# Text-To-Speech

vLLM-Omni supports several autoregressive TTS models. They share a mostly
common CLI shape (`--text`, `--ref-audio`, `--ref-text`, plus an
output-path flag — `--output-dir` for most, `--output` for OmniVoice) and
live together in this hub. Each model has its own subdirectory containing
a single `end2end.py` script; this README is the single doc entry point.

For online serving, see [`examples/online_serving/text_to_speech/`](../../online_serving/text_to_speech/README.md). For the full
list of supported architectures across all modalities, see
[Supported Models](../../../docs/models/supported_models.md).

## Supported Models

| Model | HuggingFace repo | Stages | Voice cloning | Streaming | Special modes | Sample rate |
|---|---|---|---|---|---|---|
| CosyVoice3 | `FunAudioLLM/Fun-CosyVoice3-0.5B-2512` | 2 (talker + code2wav) | ✓ | ✓ | — | 24 kHz |
| Fish Speech S2 Pro | `fishaudio/s2-pro` | dual-AR | ✓ | ✓ | — | 44.1 kHz |
| Gepard-1.0 | `nineninesix/gepard-1.0` | single (native AR + NanoCodec) | — (zero-shot; cloning WIP) | — (serving WIP) | zero-shot | 22.05 kHz |
| GLM-TTS | `zai-org/GLM-TTS` | 2 (AR + DiT) | ✓ (required) | ✓ | — | 24 kHz |
| Ming-omni-tts | `inclusionAI/Ming-omni-tts-0.5B` | 2 (AR + audio VAE) | ✓ | ✓ | style / IP / dialect / TTA / podcast | 44.1 kHz |
| Ming-flash-omni-TTS | `Jonathan1909/Ming-flash-omni-2.0` | single (talker only) | — (caption-controlled) | — | style / IP / basic captions | 44.1 kHz |
| MOSS-TTS-Nano | `OpenMOSS-Team/MOSS-TTS-Nano` | single (AR + codec) | ✓ (required) | ✓ | voice_clone, continuation | 48 kHz |
| 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 |

## Common Quick Start

Most models share this invocation shape:

```bash
python examples/offline_inference/text_to_speech/<model>/end2end.py \
    --text "Hello, this is a test." \
    --ref-audio /path/to/reference.wav \
    --ref-text  "Transcript of the reference audio."
```

`--ref-audio` and `--ref-text` are optional (text-only synthesis works without them) and must be provided together for voice cloning. The exotic scripts — Qwen3-TTS, Voxtral TTS, CosyVoice3 — accept additional model-specific flags documented in their per-model section below. Qwen3-TTS in particular uses its own argparse surface (`--query-type`, `--audio-path`, etc.) and does not follow the common shape; see its section.

---

## CosyVoice3

2-stage TTS pipeline (`talker` + `code2wav`) at 24 kHz.

### Prerequisites
```bash
uv pip install -e .
# Includes soundfile, onnxruntime, x-transformers, einops via requirements.
```

Download the model snapshot:
```python
from huggingface_hub import snapshot_download
snapshot_download('FunAudioLLM/Fun-CosyVoice3-0.5B-2512',
                  local_dir='pretrained_models/Fun-CosyVoice3-0.5B')
```

If your downloaded checkpoint lacks `config.json`, add it:
```json
{
    "model_type": "cosyvoice3",
    "architectures": ["CosyVoice3Model"]
}
```
This is required because `AutoConfig.register("cosyvoice3", CosyVoice3Config)` only registers the class mapping; the loader still reads `model_type` from `config.json` to select the class.

