Update inference/style_transfer.py

inference/style_transfer.py

@@ -1,404 +1,179 @@
-"""
-    Inference code of music style transfer
-    of the work "Music Mixing Style Transfer: A Contrastive Learning Approach to Disentangle Audio Effects"
-    Process : converts the mixing style of the input music recording to that of the refernce music.
-                files inside the target directory should be organized as follow 
-                    "path_to_data_directory"/"song_name_#1"/input.wav
-                    "path_to_data_directory"/"song_name_#1"/reference.wav
-                    ...
-                    "path_to_data_directory"/"song_name_#n"/input.wav
-                    "path_to_data_directory"/"song_name_#n"/reference.wav
-                where the 'input' and 'reference' should share the same names.
-"""
-import numpy as np
-from glob import glob
 import os
-import torch
-
-import sys
-currentdir = os.path.dirname(os.path.realpath(__file__))
-sys.path.append(os.path.join(os.path.dirname(currentdir), "mixing_style_transfer"))
-from networks import FXencoder, TCNModel
-from data_loader import *
-import librosa
-import pyloudnorm
-
-
-
-class Mixing_Style_Transfer_Inference:
-    def __init__(self, args, trained_w_ddp=True):
-        if torch.cuda.is_available():
-            self.device = torch.device("cuda:0")
-        else:
-            self.device = torch.device("cpu")
-        print(f"using device: {self.device} for inference")
-            
-        # inference computational hyperparameters
-        self.args = args
-        self.segment_length = args.segment_length
-        self.batch_size = args.batch_size
-        self.sample_rate = 44100    # sampling rate should be 44100
-        self.time_in_seconds = int(args.segment_length // self.sample_rate)
-
-        # directory configuration
-        self.output_dir = args.target_dir if args.output_dir==None else args.output_dir
-        self.target_dir = args.target_dir
-
-        # load model and its checkpoint weights
-        self.models = {}
-        self.models['effects_encoder'] = FXencoder(args.cfg_encoder).to(self.device)
-        self.models['mixing_converter'] = TCNModel(nparams=args.cfg_converter["condition_dimension"], \
-                                                    ninputs=2, \
-                                                    noutputs=2, \
-                                                    nblocks=args.cfg_converter["nblocks"], \
-                                                    dilation_growth=args.cfg_converter["dilation_growth"], \
-                                                    kernel_size=args.cfg_converter["kernel_size"], \
-                                                    channel_width=args.cfg_converter["channel_width"], \
-                                                    stack_size=args.cfg_converter["stack_size"], \
-                                                    cond_dim=args.cfg_converter["condition_dimension"], \
-                                                    causal=args.cfg_converter["causal"]).to(self.device)
-        
-        ckpt_paths = {'effects_encoder' : args.ckpt_path_enc, \
-                        'mixing_converter' : args.ckpt_path_conv}
-        # reload saved model weights
-        ddp = trained_w_ddp
-        self.reload_weights(ckpt_paths, ddp=ddp)
-
-        # load data loader for the inference procedure
-        inference_dataset = Song_Dataset_Inference(args)
-        self.data_loader = DataLoader(inference_dataset, \
-                                        batch_size=1, \
-                                        shuffle=False, \
-                                        num_workers=args.workers, \
-                                        drop_last=False)
-
-        ''' check stem-wise result '''
-        if not self.args.do_not_separate:
-            os.environ['MKL_THREADING_LAYER'] = 'GNU'
-            separate_file_names = [args.input_file_name, args.reference_file_name]
-            if self.args.interpolation:
-                separate_file_names.append(args.reference_file_name_2interpolate)
-            for cur_idx, cur_inf_dir in enumerate(sorted(glob(f"{args.target_dir}*/"))):
-                for cur_file_name in separate_file_names:
-                    cur_sep_file_path = os.path.join(cur_inf_dir, cur_file_name+'.wav')
-                    cur_sep_output_dir = os.path.join(cur_inf_dir, args.stem_level_directory_name)
-                    if os.path.exists(os.path.join(cur_sep_output_dir, self.args.separation_model, cur_file_name, 'drums.wav')):
-                        print(f'\talready separated current file : {cur_sep_file_path}')
-                    else:
-                        cur_cmd_line = f"demucs {cur_sep_file_path} -n {self.args.separation_model} -d {self.device} -o {cur_sep_output_dir}"
-                        os.system(cur_cmd_line)
-
-
-    # reload model weights from the target checkpoint path
-    def reload_weights(self, ckpt_paths, ddp=True):
-        for cur_model_name in self.models.keys():
-            checkpoint = torch.load(ckpt_paths[cur_model_name], map_location=self.device)
-
-            from collections import OrderedDict
-            new_state_dict = OrderedDict()
-            for k, v in checkpoint["model"].items():
-                # remove `module.` if the model was trained with DDP
-                name = k[7:] if ddp else k
-                new_state_dict[name] = v
-        
-            # load params
