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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): | |
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 = 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 | |
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") | |
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) and len(gen_text.encode('utf-8')) == len(gen_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) supported by [mrfakename](https://github.com/fakerybakery). 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, | |
) | |
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() | |