Text Generation
Transformers
Safetensors
English
llama
text-generation-inference
unsloth
conversational
Instructions to use originalTimi/Hypa-Orpheus-Step-latest-16bit with libraries, inference providers, notebooks, and local apps. Follow these links to get started.
- Libraries
- Transformers
How to use originalTimi/Hypa-Orpheus-Step-latest-16bit with Transformers:
# Use a pipeline as a high-level helper from transformers import pipeline pipe = pipeline("text-generation", model="originalTimi/Hypa-Orpheus-Step-latest-16bit") messages = [ {"role": "user", "content": "Who are you?"}, ] pipe(messages)# Load model directly from transformers import AutoTokenizer, AutoModelForCausalLM tokenizer = AutoTokenizer.from_pretrained("originalTimi/Hypa-Orpheus-Step-latest-16bit") model = AutoModelForCausalLM.from_pretrained("originalTimi/Hypa-Orpheus-Step-latest-16bit", device_map="auto") messages = [ {"role": "user", "content": "Who are you?"}, ] inputs = tokenizer.apply_chat_template( messages, add_generation_prompt=True, tokenize=True, return_dict=True, return_tensors="pt", ).to(model.device) outputs = model.generate(**inputs, max_new_tokens=40) print(tokenizer.decode(outputs[0][inputs["input_ids"].shape[-1]:])) - Notebooks
- Google Colab
- Kaggle
- Local Apps Settings
- vLLM
How to use originalTimi/Hypa-Orpheus-Step-latest-16bit with vLLM:
Install from pip and serve model
# Install vLLM from pip: pip install vllm # Start the vLLM server: vllm serve "originalTimi/Hypa-Orpheus-Step-latest-16bit" # Call the server using curl (OpenAI-compatible API): curl -X POST "http://localhost:8000/v1/chat/completions" \ -H "Content-Type: application/json" \ --data '{ "model": "originalTimi/Hypa-Orpheus-Step-latest-16bit", "messages": [ { "role": "user", "content": "What is the capital of France?" } ] }'Use Docker
docker model run hf.co/originalTimi/Hypa-Orpheus-Step-latest-16bit
- SGLang
How to use originalTimi/Hypa-Orpheus-Step-latest-16bit with SGLang:
Install from pip and serve model
# Install SGLang from pip: pip install sglang # Start the SGLang server: python3 -m sglang.launch_server \ --model-path "originalTimi/Hypa-Orpheus-Step-latest-16bit" \ --host 0.0.0.0 \ --port 30000 # Call the server using curl (OpenAI-compatible API): curl -X POST "http://localhost:30000/v1/chat/completions" \ -H "Content-Type: application/json" \ --data '{ "model": "originalTimi/Hypa-Orpheus-Step-latest-16bit", "messages": [ { "role": "user", "content": "What is the capital of France?" } ] }'Use Docker images
docker run --gpus all \ --shm-size 32g \ -p 30000:30000 \ -v ~/.cache/huggingface:/root/.cache/huggingface \ --env "HF_TOKEN=<secret>" \ --ipc=host \ lmsysorg/sglang:latest \ python3 -m sglang.launch_server \ --model-path "originalTimi/Hypa-Orpheus-Step-latest-16bit" \ --host 0.0.0.0 \ --port 30000 # Call the server using curl (OpenAI-compatible API): curl -X POST "http://localhost:30000/v1/chat/completions" \ -H "Content-Type: application/json" \ --data '{ "model": "originalTimi/Hypa-Orpheus-Step-latest-16bit", "messages": [ { "role": "user", "content": "What is the capital of France?" } ] }' - Unsloth Studio