### Quick start
```bash
python examples/offline_inference/text_to_speech/cosyvoice3/end2end.py \
    --model pretrained_models/Fun-CosyVoice3-0.5B \
    --tokenizer pretrained_models/Fun-CosyVoice3-0.5B/CosyVoice-BlankEN
```

### Voice cloning
If `--ref-audio` is omitted, the script downloads the upstream
[`zero_shot_prompt.wav`](https://github.com/FunAudioLLM/CosyVoice/blob/main/asset/zero_shot_prompt.wav) from the CosyVoice repo into the current directory.
To use your own clip, pass `--ref-audio /path/to/reference.wav`, and modify `--prompt-text` correspondingly.
```bash
python examples/offline_inference/text_to_speech/cosyvoice3/end2end.py \
    --model pretrained_models/Fun-CosyVoice3-0.5B \
    --tokenizer pretrained_models/Fun-CosyVoice3-0.5B/CosyVoice-BlankEN \
    --ref-audio /path/to/reference.wav \
    --prompt-text "You are a helpful assistant.<|endofprompt|>Trascript in your ref audio clip"
```

### Streaming
Streaming is enabled by default via `async_chunk: true` in `vllm_omni/deploy/cosyvoice3.yaml`. Pass `--no-async-chunk` on `vllm serve` to switch to the legacy synchronous path.

### Notes
- Stage 0 (`talker`) emits speech tokens; stage 1 (`code2wav`) runs flow matching + HiFiGAN to synthesize waveform.
- Deploy config auto-loads from `vllm_omni/deploy/cosyvoice3.yaml` based on HF `model_type`. Pass `--deploy-config <path>` to override.

---

## GLM-TTS

2-stage TTS pipeline (AR + DiT flow-matching) at 24 kHz. Every request requires reference audio and its transcript for zero-shot voice cloning.

### Quick start
```bash
python examples/offline_inference/text_to_speech/glm_tts/end2end.py \
    --model zai-org/GLM-TTS \
    --text "你好，这是语音合成测试。" \
    --ref-audio /path/to/reference.wav \
    --ref-text "这是参考音频的文本内容。" \
    --output-dir ./output
```

### Architecture
```
Text → [Stage 0: AR] → Speech Tokens → [Stage 1: DiT + HiFT] → Audio (24 kHz)
        (Llama-based)    (32k vocab)      (Flow Matching)
```

### Notes
- `--ref-audio` and `--ref-text` are **required** together; GLM-TTS does not support text-only synthesis.
- Reference audio should be 3-10 seconds.
- First run may be slow due to lazy loading of WhisperVQ tokenizer and CampPlus ONNX speaker embedder.
- Default sampling: temperature=1.0, top_k=25, top_p=0.8 (RAS method).
- The `--model` path should point to the repository root (not `llm/` subdirectory).

---

## Fish Speech S2 Pro

4B dual-AR text-to-speech model from FishAudio with the DAC codec at 44.1 kHz.

### Prerequisites
```bash
pip install fish-speech
```

### Quick start
```bash
python examples/offline_inference/text_to_speech/fish_speech/end2end.py \
    --text "Hello, this is a test of the Fish Speech text to speech system."
```

### Voice cloning
```bash
python examples/offline_inference/text_to_speech/fish_speech/end2end.py \
    --text "Hello, this is a cloned voice." \
    --ref-audio /path/to/reference.wav \
    --ref-text  "Transcript of the reference audio."
```

### Streaming
```bash
python examples/offline_inference/text_to_speech/fish_speech/end2end.py \
    --text "Hello, this is a streaming test." \
    --streaming
```
Streaming requires `async_chunk: true` in the deploy config.

### Notes
- Output: 44.1 kHz mono WAV.
- DAC codec weights (`codec.pth`) are loaded lazily from the model directory.

---

## Ming-omni-tts

Dense 0.5B two-stage TTS pipeline (`AR + flow` + audio VAE) at 44.1 kHz. The example covers style, IP voice, music-only generation, text-to-audio events, emotion, dialect, zero-shot cloning, podcast, speech+BGM, and speech+environment-sound cases.

