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README.md

@@ -1,13 +1,196 @@
----
-title: F5-TTS
-emoji: 🗣️
-colorFrom: green
-colorTo: green
-sdk: gradio
-app_file: app.py
-pinned: true
-short_description: 'F5-TTS & E2-TTS: Zero-Shot Voice Cloning (Unofficial Demo)'
-sdk_version: 5.1.0
----
-
-Check out the configuration reference at https://huggingface.co/docs/hub/spaces-config-reference
+# F5-TTS: A Fairytaler that Fakes Fluent and Faithful Speech with Flow Matching
+
+[![python](https://img.shields.io/badge/Python-3.10-brightgreen)](https://github.com/SWivid/F5-TTS)
+[![arXiv](https://img.shields.io/badge/arXiv-2410.06885-b31b1b.svg?logo=arXiv)](https://arxiv.org/abs/2410.06885)
+[![demo](https://img.shields.io/badge/GitHub-Demo%20page-blue.svg)](https://swivid.github.io/F5-TTS/)
+[![space](https://img.shields.io/badge/🤗-Space%20demo-yellow)](https://huggingface.co/spaces/mrfakename/E2-F5-TTS)
+
+**F5-TTS**: Diffusion Transformer with ConvNeXt V2, faster trained and inference.
+
+**E2 TTS**: Flat-UNet Transformer, closest reproduction.
+
+**Sway Sampling**: Inference-time flow step sampling strategy, greatly improves performance
+
+## Installation
+
+Clone the repository:
+
+```bash
+git clone https://github.com/SWivid/F5-TTS.git
+cd F5-TTS
+```
+
+Install torch with your CUDA version, e.g. :
+
+```bash
+pip install torch==2.3.0+cu118 --extra-index-url https://download.pytorch.org/whl/cu118
+pip install torchaudio==2.3.0+cu118 --extra-index-url https://download.pytorch.org/whl/cu118
+```
+
+Install other packages:
+
+```bash
+pip install -r requirements.txt
+```
+
+## Prepare Dataset
+
+Example data processing scripts for Emilia and Wenetspeech4TTS, and you may tailor your own one along with a Dataset class in `model/dataset.py`.
+
+```bash
+# prepare custom dataset up to your need
+# download corresponding dataset first, and fill in the path in scripts
+
+# Prepare the Emilia dataset
+python scripts/prepare_emilia.py
+
+# Prepare the Wenetspeech4TTS dataset
+python scripts/prepare_wenetspeech4tts.py
+```
+
+## Training
+
+Once your datasets are prepared, you can start the training process.
+
+```bash
+# setup accelerate config, e.g. use multi-gpu ddp, fp16
+# will be to: ~/.cache/huggingface/accelerate/default_config.yaml     
+accelerate config
+accelerate launch train.py
+```
+An initial guidance on Finetuning [#57](https://github.com/SWivid/F5-TTS/discussions/57).
+
+## Inference
+
+To run inference with pretrained models, download the checkpoints from [🤗 Hugging Face](https://huggingface.co/SWivid/F5-TTS), or automatically downloaded with `inference-cli` and `gradio_app`.
+
+Currently support 30s for a single generation, which is the **TOTAL** length of prompt audio and the generated. Batch inference with chunks is supported by `inference-cli` and `gradio_app`. 
+- To avoid possible inference failures, make sure you have seen through the following instructions.
+- A longer prompt audio allows shorter generated output. The part longer than 30s cannot be generated properly. Consider using a prompt audio <15s.
+- Uppercased letters will be uttered letter by letter, so use lowercased letters for normal words. 
+- Add some spaces (blank: " ") or punctuations (e.g. "," ".") to explicitly introduce some pauses. If first few words skipped in code-switched generation (cuz different speed with different languages), this might help.