-            self.models[cur_model_name].load_state_dict(new_state_dict)
-
-            print(f"---reloaded checkpoint weights : {cur_model_name} ---")
-
-
-    # Inference whole song
-    def inference(self, input_track_path, reference_track_path):
-        print("\n======= Start to inference music mixing style transfer =======")
-        # normalized input
-        output_name_tag = 'output' if self.args.normalize_input else 'output_notnormed'
-
-        for step, (input_stems, reference_stems, dir_name) in enumerate(self.data_loader):
-            print(f"---inference file name : {dir_name[0]}---")
-            cur_out_dir = dir_name[0].replace(self.target_dir, self.output_dir)
-            os.makedirs(cur_out_dir, exist_ok=True)
-            ''' stem-level inference '''
-            inst_outputs = []
-            for cur_inst_idx, cur_inst_name in enumerate(self.args.instruments):
-                print(f'\t{cur_inst_name}...')
-                ''' segmentize whole songs into batch '''
-                if len(input_stems[0][cur_inst_idx][0]) > self.args.segment_length:
-                    cur_inst_input_stem = self.batchwise_segmentization(input_stems[0][cur_inst_idx], \
-                                                                                dir_name[0], \
-                                                                                segment_length=self.args.segment_length, \
-                                                                                discard_last=False)
-                else:
-                    cur_inst_input_stem = [input_stems[:, cur_inst_idx]]
-                if len(reference_stems[0][cur_inst_idx][0]) > self.args.segment_length*2:
-                    cur_inst_reference_stem = self.batchwise_segmentization(reference_stems[0][cur_inst_idx], \
-                                                                                dir_name[0], \
-                                                                                segment_length=self.args.segment_length_ref, \
-                                                                                discard_last=False)
-                else:
-                    cur_inst_reference_stem = [reference_stems[:, cur_inst_idx]]
-
-                ''' inference '''
-                # first extract reference style embedding
-                infered_ref_data_list = []
-                for cur_ref_data in cur_inst_reference_stem:
-                    cur_ref_data = cur_ref_data.to(self.device)
-                    # Effects Encoder inference
-                    with torch.no_grad():
-                        self.models["effects_encoder"].eval()
-                        reference_feature = self.models["effects_encoder"](cur_ref_data)
-                    infered_ref_data_list.append(reference_feature)
-                # compute average value from the extracted exbeddings
-                infered_ref_data = torch.stack(infered_ref_data_list)
-                infered_ref_data_avg = torch.mean(infered_ref_data.reshape(infered_ref_data.shape[0]*infered_ref_data.shape[1], infered_ref_data.shape[2]), axis=0)
-
-                # mixing style converter
-                infered_data_list = []
-                for cur_data in cur_inst_input_stem:
-                    cur_data = cur_data.to(self.device)
-                    with torch.no_grad():
-                        self.models["mixing_converter"].eval()
-                        infered_data = self.models["mixing_converter"](cur_data, infered_ref_data_avg.unsqueeze(0))
-                    infered_data_list.append(infered_data.cpu().detach())
-
-                # combine back to whole song
-                for cur_idx, cur_batch_infered_data in enumerate(infered_data_list):
-                    cur_infered_data_sequential = torch.cat(torch.unbind(cur_batch_infered_data, dim=0), dim=-1)
-                    fin_data_out = cur_infered_data_sequential if cur_idx==0 else torch.cat((fin_data_out, cur_infered_data_sequential), dim=-1)
-                # final output of current instrument
-                fin_data_out_inst = fin_data_out[:, :input_stems[0][cur_inst_idx].shape[-1]].numpy()
-
-                inst_outputs.append(fin_data_out_inst)
-                # save output of each instrument
-                if self.args.save_each_inst:
-                    sf.write(os.path.join(cur_out_dir, f"{cur_inst_name}_{output_name_tag}.wav"), fin_data_out_inst.transpose(-1, -2), self.args.sample_rate, 'PCM_16')
-                    
-            # remix
-            fin_data_out_mix = sum(inst_outputs)
-            
-            # loudness adjusting for mastering purpose
-            if self.args.match_output_loudness:
-                meter = pyloudnorm.Meter(44100)
-                loudness_out = meter.integrated_loudness(fin_data_out_mix.transpose(-1, -2))
-                reference_aud = load_wav_segment(reference_track_path, axis=1)
-                loudness_ref = meter.integrated_loudness(reference_aud)
-                # adjust output loudness to that of the reference
-                fin_data_out_mix = pyloudnorm.normalize.loudness(fin_data_out_mix, loudness_out, loudness_ref)
-                fin_data_out_mix = np.clip(fin_data_out_mix, -1., 1.)
-            
-            # save output
-            fin_output_path = os.path.join(cur_out_dir, f"mixture_{output_name_tag}.wav")
-            sf.write(fin_output_path, fin_data_out_mix.transpose(-1, -2), self.args.sample_rate, 'PCM_16')
+import binascii
+import warnings
 
