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# ComfyUI-Orpheus Node
2 custom nodes for ComfyUI that enables text-to-speech generation using the GGUF [Orpheus](https://github.com/canopyai/Orpheus-TTS) model with emotional speech capabilities.
<img src="doc/demo.png" width="100%">
## Features
- High-quality text-to-speech synthesis
- Multiple voice options (24 different voices, depend language used) in **English, French, Spanish, Italian, Chinese, Korean, German, Hindi**
- Emotional speech capabilities
- Seamless integration with ComfyUI workflow
## Available Voices
- **English Voices**:
supported tags : chuckle, cough, gasp, groan, laugh, sigh, sniffle, yawn
- `tara` - Female voice
- `leah` - Female voice
- `jess` - Female voice
- `leo` - Male voice
- `dan` - Male voice
- `mia` - Female voice
- `zac` - Male voice
- `zoe` - Female voice
- **French Voices**:
supported tags : chuckle, cough, gasp, groan, laugh, sigh, sniffle, whimper, yawn
- `pierre` - Male voice
- `amelie` - Female voice
- `marie` - Female voice (doesn't works well)
- **German Voices**:
supported tags : chuckle, cough, gasp, groan, laugh, sigh, sniffle, yawn
- `jana` - Female voice
- `thomas` - Male voice
- `max` - Male voice
- **Korean Voices**:
supported tags : 한숨, 헐, 헛기침, 훌쩍, 하품, 낄낄, 신음, 작은 웃음, 기침, 으르렁
- `유나` - ?^^
- `준서` - ?^^
- **Chinese Voices**:
supported tags : 嬉笑, 轻笑, 呻吟, 大笑, 咳嗽, 抽鼻子, 咳
- `长乐` - ?^^
- `白芷` - ?^^
- **Hindi**:
supported tags : unknow
- `ऋतिका` - ? ^^
- **Spanish Voices**:
supported tags : groan, chuckle, gasp, resoplido, laugh, yawn, cough
- `javi` - Male voice
- `sergio` - Male voice
- `maria` - Female voice
- **Italian Voices**:
supported tags : sigh, laugh, cough, sniffle, groan, yawn, gemito, gasp
- `pietro` - Male voice
- `giulia` - Female voice
- `carlo` - Male voice
## Requirements
- Last ComfyUI version with python 3.12.9 (may be works with older versions but I haven't test it)
## Installation
1. Clone this repository into your ComfyUI custom nodes directory:
```bash
cd ComfyUI/custom_nodes
git clone https://github.com/your-repo/ComfyUI-Orpheus.git
```
2. Install the required dependencies:
load venv and :
```bash
pip install -r ComfyUI-Orpheus/requirements.txt
```
use python_embeded :
```bash
python_embeded\python.exe -m pip install -r ComfyUI-Orpheus/requirements.txt
```
3. Download the required [GGUF model](https://huggingface.co/freddyaboulton) from [FreddyAboulton](https://huggingface.co/freddyaboulton) huggingface page, and place it in your ComfyUI models directory under `models/unet/`. (Sorry I don't know where to find Italian one)
<img src="doc/models.png" width="100%">
4. **GPU Support**
On windows, default installation of llama-cpp-python doesn't take GPU support. If you want GPU Support you need
to locate `nvcc.exe` folder and:
```bash
set CMAKE_ARGS="-DGGML_CUDA=on"
set CUDA_CXX="YOUR_CUDA_DIR\v12.6.3\bin\nvcc.exe"
python_embeded\python.exe -m pip install llama-cpp-python[server] --upgrade --force-reinstall --no-cache-dir
```
Be patient, it takes time...