How to use originalTimi/Hypa-Orpheus-Step-latest-16bit with Unsloth Studio:
Install Unsloth Studio (macOS, Linux, WSL)
curl -fsSL https://unsloth.ai/install.sh | sh # Run unsloth studio unsloth studio -H 0.0.0.0 -p 8888 # Then open http://localhost:8888 in your browser # Search for originalTimi/Hypa-Orpheus-Step-latest-16bit to start chatting
Install Unsloth Studio (Windows)
irm https://unsloth.ai/install.ps1 | iex # Run unsloth studio unsloth studio -H 0.0.0.0 -p 8888 # Then open http://localhost:8888 in your browser # Search for originalTimi/Hypa-Orpheus-Step-latest-16bit to start chatting
Using HuggingFace Spaces for Unsloth
# No setup required # Open https://huggingface.co/spaces/unsloth/studio in your browser # Search for originalTimi/Hypa-Orpheus-Step-latest-16bit to start chatting
Load model with FastModel
pip install unsloth from unsloth import FastModel model, tokenizer = FastModel.from_pretrained( model_name="originalTimi/Hypa-Orpheus-Step-latest-16bit", max_seq_length=2048, ) - Docker Model Runner
How to use originalTimi/Hypa-Orpheus-Step-latest-16bit with Docker Model Runner:
docker model run hf.co/originalTimi/Hypa-Orpheus-Step-latest-16bit
File size: 14,904 Bytes
9456b8e 34e437f 9456b8e b19e04c 9456b8e b180cb0 | 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 242 243 244 245 246 247 248 249 250 251 252 253 254 255 256 257 258 259 260 261 262 263 264 265 266 267 268 269 270 271 272 273 274 275 276 277 278 279 280 281 282 283 284 285 286 287 288 289 290 291 292 293 294 295 296 297 298 299 300 301 302 303 304 305 306 307 308 309 310 311 312 313 314 315 316 317 318 319 320 321 322 323 324 325 326 327 | """
HF Inference Endpoint handler — Hypa Orpheus TTS + Voice Cloning (merged 16-bit).
Task matrix (routed by `parameters`):
task="tts", mode="vanilla" : {speaker}: text -> speech
task="tts", mode="translate" : {speaker} - {Language}: text -> speech in Language
task="vc", mode="vanilla" : reference (text+audio) + target text -> speech in reference voice
task="vc", mode="translate" : + language tag on target text -> cross-lingual cloning
VC method="m1" (in-context) | method="m2" (continue-speaking)
Output parity with the legacy endpoint: `audio_b64` is base64 of the RAW
float32 little-endian mono PCM buffer at 24000 Hz (NO WAV/RIFF container),
so existing products decode with: np.frombuffer(base64.b64decode(s), dtype=np.int16)
[If the legacy endpoint used int16, change RAW_DTYPE to np.int16 below.]
Prompts are byte-identical to Step-III training (_encode_text / build_tts /
build_vc_both), reference codes are frame-deduped, and prompts reach vLLM as
token ids (never a decoded string).
"""
import io
import os
import base64
import tempfile
import traceback
import numpy as np
import torch
import soundfile as sf
import librosa
from transformers import AutoTokenizer
from snac import SNAC
from vllm import LLM, SamplingParams
class EndpointHandler:
# ---- Orpheus special tokens (fixed by the model) ----
TOKENISER_LEN = 128256
START_OF_TEXT = 128000
END_OF_TEXT = 128009
START_OF_SPEECH = TOKENISER_LEN + 1 # 128257
END_OF_SPEECH = TOKENISER_LEN + 2 # 128258
START_OF_HUMAN = TOKENISER_LEN + 3 # 128259
END_OF_HUMAN = TOKENISER_LEN + 4 # 128260
START_OF_AI = TOKENISER_LEN + 5 # 128261
END_OF_AI = TOKENISER_LEN + 6 # 128262
AUDIO_OFFSET = 128266
# NOTE: fine-tune data capped at 2048 tokens; 4096 kept so M1-VC prompts
# (ref codes + two texts, often 1000-2000 tokens) retain a generation
# budget. Base Llama-3 RoPE supports these positions natively; expect the
# best quality when prompt+generation stays near the trained ~2048.
MAX_MODEL_LEN = 4096
MAX_REF_SECONDS = 30
SNAC_SR = 24000
RAW_DTYPE = np.int16 # legacy raw-PCM dtype (see docstring)
LANG_DISPLAY = {
"en": "English", "es": "Spanish", "fr": "French", "ha": "Hausa",
"yo": "Yoruba", "sw": "Swahili", "ar": "Arabic", "pt": "Portuguese",
"ann": "Annang", "ebi": "Ebira", "efi": "Efik", "ego": "Eggon",
"urh": "Urhobo", "ibb": "Ibibio", "idm": "Idoma", "igl": "Igala",
"ig": "Igbo", "nup": "Nupe", "tiv": "Tiv", "pg": "Pidgin",
}
# ------------------------------------------------------------------ init
def __init__(self, path=""):
self.device = "cuda" if torch.cuda.is_available() else "cpu"
# SNAC first (tiny, ~80 MB) so it never contends with vLLM's reservation.
self.snac = SNAC.from_pretrained("hubertsiuzdak/snac_24khz").to(self.device).eval()
self.model = LLM(
path,
max_model_len=self.MAX_MODEL_LEN,
gpu_memory_utilization=0.75,
model_impl="transformers",
)
self.tokenizer = AutoTokenizer.from_pretrained(path)
# ------------------------------------------------------- text encoding
def _lang_display(self, x):
if x is None:
return None
k = str(x).strip().lower()
return self.LANG_DISPLAY.get(k, k.capitalize() if k else None)
def _encode_text(self, text, speaker=None, lang_tag=None, add_bos=True):
text = "" if text is None else str(text).strip()
spk = speaker if (speaker and str(speaker).strip().lower() not in ("", "random", "none")) else None
if spk and lang_tag:
prompt = f"{spk} - {lang_tag}: {text}"
elif spk:
prompt = f"{spk}: {text}"
elif lang_tag:
prompt = f"{lang_tag}: {text}"
else:
prompt = text
ids = self.tokenizer.encode(prompt, add_special_tokens=add_bos)
ids.append(self.END_OF_TEXT)
return ids
# ------------------------------------------------------ audio encoding
def _b64_to_wave(self, b64_str):
raw = base64.b64decode(b64_str)
if not raw:
raise ValueError("reference_audio is empty.")
try:
arr, sr = sf.read(io.BytesIO(raw), dtype="float32")
except Exception:
# temp-file fallback: librosa/audioread handles containers
# libsndfile can't open, but needs a real file path for some codecs.
tmp = None
try:
with tempfile.NamedTemporaryFile(delete=False, suffix=".audio") as f:
f.write(raw)
tmp = f.name
arr, sr = librosa.load(tmp, sr=None, mono=False)
arr = np.asarray(arr, dtype=np.float32)
if arr.ndim > 1:
arr = arr.T # librosa returns (ch, n)
finally:
if tmp and os.path.exists(tmp):
os.remove(tmp)
if arr.ndim > 1:
arr = arr.mean(axis=1)
if arr.size == 0 or not np.isfinite(arr).all():
raise ValueError("Reference audio is empty or contains invalid samples.")
if sr != self.SNAC_SR:
arr = librosa.resample(arr.astype(np.float32), orig_sr=sr, target_sr=self.SNAC_SR)
dur = len(arr) / self.SNAC_SR
if dur > self.MAX_REF_SECONDS:
raise ValueError(f"Reference audio is {dur:.1f}s; max is {self.MAX_REF_SECONDS}s. "
f"Send a shorter clip.")
