import os import re import torch import torchaudio import gradio as gr import numpy as np import tempfile from einops import rearrange from vocos import Vocos from pydub import AudioSegment, silence from model import CFM, UNetT, DiT, MMDiT from cached_path import cached_path from model.utils import ( load_checkpoint, get_tokenizer, convert_char_to_pinyin, save_spectrogram, ) from transformers import pipeline import librosa import click import soundfile as sf SPLIT_WORDS = [ "but", "however", "nevertheless", "yet", "still", "therefore", "thus", "hence", "consequently", "moreover", "furthermore", "additionally", "meanwhile", "alternatively", "otherwise", "namely", "specifically", "for example", "such as", "in fact", "indeed", "notably", "in contrast", "on the other hand", "conversely", "in conclusion", "to summarize", "finally" ] device = ( "cuda" if torch.cuda.is_available() else "mps" if torch.backends.mps.is_available() else "cpu" ) print(f"Using {device} device") pipe = pipeline( "automatic-speech-recognition", model="openai/whisper-large-v3-turbo", torch_dtype=torch.float16, device=device, ) # --------------------- Settings -------------------- # target_sample_rate = 24000 n_mel_channels = 100 hop_length = 256 target_rms = 0.1 nfe_step = 32 # 16, 32 cfg_strength = 2.0 ode_method = "euler" sway_sampling_coef = -1.0 speed = 1.0 # fix_duration = 27 # None or float (duration in seconds) fix_duration = None def load_model(exp_name, model_cls, model_cfg, ckpt_step): ckpt_path = str(cached_path(f"hf://SWivid/F5-TTS/{exp_name}/model_{ckpt_step}.safetensors")) # ckpt_path = f"ckpts/{exp_name}/model_{ckpt_step}.pt" # .pt | .safetensors vocab_char_map, vocab_size = get_tokenizer("Emilia_ZH_EN", "pinyin") model = CFM( transformer=model_cls( **model_cfg, text_num_embeds=vocab_size, mel_dim=n_mel_channels ), mel_spec_kwargs=dict( target_sample_rate=target_sample_rate, n_mel_channels=n_mel_channels, hop_length=hop_length, ), odeint_kwargs=dict( method=ode_method, ), vocab_char_map=vocab_char_map, ).to(device) model = load_checkpoint(model, ckpt_path, device, use_ema = True) return model # load models F5TTS_model_cfg = dict( dim=1024, depth=22, heads=16, ff_mult=2, text_dim=512, conv_layers=4 ) E2TTS_model_cfg = dict(dim=1024, depth=24, heads=16, ff_mult=4) F5TTS_ema_model = load_model( "F5TTS_Base", DiT, F5TTS_model_cfg, 1200000 ) E2TTS_ema_model = load_model( "E2TTS_Base", UNetT, E2TTS_model_cfg, 1200000 ) def split_text_into_batches(text, max_chars=200, split_words=SPLIT_WORDS): sentences = re.split('([。.!?!?])', text) sentences = [''.join(i) for i in zip(sentences[0::2], sentences[1::2])] batches = [] current_batch = "" def split_by_words(text): words = text.split() current_word_part = "" word_batches = [] for word in words: if len(current_word_part.encode('utf-8')) + len(word.encode('utf-8')) + 1 <= max_chars: current_word_part += word + ' ' else: if current_word_part: # Try to find a suitable split word for split_word in split_words: split_index = current_word_part.rfind(' ' + split_word + ' ') if split_index != -1: word_batches.append(current_word_part[:split_index].strip()) current_word_part = current_word_part[split_index:].strip() + ' ' break else: # If no suitable split word found, just append the current part word_batches.append(current_word_part.strip()) current_word_part = "" current_word_part += word + ' ' if current_word_part: word_batches.append(current_word_part.strip()) return word_batches for sentence in sentences: if len(current_batch.encode('utf-8')) + len(sentence.encode('utf-8')) <= max_chars: current_batch += sentence else: # If adding this sentence would exceed the limit if current_batch: batches.append(current_batch) current_batch = "" # If the sentence itself is longer than max_chars, split it if len(sentence.encode('utf-8')) > max_chars: # First, try to split by colon colon_parts = sentence.split(':') if len(colon_parts) > 1: for