## Usage
1. In ComfyUI, locate the "Orpheus ⛓️" node in the node menu.
2. Configure the node parameters:
- `model_name`: Select your GGUF model
- `voice`: Choose from available voices
- `prompt`: Enter the text you want to convert to speech, you can add emotive tags :\<laugh>, \<chuckle>, \<sigh>, \<cough>, \<sniffle>, \<groan>, \<yawn>, \<gasp>
3. Connect the node outputs:
- `audio`: Contains the generated audio waveform and sample rate
## Limitations
- Maximum text length determined by MAX_TOKENS
- Processing speed depends on GPU capabilities
- Requires CUDA support for optimal performance
## Credits
- Original [Orpheus](https://github.com/canopyai/Orpheus-TTS) implementation
- [Freddy Aboulton](https://huggingface.co/freddyaboulton)
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from .node.orpheus import NODE_CLASS_MAPPINGS, NODE_DISPLAY_NAME_MAPPINGS
__all__ = ["NODE_CLASS_MAPPINGS", "NODE_DISPLAY_NAME_MAPPINGS"]
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from snac import SNAC
import numpy as np
import torch
import asyncio
import threading
import queue
model = SNAC.from_pretrained("hubertsiuzdak/snac_24khz").eval()
snac_device = "cuda"
model = model.to(snac_device)
def convert_to_audio(multiframe, count):
frames = []
if len(multiframe) < 7:
return
codes_0 = torch.tensor([], device=snac_device, dtype=torch.int32)
codes_1 = torch.tensor([], device=snac_device, dtype=torch.int32)
codes_2 = torch.tensor([], device=snac_device, dtype=torch.int32)
num_frames = len(multiframe) // 7
frame = multiframe[:num_frames*7]
for j in range(num_frames):
i = 7*j
if codes_0.shape[0] == 0:
codes_0 = torch.tensor([frame[i]], device=snac_device, dtype=torch.int32)
else:
codes_0 = torch.cat([codes_0, torch.tensor([frame[i]], device=snac_device, dtype=torch.int32)])
if codes_1.shape[0] == 0:
codes_1 = torch.tensor([frame[i+1]], device=snac_device, dtype=torch.int32)
codes_1 = torch.cat([codes_1, torch.tensor([frame[i+4]], device=snac_device, dtype=torch.int32)])
else:
codes_1 = torch.cat([codes_1, torch.tensor([frame[i+1]], device=snac_device, dtype=torch.int32)])
codes_1 = torch.cat([codes_1, torch.tensor([frame[i+4]], device=snac_device, dtype=torch.int32)])
if codes_2.shape[0] == 0:
codes_2 = torch.tensor([frame[i+2]], device=snac_device, dtype=torch.int32)
codes_2 = torch.cat([codes_2, torch.tensor([frame[i+3]], device=snac_device, dtype=torch.int32)])
codes_2 = torch.cat([codes_2, torch.tensor([frame[i+5]], device=snac_device, dtype=torch.int32)])
codes_2 = torch.cat([codes_2, torch.tensor([frame[i+6]], device=snac_device, dtype=torch.int32)])
else:
codes_2 = torch.cat([codes_2, torch.tensor([frame[i+2]], device=snac_device, dtype=torch.int32)])
codes_2 = torch.cat([codes_2, torch.tensor([frame[i+3]], device=snac_device, dtype=torch.int32)])
codes_2 = torch.cat([codes_2, torch.tensor([frame[i+5]], device=snac_device, dtype=torch.int32)])
codes_2 = torch.cat([codes_2, torch.tensor([frame[i+6]], device=snac_device, dtype=torch.int32)])
codes = [codes_0.unsqueeze(0), codes_1.unsqueeze(0), codes_2.unsqueeze(0)]
# check that all tokens are between 0 and 4096 otherwise return *
if torch.any(codes[0] < 0) or torch.any(codes[0] > 4096) or torch.any(codes[1] < 0) or torch.any(codes[1] > 4096) or torch.any(codes[2] < 0) or torch.any(codes[2] > 4096):
return
with torch.inference_mode():
audio_hat = model.decode(codes)
audio_slice = audio_hat[:, :, 2048:4096]
detached_audio = audio_slice.detach().cpu()
audio_np = detached_audio.numpy()
audio_int16 = (audio_np * 32767).astype(np.int16)
audio_bytes = audio_int16.tobytes()
return audio_bytes
def turn_token_into_id(token_string, index):
# Strip whitespace
token_string = token_string.strip()
# Find the last token in the string
last_token_start = token_string.rfind("<custom_token_")