return arr.astype(np.float32)
@torch.inference_mode()
def _audio_to_codes(self, arr):
wav = torch.from_numpy(arr).to(self.device)[None, None]
codes = self.snac.encode(wav)
c0, c1, c2 = codes[0][0].tolist(), codes[1][0].tolist(), codes[2][0].tolist()
n = min(len(c0), len(c1) // 2, len(c2) // 4)
out = []
for i in range(n):
out += [
c0[i] + self.AUDIO_OFFSET,
c1[2 * i] + self.AUDIO_OFFSET + 4096,
c2[4 * i] + self.AUDIO_OFFSET + 2 * 4096,
c2[4 * i + 1] + self.AUDIO_OFFSET + 3 * 4096,
c1[2 * i + 1] + self.AUDIO_OFFSET + 4 * 4096,
c2[4 * i + 2] + self.AUDIO_OFFSET + 5 * 4096,
c2[4 * i + 3] + self.AUDIO_OFFSET + 6 * 4096,
]
return out
@staticmethod
def _dedup_frames(codes):
if not codes:
return codes
codes = codes[: (len(codes) // 7) * 7]
if len(codes) < 7:
return codes
result = codes[:7]
for i in range(7, len(codes), 7):
if codes[i] != result[-7]:
result.extend(codes[i:i + 7])
return result
# ------------------------------------------------------ prompt builders
def build_tts_prompt(self, text, speaker, mode, language):
lang_tag = self._lang_display(language) if mode == "translate" else None
tt = self._encode_text(text, speaker, lang_tag, add_bos=True)
return [self.START_OF_HUMAN] + tt + [self.END_OF_HUMAN,
self.START_OF_AI, self.START_OF_SPEECH]
def build_vc_prompt(self, ref_text, ref_codes, target_text, mode, language, method):
tag2 = self._lang_display(language) if mode == "translate" else None
tt1 = self._encode_text(ref_text, None, None, add_bos=True)
tt2 = self._encode_text(target_text, None, tag2, add_bos=False)
if method == "m1":
return ([self.START_OF_HUMAN] + tt1 + [self.END_OF_HUMAN,
self.START_OF_AI, self.START_OF_SPEECH] + ref_codes +
[self.END_OF_SPEECH, self.END_OF_AI,
self.START_OF_HUMAN] + tt2 + [self.END_OF_HUMAN,
self.START_OF_AI, self.START_OF_SPEECH])
return ([self.START_OF_HUMAN] + tt1 + tt2 + [self.END_OF_HUMAN,
self.START_OF_AI, self.START_OF_SPEECH] + ref_codes)
# --------------------------------------------------------- generation
def _generate(self, prompt_ids, params):
sampling = SamplingParams(
temperature = params["temperature"],
top_p = params["top_p"],
top_k = params["top_k"],
max_tokens = params["max_new_tokens"],
repetition_penalty = params["repetition_penalty"],
stop_token_ids = [self.END_OF_SPEECH, self.END_OF_AI],
detokenize = False,
)
outputs = self.model.generate({"prompt_token_ids": prompt_ids}, sampling)
return list(outputs[0].outputs[0].token_ids)
# ----------------------------------------------------------- decoding
@torch.inference_mode()
def _codes_to_wave(self, gen_ids):
"""Frame-validating SNAC decode: accepts only well-formed 7-token frames
(token k in slot-k range), resyncs on malformed spans."""