+
+### CLI Inference
+
+Either you can specify everything in `inference-cli.toml` or override with flags. Leave `--ref_text ""` will have ASR model transcribe the reference audio automatically (use extra GPU memory). If encounter network error, consider use local ckpt, just set `ckpt_path` in `inference-cli.py`
+
+```bash
+python inference-cli.py \
+--model "F5-TTS" \
+--ref_audio "tests/ref_audio/test_en_1_ref_short.wav" \
+--ref_text "Some call me nature, others call me mother nature." \
+--gen_text "I don't really care what you call me. I've been a silent spectator, watching species evolve, empires rise and fall. But always remember, I am mighty and enduring. Respect me and I'll nurture you; ignore me and you shall face the consequences."
+
+python inference-cli.py \
+--model "E2-TTS" \
+--ref_audio "tests/ref_audio/test_zh_1_ref_short.wav" \
+--ref_text "对，这就是我，万人敬仰的太乙真人。" \
+--gen_text "突然，身边一阵笑声。我看着他们，意气风发地挺直了胸膛，甩了甩那稍显肉感的双臂，轻笑道，我身上的肉，是为了掩饰我爆棚的魅力，否则，岂不吓坏了你们呢？"
+```
+
+### Gradio App
+Currently supported features:
+- Chunk inference
+- Podcast Generation
+- Multiple Speech-Type Generation
+
+You can launch a Gradio app (web interface) to launch a GUI for inference (will load ckpt from Huggingface, you may set `ckpt_path` to local file in `gradio_app.py`). Currently load ASR model, F5-TTS and E2 TTS all in once, thus use more GPU memory than `inference-cli`.
+
+```bash
+python gradio_app.py
+```
+
+You can specify the port/host:
+
+```bash
+python gradio_app.py --port 7860 --host 0.0.0.0
+```
+
+Or launch a share link:
+
+```bash
+python gradio_app.py --share
+```
+
+### Speech Editing
+
+To test speech editing capabilities, use the following command.
+
+```bash
+python speech_edit.py
+```
+
+## Evaluation
+
+### Prepare Test Datasets
+
+1. Seed-TTS test set: Download from [seed-tts-eval](https://github.com/BytedanceSpeech/seed-tts-eval).
+2. LibriSpeech test-clean: Download from [OpenSLR](http://www.openslr.org/12/).
+3. Unzip the downloaded datasets and place them in the data/ directory.
+4. Update the path for the test-clean data in `scripts/eval_infer_batch.py`
+5. Our filtered LibriSpeech-PC 4-10s subset is already under data/ in this repo
+
+### Batch Inference for Test Set
+
+To run batch inference for evaluations, execute the following commands:
+
+```bash
+# batch inference for evaluations
+accelerate config  # if not set before
+bash scripts/eval_infer_batch.sh
+```
+
+### Download Evaluation Model Checkpoints
+
+1. Chinese ASR Model: [Paraformer-zh](https://huggingface.co/funasr/paraformer-zh)
+2. English ASR Model: [Faster-Whisper](https://huggingface.co/Systran/faster-whisper-large-v3)
+3. WavLM Model: Download from [Google Drive](https://drive.google.com/file/d/1-aE1NfzpRCLxA4GUxX9ITI3F9LlbtEGP/view).