-            return fin_output_path, fin_data_out_mix
+import json
+import argparse
+import copy
 
+import numpy as np
+import matplotlib.pyplot as plt
+import torch
+import tqdm
+import librosa
+import soundfile as sf
+import gradio as gr
+import pytube as pt
 
-    # Inference whole song
-    def inference_interpolation(self, ):
-        print("\n======= Start to inference interpolation examples =======")
-        # normalized input
-        output_name_tag = 'output_interpolation' if self.args.normalize_input else 'output_notnormed_interpolation'
-
-        for step, (input_stems, reference_stems_A, reference_stems_B, dir_name) in enumerate(self.data_loader):
-            print(f"---inference file name : {dir_name[0]}---")
-            cur_out_dir = dir_name[0].replace(self.target_dir, self.output_dir)
-            os.makedirs(cur_out_dir, exist_ok=True)
-            ''' stem-level inference '''
-            inst_outputs = []
-            for cur_inst_idx, cur_inst_name in enumerate(self.args.instruments):
-                print(f'\t{cur_inst_name}...')
-                ''' segmentize whole song '''
-                # segmentize input according to number of interpolating segments
-                interpolate_segment_length = input_stems[0][cur_inst_idx].shape[1] // self.args.interpolate_segments + 1
-                cur_inst_input_stem = self.batchwise_segmentization(input_stems[0][cur_inst_idx], \
-                                                                                dir_name[0], \
-                                                                                segment_length=interpolate_segment_length, \
-                                                                                discard_last=False)
-                # batchwise segmentize 2 reference tracks
-                if len(reference_stems_A[0][cur_inst_idx][0]) > self.args.segment_length_ref:
-                    cur_inst_reference_stem_A = self.batchwise_segmentization(reference_stems_A[0][cur_inst_idx], \
-                                                                                        dir_name[0], \
-                                                                                        segment_length=self.args.segment_length_ref, \
-                                                                                        discard_last=False)
-                else:
-                    cur_inst_reference_stem_A = [reference_stems_A[:, cur_inst_idx]]
-                if len(reference_stems_B[0][cur_inst_idx][0]) > self.args.segment_length_ref:
-                    cur_inst_reference_stem_B = self.batchwise_segmentization(reference_stems_B[0][cur_inst_idx], \
-                                                                                        dir_name[0], \
-                                                                                        segment_length=self.args.segment_length, \
-                                                                                        discard_last=False)
-                else:
-                    cur_inst_reference_stem_B = [reference_stems_B[:, cur_inst_idx]]
-
-                ''' inference '''
-                # first extract reference style embeddings
-                # reference A
-                infered_ref_data_list = []
-                for cur_ref_data in cur_inst_reference_stem_A:
-                    cur_ref_data = cur_ref_data.to(self.device)
-                    # Effects Encoder inference
-                    with torch.no_grad():
-                        self.models["effects_encoder"].eval()
-                        reference_feature = self.models["effects_encoder"](cur_ref_data)
-                    infered_ref_data_list.append(reference_feature)
-                # compute average value from the extracted exbeddings
-                infered_ref_data = torch.stack(infered_ref_data_list)
-                infered_ref_data_avg_A = torch.mean(infered_ref_data.reshape(infered_ref_data.shape[0]*infered_ref_data.shape[1], infered_ref_data.shape[2]), axis=0)
-
-                # reference B
-                infered_ref_data_list = []
-                for cur_ref_data in cur_inst_reference_stem_B:
-                    cur_ref_data = cur_ref_data.to(self.device)