### Quick start
```bash
python examples/offline_inference/text_to_speech/ming_tts/end2end.py \
    --case style \
    --deploy-config vllm_omni/deploy/ming_tts.yaml \
    --enforce-eager
```

### Voice cloning
```bash
python examples/offline_inference/text_to_speech/ming_tts/end2end.py \
    --case zero_shot \
    --ref-audio /path/to/reference.wav \
    --ref-text "在此奉劝大家别乱打美白针。" \
    --deploy-config vllm_omni/deploy/ming_tts.yaml \
    --enforce-eager
```

### Streaming
```bash
python examples/offline_inference/text_to_speech/ming_tts/end2end.py \
    --case basic \
    --ref-audio /path/to/reference.wav \
    --streaming \
    --deploy-config vllm_omni/deploy/ming_tts.yaml \
    --enforce-eager
```

### Notes
- `style`, `ip`, `bgm`, and `tta` do not require reference audio.
- Reference-audio cases use `--ref-audio`; `zero_shot` also requires `--ref-text`.
- `podcast` uses multiple references via `--ref-audio-paths`.
- Full case details live in [`ming_tts/README.md`](ming_tts/README.md).

---

## Ming-flash-omni-TTS

Standalone talker-only deployment of Ming-flash-omni-2.0 at 44.1 kHz. Voice is controlled through caption fields (`风格` / `IP` / `语速`/`基频`/`音量`) rather than reference audio.

### Prerequisites
The example calls into `vllm_omni.model_executor.models.ming_flash_omni.prompt_utils` for the default prompt and instruction builder; no extra pip install on top of the base vLLM-Omni install.

### Quick start
```bash
python examples/offline_inference/text_to_speech/ming_flash_omni_tts/end2end.py --case style
```

### Cases
```bash
# ASMR-style whisper (caption-driven)
python examples/offline_inference/text_to_speech/ming_flash_omni_tts/end2end.py --case style

# IP voice (preset character voice via caption)
python examples/offline_inference/text_to_speech/ming_flash_omni_tts/end2end.py --case ip

# Basic speed/pitch/volume control
python examples/offline_inference/text_to_speech/ming_flash_omni_tts/end2end.py --case basic
```

Override the default text per case with `--text`, write to a custom path with `--output`.

### Notes
- Talker-only deployment — for the multimodal Ming-flash-omni example, see [`examples/offline_inference/ming_flash_omni/`](../../ming_flash_omni/).
- Deploy config: `vllm_omni/deploy/ming_flash_omni_tts.yaml` (single GPU, `enforce_eager`, `max_num_seqs: 1`).
- Decode defaults from the Ming cookbook: `max_decode_steps=200`, `cfg=2.0`, `sigma=0.25`, `temperature=0.0`, `use_zero_spk_emb=True`.

---

## MOSS-TTS-Nano

Single-stage 0.1B AR LM + MOSS-Audio-Tokenizer-Nano codec at 48 kHz mono (mixed down from upstream stereo). ZH / EN / JA. Every request requires a reference clip via `--ref-audio`.

> **No built-in speaker presets.** `--ref-audio` is required on every call. Default `--mode voice_clone` matches upstream's recommended workflow; `--mode continuation` is exposed for completeness but upstream's continuation-with-prompt path emits very short / near-silent output, so it is rarely useful in practice. Sample reference clips ship in the upstream repo under [`assets/audio/`](https://github.com/OpenMOSS/MOSS-TTS-Nano/tree/main/assets/audio) (e.g. `zh_1.wav`, `en_2.wav`, `jp_2.wav`).

### Quick start
```bash
# Fetch a sample reference clip (one-off, user-scoped cache).
REF_DIR="${XDG_CACHE_HOME:-$HOME/.cache}/moss-tts-nano"
mkdir -p "$REF_DIR"
[ -s "$REF_DIR/zh_1.wav" ] || \
    curl -L -o "$REF_DIR/zh_1.wav" https://raw.githubusercontent.com/OpenMOSS/MOSS-TTS-Nano/main/assets/audio/zh_1.wav

python examples/offline_inference/text_to_speech/moss_tts_nano/end2end.py \
    --text "你好，这是MOSS-TTS-Nano的语音合成演示。" \
    --ref-audio "$REF_DIR/zh_1.wav"
```
The first run downloads `OpenMOSS-Team/MOSS-TTS-Nano` and `OpenMOSS-Team/MOSS-Audio-Tokenizer-Nano` from Hugging Face.