+
+### Objective Evaluation
+
+**Some Notes**
+
+For faster-whisper with CUDA 11:
+
+```bash
+pip install --force-reinstall ctranslate2==3.24.0
+```
+
+(Recommended) To avoid possible ASR failures, such as abnormal repetitions in output:
+
+```bash
+pip install faster-whisper==0.10.1
+```
+
+Update the path with your batch-inferenced results, and carry out WER / SIM evaluations:
+```bash
+# Evaluation for Seed-TTS test set
+python scripts/eval_seedtts_testset.py
+
+# Evaluation for LibriSpeech-PC test-clean (cross-sentence)
+python scripts/eval_librispeech_test_clean.py
+```
+
+## Acknowledgements
+
+- [E2-TTS](https://arxiv.org/abs/2406.18009) brilliant work, simple and effective
+- [Emilia](https://arxiv.org/abs/2407.05361), [WenetSpeech4TTS](https://arxiv.org/abs/2406.05763) valuable datasets
+- [lucidrains](https://github.com/lucidrains) initial CFM structure with also [bfs18](https://github.com/bfs18) for discussion
+- [SD3](https://arxiv.org/abs/2403.03206) & [Hugging Face diffusers](https://github.com/huggingface/diffusers) DiT and MMDiT code structure
+- [torchdiffeq](https://github.com/rtqichen/torchdiffeq) as ODE solver, [Vocos](https://huggingface.co/charactr/vocos-mel-24khz) as vocoder
+- [mrfakename](https://x.com/realmrfakename) huggingface space demo ~
+- [FunASR](https://github.com/modelscope/FunASR), [faster-whisper](https://github.com/SYSTRAN/faster-whisper), [UniSpeech](https://github.com/microsoft/UniSpeech) for evaluation tools
+- [ctc-forced-aligner](https://github.com/MahmoudAshraf97/ctc-forced-aligner) for speech edit test
+
+## Citation
+```
+@article{chen-etal-2024-f5tts,
+      title={F5-TTS: A Fairytaler that Fakes Fluent and Faithful Speech with Flow Matching}, 
+      author={Yushen Chen and Zhikang Niu and Ziyang Ma and Keqi Deng and Chunhui Wang and Jian Zhao and Kai Yu and Xie Chen},
+      journal={arXiv preprint arXiv:2410.06885},
+      year={2024},
+}
+```
+## License
+
+Our code is released under MIT License.

gradio_app.py

@@ -0,0 +1,824 @@
+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
+
+try:
+    import spaces
+    USING_SPACES = True
+except ImportError:
+    USING_SPACES = False
+
+def gpu_decorator(func):
+    if USING_SPACES:
+        return spaces.GPU(func)
+    else:
+        return func
+
+
+
+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,
+)
+vocos = Vocos.from_pretrained("charactr/vocos-mel-24khz")
+
+# --------------------- 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(repo_name, exp_name, model_cls, model_cfg, ckpt_step):
+    ckpt_path = str(cached_path(f"hf://SWivid/{repo_name}/{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(
+    "F5-TTS", "F5TTS_Base", DiT, F5TTS_model_cfg, 1200000
+)
+E2TTS_ema_model = load_model(
+    "E2-TTS", "E2TTS_Base", UNetT, E2TTS_model_cfg, 1200000
+)
+
+def split_text_into_batches(text, max_chars=200, split_words=SPLIT_WORDS):
+    if len(text.encode('utf-8')) <= max_chars:
+        return [text]
+    if text[-1] not in ['。', '.', '!', '！', '?', '？']:
+        text += '.'
