[Model][TTS][WIP] Add dots.tts (rednote-hilab) — continuous-AR 48kHz TTS (#4765)

Signed-off-by: Moore-Z <zhumoore9661996@gmail.com>
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Lucas(Ruijie) authored and GitHub committed 2026-08-13 17:58:33 +00:00
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@@ -23,6 +23,7 @@ list of supported architectures across all modalities, see
| OmniVoice | `k2-fsa/OmniVoice` | 2 (gen + dec) | ✓ | — | voice design, language hint | 24 kHz |
| Qwen3-TTS | `Qwen/Qwen3-TTS-12Hz-1.7B-{CustomVoice,VoiceDesign,Base}` | 2 (talker + code2wav) | ✓ (Base) | ✓ | 3 task variants | 24 kHz |
| VoxCPM2 | `openbmb/VoxCPM2` | single (native AR) | ✓ | ✓ (online) | continuation | 48 kHz |
| dots.tts | `rednote-hilab/dots.tts-soar` | single (native AR) | — (not wired yet) | — | — | 48 kHz |
| IndexTTS-2 | `IndexTeam/IndexTTS-2` | 2 (AR talker + S2Mel DiT + BigVGAN) | ✓ (required) | — | emotion control (`--emo-audio`, `--emo-text`, `--emo-vector`) | 22.05 kHz |
| IndexTTS-2.5 | native `checkpoints/` bundle | 2 (AR talker + EnhancedCodec + S2Mel DiT + BigVGAN) | ✓ (required) | — | multilingual (`--lang`) + emotion control | 22.05 kHz |
| Voxtral TTS | `mistralai/Voxtral-4B-TTS-2603` | varies | ✓ | ✓ | voice presets | 24 kHz |
@@ -430,6 +431,31 @@ Streaming is exposed through the online OpenAI Speech API (`stream=true`). See [
---
## dots.tts
Single-stage native AR TTS at 48 kHz (rednote-hilab). Pipeline: `Qwen2.5-1.5B base LM → DiT (10-step Euler flow matching) → patch_encoder AR loopback → AudioVAE (streaming decode)`. Same "vLLM-native base LM + side-path computation" pattern as VoxCPM2, with a plain Qwen2 backbone instead of MiniCPM4 and no FSQ / residual LM stage.
### Quick start
```bash
python examples/offline_inference/text_to_speech/dots_tts/end2end.py \
--model rednote-hilab/dots.tts-soar \
--text "Hello, this is a test of dots TTS running on vLLM Omni."
```
### Voice cloning
Not wired in this release — generation is zero-shot only. The CAM++ x-vector speaker encoder weights load, but `end2end.py` has no `--ref-audio`/`--ref-text` flags yet.
### Streaming
The AudioVAE decoder has an internal streaming path (`init_stream_state` / `stream_step` / `stream_flush`) used to avoid boundary artifacts between 160 ms patches, but it is not yet exposed through an online serving endpoint or example.
### Notes
- Output: 48 kHz mono WAV.
- Deploy config: `vllm_omni/deploy/dots_tts.yaml` (auto-loaded by HF `model_type`).
- Checkpoints: `rednote-hilab/dots.tts-soar` is the validated default. `dots.tts-base` shares the same architecture but is unvalidated in this repo. `dots.tts-mf` (MeanFlow, 2-4 step) is not supported yet.
- Known limitation: no CUDA graph capture and no batched side-path computation yet, so concurrent requests do not currently scale (each request's DiT Euler steps run serially). See `recipes/rednote-hilab/dots.tts.md` for details and the roadmap.
---
## IndexTTS-2 and IndexTTS-2.5
Both versions use a 2-stage TTS pipeline and produce 22.05 kHz mono audio. Stage 0 is a GPT AR talker. IndexTTS-2 Stage 1 uses RepCodec semantic embeddings, GPT latent, S2Mel CFM/DiT, and BigVGAN. IndexTTS-2.5 replaces RepCodec with EnhancedCodec and uses the official code-only path (`use_gpt_latent=false`). Every request requires reference audio for zero-shot voice cloning.