-                    # Effects Encoder inference
-                    with torch.no_grad():
-                        self.models["effects_encoder"].eval()
-                        reference_feature = self.models["effects_encoder"](cur_ref_data)
-                    infered_ref_data_list.append(reference_feature)
-                # compute average value from the extracted exbeddings
-                infered_ref_data = torch.stack(infered_ref_data_list)
-                infered_ref_data_avg_B = torch.mean(infered_ref_data.reshape(infered_ref_data.shape[0]*infered_ref_data.shape[1], infered_ref_data.shape[2]), axis=0)
-
-                # mixing style converter
-                infered_data_list = []
-                for cur_idx, cur_data in enumerate(cur_inst_input_stem):
-                    cur_data = cur_data.to(self.device)
-                    # perform linear interpolation on embedding space
-                    cur_weight = (self.args.interpolate_segments-1-cur_idx) / (self.args.interpolate_segments-1)
-                    cur_ref_emb = cur_weight * infered_ref_data_avg_A +  (1-cur_weight) * infered_ref_data_avg_B
-                    with torch.no_grad():
-                        self.models["mixing_converter"].eval()
-                        infered_data = self.models["mixing_converter"](cur_data, cur_ref_emb.unsqueeze(0))
-                    infered_data_list.append(infered_data.cpu().detach())
-
-                # combine back to whole song
-                for cur_idx, cur_batch_infered_data in enumerate(infered_data_list):
-                    cur_infered_data_sequential = torch.cat(torch.unbind(cur_batch_infered_data, dim=0), dim=-1)
-                    fin_data_out = cur_infered_data_sequential if cur_idx==0 else torch.cat((fin_data_out, cur_infered_data_sequential), dim=-1)
-                # final output of current instrument
-                fin_data_out_inst = fin_data_out[:, :input_stems[0][cur_inst_idx].shape[-1]].numpy()
-                inst_outputs.append(fin_data_out_inst)
-
-                # save output of each instrument
-                if self.args.save_each_inst:
-                    sf.write(os.path.join(cur_out_dir, f"{cur_inst_name}_{output_name_tag}.wav"), fin_data_out_inst.transpose(-1, -2), self.args.sample_rate, 'PCM_16')
-            # remix
-            fin_data_out_mix = sum(inst_outputs)
-            # fin_output_path = os.path.join(cur_out_dir, f"mixture_{output_name_tag}.wav")
-            # sf.write(fin_output_path, fin_data_out_mix.transpose(-1, -2), self.args.sample_rate, 'PCM_16')
-
-            # return fin_output_path, fin_data_out_mix
-            return fin_data_out_mix
-
-
-    # function that segmentize an entire song into batch
-    def batchwise_segmentization(self, target_song, song_name, segment_length, discard_last=False):
-        assert target_song.shape[-1] >= self.args.segment_length, \
-                f"Error : Insufficient duration!\n\t \
-                Target song's length is shorter than segment length.\n\t \
-                Song name : {song_name}\n\t \
-                Consider changing the 'segment_length' or song with sufficient duration"
-
-        # discard restovers (last segment)
-        if discard_last:
-            target_length = target_song.shape[-1] - target_song.shape[-1] % segment_length
-            target_song = target_song[:, :target_length]
-        # pad last segment
-        else:
-            pad_length = segment_length - target_song.shape[-1] % segment_length
-            target_song = torch.cat((target_song, torch.zeros(2, pad_length)), axis=-1)
-
-        # segmentize according to the given segment_length
-        whole_batch_data = []
-        batch_wise_data = []
-        for cur_segment_idx in range(target_song.shape[-1]//segment_length):
-            batch_wise_data.append(target_song[..., cur_segment_idx*segment_length:(cur_segment_idx+1)*segment_length])
-            if len(batch_wise_data)==self.args.batch_size:
-                whole_batch_data.append(torch.stack(batch_wise_data, dim=0))
-                batch_wise_data = []
-        if batch_wise_data:
-            whole_batch_data.append(torch.stack(batch_wise_data, dim=0))
-
-        return whole_batch_data
-
+from pytube.exceptions import VideoUnavailable
 