+        
+    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 = 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
+        if len(ref_text[-1].encode('utf-8')) == 1:
+            ref_text = ref_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")
+        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
+    audio, sr = torchaudio.load(ref_audio)
+    max_chars = int(len(ref_text.encode('utf-8')) / (audio.shape[-1] / sr) * (30 - audio.shape[-1] / sr))
+    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((audio, sr), 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
+
+def parse_speechtypes_text(gen_text):
+    # Pattern to find (Emotion)
+    pattern = r'\((.*?)\)'
+
+    # Split the text by the pattern
+    tokens = re.split(pattern, gen_text)
+
+    segments = []
+
+    current_emotion = 'Regular'
+
+    for i in range(len(tokens)):
+        if i % 2 == 0:
+            # This is text
+            text = tokens[i].strip()
+            if text:
+                segments.append({'emotion': current_emotion, 'text': text})
+        else:
+            # This is emotion
+            emotion = tokens[i].strip()
+            current_emotion = emotion
+
+    return segments
+
+def update_speed(new_speed):
+    global speed
+    speed = new_speed
+    return f"Speed set to: {speed}"
+
+with gr.Blocks() as app_credits:
+    gr.Markdown("""
+# Credits
+
+* [mrfakename](https://github.com/fakerybakery) for the original [online demo](https://huggingface.co/spaces/mrfakename/E2-F5-TTS)
+* [RootingInLoad](https://github.com/RootingInLoad) for the podcast generation
+""")
+with gr.Blocks() as app_tts:
+    gr.Markdown("# Batched TTS")
+    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,
+        )
+        speed_slider = gr.Slider(
+            label="Speed",
+            minimum=0.3,
+            maximum=2.0,
+            value=speed,
+            step=0.1,
+            info="Adjust the speed of the audio.",
+        )
+    speed_slider.change(update_speed, inputs=speed_slider)
+
+    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],
+    )
+    
+with gr.Blocks() as app_podcast:
+    gr.Markdown("# 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,
+    )
+
+def parse_emotional_text(gen_text):
+    # Pattern to find (Emotion)
+    pattern = r'\((.*?)\)'
+
+    # Split the text by the pattern
+    tokens = re.split(pattern, gen_text)
+
+    segments = []
+
+    current_emotion = 'Regular'
+
+    for i in range(len(tokens)):
+        if i % 2 == 0:
+            # This is text
+            text = tokens[i].strip()
+            if text:
+                segments.append({'emotion': current_emotion, 'text': text})
+        else:
+            # This is emotion
+            emotion = tokens[i].strip()
+            current_emotion = emotion
+
+    return segments
+
+with gr.Blocks() as app_emotional:
+    # New section for emotional generation
+    gr.Markdown(
+        """
+    # Multiple Speech-Type Generation
+
+    This section allows you to upload different audio clips for each speech type. 'Regular' emotion is mandatory. You can add additional speech types by clicking the "Add Speech Type" button. Enter your text in the format shown below, and the system will generate speech using the appropriate emotions. If unspecified, the model will use the regular speech type. The current speech type will be used until the next speech type is specified.
+
+    **Example Input:**
+
+    (Regular) Hello, I'd like to order a sandwich please. (Surprised) What do you mean you're out of bread? (Sad) I really wanted a sandwich though... (Angry) You know what, darn you and your little shop, you suck! (Whisper) I'll just go back home and cry now. (Shouting) Why me?!
+    """
+    )
+
+    gr.Markdown("Upload different audio clips for each speech type. 'Regular' emotion is mandatory. You can add additional speech types by clicking the 'Add Speech Type' button.")
+
+    # Regular speech type (mandatory)
+    with gr.Row():
+        regular_name = gr.Textbox(value='Regular', label='Speech Type Name', interactive=False)
+        regular_audio = gr.Audio(label='Regular Reference Audio', type='filepath')
+        regular_ref_text = gr.Textbox(label='Reference Text (Regular)', lines=2)
+
+    # Additional speech types (up to 9 more)
+    max_speech_types = 10
+    speech_type_names = []
+    speech_type_audios = []
+    speech_type_ref_texts = []
+    speech_type_delete_btns = []
+
+    for i in range(max_speech_types - 1):
+        with gr.Row():
+            name_input = gr.Textbox(label='Speech Type Name', visible=False)
+            audio_input = gr.Audio(label='Reference Audio', type='filepath', visible=False)
+            ref_text_input = gr.Textbox(label='Reference Text', lines=2, visible=False)
+            delete_btn = gr.Button("Delete", variant="secondary", visible=False)
+        speech_type_names.append(name_input)
+        speech_type_audios.append(audio_input)
+        speech_type_ref_texts.append(ref_text_input)
+        speech_type_delete_btns.append(delete_btn)
+
+    # Button to add speech type
+    add_speech_type_btn = gr.Button("Add Speech Type")
+
+    # Keep track of current number of speech types
+    speech_type_count = gr.State(value=0)
+
+    # Function to add a speech type
+    def add_speech_type_fn(speech_type_count):
+        if speech_type_count < max_speech_types - 1:
+            speech_type_count += 1
+            # Prepare updates for the components
+            name_updates = []
+            audio_updates = []
+            ref_text_updates = []
+            delete_btn_updates = []
+            for i in range(max_speech_types - 1):
+                if i < speech_type_count:
+                    name_updates.append(gr.update(visible=True))
+                    audio_updates.append(gr.update(visible=True))
+                    ref_text_updates.append(gr.update(visible=True))
+                    delete_btn_updates.append(gr.update(visible=True))
+                else:
+                    name_updates.append(gr.update())
+                    audio_updates.append(gr.update())
+                    ref_text_updates.append(gr.update())
+                    delete_btn_updates.append(gr.update())
+        else:
+            # Optionally, show a warning
+            # gr.Warning("Maximum number of speech types reached.")