frames, i, n, resyncs = [], 0, len(gen_ids), 0
while i <= n - 7:
vals, ok = [], True
for k in range(7):
lo = self.AUDIO_OFFSET + k * 4096
t = gen_ids[i + k]
if not (lo <= t < lo + 4096):
ok = False
break
vals.append(t - lo)
if ok:
frames.append(vals)
i += 7
else:
i += 1
resyncs += 1
self._last_resyncs = resyncs
if not frames:
return None, 0
l1 = [f[0] for f in frames]
l2, l3 = [], []
for f in frames:
l2.append(f[1]); l3.append(f[2]); l3.append(f[3])
l2.append(f[4]); l3.append(f[5]); l3.append(f[6])
tensors = [
torch.tensor(l1)[None].to(self.device),
torch.tensor(l2)[None].to(self.device),
torch.tensor(l3)[None].to(self.device),
]
wav = self.snac.decode(tensors).squeeze().detach().cpu().numpy()
return wav, len(frames)
def _wave_to_b64_raw(self, wav):
wav = np.clip(wav, -1.0, 1.0)
pcm16 = (wav * 32767.0).astype(self.RAW_DTYPE)
return base64.b64encode(np.ascontiguousarray(pcm16).tobytes()).decode("utf-8")
# -------------------------------------------------------------- entry
def __call__(self, data):
try:
target_text = data.get("inputs")
if not target_text:
return {"error": "Missing 'inputs' (target text)."}
p = data.get("parameters", {}) or {}
task = str(p.get("task", "tts")).lower()
mode = str(p.get("mode", "vanilla")).lower()
method = str(p.get("method", "m2")).lower()
if mode in ("translation", "trans"):
mode = "translate"
if task not in ("tts", "vc"):
return {"error": "parameters.task must be 'tts' or 'vc'."}
if mode not in ("vanilla", "translate"):
return {"error": "parameters.mode must be 'vanilla' or 'translate'."}
if mode == "translate" and not p.get("language"):
return {"error": "parameters.language is required for translate mode."}
gen_params = {
"temperature": float(p.get("temperature", 0.6)),
"top_p": float(p.get("top_p", 0.95)),
"top_k": int(p.get("top_k", 50)),
"max_new_tokens": int(p.get("max_new_tokens", 1200)),
"repetition_penalty": float(p.get("repetition_penalty", 1.1)),
}
if not 0 < gen_params["top_p"] <= 1:
return {"error": "top_p must be within (0, 1]."}
if not (gen_params["top_k"] == -1 or gen_params["top_k"] > 0):
return {"error": "top_k must be -1 (disabled) or a positive integer."}
if not 0 < gen_params["repetition_penalty"] <= 2:
return {"error": "repetition_penalty must be within (0, 2]."}
if gen_params["max_new_tokens"] <= 0:
return {"error": "max_new_tokens must be positive."}
if task == "vc":
ref_text = p.get("reference_text")
ref_audio = p.get("reference_audio")
if not ref_text or not ref_audio:
return {"error": "VC requires parameters.reference_text and "
"parameters.reference_audio (base64)."}
if method not in ("m1", "m2"):
return {"error": "parameters.method must be 'm1' or 'm2'."}
ref_wave = self._b64_to_wave(ref_audio)
ref_codes = self._dedup_frames(self._audio_to_codes(ref_wave))
if not ref_codes:
return {"error": "Reference audio produced no SNAC codes."}
prompt_ids = self.build_vc_prompt(
ref_text, ref_codes, target_text, mode, p.get("language"), method)
else:
prompt_ids = self.build_tts_prompt(
target_text, p.get("voice") or p.get("speaker"),
mode, p.get("language"))
budget = self.MAX_MODEL_LEN - gen_params["max_new_tokens"]
if len(prompt_ids) > budget:
return {"error": f"Prompt is {len(prompt_ids)} tokens; exceeds budget "
f"{budget} (max_model_len - max_new_tokens). "
f"Shorten the reference clip or text."}
gen_ids = self._generate(prompt_ids, gen_params)
wav, n_frames = self._codes_to_wave(gen_ids)
if wav is None:
return {"error": "Model generated no audio tokens.",
"input_tokens": len(prompt_ids),
"generated_tokens": len(gen_ids)}
return {
"audio_b64": self._wave_to_b64_raw(wav), # RAW float32 PCM (legacy parity)
"audio_dtype": np.dtype(self.RAW_DTYPE).name,
"sample_rate": self.SNAC_SR,
"duration_seconds": round(len(wav) / self.SNAC_SR, 3),
"audio_frames": n_frames,
"input_tokens": len(prompt_ids),
"generated_tokens": len(gen_ids),
"task": task, "mode": mode,
"method": method if task == "vc" else None,
"decode_resyncs": getattr(self, "_last_resyncs", 0),
}
except ValueError as e:
return {"error": str(e)}
except Exception as e:
traceback.print_exc()
return {"error": str(e)} |