app.py

import os
import re
import io
import torch
import librosa
import zipfile
import requests
import torchaudio
import numpy as np
import gradio as gr
from uroman import uroman
import concurrent.futures
from pydub import AudioSegment
from datasets import load_dataset
from IPython.display import Audio
from scipy.signal import butter, lfilter
from speechbrain.pretrained import EncoderClassifier
from transformers import SpeechT5Processor, SpeechT5ForTextToSpeech, SpeechT5HifiGan

# Variables
spk_model_name = "speechbrain/spkrec-xvect-voxceleb"
dataset_name = "truong-xuan-linh/vi-xvector-speechbrain"
cache_dir="temp/"
default_model_name = "truong-xuan-linh/speecht5-vietnamese-voiceclone-lsvsc"
speaker_id = "speech_dataset_denoised"

# Active device
device = "cuda" if torch.cuda.is_available() else "cpu"

# Load models and datasets
speaker_model = EncoderClassifier.from_hparams(
    source=spk_model_name,
    run_opts={"device": device},
    savedir=os.path.join("/tmp", spk_model_name),
)
dataset = load_dataset(
    dataset_name, 
    download_mode="force_redownload", 
    verification_mode="no_checks", 
    cache_dir=cache_dir,
    revision="5ea5e4345258333cbc6d1dd2544f6c658e66a634"
)
dataset = dataset["train"].to_list()
dataset_dict = {}
for rc in dataset:
    dataset_dict[rc["speaker_id"]] = rc["embedding"]
vocoder = SpeechT5HifiGan.from_pretrained("microsoft/speecht5_hifigan")

# Model utility functions
def remove_special_characters(sentence):
    # Use regular expression to keep only letters, periods, and commas
    sentence_after_removal =  re.sub(r'[^a-zA-Z\s,.\u00C0-\u1EF9]', ' ,', sentence)
    return sentence_after_removal

def create_speaker_embedding(waveform):
    with torch.no_grad():
        speaker_embeddings = speaker_model.encode_batch(waveform)
        speaker_embeddings = torch.nn.functional.normalize(speaker_embeddings, dim=-1)
    return speaker_embeddings

def butter_bandpass(lowcut, highcut, fs, order=5):
    nyq = 0.5 * fs
    low = lowcut / nyq
    high = highcut / nyq
    b, a = butter(order, [low, high], btype='band')
    return b, a

def butter_bandpass_filter(data, lowcut, highcut, fs, order=5):
    b, a = butter_bandpass(lowcut, highcut, fs, order=order)
    y = lfilter(b, a, data)
    return y

def korean_splitter(string):
    pattern = re.compile('[가-힣]+')
    matches = pattern.findall(string)
    return matches

def uroman_normalization(string):
    korean_inputs = korean_splitter(string)
    for korean_input in korean_inputs:
        korean_roman = uroman(korean_input)
        string = string.replace(korean_input, korean_roman)
    return string

# Model class
class Model():
    def __init__(self, model_name, speaker_url=""):
        self.model_name = model_name
        self.processor = SpeechT5Processor.from_pretrained(model_name)
        self.model = SpeechT5ForTextToSpeech.from_pretrained(model_name)

        self.model.eval()
        self.speaker_url = speaker_url
        if speaker_url:
            print(f"download speaker_url")
            response = requests.get(speaker_url)
            audio_stream = io.BytesIO(response.content)
            audio_segment = AudioSegment.from_file(audio_stream, format="wav")
            audio_segment = audio_segment.set_channels(1)
            audio_segment = audio_segment.set_frame_rate(16000)
            audio_segment = audio_segment.set_sample_width(2)
            wavform, _ = torchaudio.load(audio_segment.export())
            self.speaker_embeddings = create_speaker_embedding(wavform)[0]
        else:
            self.speaker_embeddings = None
        
        if model_name == "truong-xuan-linh/speecht5-vietnamese-commonvoice" or model_name == "truong-xuan-linh/speecht5-irmvivoice":
            self.speaker_embeddings = torch.zeros((1, 512))  # or load xvectors from a file
            
    def inference(self, text, speaker_id=None):
        if "voiceclone" in self.model_name:
            if not self.speaker_url:
                self.speaker_embeddings = torch.tensor(dataset_dict[speaker_id])
            
        with torch.no_grad():
            full_speech = []
            separators = r";|\.|!|\?|\n"
            text = uroman_normalization(text)
            text = remove_special_characters(text)
            text = text.replace(" ", "▁")
            split_texts = re.split(separators, text)
            
            for split_text in split_texts:
                if split_text != "▁":
                    split_text = split_text.lower() + "▁"
                    print(split_text)
                    inputs = self.processor.tokenizer(text=split_text, return_tensors="pt")
                    speech = self.model.generate_speech(inputs["input_ids"], threshold=0.5, speaker_embeddings=self.speaker_embeddings, vocoder=vocoder)
                    full_speech.append(speech.numpy())
            return np.concatenate(full_speech)
    
