from typing import Union from argparse import ArgumentParser from pathlib import Path import subprocess import librosa import os import time import random import yt_dlp from search import get_youtube, download_random import soundfile import matplotlib.pyplot as plt import numpy as np from PIL import Image, ImageDraw, ImageFont from moviepy.editor import * from moviepy.video.io.VideoFileClip import VideoFileClip import moviepy.editor as mpe import asyncio import json import hashlib from os import path, getenv from pydub import AudioSegment import gradio as gr import torch import edge_tts from datetime import datetime from scipy.io.wavfile import write import config import util from infer_pack.models import ( SynthesizerTrnMs768NSFsid, SynthesizerTrnMs768NSFsid_nono ) from vc_infer_pipeline import VC # music search def auto_search(name): save_music_path = '/tmp/downloaded' if not os.path.exists(save_music_path): os.makedirs(save_music_path) config = {'logfilepath': 'musicdl.log', save_music_path: save_music_path, 'search_size_per_source': 5, 'proxies': {}} save_path = os.path.join(save_music_path, name + '.mp3') # youtube get_youtube(name, os.path.join(save_music_path, name)) # task1 = threading.Thread( # target=get_youtube, # args=(name, os.path.join(save_music_path, name)) # ) # task1.start() # task2 = threading.Thread( # target=download_random, # args=(name, config, save_path) # ) # task2.start() # task1.join(timeout=20) # task2.join(timeout=10) if not os.path.exists(save_path): return "Not Found", None signal, sampling_rate = soundfile.read(save_path, dtype=np.int16) # signal, sampling_rate = open_audio(save_path) return (sampling_rate, signal) # SadTalker import os, sys from src.gradio_demo import SadTalker try: import webui # in webui in_webui = True except: in_webui = False def toggle_audio_file(choice): if choice == False: return gr.update(visible=True), gr.update(visible=False) else: return gr.update(visible=False), gr.update(visible=True) def ref_video_fn(path_of_ref_video): if path_of_ref_video is not None: return gr.update(value=True) else: return gr.update(value=False) sad_talker = SadTalker("checkpoints", "src/config", lazy_load=True) # combine video with music def combine_music(video, audio): my_clip = mpe.VideoFileClip(video) audio_background = mpe.AudioFileClip(audio) final_audio = mpe.CompositeAudioClip([my_clip.audio, audio_background]) final_clip = my_clip.set_audio(final_audio) final_clip.write_videofile("video.mp4") return "video.mp4" # Reference: https://huggingface.co/spaces/zomehwh/rvc-models/blob/main/app.py#L21 # noqa in_hf_space = getenv('SYSTEM') == 'spaces' high_quality = True # Argument parsing arg_parser = ArgumentParser() arg_parser.add_argument( '--hubert', default=getenv('RVC_HUBERT', 'hubert_base.pt'), help='path to hubert base model (default: hubert_base.pt)' ) arg_parser.add_argument( '--config', default=getenv('RVC_MULTI_CFG', 'multi_config.json'), help='path to config file (default: multi_config.json)' ) arg_parser.add_argument( '--api', action='store_true', help='enable api endpoint' ) arg_parser.add_argument( '--cache-examples', action='store_true', help='enable example caching, please remember delete gradio_cached_examples folder when example config has been modified' # noqa ) args = arg_parser.parse_args() app_css = ''' #model_info img { max-width: 100px; max-height: 100px; float: right; } #model_info p { margin: unset; } ''' app = gr.Blocks( theme=gr.themes.Soft(primary_hue="orange", secondary_hue="slate"), css=app_css, analytics_enabled=False ) # Load hubert model hubert_model = util.load_hubert_model(config.device, args.hubert) hubert_model.eval() # Load models multi_cfg = json.load(open(args.config, 'r')) loaded_models = [] for model_name in multi_cfg.get('models'): print(f'Loading model: {model_name}') # Load model info model_info = json.load( open(path.join('model', model_name, 'config.json'), 'r') ) # Load RVC checkpoint cpt = torch.load( path.join('model', model_name, model_info['model']), map_location='cpu' ) tgt_sr = cpt['config'][-1] cpt['config'][-3] = cpt['weight']['emb_g.weight'].shape[0] # n_spk if_f0 = cpt.get('f0', 1) net_g: Union[SynthesizerTrnMs768NSFsid, SynthesizerTrnMs768NSFsid_nono] if