@@ -0,0 +1,125 @@
"""Offline dots.tts inference example (single-stage native AR pipeline).
Uses the single-stage native AR config (vllm_omni/deploy/dots_tts.yaml).
Reference checkpoint: rednote-hilab/dots.tts-soar.
"""
from __future__ import annotations
import os
import time
from pathlib import Path
import soundfile as sf
import torch
from vllm_omni import Omni
from vllm_omni.utils.tracking_parser import TrackingArgumentParser
REPO_ROOT = Path(__file__).resolve().parents[4]
SAMPLE_RATE = 48_000
def parse_args():
parser = TrackingArgumentParser(description="Offline dots.tts native AR inference")
parser.add_argument(
"--model",
type=str,
default="rednote-hilab/dots.tts-soar",
help="dots.tts model path or HuggingFace repo ID.",
)
parser.add_argument(
"--text",
type=str,
default="Hello, this is a test of dots TTS running on vLLM Omni.",
help="Text to synthesize.",
)
parser.add_argument(
"--output-dir",
type=str,
default="output_audio",
help="Directory for output WAV files.",
)
parser.add_argument(
"--deploy-config",
type=str,
default=None,
help="Override the deploy config path. If unset, auto-loads "
"vllm_omni/deploy/dots_tts.yaml based on the HF model_type.",
)
return parser.parse_args()
def extract_audio(multimodal_output: dict) -> torch.Tensor:
"""Extract the final complete audio tensor from multimodal output.
The output processor concatenates per-step delta tensors under
``model_outputs``. Falls back to ``audio`` for backwards compat.
"""
audio = multimodal_output.get("model_outputs")
if audio is None:
audio = multimodal_output.get("audio")
if audio is None:
raise ValueError(f"No audio key in multimodal_output: {list(multimodal_output.keys())}")
if isinstance(audio, list):
valid = [torch.as_tensor(a).float().cpu().reshape(-1) for a in audio if a is not None]
if not valid:
raise ValueError("Audio list is empty or all elements are None.")
return torch.cat(valid, dim=0) if len(valid) > 1 else valid[0]
return torch.as_tensor(audio).float().cpu().reshape(-1)
def main():
args = parse_args()
output_dir = Path(args.output_dir)
output_dir.mkdir(parents=True, exist_ok=True)
engine = Omni(
model=args.model,
deploy_config=args.deploy_config,
)
from transformers import AutoTokenizer
from vllm_omni.model_executor.models.dots_tts.dots_tts_prompt import (
build_dots_tts_prompt,
)
tokenizer = AutoTokenizer.from_pretrained(args.model, trust_remote_code=True)
prompt = build_dots_tts_prompt(
tokenizer=tokenizer,
text=args.text,
)
print(f"Model : {args.model}")
print(f"Text : {args.text}")
print(f"Output dir : {output_dir}")
print(f"Prompt len : {len(prompt['prompt_token_ids'])} tokens")
t_start = time.perf_counter()
outputs = engine.generate([prompt])
elapsed = time.perf_counter() - t_start
request_output = outputs[0]
mm = request_output.outputs[0].multimodal_output
audio = extract_audio(mm)
duration = audio.numel() / SAMPLE_RATE
rtf = elapsed / duration if duration > 0 else float("inf")
output_path = output_dir / "output.wav"
sf.write(str(output_path), audio.numpy(), SAMPLE_RATE, format="WAV")
print(f"Saved : {output_path}")
print(f"Duration : {duration:.2f}s")
print(f"Inference : {elapsed:.2f}s")
print(f"RTF : {rtf:.3f}")
if __name__ == "__main__":
os.environ["VLLM_WORKER_MULTIPROC_METHOD"] = "spawn"
main()