+            name_updates = [gr.update() for _ in range(max_speech_types - 1)]
+            audio_updates = [gr.update() for _ in range(max_speech_types - 1)]
+            ref_text_updates = [gr.update() for _ in range(max_speech_types - 1)]
+            delete_btn_updates = [gr.update() for _ in range(max_speech_types - 1)]
+        return [speech_type_count] + name_updates + audio_updates + ref_text_updates + delete_btn_updates
+
+    add_speech_type_btn.click(
+        add_speech_type_fn,
+        inputs=speech_type_count,
+        outputs=[speech_type_count] + speech_type_names + speech_type_audios + speech_type_ref_texts + speech_type_delete_btns
+    )
+
+    # Function to delete a speech type
+    def make_delete_speech_type_fn(index):
+        def delete_speech_type_fn(speech_type_count):
+            # Prepare updates
+            name_updates = []
+            audio_updates = []
+            ref_text_updates = []
+            delete_btn_updates = []
+
+            for i in range(max_speech_types - 1):
+                if i == index:
+                    name_updates.append(gr.update(visible=False, value=''))
+                    audio_updates.append(gr.update(visible=False, value=None))
+                    ref_text_updates.append(gr.update(visible=False, value=''))
+                    delete_btn_updates.append(gr.update(visible=False))
+                else:
+                    name_updates.append(gr.update())
+                    audio_updates.append(gr.update())
+                    ref_text_updates.append(gr.update())
+                    delete_btn_updates.append(gr.update())
+
+            speech_type_count = max(0, speech_type_count - 1)
+
+            return [speech_type_count] + name_updates + audio_updates + ref_text_updates + delete_btn_updates
+
+        return delete_speech_type_fn
+
+    for i, delete_btn in enumerate(speech_type_delete_btns):
+        delete_fn = make_delete_speech_type_fn(i)
+        delete_btn.click(
+            delete_fn,
+            inputs=speech_type_count,
+            outputs=[speech_type_count] + speech_type_names + speech_type_audios + speech_type_ref_texts + speech_type_delete_btns
+        )
+
+    # Text input for the prompt
+    gen_text_input_emotional = gr.Textbox(label="Text to Generate", lines=10)
+
+    # Model choice
+    model_choice_emotional = gr.Radio(
+        choices=["F5-TTS", "E2-TTS"], label="Choose TTS Model", value="F5-TTS"
+    )
+
+    with gr.Accordion("Advanced Settings", open=False):
+        remove_silence_emotional = gr.Checkbox(
+            label="Remove Silences",
+            value=True,
+        )
+
+    # Generate button
+    generate_emotional_btn = gr.Button("Generate Emotional Speech", variant="primary")
+
+    # Output audio
+    audio_output_emotional = gr.Audio(label="Synthesized Audio")
+
+    def generate_emotional_speech(
+        regular_audio,
+        regular_ref_text,
+        gen_text,
+        *args,
+    ):
+        num_additional_speech_types = max_speech_types - 1
+        speech_type_names_list = args[:num_additional_speech_types]
+        speech_type_audios_list = args[num_additional_speech_types:2 * num_additional_speech_types]
+        speech_type_ref_texts_list = args[2 * num_additional_speech_types:3 * num_additional_speech_types]
+        model_choice = args[3 * num_additional_speech_types]
+        remove_silence = args[3 * num_additional_speech_types + 1]
+
+        # Collect the speech types and their audios into a dict
+        speech_types = {'Regular': {'audio': regular_audio, 'ref_text': regular_ref_text}}
+