### Reproducible runs
```bash
python examples/offline_inference/text_to_speech/moss_tts_nano/end2end.py \
    --text "Deterministic test." \
    --ref-audio "$REF_DIR/en_2.wav" \
    --seed 42
```

### Notes
- Output: 48 kHz mono WAV (the tokenizer is internally stereo at 48 kHz; the wrapper averages to mono before reaching the engine).
- Deploy config: `vllm_omni/deploy/moss_tts_nano.yaml` (auto-loaded; override with `--deploy-config`).
- Default `--max-new-frames 375` ≈ 14 s of audio; raise for longer outputs.
- `--ref-text` is rejected in `voice_clone` mode and required only with `--mode continuation`.
- Run `--help` for the full sampling-knob surface (`--audio-temperature`, `--audio-top-k`, `--audio-top-p`, `--text-temperature`).

---

## OmniVoice

Zero-shot multilingual TTS supporting 600+ languages, with three modes (auto / clone / design).

### Prerequisites
```bash
huggingface-cli download k2-fsa/OmniVoice
```
Voice cloning requires `transformers>=5.3.0`. Auto and design modes work with `transformers>=4.57.0`.

### Quick start (auto voice)
```bash
python examples/offline_inference/text_to_speech/omnivoice/end2end.py \
    --model k2-fsa/OmniVoice \
    --text "Hello, this is a test."
```

### Voice cloning
```bash
python examples/offline_inference/text_to_speech/omnivoice/end2end.py \
    --model k2-fsa/OmniVoice \
    --text "Hello, this is a test." \
    --ref-audio ref.wav \
    --ref-text  "This is the reference transcription."
```

### Voice design
```bash
python examples/offline_inference/text_to_speech/omnivoice/end2end.py \
    --model k2-fsa/OmniVoice \
    --text "Hello, this is a test." \
    --instruct "female, low pitch, british accent"
```

### Language hint
```bash
python examples/offline_inference/text_to_speech/omnivoice/end2end.py \
    --model k2-fsa/OmniVoice \
    --text "你好，这是一个测试。" \
    --lang zh
```

### Seed for Reproducibility
```bash
python examples/offline_inference/text_to_speech/omnivoice/end2end.py \
    --model k2-fsa/OmniVoice \
    --text "Hello, this is a test." \
    --seed 42
```

### Notes
- Stage 0 (Generator): Qwen3-0.6B with 32-step iterative unmasking.
- Stage 1 (Decoder): HiggsAudioV2 RVQ + DAC at 24 kHz.

---

## Qwen3-TTS

3-task-variant TTS with 24 kHz output. Has its own argparse surface (this script does not follow the common `--text` / `--ref-audio` shape).

### Prerequisites
For ROCm builds, replace `onnxruntime` with `onnxruntime-rocm`:
```bash
pip uninstall onnxruntime
pip install onnxruntime-rocm
```

### Task variants
- `CustomVoice`: predefined speaker (speaker ID) with optional style instruction.
- `VoiceDesign`: text + descriptive instruction designs a new voice.
- `Base`: voice cloning from reference audio + transcript.

```bash
# Single sample
python examples/offline_inference/text_to_speech/qwen3_tts/end2end.py --query-type CustomVoice
python examples/offline_inference/text_to_speech/qwen3_tts/end2end.py --query-type VoiceDesign
python examples/offline_inference/text_to_speech/qwen3_tts/end2end.py --query-type Base

# Base with a custom reference audio (Qwen3-TTS uses --audio-path, not --ref-audio):
python examples/offline_inference/text_to_speech/qwen3_tts/end2end.py \
    --query-type Base --audio-path /path/to/reference.wav

# Base variant has an additional mode flag:
python examples/offline_inference/text_to_speech/qwen3_tts/end2end.py --query-type Base --mode-tag icl       # default
python examples/offline_inference/text_to_speech/qwen3_tts/end2end.py --query-type Base --mode-tag xvec_only # x_vector_only_mode

# Batch (multiple prompts in one run)
python examples/offline_inference/text_to_speech/qwen3_tts/end2end.py --query-type CustomVoice --use-batch-sample
```

### Streaming
```bash
python examples/offline_inference/text_to_speech/qwen3_tts/end2end.py \
    --query-type CustomVoice \
    --streaming \
    --output-dir /tmp/out_stream
```
Streaming requires `async_chunk: true` in the deploy config.