    @staticmethod
    def moving_average(data, window_size):
        return np.convolve(data, np.ones(window_size)/window_size, mode='same')

# Initialize model
model = Model(
    model_name=default_model_name, 
    speaker_url=""
)

# Audio processing functions
def read_srt(file_path):
    subtitles = []
    with open(file_path, 'r', encoding='utf-8') as file:
        lines = file.readlines()
        
        for i in range(0, len(lines), 4):
            if i+2 < len(lines):
                start_time, end_time = lines[i+1].strip().split('-->')
                start_time = start_time.strip()
                end_time = end_time.strip()
                text = lines[i+2].strip()
                
                # Delete trailing dots
                while text.endswith('.'):
                    text = text[:-1]
                
                subtitles.append((start_time, end_time, text))
    
    return subtitles

def is_valid_srt(file_path):
    try:
        read_srt(file_path)
        return True
    except:
        return False

def time_to_seconds(time_str):
    h, m, s = time_str.split(':')
    seconds = int(h) * 3600 + int(m) * 60 + float(s.replace(',', '.'))
    return seconds

def closest_speedup_factor(factor, allowed_factors):
    return min(allowed_factors, key=lambda x: abs(x - factor)) + 0.1

def lowpass_filter(audio_data, cutoff=4000, fs=16000, order=4):
    """
    Áp dụng bộ lọc thông thấp cho dữ liệu âm thanh.

    Parameters:
    - audio_data: numpy array chứa dữ liệu âm thanh.
    - cutoff: Tần số cắt (Hz).
    - fs: Tần số lấy mẫu (Hz).
    - order: Bậc của bộ lọc.

    Returns:
    - filtered_audio: numpy array của âm thanh đã được lọc.
    """
    # Tạo bộ lọc butterworth
    nyq = 0.5 * fs
    normal_cutoff = cutoff / nyq
    b, a = butter(order, normal_cutoff, btype='low', analog=False)
    
    # Áp dụng bộ lọc
    filtered_audio = lfilter(b, a, audio_data)
    return filtered_audio

def generate_audio_with_pause(srt_file_path, speaker_id, speed_of_non_edit_speech):
    subtitles = read_srt(srt_file_path)
    audio_clips = []
    # allowed_factors = [1.1, 1.2, 1.3, 1.4, 1.5, 1.6, 1.7, 1.8, 1.9, 2.0]

    for i, (start_time, end_time, text) in enumerate(subtitles):
        # print("=====================================")
        # print("Text number:", i)
        # print(f"Start: {start_time}, End: {end_time}, Text: {text}")
        
        # Generate initial audio
        audio_data = model.inference(text=text, speaker_id=speaker_id)
        audio_data = audio_data / np.max(np.abs(audio_data))
        
        # Calculate required duration
        desired_duration = time_to_seconds(end_time) - time_to_seconds(start_time)
        current_duration = len(audio_data) / 16000
        
        # print(f"Time to seconds: {time_to_seconds(start_time)}, {time_to_seconds(end_time)}")
        # print(f"Desired duration: {desired_duration}, Current duration: {current_duration}")
        
        # Adjust audio speed by speedup
        if current_duration > desired_duration:
            raw_speedup_factor = current_duration / desired_duration
            # speedup_factor = closest_speedup_factor(raw_speedup_factor, allowed_factors)
            speedup_factor = raw_speedup_factor
            audio_data = librosa.effects.time_stretch(
                y=audio_data,
                rate=speedup_factor,
                n_fft=1024,
                hop_length=256
            )
            audio_data = audio_data / np.max(np.abs(audio_data))
            audio_data = audio_data * 1.2
            
        if current_duration < desired_duration:  
            if speed_of_non_edit_speech != 1:
                audio_data = librosa.effects.time_stretch(
                    y=audio_data,
                    rate=speed_of_non_edit_speech,
                    n_fft=1024,
                    hop_length=256
                )
                audio_data = audio_data / np.max(np.abs(audio_data))
                audio_data = audio_data * 1.2
            
            current_duration = len(audio_data) / 16000
            padding = int((desired_duration - current_duration) * 16000)
            audio_data = np.concatenate([np.zeros(padding), audio_data])
            
