them into fixed-length segments, and applying padding or trimming as necessary.
-
-    Methods:
-    process(path, segment_length, sampling_rate):
-        Processes audio data by reading, padding, or segmenting it to match the specified segment length.
-    
-    Parameters:
-    path (dict): A dictionary containing file paths for 'inputs' and 'labels' audio files.
-    segment_length (int): The desired length of audio segments to extract.
-    sampling_rate (int): The target sampling rate for reading the audio files.
-    """
-
-    def process(self, path, segment_length, sampling_rate):
-        """
-        Reads input and label audio files, and ensures the audio is segmented into
-        the desired length, padding if necessary or extracting random segments if
-        the audio is longer than the target segment length.
-
-        Parameters:
-        path (dict): Dictionary containing the paths to 'inputs' and 'labels' audio files.
-        segment_length (int): Desired length of the audio segment in samples.
-        sampling_rate (int): Target sample rate for the audio.
-
-        Returns:
-        tuple: A pair of numpy arrays representing the processed input and label audio,
-               either padded to the segment length or trimmed.
-        """
-        # Read the input and label audio files using the target sampling rate.
-        wave_inputs = audioread(path['inputs'], sampling_rate)
-        wave_labels = audioread(path['labels'], sampling_rate)
-        
-        # Get the length of the label audio (assumed both inputs and labels have similar lengths).
-        len_wav = wave_labels.shape[0]
-        
-        # If the input audio is shorter than the desired segment length, pad it with zeros.
-        if wave_inputs.shape[0] < segment_length:
-            # Create zero-padded arrays for inputs and labels.
-            padded_inputs = np.zeros(segment_length, dtype=np.float32)
-            padded_labels = np.zeros(segment_length, dtype=np.float32)
-            
-            # Copy the original audio into the padded arrays.
-            padded_inputs[:wave_inputs.shape[0]] = wave_inputs
-            padded_labels[:wave_labels.shape[0]] = wave_labels
-        else:
-            # Randomly select a start index for segmenting the audio if it's longer than the segment length.
-            st_idx = random.randint(0, len_wav - segment_length)
-            
-            # Extract a segment of the desired length from the inputs and labels.
-            padded_inputs = wave_inputs[st_idx:st_idx + segment_length]
-            padded_labels = wave_labels[st_idx:st_idx + segment_length]
-        
-        # Return the processed (padded or segmented) input and label audio.
-        return padded_inputs, padded_labels
-
-class Fbank_Processor(object):
-    """
-    A class for processing input audio data into mel-filterbank (Fbank) features, 
-    including the computation of delta and delta-delta features.
-    
-    Methods:
-    process(inputs, args):
-        Processes the raw audio input and returns the mel-filterbank features 
-        along with delta and delta-delta features.
-    """
-    
-    def process(self, inputs, args):
-        # Convert frame length and shift from seconds to milliseconds.
-        frame_length = int(args.win_len / args.sampling_rate * 1000)
-        frame_shift = int(args.win_inc / args.sampling_rate * 1000)
-
-        # Set up configuration for the mel-filterbank computation.
-        fbank_config = {
-            "dither": 1.0,
-            "frame_length": frame_length,
-            "frame_shift": frame_shift,
-            "num_mel_bins": args.num_mels,
-            "sample_frequency": args.sampling_rate,
-            "window_type": args.win_type
-        }
-
-        # Convert the input audio to a FloatTensor and scale it to match the expected input range.
-        inputs = torch.FloatTensor(inputs * MAX_WAV_VALUE)
-
-        # Compute the mel-filterbank features using Kaldi's fbank function.
-        fbank = torchaudio.compliance.kaldi.fbank(inputs.unsqueeze(0), **fbank_config)
-
-        # Add delta and delta-delta features.
-        fbank_tr = torch.transpose(fbank, 0, 1)
-        fbank_delta = torchaudio.functional.compute_deltas(fbank_tr)
-        fbank_delta_delta = torchaudio.functional.compute_deltas(fbank_delta)
-        fbank_delta = torch.transpose(fbank_delta, 0, 1)
-        fbank_delta_delta = torch.transpose(fbank_delta_delta, 0, 1)
-        
-        # Concatenate the original Fbank, delta, and delta-delta features.
