app.py

import museval
from tqdm import tqdm

import numpy as np
import torch
import gradio as gr
import data.utils
import model.utils as model_utils
import utils
import soundfile as sf
import argparse
import os
from model.waveunet import Waveunet

features = 32
feature_growth = "double"
output_size = 2
sr=44100
levels=6
channels =2
instruments =["bass", "drums", "other", "vocals"]
cuda="false"

def compute_model_output(model, inputs):
    '''
    Computes outputs of model with given inputs. Does NOT allow propagating gradients! See compute_loss for training.
    Procedure depends on whether we have one model for each source or not
    :param model: Model to train with
    :param compute_grad: Whether to compute gradients
    :return: Model outputs, Average loss over batch
    '''
    all_outputs = {}

    if model.separate:
        for inst in model.instruments:
            output = model(inputs, inst)
            all_outputs[inst] = output[inst].detach().clone()
    else:
        all_outputs = model(inputs)

    return all_outputs

def predict(audio, model):
    '''
    Predict sources for a given audio input signal, with a given model. Audio is split into chunks to make predictions on each chunk before they are concatenated.
    :param audio: Audio input tensor, either Pytorch tensor or numpy array
    :param model: Pytorch model
    :return: Source predictions, dictionary with source names as keys
    '''
    if isinstance(audio, torch.Tensor):
        is_cuda = audio.is_cuda()
        audio = audio.detach().cpu().numpy()
        return_mode = "pytorch"
    else:
        return_mode = "numpy"

    expected_outputs = audio.shape[1]

    # Pad input if it is not divisible in length by the frame shift number
    output_shift = model.shapes["output_frames"]
    pad_back = audio.shape[1] % output_shift
    pad_back = 0 if pad_back == 0 else output_shift - pad_back
    if pad_back > 0:
        audio = np.pad(audio, [(0,0), (0, pad_back)], mode="constant", constant_values=0.0)

    target_outputs = audio.shape[1]
    outputs = {key: np.zeros(audio.shape, np.float32) for key in model.instruments}

    # Pad mixture across time at beginning and end so that neural network can make prediction at the beginning and end of signal
    pad_front_context = model.shapes["output_start_frame"]
    pad_back_context = model.shapes["input_frames"] - model.shapes["output_end_frame"]
    audio = np.pad(audio, [(0,0), (pad_front_context, pad_back_context)], mode="constant", constant_values=0.0)

    # Iterate over mixture magnitudes, fetch network prediction
    with torch.no_grad():
        for target_start_pos in range(0, target_outputs, model.shapes["output_frames"]):
            # Prepare mixture excerpt by selecting time interval
            curr_input = audio[:, target_start_pos:target_start_pos + model.shapes["input_frames"]] # Since audio was front-padded input of [targetpos:targetpos+inputframes] actually predicts [targetpos:targetpos+outputframes] target range

            # Convert to Pytorch tensor for model prediction
            curr_input = torch.from_numpy(curr_input).unsqueeze(0)

            # Predict
            for key, curr_targets in compute_model_output(model, curr_input).items():
                outputs[key][:,target_start_pos:target_start_pos+model.shapes["output_frames"]] = curr_targets.squeeze(0).cpu().numpy()

    # Crop to expected length (since we padded to handle the frame shift)
    outputs = {key : outputs[key][:,:expected_outputs] for key in outputs.keys()}

    if return_mode == "pytorch":
        outputs = torch.from_numpy(outputs)
        if is_cuda:
            outputs = outputs.cuda()
    return outputs

def predict_song(audio_path):
    '''
    Predicts sources for an audio file for which the file path is given, using a given model.
    Takes care of resampling the input audio to the models sampling rate and resampling predictions back to input sampling rate.
    :param args: Options dictionary
    :param audio_path: Path to mixture audio file
    :param model: Pytorch model
    :return: Source estimates given as dictionary with keys as source names
    '''
    # sr, data = audio_path
    # print(sr)
    # print(data)
    # return (sr, np.flipud(data))
    sr = 44100
    model.eval()

    # Load mixture in original sampling rate
    mix_audio, mix_sr = data.utils.load(audio_path, sr=None, mono=False)
    mix_channels = mix_audio.shape[0]
    mix_len = mix_audio.shape[1]

    # Adapt mixture channels to required input channels
    if channels == 1:
        mix_audio = np.mean(mix_audio, axis=0, keepdims=True)
    else:
        if mix_channels == 1: # Duplicate channels if input is mono but model is stereo
            mix_audio = np.tile(mix_audio, [channels, 1])
        else:
            assert(mix_channels == channels)

    # resample to model sampling rate
    mix_audio = data.utils.resample(mix_audio, mix_sr, sr)

    sources = predict(mix_audio, model)

    # Resample back to mixture sampling rate in case we had model on different sampling rate
    sources = {key : data.utils.resample(sources[key], sr, mix_sr) for key in sources.keys()}

    # In case we had to pad the mixture at the end, or we have a few samples too many due to inconsistent down- and upsamṕling, remove those samples from source prediction now
    for key in sources.keys():
        diff = sources[key].shape[1] - mix_len
        if diff > 0:
            print("WARNING: Cropping " + str(diff) + " samples")
            sources[key] = sources[key][:, :-diff]
        elif diff < 0:
            print("WARNING: Padding output by " + str(diff) + " samples")
            sources[key] = np.pad(sources[key], [(0,0), (0, -diff)], "constant", 0.0)

        # Adapt channels
        if mix_channels > channels:
            assert(channels == 1)
            # Duplicate mono predictions
            sources[key] = np.tile(sources[key], [mix_channels, 1])
        elif mix_channels < channels:
            assert(mix_channels == 1)
            # Reduce model output to mono
            sources[key] = np.mean(sources[key], axis=0, keepdims=True)

        sources[key] = np.asfortranarray(sources[key]) # So librosa does not complain if we want to save it

    data.utils.write_wav("test.wav", sources['vocals'], sr)
    return "test.wav"

# load model
num_features = [features*i for i in range(1, levels+1)] if feature_growth == "add" else \
                   [features*2**i for i in range(0, levels)]
target_outputs = int(output_size * sr)
model = Waveunet(channels, num_features, channels, instruments, kernel_size=5,
                     target_output_size=target_outputs, depth=1, strides=4,
                     conv_type="gn", res="fixed", separate=1)
load_model = 'checkpoints/waveunet/model'
state = model_utils.load_model(model, None, load_model, cuda=0)

# Create title, description and article strings
title = "Denoise Audio"
description = "Using Wave-u-net to Denoise Audio"
article = "Created at github [Wave-U-Net-Pytorch] of author Daniel Stoller(https://github.com/f90/Wave-U-Net-Pytorch)."

# Create the Gradio demo
demo = gr.Interface(fn=predict_song, # mapping function from input to output
                    inputs=gr.Audio(type="filepath"), # what are the inputs?
                    outputs=gr.File(file_count="multiple", file_types=[".wav"]), # our fn has two outputs, therefore we have two outputs
                    title=title,
                    description=description,
                    article=article)

# Launch the demo!
demo.launch() # generate a publically shareable URL?