### Word Timestamps
Word-level timestamps are currently a serving-path feature: launch
`vllm-omni serve` with `--forced-aligner` and request `word_timestamps`
(see `examples/online_serving/text_to_speech/README.md`). An offline
example is not provided in this release.

### Batched decoding
The Code2Wav stage supports batched decoding through the SpeechTokenizer. Pass multiple prompts via `--txt-prompts` and set `--batch-size` accordingly. To raise `max_num_seqs` on either stage, create an overlay based on `vllm_omni/deploy/qwen3_tts.yaml` and pass it with `--deploy-config`:
```bash
python examples/offline_inference/text_to_speech/qwen3_tts/end2end.py \
    --query-type CustomVoice \
    --txt-prompts examples/offline_inference/text_to_speech/qwen3_tts/benchmark_prompts.txt \
    --batch-size 4 \
    --deploy-config /path/to/qwen3_tts_batched.yaml
```
`--batch-size` must match a CUDA-graph capture size (1, 2, 4, 8, 16…).

### Notes
- Run `--help` for the full argument surface.
- See `qwen3_tts/end2end.py` for the prompt-length-estimation logic the Talker uses.

---

## Gepard-1.0

Single-stage native AR TTS at 22.05 kHz. Pipeline: `Qwen3.5 backbone → 32 FSQ codebook heads (one frame/step) → NeMo NanoCodec`. Backbone runs under vLLM paged attention; the 32-head sampling + learned embedding feedback ride the native-AR runner hooks. `enforce_eager` (CUDA graph is a perf follow-up).

### Prerequisites
The NanoCodec decoder needs NeMo (NVIDIA Open Model License), installed separately:

```bash
uv pip install "nemo_toolkit[tts]==2.7.3" \
    --overrides <(printf '%s\n' "transformers>=5.5.3" "huggingface-hub>=1.0")
```

The overrides are what make that command work. `nemo_toolkit[tts]` declares
`transformers~=4.57.0`, so a plain install downgrades the version
`requirements/common.txt` pins and takes `huggingface-hub` back to 0.x with it — after
which `import vllm` fails. Both packages are named here rather than left to resolve on
their own: transformers 5.x happens to require `huggingface-hub` 1.x, but relying on
that would make the second repair an accident of the first. `--constraint` cannot be
used instead, since it narrows a version range rather than overriding a dependency a
package declares, and the two sets never intersect. NeMo 2.7.3 decodes correctly
against transformers 5.x, so nothing is lost by holding it there.

(The same clash is why this is documented here rather than declared in
`pyproject.toml`: an extra resolves together with the base dependencies, so
`vllm-omni[gepard]` would be unsatisfiable rather than opt-in. SoulX-Singer English
SVS documents `nemo_toolkit[asr]` the same way.)

On a host whose CUDA toolkit cannot build kernels for the GPU — no `nvcc`/`ninja`, or a
consumer Blackwell (`sm_120`) card — also set `VLLM_USE_FLASHINFER_SAMPLER=0`. vLLM's
sampler JIT-compiles through FlashInfer on first use, and where that build fails the engine
dies inside `profile_run` long after every import has succeeded, which reads as a model
failure rather than an environment one. The PyTorch sampler it falls back to costs nothing
here.

### Quick start (zero-shot, default voice)
```bash
python examples/offline_inference/text_to_speech/gepard/end2end.py \
    --text "Hello, this is a Gepard demo."
```

### Voice cloning
Not yet — PR1 is zero-shot only (the learned `null_prefix` default voice). Reference-audio cloning (`ref_compressor` speaker prefix) is a follow-up PR.