-        fbanks = torch.cat([fbank, fbank_delta, fbank_delta_delta], dim=1)
-        
-        return fbanks.numpy()
-
-class AudioDataset(Dataset):
-    """
-    A dataset class for loading and processing audio data from different data types 
-    (train, validation, test). Supports audio processing and feature extraction 
-    (e.g., waveform processing, Fbank feature extraction).
-
-    Parameters:
-    args: Arguments containing dataset configuration (paths, sampling rate, etc.).
-    data_type (str): The type of data to load (train, val, test).
-    """
-
-    def __init__(self, args, data_type):
-        self.args = args
-        self.sampling_rate = args.sampling_rate
-        
-        # Read the list of audio files based on the data type.
-        if data_type == 'train':
-            self.wav_list = read_and_config_file(args.tr_list)
-        elif data_type == 'val':
-            self.wav_list = read_and_config_file(args.cv_list)
-        elif data_type == 'test':
-            self.wav_list = read_and_config_file(args.tt_list)
-        else:
-            print(f'Data type: {data_type} is unknown!')
-        
-        # Initialize processors for waveform and Fbank features.
-        self.wav_processor = Wave_Processor()
-        self.fbank_processor = Fbank_Processor()
-        
-        # Clip data to a fixed segment length based on the sampling rate and max length.
-        self.segment_length = self.sampling_rate * self.args.max_length
-        print(f'No. {data_type} files: {len(self.wav_list)}')
-
-    def __len__(self):
-        # Return the number of audio files in the dataset.
-        return len(self.wav_list)
-
-    def __getitem__(self, index):
-        # Get the input and label paths from the list.
-        data_info = self.wav_list[index]
-        
-        # Process the waveform inputs and labels.
-        inputs, labels = self.wav_processor.process(
-            {'inputs': data_info['inputs'], 'labels': data_info['labels']}, 
-            self.segment_length, 
-            self.sampling_rate
-        )
-        
-        # Optionally load Fbank features if specified.
-        if self.args.load_fbank is not None:
-            fbanks = self.fbank_processor.process(inputs, self.args)
-            return inputs * MAX_WAV_VALUE, labels * MAX_WAV_VALUE, fbanks
-        
-        return inputs, labels
-
-def zero_pad_concat(self, inputs):
-    """
-    Concatenates a list of input arrays, applying zero-padding as needed to ensure 
-    they all match the length of the longest input.
-
-    Parameters:
-    inputs (list of numpy arrays): List of input arrays to be concatenated.
-
-    Returns:
-    numpy.ndarray: A zero-padded array with concatenated inputs.
-    """
-    
-    # Get the maximum length among all inputs.
-    max_t = max(inp.shape[0] for inp in inputs)
-    
-    # Determine the shape of the output based on the input dimensions.
-    shape = None
-    if len(inputs[0].shape) == 1:
-        shape = (len(inputs), max_t)
-    elif len(inputs[0].shape) == 2:
-        shape = (len(inputs), max_t, inputs[0].shape[1])
-    
-    # Initialize an array with zeros to hold the concatenated inputs.
-    input_mat = np.zeros(shape, dtype=np.float32)
-    
-    # Copy the input data into the zero-padded array.
-    for e, inp in enumerate(inputs):
-        if len(inp.shape) == 1:
-            input_mat[e, :inp.shape[0]] = inp
-        elif len(inp.shape) == 2:
-            input_mat[e, :inp.shape[0], :] = inp
-    
-    return input_mat
-
-def collate_fn_2x_wavs(data):
-    """
-    A custom collate function for combining batches of waveform input and label pairs.
-
-    Parameters:
-    data (list): List of tuples (inputs, labels).
-
-    Returns:
-    tuple: Batched inputs and labels as torch.FloatTensors.
-    """
-    inputs, labels = zip(*data)
-    x = torch.FloatTensor(inputs)
-    y = torch.FloatTensor(labels)
-    return x, y
-
-def collate_fn_2x_wavs_fbank(data):
-    """
-    A custom collate function for combining batches of waveform inputs, labels, and Fbank features.
-
-    Parameters:
-    data (list): List of tuples (inputs, labels, fbanks).
-
-    Returns:
-    tuple: Batched inputs, labels, and Fbank features as torch.FloatTensors.