+from inference.style_transfer import *
 
-def trim_audio(target_file_path, start_point_in_second=0, duration_in_second=30, sample_rate=44100):
-    # insure format
-    cur_aud, _ = librosa.load(target_file_path, sr=sample_rate, mono=False)
-    sf.write(target_file_path, cur_aud.transpose(-1, -2), sample_rate, 'PCM_16')
-    # trim if possible
-    cur_wav_length = load_wav_length(target_file_path)
-    if cur_wav_length < duration_in_second*sample_rate:
-        return
-    if cur_wav_length-start_point_in_second*sample_rate < duration_in_second*sample_rate:
-        trimmed_audio = load_wav_segment(target_file_path, start_point=int(start_point_in_second*sample_rate), axis=1)
-    else:
-        trimmed_audio = load_wav_segment(target_file_path, start_point=int(start_point_in_second*sample_rate), duration=int(duration_in_second*sample_rate), axis=1)
-    sf.write(target_file_path, trimmed_audio, sample_rate, 'PCM_16')
 
+yt_video_dir = f"./yt_dir/0"
+os.makedirs(yt_video_dir, exist_ok=True)
 
-def set_up(start_point_in_second=0, duration_in_second=30):
-    os.environ['MASTER_ADDR'] = '127.0.0.1'
-    os.environ["CUDA_VISIBLE_DEVICES"] = '0'
-    os.environ['MASTER_PORT'] = '8888'
 
-    def str2bool(v):
-        if v.lower() in ('yes', 'true', 't', 'y', '1'):
-            return True
-        elif v.lower() in ('no', 'false', 'f', 'n', '0'):
-            return False
-        else:
-            raise argparse.ArgumentTypeError('Boolean value expected.')
+def get_audio_from_yt_video_input(yt_link: str, start_point_in_second=0, duration_in_second=30):
+    try:
+        yt = pt.YouTube(yt_link)
+        t = yt.streams.filter(only_audio=True)
+        filename_in = os.path.join(yt_video_dir, "input.wav")
+        t[0].download(filename=filename_in)
+    except VideoUnavailable as e:
+        warnings.warn(f"Video Not Found at {yt_link} ({e})")
+        filename_in = None
 
-    ''' Configurations for music mixing style transfer '''
-    currentdir = os.path.dirname(os.path.realpath(__file__))
-    default_ckpt_path_enc = os.path.join(os.path.dirname(currentdir), 'weights', 'FXencoder_ps.pt')
-    default_ckpt_path_conv = os.path.join(os.path.dirname(currentdir), 'weights', 'MixFXcloner_ps.pt')
-    default_norm_feature_path = os.path.join(os.path.dirname(currentdir), 'weights', 'musdb18_fxfeatures_eqcompimagegain.npy')
+    # trim audio length - due to computation time on HuggingFace environment
+    trim_audio(target_file_path=filename_in, start_point_in_second=start_point_in_second, duration_in_second=duration_in_second)
+    
+    return filename_in, filename_in
+
+def get_audio_from_yt_video_ref(yt_link: str, start_point_in_second=0, duration_in_second=30):
+    try:
+        yt = pt.YouTube(yt_link)
+        t = yt.streams.filter(only_audio=True)
+        filename_ref = os.path.join(yt_video_dir, "reference.wav")
+        t[0].download(filename=filename_ref)
+    except VideoUnavailable as e:
+        warnings.warn(f"Video Not Found at {yt_link} ({e})")
+        filename_ref = None
+
+    # trim audio length - due to computation time on HuggingFace environment
+    trim_audio(target_file_path=filename_ref, start_point_in_second=start_point_in_second, duration_in_second=duration_in_second)
+    
+    return filename_ref, filename_ref
 