        # print(f"Final audio duration: {len(audio_data) / 16000}")
        # print("=====================================")

        audio_clips.append(lowpass_filter(audio_data))

        # Add pause
        if i < len(subtitles) - 1:
            next_start_time = subtitles[i + 1][0]
            pause_duration = time_to_seconds(next_start_time) - time_to_seconds(end_time)
            if pause_duration:
                pause_samples = int(pause_duration * 16000)
                audio_clips.append(np.zeros(pause_samples))

    final_audio = np.concatenate(audio_clips)
    
    return final_audio

def check_input_files(srt_files):
    if not srt_files:
        return None
    
    invalid_files = []
    for srt_file in srt_files:
        if not is_valid_srt(srt_file.name):
            invalid_files.append(srt_file.name)
    if invalid_files:
        raise gr.Warning(f"Invalid SRT files: {', '.join(invalid_files)}")

def srt_to_audio_multi(srt_files, speaker_id, speed_of_non_edit_speech):
    output_paths = []
    invalid_files = []
    
    def process_file(srt_file):
        if not is_valid_srt(srt_file.name):
            invalid_files.append(srt_file.name)
            return None
        audio_data = generate_audio_with_pause(srt_file.name, speaker_id, speed_of_non_edit_speech)
        output_path = os.path.join(cache_dir, f'output_{os.path.basename(srt_file.name)}.wav')
        torchaudio.save(output_path, torch.tensor(audio_data).unsqueeze(0), 16000)
        return output_path
    
    with concurrent.futures.ThreadPoolExecutor() as executor:
        futures = [executor.submit(process_file, srt_file) for srt_file in srt_files]
        for future in concurrent.futures.as_completed(futures):
            result = future.result()
            if result:
                output_paths.append(result)
    
    if invalid_files:
        raise gr.Warning(f"Invalid SRT files: {', '.join(invalid_files)}")
    
    return output_paths

def download_all(outputs):
    # If no outputs, return None
    if not outputs:
        raise gr.Warning("No files available for download.")
    
    zip_path = os.path.join(cache_dir, "all_outputs.zip")
    with zipfile.ZipFile(zip_path, 'w') as zipf:
        for file_path in outputs:
            zipf.write(file_path, os.path.basename(file_path))
    return zip_path

# Initialize model
model = Model(
    model_name=default_model_name, 
    speaker_url=""
)

# UI display
css = '''
#title{text-align: center}
#container{display: flex; justify-content: space-between; align-items: center;}
#setting-box{padding: 10px; border: 1px solid #ccc; border-radius: 5px;}
#setting-heading{margin-bottom: 10px; text-align: center;}
'''

with gr.Blocks(css=css) as demo:
    title = gr.HTML(
        """<h1>SRT to Audio Tool</h1>""",
        elem_id="title",
    )
    with gr.Column(elem_id="setting-box"):
        heading = gr.HTML("<h2>Settings</h2>", elem_id="setting-heading")
        with gr.Row():
            speaker_id = gr.Dropdown(
                label="Speaker ID",
                choices=list(dataset_dict.keys()),
                value=speaker_id
            )
            speed_of_non_edit_speech = gr.Slider(
                label="Speed of non-edit speech",
                minimum=1,
                maximum=2.0,
                step=0.1,
                value=1.2
            )
        
        
    with gr.Row(elem_id="container"):
        inp_srt = gr.File(
            label="Upload SRT files",
            file_count="multiple",
            type="filepath",
            file_types=["srt"],
            height=600
        )
        out = gr.File(
            label="Generated Audio Files",
            file_count="multiple",
            type="filepath",
            height=600,
            interactive=False
        )
    
    btn = gr.Button("Generate")
    download_btn = gr.Button("Download All")
    download_out = gr.File(
        label="Download ZIP", 
        interactive=False,
        height=100
    )

    inp_srt.change(check_input_files, inputs=inp_srt)
    btn.click(
        fn=srt_to_audio_multi, 
        inputs=[inp_srt, speaker_id, speed_of_non_edit_speech], 
        outputs=out
    )
    download_btn.click(fn=download_all, inputs=out, outputs=download_out)

if __name__ == "__main__":
    demo.launch()