if last_token_start == -1:
print("No token found in the string")
return None
# Extract the last token
last_token = token_string[last_token_start:]
# Process the last token
if last_token.startswith("<custom_token_") and last_token.endswith(">"):
try:
number_str = last_token[14:-1]
return int(number_str) - 10 - ((index % 7) * 4096)
except ValueError:
return None
else:
return None
async def tokens_decoder(token_gen):
buffer = []
count = 0
async for token_sim in token_gen:
token = turn_token_into_id(token_sim, count)
if token is None:
pass
else:
if token > 0:
buffer.append(token)
count += 1
if count % 7 == 0 and count > 27:
buffer_to_proc = buffer[-28:]
audio_samples = convert_to_audio(buffer_to_proc, count)
if audio_samples is not None:
yield audio_samples
# ------------------ Synchronous Tokens Decoder Wrapper ------------------ #
def tokens_decoder_sync(syn_token_gen):
audio_queue = queue.Queue()
# Convert the synchronous token generator into an async generator.
async def async_token_gen():
for token in syn_token_gen:
yield token
async def async_producer():
# tokens_decoder.tokens_decoder is assumed to be an async generator that processes tokens.
async for audio_chunk in tokens_decoder(async_token_gen()):
audio_queue.put(audio_chunk)
audio_queue.put(None) # Sentinel
def run_async():
asyncio.run(async_producer())
thread = threading.Thread(target=run_async)
thread.start()
while True:
audio = audio_queue.get()
if audio is None:
break
yield audio
thread.join()
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import os
import time
import logging
import wave
import folder_paths
import hashlib
import torchaudio
from .decoder import convert_to_audio as orpheus_convert_to_audio
from llama_cpp import Llama
def update_folder_names_and_paths(key, targets=[]):
# check for existing key
base = folder_paths.folder_names_and_paths.get(key, ([], {}))
base = base[0] if isinstance(base[0], (list, set, tuple)) else []
# find base key & add w/ fallback, sanity check + warning
target = next((x for x in targets if x in folder_paths.folder_names_and_paths), targets[0])
orig, _ = folder_paths.folder_names_and_paths.get(target, ([], {}))
folder_paths.folder_names_and_paths[key] = (orig or base, {".gguf"})
if base and base != orig:
logging.warning(f"Unknown file list already present on key {key}: {base}")
# Add a custom keys for files ending in .gguf
update_folder_names_and_paths("unet_gguf", ["diffusion_models", "unet"])
BOOLEAN = ("BOOLEAN", {"default": True})
STRING = ("STRING", {"default": ""})
# Model parameters
MAX_TOKENS = 8192
TEMPERATURE = 0.6
TOP_P = 0.9
REPETITION_PENALTY = 1.1
SAMPLE_RATE = 24000 # SNAC model uses 24kHz
# Available voices based on the Orpheus-TTS repository
AVAILABLE_VOICES = ["tara", "leah", "jess", "leo", "dan", "mia", "zac", "zoe", "pierre", "amelie", "marie","jana", "thomas", "max", "유나", "준서", "长乐", "白芷" "javi", "sergio", "maria", "pietro", "giulia", "carlo"]
DEFAULT_VOICE = "pierre" # Best voice according to documentation
CUSTOM_TOKEN_PREFIX = "<custom_token_"
def format_prompt(prompt, voice=DEFAULT_VOICE):
"""Format prompt for Orpheus model with voice prefix and special tokens."""
if voice not in AVAILABLE_VOICES:
print(f"Warning: Voice '{voice}' not recognized. Using '{DEFAULT_VOICE}' instead.")
voice = DEFAULT_VOICE
# Format similar to how engine_class.py does it with special tokens
formatted_prompt = f"{voice}: {prompt}"
# Add special token markers for the LM Studio API
special_start = "<|audio|>" # Using the additional_special_token from config
special_end = "<|eot_id|>" # Using the eos_token from config
return f"{special_start}{formatted_prompt}{special_end}"
def turn_token_into_id(token_string, index):
"""Convert token string to numeric ID for audio processing."""