-    """
-    inputs, labels, fbanks = zip(*data)
-    x = torch.FloatTensor(inputs)
-    y = torch.FloatTensor(labels)
-    z = torch.FloatTensor(fbanks)
-    return x, y, z
-
-class DistributedSampler(data.Sampler):
-    """
-    Sampler for distributed training. Divides the dataset among multiple replicas (processes), 
-    ensuring that each process gets a unique subset of the data. It also supports shuffling 
-    and managing epochs.
-
-    Parameters:
-    dataset (Dataset): The dataset to sample from.
-    num_replicas (int): Number of processes participating in the training.
-    rank (int): Rank of the current process.
-    shuffle (bool): Whether to shuffle the data or not.
-    seed (int): Random seed for reproducibility.
-    """
-
-    def __init__(self, dataset, num_replicas=None, rank=None, shuffle=True, seed=0):
-        if num_replicas is None:
-            if not dist.is_available():
-                raise RuntimeError("Requires distributed package to be available")
-            num_replicas = dist.get_world_size()
-        if rank is None:
-            if not dist.is_available():
-                raise RuntimeError("Requires distributed package to be available")
-            rank = dist.get_rank()
-        
-        self.dataset = dataset
-        self.num_replicas = num_replicas
-        self.rank = rank
-        self.epoch = 0
-        self.num_samples = int(math.ceil(len(self.dataset) * 1.0 / self.num_replicas))
-        self.total_size = self.num_samples * self.num_replicas
-        self.shuffle = shuffle
-        self.seed = seed
-
-    def __iter__(self):
-        # Shuffle the indices based on the epoch and seed.
-        if self.shuffle:
-            g = torch.Generator()
-            g.manual_seed(self.seed + self.epoch)
-            ind = torch.randperm(int(len(self.dataset) / self.num_replicas), generator=g) * self.num_replicas
-            indices = []
-            for i in range(self.num_replicas):
-                indices = indices + (ind + i).tolist()
-        else:
-            indices = list(range(len(self.dataset)))
-        
-        # Add extra samples to make the dataset evenly divisible.
-        indices += indices[:(self.total_size - len(indices))]
-        assert len(indices) == self.total_size
-
-        # Subsample for the current process.
-        indices = indices[self.rank * self.num_samples:(self.rank + 1) * self.num_samples]
-        assert len(indices) == self.num_samples
-
-        return iter(indices)
-
-    def __len__(self):
-        return self.num_samples
-
-    def set_epoch(self, epoch):
-        self.epoch = epoch
-
-def get_dataloader(args, data_type):
-    """
-    Creates and returns a data loader and sampler for the specified dataset type (train, validation, or test).
-    
-    Parameters:
-    args (Namespace): Configuration arguments containing details such as batch size, sampling rate, 
-                      network type, and whether distributed training is used.
-    data_type (str): The type of dataset to load ('train', 'val', 'test').
-    
-    Returns:
-    sampler (DistributedSampler or None): The sampler for distributed training, or None if not used.
-    generator (DataLoader): The PyTorch DataLoader for the specified dataset.
-    """
-    
-    # Initialize the dataset based on the given arguments and dataset type (train, val, or test).
-    datasets = AudioDataset(args=args, data_type=data_type)
-
-    # Create a distributed sampler if distributed training is enabled; otherwise, use no sampler.
-    sampler = DistributedSampler(
-        datasets,
-        num_replicas=args.world_size,  # Number of replicas in distributed training.
-        rank=args.local_rank  # Rank of the current process.
-    ) if args.distributed else None
-
-    # Select the appropriate collate function based on the network type.
-    if args.network == 'FRCRN_SE_16K' or args.network == 'MossFormerGAN_SE_16K':
-        # Use the collate function for two-channel waveform data (inputs and labels).
-        collate_fn = collate_fn_2x_wavs
-    elif args.network == 'MossFormer2_SE_48K':
-        # Use the collate function for waveforms along with Fbank features.
-        collate_fn = collate_fn_2x_wavs_fbank
-    else:
-        # Print an error message if the network type is unknown.
-        print(f'in dataloader, please specify a correct network type using args.network!')
-        return
-
-    # Create a DataLoader with the specified dataset, batch size, and worker configuration.