### Notes
- Output: 22.05 kHz mono WAV.
- First run downloads two checkpoints, not one: the model itself, and the NanoCodec decoder it names in `codec_id` (`nvidia/nemo-nano-codec-22khz-1.89kbps-21.5fps` by default). Both are fetched automatically; expect the first startup to be correspondingly slower.
- Offline only for now; the `/v1/audio/speech` adapter is a follow-up PR.
- Deploy config: `vllm_omni/deploy/gepard.yaml` (the example's default; copy it and pass `--deploy-config` to change it).
- Generation length and reproducibility are stage settings, not script flags. One output token is one audio frame, so `max_tokens` in the YAML is the frame budget; `seed` makes the in-model 32-head sampling reproducible. The script deliberately passes no `SamplingParams`: one supplied by a caller replaces the stage defaults wholesale rather than merging, which would drop the pipeline's stop token and run every request to `max_tokens`.
- Text length: short texts are repeated internally to match the training layout (the checkpoint's `text_repetition` block); the upper bound is the stage's `max_model_len`, enforced by the engine. Empty text is rejected rather than voiced.
- `VLLM_GEPARD_GREEDY=1` swaps the 32-head Gumbel-max sampling for argmax, for reproducible comparisons that do not depend on a seed.

## VoxCPM2

Single-stage native AR TTS at 48 kHz. Pipeline: `feat_encoder → MiniCPM4 → FSQ → residual_lm → LocDiT → AudioVAE`.

### Prerequisites
```bash
pip install voxcpm
# or, for a local source checkout:
export VLLM_OMNI_VOXCPM_CODE_PATH=/path/to/voxcpm
```

### Quick start
```bash
python examples/offline_inference/text_to_speech/voxcpm2/end2end.py \
    --model openbmb/VoxCPM2 \
    --text "Hello, this is a VoxCPM2 demo."
```

### Voice cloning
Pass a reference audio for isolated cloning, or both `--ref-audio` + `--ref-text` for prompt continuation:
```bash
python examples/offline_inference/text_to_speech/voxcpm2/end2end.py \
    --text "Hello, this is a voice clone demo." \
    --ref-audio /path/to/reference.wav \
    --ref-text  "Transcript of the reference audio."
```

### Streaming
Streaming is exposed through the online OpenAI Speech API (`stream=true`). See [`examples/online_serving/text_to_speech/voxcpm2/gradio_demo.py`](../../online_serving/text_to_speech/voxcpm2/gradio_demo.py) for an AudioWorklet-based gapless streaming player; the offline `end2end.py` script does not expose a streaming path.

### Notes
- Output: 48 kHz mono WAV.
- Deploy config: `vllm_omni/deploy/voxcpm2.yaml` (auto-loaded by HF `model_type`).

---

## 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.

### Quick start

IndexTTS-2:

```bash
python examples/offline_inference/text_to_speech/indextts2/end2end.py \
    --model IndexTeam/IndexTTS-2 \
    --text "你好，这是一个语音合成测试。" \
    --ref-audio /path/to/reference.wav
```

IndexTTS-2 uses its existing frontend and does not require the IndexTTS-2.5 text dependencies. Before running IndexTTS-2.5, install its optional frontend dependencies:

```bash
pip install 'vllm-omni[indextts2]'
```

IndexTTS-2.5 with a native repository/lab bundle whose model root contains `checkpoints/`:

```bash
python examples/offline_inference/text_to_speech/indextts2/end2end.py \
    --model /path/to/indextts-2.5 \
    --model-version 2.5 \
    --lang ja \
    --text "こんにちは、音声合成のテストです。" \
    --ref-audio /path/to/reference.wav