-    import argparse
-    import yaml
-    parser = argparse.ArgumentParser()
+def inference(file_uploaded_in, file_uploaded_ref):
+    # clear out previously separated results
+    os.system(f"rm -r {yt_video_dir}/separated")
+    # change file path name
+    # os.system(f"cp {file_uploaded_in} {yt_video_dir}/input.wav")
+    # os.system(f"cp {file_uploaded_ref} {yt_video_dir}/reference.wav")
 
-    directory_args = parser.add_argument_group('Directory args')
-    # directory paths
-    directory_args.add_argument('--target_dir', type=str, default='./yt_dir/')
-    directory_args.add_argument('--output_dir', type=str, default=None, help='if no output_dir is specified (None), the results will be saved inside the target_dir')
-    directory_args.add_argument('--input_file_name', type=str, default='input')
-    directory_args.add_argument('--reference_file_name', type=str, default='reference')
-    directory_args.add_argument('--reference_file_name_2interpolate', type=str, default='reference_B')
-    # saved weights
-    directory_args.add_argument('--ckpt_path_enc', type=str, default=default_ckpt_path_enc)
-    directory_args.add_argument('--ckpt_path_conv', type=str, default=default_ckpt_path_conv)
-    directory_args.add_argument('--precomputed_normalization_feature', type=str, default=default_norm_feature_path)
+    sample_rate, data = file_uploaded_in
+    sf.write(f"{yt_video_dir}/input.wav", data, sample_rate)
+    sample_rate, data = file_uploaded_ref
+    sf.write(f"{yt_video_dir}/reference.wav", data, sample_rate)
 
-    inference_args = parser.add_argument_group('Inference args')
-    inference_args.add_argument('--sample_rate', type=int, default=44100)
-    inference_args.add_argument('--segment_length', type=int, default=2**19)        # segmentize input according to this duration
-    inference_args.add_argument('--segment_length_ref', type=int, default=2**19)    # segmentize reference according to this duration
-    # stem-level instruments & separation
-    inference_args.add_argument('--instruments', type=str2bool, default=["drums", "bass", "other", "vocals"], help='instrumental tracks to perform style transfer')
-    inference_args.add_argument('--stem_level_directory_name', type=str, default='separated')
-    inference_args.add_argument('--save_each_inst', type=str2bool, default=False)
-    inference_args.add_argument('--do_not_separate', type=str2bool, default=False)
-    inference_args.add_argument('--separation_model', type=str, default='htdemucs')
-    # FX normalization
-    inference_args.add_argument('--normalize_input', type=str2bool, default=True)
-    inference_args.add_argument('--normalization_order', type=str2bool, default=['loudness', 'eq', 'compression', 'imager', 'loudness']) # Effects to be normalized, order matters
-    inference_args.add_argument('--match_output_loudness', type=str2bool, default=False)
-    # interpolation
-    inference_args.add_argument('--interpolation', type=str2bool, default=False)
-    inference_args.add_argument('--interpolate_segments', type=int, default=30)
     