# Strip whitespace
token_string = token_string.strip()
# Find the last token in the string
last_token_start = token_string.rfind(CUSTOM_TOKEN_PREFIX)
if last_token_start == -1:
return None
# Extract the last token
last_token = token_string[last_token_start:]
# Process the last token
if last_token.startswith(CUSTOM_TOKEN_PREFIX) and last_token.endswith(">"):
#print(f"Last token: {last_token}")
try:
number_str = last_token[14:-1]
# print(f"Number string: {number_str}")
token_id = int(number_str) - 10 - ((index % 7) * 4096)
# print(f"Token ID: {token_id}")
return token_id
except ValueError:
return None
else:
return None
def convert_to_audio(multiframe, count):
"""Convert token frames to audio."""
# Import here to avoid circular imports
return orpheus_convert_to_audio(multiframe, count)
def my_tokens_decoder(token_gen):
"""Asynchronous token decoder that converts token stream to audio stream."""
buffer = []
count = 0
audio_samples_buffer = []
#print(token_gen)
#tgen = extract_custom_tokens(token_gen)
for token_text in token_gen:
#print("Token text:", token_text)
token = turn_token_into_id(token_text, count)
#print("Token ID:", token)
if token is not None and token > 0:
buffer.append(token)
count += 1
# Convert to audio when we have enough tokens
if count % 7 == 0 and count > 27:
buffer_to_proc = buffer[-28:]
audio_samples = convert_to_audio(buffer_to_proc, count)
if audio_samples is not None:
audio_samples_buffer.append(audio_samples)
return audio_samples_buffer
def extract_custom_tokens(token_string):
"""
Extract all custom tokens from a string and return them as an array.
Example: "<custom_token_465456><custom_token_7867>" -> ["<custom_token_465456>", "<custom_token_7867>"]
"""
#print(token_string)
tokens = []
i = 0
while i < len(token_string):
# Find the start of a custom token
start_pos = token_string.find(CUSTOM_TOKEN_PREFIX, i)
if start_pos == -1:
break
# Find the end of this token
end_pos = token_string.find(">", start_pos)
if end_pos == -1:
break
# Extract the complete token
token = token_string[start_pos:end_pos+1]
tokens.append(token)
# Move past this token
i = end_pos + 1
return tokens
def tokens_decoder_sync(syn_token_gen, output_file=None):
wav_file = None
if output_file:
# Create directory if it doesn't exist
os.makedirs(os.path.dirname(os.path.abspath(output_file)), exist_ok=True)
wav_file = wave.open(output_file, "wb")
wav_file.setnchannels(1)
wav_file.setsampwidth(2)
wav_file.setframerate(SAMPLE_RATE)
def my_producer(syn_token_gen):
return my_tokens_decoder(extract_custom_tokens(syn_token_gen))
tokens = my_producer(syn_token_gen)
for audio_chunk in tokens:
wav_file.writeframes(audio_chunk)
if wav_file:
wav_file.close()
def load_model(model_path):
llm = Llama(
model_path=model_path,
n_ctx=4096, # Context length to use
n_threads=12, # Number of CPU threads to use
n_gpu_layers=32,
verbose=False
)
return llm
class orpheus:
def __init__(self):
pass
@classmethod
def INPUT_TYPES(cls):
unet_names = [x for x in folder_paths.get_filename_list("unet_gguf")]
#print(unet_names)
return {
"required": {
"model_name": (unet_names, ),
"voice": (AVAILABLE_VOICES,),
"prompt": ("STRING", {"default": "Hello, I am Orpheus, an AI assistant with emotional speech capabilities.","multiline": True})
},
}