part in colon_parts: if len(part.encode('utf-8')) <= max_chars: batches.append(part) else: # If colon part is still too long, split by comma comma_parts = re.split('[,,]', part) if len(comma_parts) > 1: current_comma_part = "" for comma_part in comma_parts: if len(current_comma_part.encode('utf-8')) + len(comma_part.encode('utf-8')) <= max_chars: current_comma_part += comma_part + ',' else: if current_comma_part: batches.append(current_comma_part.rstrip(',')) current_comma_part = comma_part + ',' if current_comma_part: batches.append(current_comma_part.rstrip(',')) else: # If no comma, split by words batches.extend(split_by_words(part)) else: # If no colon, split by comma comma_parts = re.split('[,,]', sentence) if len(comma_parts) > 1: current_comma_part = "" for comma_part in comma_parts: if len(current_comma_part.encode('utf-8')) + len(comma_part.encode('utf-8')) <= max_chars: current_comma_part += comma_part + ',' else: if current_comma_part: batches.append(current_comma_part.rstrip(',')) current_comma_part = comma_part + ',' if current_comma_part: batches.append(current_comma_part.rstrip(',')) else: # If no comma, split by words batches.extend(split_by_words(sentence)) else: current_batch = sentence if current_batch: batches.append(current_batch) return batches def infer_batch(ref_audio, ref_text, gen_text_batches, exp_name, remove_silence, progress=gr.Progress()): if exp_name == "F5-TTS": ema_model = F5TTS_ema_model elif exp_name == "E2-TTS": ema_model = E2TTS_ema_model audio, sr = torchaudio.load(ref_audio) if audio.shape[0] > 1: audio = torch.mean(audio, dim=0, keepdim=True) rms = torch.sqrt(torch.mean(torch.square(audio))) if rms < target_rms: audio = audio * target_rms / rms if sr != target_sample_rate: resampler = torchaudio.transforms.Resample(sr, target_sample_rate) audio = resampler(audio) audio = audio.to(device) generated_waves = [] spectrograms = [] for i, gen_text in enumerate(progress.tqdm(gen_text_batches)): # Prepare the text text_list = [ref_text + gen_text] final_text_list = convert_char_to_pinyin(text_list) # Calculate duration ref_audio_len = audio.shape[-1] // hop_length zh_pause_punc = r"。,、;:?!" ref_text_len = len(ref_text.encode('utf-8')) + 3 * len(re.findall(zh_pause_punc, ref_text)) gen_text_len = len(gen_text.encode('utf-8')) + 3 * len(re.findall(zh_pause_punc, gen_text)) duration = ref_audio_len + int(ref_audio_len / ref_text_len * gen_text_len / speed) # inference with torch.inference_mode(): generated, _ = ema_model.sample( cond=audio, text=final_text_list, duration=duration, steps=nfe_step, cfg_strength=cfg_strength, sway_sampling_coef=sway_sampling_coef, ) generated = generated[:, ref_audio_len:, :] generated_mel_spec = rearrange(generated, "1 n d -> 1 d n") vocos = Vocos.from_pretrained("charactr/vocos-mel-24khz") generated_wave = vocos.decode(generated_mel_spec.cpu()) if rms < target_rms: generated_wave = generated_wave * rms / target_rms # wav -> numpy generated_wave = generated_wave.squeeze().cpu().numpy() generated_waves.append(generated_wave) spectrograms.append(generated_mel_spec[0].cpu().numpy()) # Combine all generated waves final_wave = np.concatenate(generated_waves) # Remove silence if remove_silence: with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as f: sf.write(f.name, final_wave, target_sample_rate) aseg = AudioSegment.from_file(f.name) non_silent_segs = silence.split_on_silence(aseg, min_silence_len=1000, silence_thresh=-50, keep_silence=500) non_silent_wave = AudioSegment.silent(duration=0) for non_silent_seg in non_silent_segs: non_silent_wave += non_silent_seg aseg = non_silent_wave aseg.export(f.name, format="wav") final_wave, _ = torchaudio.load(f.name) final_wave = final_wave.squeeze().cpu().numpy() # Create a combined spectrogram combined_spectrogram = np.concatenate(spectrograms, axis=1) with tempfile.NamedTemporaryFile(suffix=".png", delete=False) as tmp_spectrogram: spectrogram_path = tmp_spectrogram.name save_spectrogram(combined_spectrogram, spectrogram_path) return (target_sample_rate, final_wave), spectrogram_path def infer(ref_audio_orig, ref_text, gen_text, exp_name, remove_silence, custom_split_words): if not custom_split_words.strip(): custom_words = [word.strip() for word in custom_split_words.split(',')] global SPLIT_WORDS SPLIT_WORDS = custom_words print(gen_text) gr.Info("Converting audio...") with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as f: aseg = AudioSegment.from_file(ref_audio_orig) non_silent_segs = silence.split_on_silence(aseg, min_silence_len=1000, silence_thresh=-50, keep_silence=500) non_silent_wave = AudioSegment.silent(duration=0) for non_silent_seg in non_silent_segs: non_silent_wave += non_silent_seg aseg = non_silent_wave audio_duration = len(aseg) if audio_duration > 15000: gr.Warning("Audio is over 15s, clipping to only first 15s.") aseg = aseg[:15000] aseg.export(f.name, format="wav") ref_audio = f.name if not ref_text.strip(): gr.Info("No reference text provided, transcribing reference audio...") ref_text = pipe( ref_audio, chunk_length_s=30, batch_size=128, generate_kwargs={"task": "transcribe"}, return_timestamps=False, )["text"].strip() gr.Info("Finished transcription") else: gr.Info("Using custom reference text...") # Split the input text into batches if len(ref_text.encode('utf-8')) == len(ref_text): max_chars = 400-len(ref_text.encode('utf-8')) else: max_chars = 300-len(ref_text.encode('utf-8')) gen_text_batches = split_text_into_batches(gen_text, max_chars=max_chars) print('ref_text', ref_text) for i, gen_text in enumerate(gen_text_batches): print(f'gen_text {i}', gen_text) gr.Info(f"Generating audio using {exp_name} in {len(gen_text_batches)} batches") return infer_batch(ref_audio, ref_text, gen_text_batches, exp_name, remove_silence) def generate_podcast(script, speaker1_name, ref_audio1, ref_text1, speaker2_name, ref_audio2, ref_text2, exp_name, remove_silence): # Split the script into speaker blocks speaker_pattern = re.compile(f"^({re.escape(speaker1_name)}|{re.escape(speaker2_name)}):", re.MULTILINE) speaker_blocks = speaker_pattern.split(script)[1:] # Skip the first empty element generated_audio_segments = [] for i in range(0, len(speaker_blocks), 2): speaker = speaker_blocks[i] text = speaker_blocks[i+1].strip() # Determine which speaker is talking if speaker == speaker1_name: ref_audio = ref_audio1 ref_text = ref_text1 elif speaker == speaker2_name: ref_audio = ref_audio2 ref_text = ref_text2 else: continue # Skip if the speaker is neither speaker1 nor speaker2 # Generate audio for this block audio, _ = infer(ref_audio, ref_text, text, exp_name, remove_silence) # Convert the generated audio to a numpy array sr, audio_data = audio # Save the audio data as a WAV file with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as temp_file: sf.write(temp_file.name, audio_data, sr) audio_segment = AudioSegment.from_wav(temp_file.name) generated_audio_segments.append(audio_segment) # Add a short pause between speakers pause = AudioSegment.silent(duration=500) # 500ms pause generated_audio_segments.append(pause) # Concatenate all audio segments final_podcast = sum(generated_audio_segments) # Export the final podcast with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as temp_file: podcast_path = temp_file.name final_podcast.export(podcast_path, format="wav") return podcast_path with gr.Blocks() as app: gr.Markdown( """ # E2/F5 TTS with Advanced Batch Processing This is a local web UI for F5 TTS with advanced batch processing support, based on the unofficial [online demo](https://huggingface.co/spaces/mrfakename/E2-F5-TTS). This app supports the following TTS models: * [F5-TTS](https://arxiv.org/abs/2410.06885) (A Fairytaler that Fakes Fluent and Faithful Speech with Flow Matching) * [E2-TTS](https://arxiv.org/abs/2406.18009) (Embarrassingly Easy Fully Non-Autoregressive Zero-Shot TTS) The checkpoints support English and Chinese. If you're having issues, try converting your reference audio to WAV or MP3, clipping it to 15s, and shortening your prompt. **NOTE: Reference text will be automatically transcribed with Whisper if not provided. For best results, keep your reference clips short (<15s). Ensure the audio is fully uploaded before generating.