```

### Emotion control
```bash
# Emotion from reference audio
python examples/offline_inference/text_to_speech/indextts2/end2end.py \
    --model IndexTeam/IndexTTS-2 \
    --text "今天天气真好！" \
    --ref-audio /path/to/ref.wav \
    --emo-audio /path/to/happy.wav

# Emotion from 8-dim vector (happy angry sad afraid disgusted melancholy surprised calm)
python examples/offline_inference/text_to_speech/indextts2/end2end.py \
    --model IndexTeam/IndexTTS-2 \
    --text "今天天气真好！" \
    --ref-audio /path/to/ref.wav \
    --emo-vector 1.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0

# Emotion from text description
python examples/offline_inference/text_to_speech/indextts2/end2end.py \
    --model IndexTeam/IndexTTS-2 \
    --text "今天天气真好！" \
    --ref-audio /path/to/ref.wav \
    --emo-text "happy and excited"
```

### Notes
- `--ref-audio` is **required** — neither version provides text-only synthesis.
- IndexTTS-2.5 Stage 0 (AR Talker) generates mel codes from text + reference audio using plain vLLM sampling. It does not reproduce the official default `num_beams=3` beam search. For parity comparisons, run upstream with `num_beams=1`; output quality can differ from the official beam-search result.
- IndexTTS-2.5 accepts language codes such as `zh`, `en`, `zhen` (mixed Chinese/English), `ja`, and `yue`. `Mandarin` is a vLLM-Omni convenience alias for `zh`. Japanese (`ja`) uses `fugashi` tokenization and produces audio, but it does not automatically expand numbers, dates, or percentages; write those inputs as readable Japanese text before calling the model. Use `--no-text-normalization` only when the input is already normalized.
- A seed controls Stage 0 AR sampling and per-request CFM noise. Different concurrent batch compositions do not guarantee a bit-identical waveform.
- Stage 1 (semantic codec + S2Mel + BigVGAN): CFM/DiT converts the complete semantic-code sequence to waveform.
- Deploy configs: `vllm_omni/deploy/indextts2.yaml` and `indextts2_5.yaml`.
- `async_chunk=false` is intentional for IndexTTS-2.5 correctness: S2Mel consumes the completed semantic sequence.

---

## Voxtral TTS

Voxtral-4B-TTS (Mistral). Has its own argparse surface; uses voice presets and the `mistral_common` `SpeechRequest` protocol.

### Prerequisites
Latest `mistral_common` with `SpeechRequest` support:
```bash
pip install -e /path/to/mistral-common  # or upgrade from PyPI when available
```

### Quick start (voice preset)
```bash
python examples/offline_inference/text_to_speech/voxtral_tts/end2end.py \
    --write-audio --voice cheerful_female \
    --model mistralai/Voxtral-4B-TTS-2603 \
    --text "That eerie silence after the first storm was just the calm before another round of chaos, wasn't it?"
```

### Voice cloning (capability gated upstream)
```bash
python examples/offline_inference/text_to_speech/voxtral_tts/end2end.py \
    --write-audio \
    --model mistralai/Voxtral-4B-TTS-2603 \
    --text "This is a test message." \
    --ref-audio path/to/reference_audio.wav
```

### Streaming + concurrency
```bash
python examples/offline_inference/text_to_speech/voxtral_tts/end2end.py \
    --num-prompts 32 --concurrency 8 --streaming --write-audio --voice neutral_female \
    --model mistralai/Voxtral-4B-TTS-2603 \
    --text "..."
```
Available voice presets are listed on the HF model card (`mistralai/Voxtral-4B-TTS-2603`).

### Notes
- `--num-prompts N` replicates the prompt for performance measurement.
- `--concurrency M` requires `--streaming` and must evenly divide `--num-prompts`.
- Run `--help` for the full argument surface.

---

## SoulX-Singer

Singing voice synthesis (SVS) and conversion (SVC) at 24 kHz. Script: `soulxsinger/end2end.py`. Deploy: `vllm_omni/deploy/soulxsinger_svs.yaml` or `soulxsinger_svc.yaml`.