if_f0 == 1: net_g = SynthesizerTrnMs768NSFsid( *cpt['config'], is_half=util.is_half(config.device) ) else: net_g = SynthesizerTrnMs768NSFsid_nono(*cpt['config']) del net_g.enc_q # According to original code, this thing seems necessary. print(net_g.load_state_dict(cpt['weight'], strict=False)) net_g.eval().to(config.device) net_g = net_g.half() if util.is_half(config.device) else net_g.float() vc = VC(tgt_sr, config) loaded_models.append(dict( name=model_name, metadata=model_info, vc=vc, net_g=net_g, if_f0=if_f0, target_sr=tgt_sr )) print(f'Models loaded: {len(loaded_models)}') # Edge TTS speakers tts_speakers_list = asyncio.get_event_loop().run_until_complete(edge_tts.list_voices()) # noqa # Make MV def make_bars_image(height_values, index, new_height): # Define the size of the image width = 512 height = new_height # Create a new image with a transparent background image = Image.new('RGBA', (width, height), color=(0, 0, 0, 0)) # Get the image drawing context draw = ImageDraw.Draw(image) # Define the rectangle width and spacing rect_width = 2 spacing = 2 # Define the list of height values for the rectangles #height_values = [20, 40, 60, 80, 100, 80, 60, 40] num_bars = len(height_values) # Calculate the total width of the rectangles and the spacing total_width = num_bars * rect_width + (num_bars - 1) * spacing # Calculate the starting position for the first rectangle start_x = int((width - total_width) / 2) # Define the buffer size buffer_size = 80 # Draw the rectangles from left to right x = start_x for i, height in enumerate(height_values): # Define the rectangle coordinates y0 = buffer_size y1 = height + buffer_size x0 = x x1 = x + rect_width # Draw the rectangle draw.rectangle([x0, y0, x1, y1], fill='white') # Move to the next rectangle position if i < num_bars - 1: x += rect_width + spacing # Rotate the image by 180 degrees image = image.rotate(180) # Mirror the image image = image.transpose(Image.FLIP_LEFT_RIGHT) # Save the image image.save('audio_bars_'+ str(index) + '.png') return 'audio_bars_'+ str(index) + '.png' def db_to_height(db_value): # Scale the dB value to a range between 0 and 1 scaled_value = (db_value + 80) / 80 # Convert the scaled value to a height between 0 and 100 height = scaled_value * 50 return height def infer(title, audio_in, image_in): # Load the audio file audio_path = audio_in audio_data, sr = librosa.load(audio_path) # Get the duration in seconds duration = librosa.get_duration(y=audio_data, sr=sr) # Extract the audio data for the desired time start_time = 0 # start time in seconds end_time = duration # end time in seconds start_index = int(start_time * sr) end_index = int(end_time * sr) audio_data = audio_data[start_index:end_index] # Compute the short-time Fourier transform hop_length = 512 stft = librosa.stft(audio_data, hop_length=hop_length) spectrogram = librosa.amplitude_to_db(np.abs(stft), ref=np.max) # Get the frequency values freqs = librosa.fft_frequencies(sr=sr, n_fft=stft.shape[0]) # Select the indices of the frequency values that correspond to the desired frequencies n_freqs = 114 freq_indices = np.linspace(0, len(freqs) - 1, n_freqs, dtype=int) # Extract the dB values for the desired frequencies db_values = [] for i in range(spectrogram.shape[1]): db_values.append(list(zip(freqs[freq_indices], spectrogram[freq_indices, i]))) # Print the dB values for the first time frame print(db_values[0]) proportional_values = [] for frame in db_values: proportional_frame = [db_to_height(db) for f, db in frame] proportional_values.append(proportional_frame) print(proportional_values[0]) print("AUDIO CHUNK: " + str(len(proportional_values))) # Open the background image background_image = Image.open(image_in) # Resize the image while keeping its aspect ratio bg_width, bg_height = background_image.size aspect_ratio = bg_width / bg_height new_width = 512 new_height = int(new_width / aspect_ratio) resized_bg = background_image.resize((new_width, new_height)) # Apply black cache for better visibility of the white text bg_cache = Image.open('black_cache.png') resized_bg.paste(bg_cache, (0, resized_bg.height - bg_cache.height), mask=bg_cache) # Create a new ImageDraw object draw = ImageDraw.Draw(resized_bg) # Define the text to be added text = title font = ImageFont.truetype("NotoSansSC-Regular.otf", 16) text_color = (255, 255, 255) # white color # Calculate the position of the text text_width, text_height = draw.textsize(text, font=font) x = 30 y = new_height - 70 # Draw the text on the image draw.text((x, y), text, fill=text_color, font=font) # Save the resized image resized_bg.save('resized_background.jpg') generated_frames = [] for i, frame in enumerate(proportional_values): bars_img = make_bars_image(frame, i, new_height) bars_img = Image.open(bars_img) # Paste the audio bars image on top of the background image fresh_bg = Image.open('resized_background.jpg') fresh_bg.paste(bars_img, (0, 0), mask=bars_img) # Save the image fresh_bg.save('audio_bars_with_bg' + str(i) + '.jpg') generated_frames.append('audio_bars_with_bg' + str(i) + '.jpg') print(generated_frames) # Create a video clip from the images clip = ImageSequenceClip(generated_frames, fps=len(generated_frames)/(end_time-start_time)) audio_clip = AudioFileClip(audio_in) clip = clip.set_audio(audio_clip) # Set the output codec codec = 'libx264' audio_codec = 'aac' # Save the video to a file clip.write_videofile("my_video.mp4", codec=codec, audio_codec=audio_codec) retimed_clip = VideoFileClip("my_video.mp4") # Set the desired frame rate new_fps = 25 # Create a new clip with the new frame rate new_clip = retimed_clip.set_fps(new_fps) # Save the new clip as a new video file new_clip.write_videofile("my_video_retimed.mp4", codec=codec, audio_codec=audio_codec) return "my_video_retimed.mp4" # mix vocal and non-vocal def mix(audio1, audio2): sound1 = AudioSegment.from_file(audio1) sound2 = AudioSegment.from_file(audio2) length = len(sound1) mixed = sound1[:length].overlay(sound2) mixed.export("song.wav", format="wav") return "song.wav" # Bilibili def youtube_downloader( video_identifier, start_time, end_time, is_full_song, output_filename="track.wav", num_attempts=5, url_base="", quiet=False, force=True, ): if is_full_song: ydl_opts = { 'noplaylist': True, 'format': 'bestaudio/best', 'postprocessors': [{ 'key': 'FFmpegExtractAudio', 'preferredcodec': 'wav', }], "outtmpl": 'dl_audio/youtube_audio', } with yt_dlp.YoutubeDL(ydl_opts) as ydl: ydl.download([video_identifier]) audio_path = "dl_audio/youtube_audio.wav" return audio_path else: output_path = Path(output_filename) if output_path.exists(): if not force: return output_path else: output_path.unlink() quiet = "--quiet --no-warnings" if quiet else "" command = f""" yt-dlp {quiet} -x --audio-format wav -f bestaudio -o "{output_filename}" --download-sections "*{start_time}-{end_time}" "{url_base}{video_identifier}" # noqa: E501 """.strip() attempts = 0 while True: try: _ = subprocess.check_output(command, shell=True, stderr=subprocess.STDOUT) except subprocess.CalledProcessError: attempts += 1 if attempts == num_attempts: return None else: break if output_path.exists(): return output_path else: return None def audio_separated(audio_input, progress=gr.Progress()): # start progress progress(progress=0, desc="Starting...") time.sleep(0.1) # check file input if audio_input is None: # show progress for i in progress.tqdm(range(100), desc="Please wait..."): time.sleep(0.01) return (None, None, 'Please input audio.') # create filename filename = str(random.randint(10000,99999))+datetime.now().strftime("%d%m%Y%H%M%S") # progress progress(progress=0.10, desc="Please wait...") # make dir output os.makedirs("output", exist_ok=True) # progress progress(progress=0.20, desc="Please wait...") # write if high_quality: write(filename+".wav", audio_input[0], audio_input[1]) else: write(filename+".mp3", audio_input[0], audio_input[1]) # progress progress(progress=0.50, desc="Please wait...") # demucs process if high_quality: command_demucs = "python3 -m demucs --two-stems=vocals -d cpu "+filename+".wav -o output" else: command_demucs = "python3 -m demucs --two-stems=vocals --mp3 --mp3-bitrate 128 -d cpu "+filename+".mp3 -o output" os.system(command_demucs) # progress progress(progress=0.70, desc="Please wait...") # remove file audio if high_quality: command_delete = "rm -v ./"+filename+".wav" else: command_delete = "rm -v ./"+filename+".mp3" os.system(command_delete) # progress progress(progress=0.80, desc="Please wait...") # progress for i in progress.tqdm(range(80,100), desc="Please wait..."): time.sleep(0.1) if high_quality: return "./output/htdemucs/"+filename+"/vocals.wav","./output/htdemucs/"+filename+"/no_vocals.wav","Successfully..." else: return "./output/htdemucs/"+filename+"/vocals.mp3","./output/htdemucs/"+filename+"/no_vocals.mp3","Successfully..." # https://github.com/fumiama/Retrieval-based-Voice-Conversion-WebUI/blob/main/infer-web.py#L118 # noqa def vc_func( input_audio, model_index, pitch_adjust, f0_method, feat_ratio, filter_radius, rms_mix_rate, resample_option ): if input_audio is None: return (None, 'Please provide input audio.') if model_index is None: return (None, 'Please select a model.') model = loaded_models[model_index] # Reference: so-vits (audio_samp, audio_npy) = input_audio # https://huggingface.co/spaces/zomehwh/rvc-models/blob/main/app.py#L49 # Can be change well, we will see if (audio_npy.shape[0] / audio_samp) > 600 and in_hf_space: return (None, 'Input audio is longer than 600 secs.') # Bloody hell: https://stackoverflow.com/questions/26921836/ if audio_npy.dtype != np.float32: # :thonk: audio_npy = ( audio_npy / np.iinfo(audio_npy.dtype).max ).astype(np.float32) if len(audio_npy.shape) > 1: audio_npy = librosa.to_mono(audio_npy.transpose(1, 0)) if audio_samp != 16000: audio_npy = librosa.resample( audio_npy, orig_sr=audio_samp, target_sr=16000 ) pitch_int = int(pitch_adjust) resample = ( 0 if resample_option == 'Disable resampling' else int(resample_option) ) times = [0, 0, 0] checksum = hashlib.sha512() checksum.update(audio_npy.tobytes()) output_audio = model['vc'].pipeline( hubert_model, model['net_g'], model['metadata'].get('speaker_id', 0), audio_npy, checksum.hexdigest(), times, pitch_int, f0_method, path.join('model', model['name'], model['metadata']['feat_index']), feat_ratio, model['if_f0'], filter_radius, model['target_sr'], resample, rms_mix_rate, 'v2' ) out_sr = ( resample if resample >= 16000 and model['target_sr'] != resample else model['target_sr'] ) print(f'npy: {times[0]}s, f0: {times[1]}s, infer: {times[2]}s') return ((out_sr, output_audio), 'Success') async def edge_tts_vc_func( input_text, model_index, tts_speaker, pitch_adjust, f0_method, feat_ratio, filter_radius, rms_mix_rate, resample_option ): if input_text is None: return (None, 'Please provide TTS text.') if tts_speaker is None: return (None, 'Please select TTS speaker.') if model_index is None: return (None, 'Please select a model.') speaker = tts_speakers_list[tts_speaker]['ShortName'] (tts_np, tts_sr) = await util.call_edge_tts(speaker, input_text) return vc_func( (tts_sr, tts_np), model_index, pitch_adjust, f0_method, feat_ratio, filter_radius, rms_mix_rate, resample_option ) def update_model_info(model_index): if model_index is None: return str( '### Model info\n' 'Please select a model from dropdown above.' ) model = loaded_models[model_index] model_icon = model['metadata'].get('icon', '') return str( '### Model info\n' '![model icon]({icon})' '**{name}**\n\n' 'Author: {author}\n\n' 'Source: {source}\n\n' '{note}' ).format( name=model['metadata'].get('name'), author=model['metadata'].get('author', 'Anonymous'), source=model['metadata'].get('source', 'Unknown'), note=model['metadata'].get('note', ''), icon=( model_icon if model_icon.startswith(('http://', 'https://')) else '/file/model/%s/%s' % (model['name'], model_icon) ) ) def _example_vc( input_audio, model_index, pitch_adjust, f0_method, feat_ratio, filter_radius, rms_mix_rate, resample_option ): (audio, message) = vc_func( input_audio, model_index, pitch_adjust, f0_method, feat_ratio, filter_radius, rms_mix_rate, resample_option ) return ( audio, message, update_model_info(model_index) ) async def _example_edge_tts( input_text, model_index, tts_speaker, pitch_adjust, f0_method, feat_ratio, filter_radius, rms_mix_rate, resample_option ): (audio, message) = await edge_tts_vc_func( input_text, model_index, tts_speaker, pitch_adjust, f0_method, feat_ratio, filter_radius, rms_mix_rate, resample_option ) return ( audio, message, update_model_info(model_index) ) with app: gr.HTML("