-    generator = data.DataLoader(
-        datasets,
-        batch_size=args.batch_size,  # Batch size for training.
-        shuffle=(sampler is None),  # Shuffle the data only if no sampler is used.
-        collate_fn=collate_fn,  # Use the selected collate function for batching data.
-        num_workers=args.num_workers,  # Number of workers for data loading.
-        sampler=sampler  # Use the distributed sampler if applicable.
-    )
-    
-    # Return both the sampler and DataLoader (generator).
-    return sampler, generator
-

dataloader/misc.py

@@ -1,84 +0,0 @@
-
-#!/usr/bin/env python -u
-# -*- coding: utf-8 -*-
-
-from __future__ import absolute_import
-from __future__ import division
-from __future__ import print_function
-import torch 
-import torch.nn as nn
-import numpy as np
-import os 
-import sys
-import librosa
-
-def read_and_config_file(args, input_path, decode=0):
-    """
-    Reads and processes the input file or directory to extract audio file paths or configuration data.
-    
-    Parameters:
-    args: The args
-    input_path (str): Path to a file or directory containing audio data or file paths.
-    decode (bool): If True (decode=1) for decoding, process the input as audio files directly (find .wav or .flac files) or from a .scp file.
-                   If False (decode=0) for training, assume the input file contains lines with paths to audio files.
-    
-    Returns:
-    processed_list (list): A list of processed file paths or a list of dictionaries containing input 
-                           and optional condition audio paths.
-    """
-    processed_list = []  # Initialize list to hold processed file paths or configurations
-
-    if decode:
-        if args.task == 'target_speaker_extraction':
-            if args.network_reference.cue== 'lip':
-                # If decode is True, find video files in a directory or single file
-                if os.path.isdir(input_path):
-                    # Find all .mp4 , mov .avi files in the input directory
-                    processed_list = librosa.util.find_files(input_path, ext="mp4")
-                    processed_list += librosa.util.find_files(input_path, ext="avi")
-                    processed_list += librosa.util.find_files(input_path, ext="mov")
-                    processed_list += librosa.util.find_files(input_path, ext="MOV")
-                else:
-                    # If it's a single file and it's a .wav or .flac, add to processed list
-                    if input_path.lower().endswith(".mp4") or input_path.lower().endswith(".avi") or input_path.lower().endswith(".mov") or input_path.lower().endswith(".webm"):
-                        processed_list.append(input_path)
-                    else:
-                        # Read file paths from the input text file (one path per line)
-                        with open(input_path) as fid:
-                            for line in fid:
-                                path_s = line.strip().split()  # Split paths (space-separated)
-                                processed_list.append(path_s[0])  # Add the first path (input audio path)
-                return processed_list
-
-        # If decode is True, find audio files in a directory or single file
-        if os.path.isdir(input_path):
-            # Find all .wav files in the input directory
-            processed_list = librosa.util.find_files(input_path, ext="wav")
-            if len(processed_list) == 0:
-                # If no .wav files, look for .flac files
-                processed_list = librosa.util.find_files(input_path, ext="flac")
-        else:
-            # If it's a single file and it's a .wav or .flac, add to processed list
-            if input_path.lower().endswith(".wav") or input_path.lower().endswith(".flac"):
-                processed_list.append(input_path)
-            else:
-                # Read file paths from the input text file (one path per line)
-                with open(input_path) as fid:
-                    for line in fid:
-                        path_s = line.strip().split()  # Split paths (space-separated)
-                        processed_list.append(path_s[0])  # Add the first path (input audio path)
-        return processed_list
-
-    # If decode is False, treat the input file as a configuration file
-    with open(input_path) as fid:
-        for line in fid:
-            tmp_paths = line.strip().split()  # Split paths (space-separated)
-            if len(tmp_paths) == 2:
-                # If two paths per line, treat the second as 'condition_audio'
-                sample = {'inputs': tmp_paths[0], 'condition_audio': tmp_paths[1]}
-            elif len(tmp_paths) == 1:
-                # If only one path per line, treat it as 'inputs'
-                sample = {'inputs': tmp_paths[0]}
-            processed_list.append(sample)  # Append processed sample to list
-    return processed_list
-