RETURN_TYPES = ("AUDIO",)
RETURN_NAMES = ("audio",)
FUNCTION = "execute"
CATEGORY = "Orpheus ⛓️"
@classmethod
def IS_CHANGED(s, model_name, voice,prompt, **kwargs):
m = hashlib.sha256()
m.update(model_name.encode() + voice.encode() + prompt.encode())
return m.digest().hex()
def execute(self, model_name, voice, prompt, **kwargs):
timestamp = time.strftime("%Y%m%d_%H%M%S")
output_file = os.path.join(folder_paths.get_output_directory(), f"orpheus_{voice}_{timestamp}.wav")
model_path = folder_paths.get_full_path("unet_gguf", model_name)
#print(model_path)
llm = load_model(model_path)
prompt = format_prompt(prompt, voice=voice)
generation_kwargs = {
"max_tokens": MAX_TOKENS,
"temperature": TEMPERATURE,
"top_p": TOP_P,
"repeat_penalty": REPETITION_PENALTY
}
res = llm(prompt, **generation_kwargs) # Res is a dictionary
tokens_decoder_sync(res['choices'][0]['text'], output_file)
# open file as a tensor
waveform, sample_rate = torchaudio.load(output_file)
audio = {"waveform": waveform.unsqueeze(0), "sample_rate": sample_rate}
return (audio, )
class orpheusAdvanced:
def __init__(self):
pass
@classmethod
def INPUT_TYPES(cls):
unet_names = [x for x in folder_paths.get_filename_list("unet_gguf")]
#print(unet_names)
return {
"required": {
"model_name": (unet_names, ),
"voice": (AVAILABLE_VOICES,),
"prompt": ("STRING", {"default": "Hello, I am Orpheus, an AI assistant with emotional speech capabilities.","multiline": True}),
"max_tokens" :("INT", {"default": 8192, "min": 4096, "max": 131072, "step": 1}),
"temperature" :("FLOAT", {"default": 0.6, "min": 0.0, "max": 1.0, "step": 0.01}),
"top_p" :("FLOAT", {"default": 0.9, "min": 0.0, "max": 1.0, "step": 0.01}),
"repeat_penalty" :("FLOAT", {"default": 1.1, "min": 0.0, "max": 1.5, "step": 0.01}),
},
}
RETURN_TYPES = ("AUDIO",)
RETURN_NAMES = ("audio",)
FUNCTION = "execute"
CATEGORY = "Orpheus ⛓️"
@classmethod
def IS_CHANGED(s, model_name, voice,prompt, max_tokens, temperature, top_p, repeat_penalty, **kwargs):
m = hashlib.sha256()
m.update(model_name.encode() + voice.encode() + prompt.encode() + str(max_tokens).encode() + str(temperature).encode() + str(top_p).encode() + str(repeat_penalty).encode())
return m.digest().hex()
def execute(self, model_name, voice, prompt, max_tokens, temperature, top_p, repeat_penalty, **kwargs):
timestamp = time.strftime("%Y%m%d_%H%M%S")
output_file = os.path.join(folder_paths.get_output_directory(), f"orpheus_{voice}_{timestamp}.wav")
model_path = folder_paths.get_full_path("unet_gguf", model_name)
#print(model_path)
llm = load_model(model_path)
prompt = format_prompt(prompt, voice=voice)
generation_kwargs = {
"max_tokens": max_tokens,
"temperature": temperature,
"top_p": top_p,
"repeat_penalty": repeat_penalty
}
res = llm(prompt, **generation_kwargs) # Res is a dictionary
tokens_decoder_sync(res['choices'][0]['text'], output_file)
# open file as a tensor
waveform, sample_rate = torchaudio.load(output_file)
audio = {"waveform": waveform.unsqueeze(0), "sample_rate": sample_rate}
return (audio, )
NODE_CLASS_MAPPINGS = {
"orpheus": orpheus,
"orpheusAdvanced": orpheusAdvanced,
}
NODE_DISPLAY_NAME_MAPPINGS = {
"orpheus": "Orpheus",
"orpheusAdvanced": "Orpheus Advanced",
}
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sounddevice
snac