### Prerequisites

Download DiT and preprocess weights, then set up separate SVS / SVC view directories. Copy `soulxsinger/utils/phoneme/phone_set.json` from upstream [SoulX-Singer](https://github.com/Soul-AILab/SoulX-Singer) into the model weights dir as `phoneme/phone_set.json` — HuggingFace does not ship it.

```bash
# 1. DiT weights
export BASE=path/to/SoulX-Singer
export PREPROCESS=path/to/SoulX-Singer-Preprocess
export SVC_DIR=path/to/SoulX-Singer-svc

huggingface-cli download Soul-AILab/SoulX-Singer --local-dir "$BASE"

# 2. Preprocess weights (required)
huggingface-cli download Soul-AILab/SoulX-Singer-Preprocess --local-dir "$PREPROCESS"
export SOULX_PREPROCESS_WEIGHTS_DIR="$PREPROCESS"

# 3. SVS / SVC view directories
mkdir -p "$SVC_DIR"
cp $BASE/{config.yaml,README.md,assets} $SVC_DIR
mv $BASE/model-svc.pt $SVC_DIR/model-svc.pt

cat > "$BASE/config.json" <<'EOF'
{
  "model_type": "soulxsinger",
  "architectures": ["SoulXSingerPipeline"],
  "max_num_seqs": 1
}
EOF

cat > "$SVC_DIR/config.json" <<'EOF'
{
  "model_type": "soulxsinger",
  "architectures": ["SoulXSingerSVCPipeline"],
  "max_num_seqs": 1
}
EOF
```

`config.yaml` hyper-parameters live under `$BASE`; each view's `config.json` `architectures` field is the single source of truth for SVS vs SVC. Point `--model` at the matching directory (`$BASE` for SVS, `$SVC_DIR` for SVC). Deploy YAML is chosen automatically from `config.json`; optional `--svs` / `--svc` only assert the mode matches.

**Online preprocess** is the default: pass `--prompt-audio` and `--target-audio`, and the worker runs vocal separation, F0, and (for SVS) lyrics/MIDI before DiT. Install only what your run needs:

```bash
pip install "BS-RoFormer"   # vocal sep + F0 on GPU — SVS and SVC
```

Mandarin SVS also needs FunASR and Chinese G2P; `ffmpeg` must be on `PATH`:

```bash
# install optional dependencies:
pip install -e ".[soulx-svs]"
```

English SVS adds NeMo ASR and NLTK data; pass `--language English`:

```bash
pip install "nemo_toolkit[asr]==2.6.1" lhotse==1.32.2
python -c "import nltk; nltk.download('cmudict'); nltk.download('averaged_perceptron_tagger_eng')"
```

**Precomputed metadata** is the alternative: pass both `--prompt-metadata-path` and `--target-metadata-path` and skip online ASR/ROSVOT — none of the packages above are required. JSON can be produced by integrated preprocess on a prior run, or by upstream [SoulX-Singer](https://github.com/Soul-AILab/SoulX-Singer) `preprocess/` scripts if you prefer to run that outside vLLM-Omni.

### Quick start

```bash
# SVS — default demo audio: tests/assets/soulxsinger/zh_prompt.mp3 + music.mp3
python examples/offline_inference/text_to_speech/soulxsinger/end2end.py \
    --model "$BASE" \
    --preprocess-weights-dir "$PREPROCESS" \
    --control score \
    --num-inference-steps 32 \
    -o output.wav

python examples/offline_inference/text_to_speech/soulxsinger/end2end.py \
    --model "$SVC_DIR" \
    --preprocess-weights-dir "$PREPROCESS" \
    --svc \
    --num-inference-steps 32 \
    -o output_svc.wav
```

`SOULX_PREPROCESS_WEIGHTS_DIR` makes `--preprocess-weights-dir` optional. Long SVS targets are handled in one request. See `end2end.py --help` for `--pitch-shift`, `--vocal-sep`, `--auto-shift`, and language/control options.

### Notes

- Output: 24 kHz mono WAV; batch only.
- Defaults match upstream: `--guidance-scale 3.0`, `--seed 42`, `--auto-shift` on.
- SVS `--control`: `score` or `melody`. MIDI / lyric QC: upstream `midi_editor` only.

---