Orpheus-TTS / app.py
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import spaces
from snac import SNAC
import torch
import gradio as gr
from transformers import AutoModelForCausalLM, AutoTokenizer
from huggingface_hub import snapshot_download
from dotenv import load_dotenv
load_dotenv()
# Check if CUDA is available
device = "cuda" if torch.cuda.is_available() else "cpu"
print("Loading SNAC model...")
snac_model = SNAC.from_pretrained("hubertsiuzdak/snac_24khz")
snac_model = snac_model.to(device)
model_name = "Vyvo/VyvoTTS-LFM2-Multi-Speaker"
# Download only model config and safetensors
snapshot_download(
repo_id=model_name,
allow_patterns=[
"config.json",
"*.safetensors",
"model.safetensors.index.json",
],
ignore_patterns=[
"optimizer.pt",
"pytorch_model.bin",
"training_args.bin",
"scheduler.pt",
"tokenizer.json",
"tokenizer_config.json",
"special_tokens_map.json",
"vocab.json",
"merges.txt",
"tokenizer.*"
]
)
model = AutoModelForCausalLM.from_pretrained(model_name, torch_dtype=torch.bfloat16)
model.to(device)
tokenizer = AutoTokenizer.from_pretrained(model_name)
print(f"Model loaded to {device}")
# LFM2 Special Tokens Configuration (Sizin doğru değerleriniz)
TOKENIZER_LENGTH = 64400
START_OF_TEXT = 1
END_OF_TEXT = 7
START_OF_SPEECH = TOKENIZER_LENGTH + 1
END_OF_SPEECH = TOKENIZER_LENGTH + 2
START_OF_HUMAN = TOKENIZER_LENGTH + 3
END_OF_HUMAN = TOKENIZER_LENGTH + 4
START_OF_AI = TOKENIZER_LENGTH + 5
END_OF_AI = TOKENIZER_LENGTH + 6
PAD_TOKEN = TOKENIZER_LENGTH + 7
AUDIO_TOKENS_START = TOKENIZER_LENGTH + 10
# Process text prompt (Sizin doğru formatınız)
def process_prompt(prompt, voice, tokenizer, device):
prompt = f"{voice}: {prompt}"
input_ids = tokenizer(prompt, return_tensors="pt").input_ids
start_token = torch.tensor([[START_OF_HUMAN]], dtype=torch.int64)
end_tokens = torch.tensor([[END_OF_TEXT, END_OF_HUMAN]], dtype=torch.int64)
modified_input_ids = torch.cat([start_token, input_ids, end_tokens], dim=1)
# No padding needed for single input
attention_mask = torch.ones_like(modified_input_ids)
return modified_input_ids.to(device), attention_mask.to(device)
# Parse output tokens to audio (Sizin doğru formatınız)
def parse_output(generated_ids):
token_to_find = START_OF_SPEECH
token_to_remove = END_OF_SPEECH
token_indices = (generated_ids == token_to_find).nonzero(as_tuple=True)
if len(token_indices[1]) > 0:
last_occurrence_idx = token_indices[1][-1].item()
cropped_tensor = generated_ids[:, last_occurrence_idx+1:]
else:
cropped_tensor = generated_ids
processed_rows = []
for row in cropped_tensor:
masked_row = row[row != token_to_remove]
processed_rows.append(masked_row)
code_lists = []
for row in processed_rows:
row_length = row.size(0)
new_length = (row_length // 7) * 7
trimmed_row = row[:new_length]
trimmed_row = [t - AUDIO_TOKENS_START for t in trimmed_row]
code_lists.append(trimmed_row)
return code_lists[0] # Return just the first one for single sample
# Redistribute codes for audio generation (Aynı kalıyor)
def redistribute_codes(code_list, snac_model):
device = next(snac_model.parameters()).device # Get the device of SNAC model
layer_1 = []
layer_2 = []
layer_3 = []
for i in range((len(code_list)+1)//7):
layer_1.append(code_list[7*i])
layer_2.append(code_list[7*i+1]-4096)
layer_3.append(code_list[7*i+2]-(2*4096))
layer_3.append(code_list[7*i+3]-(3*4096))
layer_2.append(code_list[7*i+4]-(4*4096))
layer_3.append(code_list[7*i+5]-(5*4096))
layer_3.append(code_list[7*i+6]-(6*4096))
# Move tensors to the same device as the SNAC model
codes = [
torch.tensor(layer_1, device=device).unsqueeze(0),
torch.tensor(layer_2, device=device).unsqueeze(0),
torch.tensor(layer_3, device=device).unsqueeze(0)
]
audio_hat = snac_model.decode(codes)
return audio_hat.detach().squeeze().cpu().numpy() # Always return CPU numpy array
# Main generation function
@spaces.GPU()
def generate_speech(text, voice, temperature, top_p, repetition_penalty, max_new_tokens, progress=gr.Progress()):
if not text.strip():
return None
try:
progress(0.1, f"Processing text with {voice} voice...")
input_ids, attention_mask = process_prompt(text, voice, tokenizer, device)
progress(0.3, "Generating speech tokens...")
with torch.no_grad():
generated_ids = model.generate(
input_ids=input_ids,
attention_mask=attention_mask,
max_new_tokens=max_new_tokens,
do_sample=True,
temperature=temperature,
top_p=top_p,
repetition_penalty=repetition_penalty,
num_return_sequences=1,
eos_token_id=END_OF_SPEECH, # Doğru EOS token
)
progress(0.6, "Processing speech tokens...")
