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vllm-omni/examples/offline_inference/fish_speech/end2end.py
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7.5 KiB
Python

"""Offline inference demo for Fish Speech S2 Pro via vLLM Omni.
Generates speech from text using the fishaudio/s2-pro model.
Supports both text-only synthesis and voice cloning with reference audio.
Usage:
# Text-only synthesis
python end2end.py --text "Hello, this is a test."
# Voice cloning with reference audio
python end2end.py --text "Hello, this is a test." --ref-audio ref.wav --ref-text "Reference text."
# Streaming mode
python end2end.py --text "Hello, this is a test." --streaming
"""
import asyncio
import logging
import os
import time
import soundfile as sf
import torch
os.environ["VLLM_WORKER_MULTIPROC_METHOD"] = "spawn"
from vllm.utils.argparse_utils import FlexibleArgumentParser
from vllm_omni import AsyncOmni, Omni
logger = logging.getLogger(__name__)
DEFAULT_MODEL = "fishaudio/s2-pro"
DEFAULT_STAGE_CONFIG = os.path.join(
os.path.dirname(__file__),
"..",
"..",
"..",
"vllm_omni",
"model_executor",
"stage_configs",
"fish_speech_s2_pro.yaml",
)
def build_prompt(
text: str,
ref_audio_path: str | None = None,
ref_text: str | None = None,
model_name: str = DEFAULT_MODEL,
) -> dict:
"""Build a prompt for Fish Speech S2 Pro.
Uses Qwen3 chat template format with <|voice|> modality marker:
<|im_start|>user\n<|speaker:0|>text<|im_end|>\n<|im_start|>assistant\n<|voice|>
The model then generates semantic tokens autoregressively.
"""
from transformers import AutoTokenizer
tokenizer = AutoTokenizer.from_pretrained(model_name, trust_remote_code=True)
# Build user message with speaker tag prefix.
user_text = f"<|speaker:0|>{text}"
# Use chat template for user turn + assistant generation prompt.
messages = [{"role": "user", "content": user_text}]
prompt_ids = tokenizer.apply_chat_template(
messages,
tokenize=True,
add_generation_prompt=True,
)
# Append <|voice|> token (151673) to signal voice generation mode.
voice_token_id = tokenizer.encode("<|voice|>", add_special_tokens=False)
prompt_ids = prompt_ids + voice_token_id
additional_information = {
"text": [text],
"max_new_tokens": [4096],
}
if ref_audio_path:
additional_information["ref_audio"] = [ref_audio_path]
if ref_text:
additional_information["ref_text"] = [ref_text]
return {
"prompt_token_ids": prompt_ids,
"additional_information": additional_information,
}
def _save_wav(output_dir: str, request_id: str, mm: dict) -> None:
"""Concatenate audio chunks and write to a wav file."""
audio_data = mm["audio"]
sr_raw = mm["sr"]
sr_val = sr_raw[-1] if isinstance(sr_raw, list) and sr_raw else sr_raw
sr = sr_val.item() if hasattr(sr_val, "item") else int(sr_val)
audio_tensor = torch.cat(audio_data, dim=-1) if isinstance(audio_data, list) else audio_data
out_wav = os.path.join(output_dir, f"output_{request_id}.wav")
sf.write(out_wav, audio_tensor.float().cpu().numpy().flatten(), samplerate=sr, format="WAV")
logger.info("Request %s: saved audio to %s (sr=%d)", request_id, out_wav, sr)
def main(args):
"""Run offline inference with Omni."""
output_dir = args.output_dir
os.makedirs(output_dir, exist_ok=True)
stage_configs_path = args.stage_configs_path or DEFAULT_STAGE_CONFIG
model_name = args.model or DEFAULT_MODEL
inputs = [
build_prompt(
text=args.text,
ref_audio_path=args.ref_audio,
ref_text=args.ref_text,
model_name=model_name,
)
]
omni = Omni(
model=model_name,
stage_configs_path=stage_configs_path,
log_stats=args.log_stats,
stage_init_timeout=args.stage_init_timeout,
)
t_start = time.perf_counter()
for stage_outputs in omni.generate(inputs):
for output in stage_outputs.request_output:
_save_wav(output_dir, output.request_id, output.outputs[0].multimodal_output)
t_end = time.perf_counter()
logger.info("Total inference time: %.1f ms", (t_end - t_start) * 1000)
async def main_streaming(args):
"""Run offline inference with AsyncOmni for streaming audio chunks."""
output_dir = args.output_dir
os.makedirs(output_dir, exist_ok=True)
stage_configs_path = args.stage_configs_path or DEFAULT_STAGE_CONFIG
model_name = args.model or DEFAULT_MODEL
prompt = build_prompt(
text=args.text,
ref_audio_path=args.ref_audio,
ref_text=args.ref_text,
model_name=model_name,
)
omni = AsyncOmni(
model=model_name,
stage_configs_path=stage_configs_path,
log_stats=args.log_stats,
stage_init_timeout=args.stage_init_timeout,
)
request_id = "0"
t_start = time.perf_counter()
t_prev = t_start
chunk_idx = 0
async for stage_output in omni.generate(prompt, request_id=request_id):
mm = stage_output.request_output.outputs[0].multimodal_output
if not stage_output.finished:
t_now = time.perf_counter()
audio = mm.get("audio")
n = len(audio) if isinstance(audio, list) else (0 if audio is None else 1)
dt_ms = (t_now - t_prev) * 1000
ttfa_ms = (t_now - t_start) * 1000
if chunk_idx == 0:
logger.info("Request %s: chunk %d samples=%d TTFA=%.1fms", request_id, chunk_idx, n, ttfa_ms)
else:
logger.info("Request %s: chunk %d samples=%d inter_chunk=%.1fms", request_id, chunk_idx, n, dt_ms)
t_prev = t_now
chunk_idx += 1
else:
t_end = time.perf_counter()
total_ms = (t_end - t_start) * 1000
logger.info("Request %s: done total=%.1fms chunks=%d", request_id, total_ms, chunk_idx)
_save_wav(output_dir, request_id, mm)
def parse_args():
parser = FlexibleArgumentParser(
description="Fish Speech S2 Pro offline inference with vLLM Omni",
)
parser.add_argument(
"--text",
type=str,
default="Hello, this is a test of the Fish Speech text to speech system.",
help="Text to synthesize.",
)
parser.add_argument(
"--ref-audio",
type=str,
default=None,
help="Path to reference audio for voice cloning.",
)
parser.add_argument(
"--ref-text",
type=str,
default=None,
help="Reference text corresponding to the reference audio.",
)
parser.add_argument(
"--model",
type=str,
default=None,
help=f"HuggingFace model path (default: {DEFAULT_MODEL}).",
)
parser.add_argument(
"--stage-configs-path",
type=str,
default=None,
help="Path to stage configs YAML.",
)
parser.add_argument(
"--output-dir",
default="output_audio",
help="Output directory for generated wav files.",
)
parser.add_argument(
"--log-stats",
action="store_true",
default=False,
help="Enable writing detailed statistics.",
)
parser.add_argument(
"--stage-init-timeout",
type=int,
default=600,
help="Timeout for initializing stages in seconds.",
)
parser.add_argument(
"--streaming",
action="store_true",
default=False,
help="Stream audio chunks as they arrive via AsyncOmni.",
)
return parser.parse_args()
if __name__ == "__main__":
logging.basicConfig(level=logging.INFO)
args = parse_args()
if args.streaming:
asyncio.run(main_streaming(args))
else:
main(args)