-    device_args = parser.add_argument_group('Device args')
-    device_args.add_argument('--workers', type=int, default=1)
-    device_args.add_argument('--batch_size', type=int, default=1)   # for processing long audio
-
-    args = parser.parse_args()
+    # Perform music mixing style transfer
+    args = set_up()
+    
+    inference_style_transfer = Mixing_Style_Transfer_Inference(args)
+    # output_wav_path, fin_data_out_mix = inference_style_transfer.inference(file_uploaded_in, file_uploaded_ref)
+    output_wav_path, fin_data_out_mix = inference_style_transfer.inference(f"{yt_video_dir}/input.wav", f"{yt_video_dir}/reference.wav")
+    print(fin_data_out_mix.shape)
+    return (44100, fin_data_out_mix.transpose())
+
+
+
+with gr.Blocks() as demo:
+    gr.HTML(
+        """
+            <div style="text-align: center; max-width: 700px; margin: 0 auto;">
+              <div
+                style="
+                  display: inline-flex;
+                  align-items: center;
+                  gap: 0.8rem;
+                  font-size: 1.75rem;
+                "
+              >
+                <h1 style="font-weight: 900; margin-bottom: 7px;">
+                  Music Mixing Style Transfer
+                </h1>
+              </div>
+        """
+    )
+    gr.Markdown(
+        """
+        This page is a Hugging Face interactive demo of the paper ["Music Mixing Style Transfer: A Contrastive Learning Approach to Disentangle Audio Effects"](https://huggingface.co/papers/2211.02247) (ICASSP 2023).
+            - [project page](https://jhtonykoo.github.io/MixingStyleTransfer/)
+            - [GitHub](https://github.com/jhtonyKoo/music_mixing_style_transfer)
+            - [supplementary](https://pale-cicada-946.notion.site/Music-Mixing-Style-Transfer-A-Contrastive-Learning-Approach-to-Disentangle-Audio-Effects-Supplemen-e6eccd9a431a4a8fa4fdd5adb2d3f219)
+        """
+    )
+    with gr.Group():
+        with gr.Column():
+            with gr.Blocks():
+                with gr.Tab("Input Music"):
+                    file_uploaded_in = gr.Audio(label="Input track (mix) to be mixing style transferred")
+                with gr.Tab("YouTube url"):
+                    with gr.Row():
+                        yt_link_in = gr.Textbox(
+                            label="Enter YouTube Link of the Video", autofocus=True, lines=3
+                        )
+                        yt_in_start_sec = gr.Number(
+                            value=0,
+                            label="starting point of the song (in seconds)"
+                        )
+                        yt_in_duration_sec = gr.Number(
+                            value=30,
+                            label="duration of the song (in seconds)"
+                        )
+                        yt_btn_in = gr.Button("Download Audio from YouTube Link", size="lg")
+                    yt_audio_path_in = gr.Audio(
+                        label="Input Audio Extracted from the YouTube Video", interactive=False
+                    )
+                    yt_btn_in.click(
+                        get_audio_from_yt_video_input,
+                        inputs=[yt_link_in, yt_in_start_sec, yt_in_duration_sec],
+                        outputs=[yt_audio_path_in, file_uploaded_in],
+                    )
+            with gr.Blocks():
+                with gr.Tab("Reference Music"):
+                    file_uploaded_ref = gr.Audio(label="Reference track (mix) to copy mixing style")
+                with gr.Tab("YouTube url"):
+                    with gr.Row():
+                        yt_link_ref = gr.Textbox(
+                            label="Enter YouTube Link of the Video", autofocus=True, lines=3
+                        )
+                        yt_ref_start_sec = gr.Number(
+                            value=0,
+                            label="starting point of the song (in seconds)"
+                        )
+                        yt_ref_duration_sec = gr.Number(
+                            value=30,
+                            label="duration of the song (in seconds)"
+                        )
+                        yt_btn_ref = gr.Button("Download Audio from YouTube Link", size="lg")
+                    yt_audio_path_ref = gr.Audio(
+                        label="Reference Audio Extracted from the YouTube Video", interactive=False
+                    )
+                    yt_btn_ref.click(
+                        get_audio_from_yt_video_ref,
+                        inputs=[yt_link_ref, yt_ref_start_sec, yt_ref_duration_sec],
+                        outputs=[yt_audio_path_ref, file_uploaded_ref],
+                    )
+                
+    with gr.Group():
+        gr.HTML(
+            """
+            <div> <h3> <center> Mixing Style Transfer. Perform stem-wise audio-effects style conversion by first source separating the input mix. The inference computation time takes longer as the input samples' duration. so plz be patient...  </h3> </div>
+            """
+        )
+        with gr.Column():
+            inference_btn = gr.Button("Run Mixing Style Transfer")
+        with gr.Row():
+            output_mix = gr.Audio(label="mixing style transferred music track", type='numpy')
+            inference_btn.click(
+                inference,
+                inputs=[file_uploaded_in, file_uploaded_ref],
+                outputs=[output_mix],
+            )
 
-    # load network configurations
-    with open(os.path.join(currentdir, 'configs.yaml'), 'r') as f:
-        configs = yaml.full_load(f)
-    args.cfg_encoder = configs['Effects_Encoder']['default']
-    args.cfg_converter = configs['TCN']['default']
 
-    return args
+    
+if __name__ == "__main__":
+    demo.launch(debug=True)