code_list = parse_output(generated_ids)
progress(0.8, "Converting to audio...")
audio_samples = redistribute_codes(code_list, snac_model)
progress(1.0, f"✅ Completed with {voice}!")
return (24000, audio_samples) # Return sample rate and audio
except Exception as e:
print(f"Error generating speech: {e}")
return None
# Examples for the UI - Genshin karakterleri ile
examples = [
["Hey there! I am ready to help you on your adventure in Teyvat.", "Tighnari", 0.6, 0.95, 1.1, 1200],
["The wind brings new adventures and ancient secrets to discover.", "Kaeya", 0.7, 0.95, 1.1, 1200],
["Let me share the wisdom of the elements with you, traveler.", "Nahida", 0.6, 0.9, 1.2, 1200],
["Every journey begins with a single step forward into the unknown.", "Noelle", 0.65, 0.9, 1.1, 1200],
["The stars above guide us through even the darkest of nights.", "Furina", 0.7, 0.95, 1.1, 1200],
["Together we can explore the mysteries of this vast world.", "Lyney", 0.65, 0.9, 1.15, 1200],
["Knowledge is power, but wisdom is knowing how to use it.", "Alhaitham", 0.7, 0.95, 1.1, 1200],
["The beauty of Sumeru never fails to take my breath away.", "Collei", 0.6, 0.95, 1.1, 1200]
]
# Available voices - Genshin karakterleri ve diğerleri
VOICES = [
"Stephen_Fry",
"Tighnari",
"Thoma",
"Shikanoin_Heizou",
"Noelle",
"Ningguang",
"Nilou",
"Neuvillette",
"Navia",
"Nahida",
"Mualani",
"Lyney",
"Lynette",
"Layla",
"Kaveh",
"Kaeya",
"Furina",
"Dehya",
"Cyno",
"Collei",
"Beidou",
"Alhaitham",
"Arataki_Itto",
"Jenny_Voice",
"Optimus_Prime"
]
# Available Emotive Tags
EMOTIVE_TAGS = ["`<laugh>`", "`<chuckle>`", "`<sigh>`", "`<cough>`", "`<sniffle>`", "`<groan>`", "`<yawn>`", "`<gasp>`"]
# Create Gradio interface
with gr.Blocks(title="VyvoTTS Multi-Speaker") as demo:
gr.Markdown(f"""
# 🎮 VyvoTTS Multi-Speaker
VyvoTTS is a text-to-speech model by Vyvo team using LFM2 architecture, trained on multiple diverse open-source datasets.
Since some datasets may contain transcription errors or quality issues, output quality can vary.
Higher quality datasets typically produce better speech synthesis results.
**Available Character Voices:**
🌟 Genshin Impact: Tighnari, Thoma, Heizou, Noelle, Ningguang, Nilou, Neuvillette, Navia, Nahida, Mualani, Lyney, Lynette, Layla, Kaveh, Kaeya, Furina, Dehya, Cyno, Collei, Beidou, Alhaitham, Itto
🎭 Others: Stephen Fry, Jenny Voice, Optimus Prime
## Tips for better prompts:
- Add paralinguistic elements like {", ".join(EMOTIVE_TAGS)} or `uhm` for more human-like speech.
- Longer text prompts generally work better than very short phrases
- Increasing `repetition_penalty` and `temperature` makes the model speak faster.
**Note:** Output quality may vary depending on the source dataset quality for each character voice.
""")
with gr.Row():
with gr.Column(scale=3):
text_input = gr.Textbox(
label="Text to speak",
placeholder="Enter your text here...",
lines=5
)
voice = gr.Dropdown(
choices=VOICES,
value="Tighnari",
label="Character Voice"
)
with gr.Accordion("Advanced Settings", open=False):
temperature = gr.Slider(
minimum=0.1, maximum=1.5, value=0.6, step=0.05,
label="Temperature",
info="Higher values (0.7-1.0) create more expressive but less stable speech"
)
top_p = gr.Slider(
minimum=0.1, maximum=1.0, value=0.95, step=0.05,
label="Top P",
info="Nucleus sampling threshold"
)
repetition_penalty = gr.Slider(
minimum=1.0, maximum=2.0, value=1.1, step=0.05,
label="Repetition Penalty",
info="Higher values discourage repetitive patterns"
)
max_new_tokens = gr.Slider(
minimum=100, maximum=2000, value=1200, step=100,
label="Max Length",
info="Maximum length of generated audio (in tokens)"
)
with gr.Row():
submit_btn = gr.Button("Generate Speech", variant="primary")
clear_btn = gr.Button("Clear")
with gr.Column(scale=2):
audio_output = gr.Audio(label="Generated Speech", type="numpy")
# Set up examples
gr.Examples(
examples=examples,
inputs=[text_input, voice, temperature, top_p, repetition_penalty, max_new_tokens],
outputs=audio_output,
fn=generate_speech,
cache_examples=True,
)
# Set up event handlers
submit_btn.click(
fn=generate_speech,
inputs=[text_input, voice, temperature, top_p, repetition_penalty, max_new_tokens],
outputs=audio_output,
show_progress=True
)
clear_btn.click(
fn=lambda: (None, None),
inputs=[],
outputs=[text_input, audio_output]
)
# Launch the app
if __name__ == "__main__":
demo.queue().launch(share=False, ssr_mode=False)