+        for name_input, audio_input, ref_text_input in zip(speech_type_names_list, speech_type_audios_list, speech_type_ref_texts_list):
+            if name_input and audio_input:
+                speech_types[name_input] = {'audio': audio_input, 'ref_text': ref_text_input}
+
+        # Parse the gen_text into segments
+        segments = parse_speechtypes_text(gen_text)
+
+        # For each segment, generate speech
+        generated_audio_segments = []
+        current_emotion = 'Regular'
+
+        for segment in segments:
+            emotion = segment['emotion']
+            text = segment['text']
+
+            if emotion in speech_types:
+                current_emotion = emotion
+            else:
+                # If emotion not available, default to Regular
+                current_emotion = 'Regular'
+
+            ref_audio = speech_types[current_emotion]['audio']
+            ref_text = speech_types[current_emotion].get('ref_text', '')
+
+            # Generate speech for this segment
+            audio, _ = infer(ref_audio, ref_text, text, model_choice, remove_silence, "")
+            sr, audio_data = audio
+
+            generated_audio_segments.append(audio_data)
+
+        # Concatenate all audio segments
+        if generated_audio_segments:
+            final_audio_data = np.concatenate(generated_audio_segments)
+            return (sr, final_audio_data)
+        else:
+            gr.Warning("No audio generated.")
+            return None
+
+    generate_emotional_btn.click(
+        generate_emotional_speech,
+        inputs=[
+            regular_audio,
+            regular_ref_text,
+            gen_text_input_emotional,
+        ] + speech_type_names + speech_type_audios + speech_type_ref_texts + [
+            model_choice_emotional,
+            remove_silence_emotional,
+        ],
+        outputs=audio_output_emotional,
+    )
+
+    # Validation function to disable Generate button if speech types are missing
+    def validate_speech_types(
+        gen_text,
+        regular_name,
+        *args
+    ):
+        num_additional_speech_types = max_speech_types - 1
+        speech_type_names_list = args[:num_additional_speech_types]
+
+        # Collect the speech types names
+        speech_types_available = set()
+        if regular_name:
+            speech_types_available.add(regular_name)
+        for name_input in speech_type_names_list:
+            if name_input:
+                speech_types_available.add(name_input)
+
+        # Parse the gen_text to get the speech types used
+        segments = parse_emotional_text(gen_text)
+        speech_types_in_text = set(segment['emotion'] for segment in segments)
+
+        # Check if all speech types in text are available
+        missing_speech_types = speech_types_in_text - speech_types_available
+
+        if missing_speech_types:
+            # Disable the generate button
+            return gr.update(interactive=False)
+        else:
+            # Enable the generate button
+            return gr.update(interactive=True)
+
+    gen_text_input_emotional.change(
+        validate_speech_types,
+        inputs=[gen_text_input_emotional, regular_name] + speech_type_names,
+        outputs=generate_emotional_btn
+    )
+with gr.Blocks() as app:
+    gr.Markdown(
+        """
+# E2/F5 TTS
+
+This is a local web UI for F5 TTS with advanced batch processing support. 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.**
+"""
+    )
+    gr.TabbedInterface([app_tts, app_podcast, app_emotional, app_credits], ["TTS", "Podcast", "Multi-Style", "Credits"])
+
+@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()