** """ ) ref_audio_input = gr.Audio(label="Reference Audio", type="filepath") gen_text_input = gr.Textbox(label="Text to Generate", lines=10) model_choice = gr.Radio( choices=["F5-TTS", "E2-TTS"], label="Choose TTS Model", value="F5-TTS" ) generate_btn = gr.Button("Synthesize", variant="primary") with gr.Accordion("Advanced Settings", open=False): ref_text_input = gr.Textbox( label="Reference Text", info="Leave blank to automatically transcribe the reference audio. If you enter text it will override automatic transcription.", lines=2, ) remove_silence = gr.Checkbox( label="Remove Silences", info="The model tends to produce silences, especially on longer audio. We can manually remove silences if needed. Note that this is an experimental feature and may produce strange results. This will also increase generation time.", value=True, ) split_words_input = gr.Textbox( label="Custom Split Words", info="Enter custom words to split on, separated by commas. Leave blank to use default list.", lines=2, ) audio_output = gr.Audio(label="Synthesized Audio") spectrogram_output = gr.Image(label="Spectrogram") generate_btn.click( infer, inputs=[ ref_audio_input, ref_text_input, gen_text_input, model_choice, remove_silence, split_words_input, ], outputs=[audio_output, spectrogram_output], ) gr.Markdown( """ # Podcast Generation Supported by [RootingInLoad](https://github.com/RootingInLoad) """ ) with gr.Tab("Podcast Generation"): speaker1_name = gr.Textbox(label="Speaker 1 Name") ref_audio_input1 = gr.Audio(label="Reference Audio (Speaker 1)", type="filepath") ref_text_input1 = gr.Textbox(label="Reference Text (Speaker 1)", lines=2) speaker2_name = gr.Textbox(label="Speaker 2 Name") ref_audio_input2 = gr.Audio(label="Reference Audio (Speaker 2)", type="filepath") ref_text_input2 = gr.Textbox(label="Reference Text (Speaker 2)", lines=2) script_input = gr.Textbox(label="Podcast Script", lines=10, placeholder="Enter the script with speaker names at the start of each block, e.g.:\nSean: How did you start studying...\n\nMeghan: I came to my interest in technology...\nIt was a long journey...\n\nSean: That's fascinating. Can you elaborate...") podcast_model_choice = gr.Radio( choices=["F5-TTS", "E2-TTS"], label="Choose TTS Model", value="F5-TTS" ) podcast_remove_silence = gr.Checkbox( label="Remove Silences", value=True, ) generate_podcast_btn = gr.Button("Generate Podcast", variant="primary") podcast_output = gr.Audio(label="Generated Podcast") def podcast_generation(script, speaker1, ref_audio1, ref_text1, speaker2, ref_audio2, ref_text2, model, remove_silence): return generate_podcast(script, speaker1, ref_audio1, ref_text1, speaker2, ref_audio2, ref_text2, model, remove_silence) generate_podcast_btn.click( podcast_generation, inputs=[ script_input, speaker1_name, ref_audio_input1, ref_text_input1, speaker2_name, ref_audio_input2, ref_text_input2, podcast_model_choice, podcast_remove_silence, ], outputs=podcast_output, ) @click.command() @click.option("--port", "-p", default=None, type=int, help="Port to run the app on") @click.option("--host", "-H", default=None, help="Host to run the app on") @click.option( "--share", "-s", default=False, is_flag=True, help="Share the app via Gradio share link", ) @click.option("--api", "-a", default=True, is_flag=True, help="Allow API access") def main(port, host, share, api): global app print(f"Starting app...") app.queue(api_open=api).launch( server_name=host, server_port=port, share=share, show_api=api ) if __name__ == "__main__": main()