[Feature] timestamps for TTS via a forced-aligner pooling (#4795)

Signed-off-by: wjinxu <1299461899@qq.com>
Co-authored-by: xurui <xurui1@modelbest.onaliyun.com>
Co-authored-by: Claude Opus 4.8 (1M context) <noreply@anthropic.com>

examples/offline_inference/text_to_speech/README.md

@@ -372,18 +372,10 @@ python examples/offline_inference/text_to_speech/qwen3_tts/end2end.py \
 Streaming requires `async_chunk: true` in the stage config.
 
 ### Word Timestamps
-Generate a WAV offline and a JSON sidecar with word-level timestamps from
-`Qwen/Qwen3-ForcedAligner-0.6B`:
-```bash
-python examples/offline_inference/text_to_speech/qwen3_tts/word_timestamps.py \
-    --model Qwen/Qwen3-TTS-12Hz-1.7B-CustomVoice \
-    --forced-aligner Qwen/Qwen3-ForcedAligner-0.6B \
-    --text "Hello world." \
-    --output-dir /tmp/qwen3_tts_timestamps
-```
-The script writes `qwen3_tts_word_timestamps.wav` and
-`qwen3_tts_word_timestamps.json`. On machines without a local CUDA toolkit,
-set `VLLM_USE_FLASHINFER_SAMPLER=0` to avoid FlashInfer sampler JIT.
+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, point `--stage-configs-path` at a stage configs YAML with the desired values (see `vllm_omni/model_executor/stage_configs/` for templates):

examples/offline_inference/text_to_speech/qwen3_tts/word_timestamps.py

@@ -1,172 +0,0 @@
-"""Offline Qwen3-TTS example with word-level timestamps.
-
-This demo runs Qwen3-TTS offline, aligns the synthesized audio in memory with
-the same shared forced aligner used by the streaming server path, then writes
-the WAV and JSON sidecar:
-
-    python examples/offline_inference/text_to_speech/qwen3_tts/word_timestamps.py \
-        --model Qwen/Qwen3-TTS-12Hz-1.7B-CustomVoice \
-        --forced-aligner Qwen/Qwen3-ForcedAligner-0.6B
-
-On machines without a local CUDA toolkit, set ``VLLM_USE_FLASHINFER_SAMPLER=0``
-to avoid FlashInfer sampler JIT during warmup.
-"""
-
-from __future__ import annotations
-
-import asyncio
-import json
-import os
-from pathlib import Path
-from typing import Any
-
-os.environ.setdefault("VLLM_WORKER_MULTIPROC_METHOD", "spawn")
-# The in-process offline aligner consumes token_classify outputs. Keeping V1
-# multiprocessing off avoids msgspec serialization issues for token-wise logits.
-os.environ.setdefault("VLLM_ENABLE_V1_MULTIPROCESSING", "0")
-
-import numpy as np
-import soundfile as sf
-import torch
-from end2end import _estimate_prompt_len
-from vllm.utils.argparse_utils import FlexibleArgumentParser
-
-from vllm_omni import Omni
-from vllm_omni.engine.arg_utils import nullify_stage_engine_defaults
-from vllm_omni.utils.forced_aligner import align, build_forced_aligner_config
-
-
-def _default_stage_config() -> str:
-    repo_root = Path(__file__).resolve().parents[4]
-    return str(repo_root / "vllm_omni" / "deploy" / "qwen3_tts.yaml")
-
-
-def _build_custom_voice_input(args: Any) -> dict[str, Any]:
-    additional_information = {
-        "task_type": ["CustomVoice"],
-        "text": [args.text],
-        "language": [args.language],
-        "speaker": [args.speaker],
-        "instruct": [args.instructions],
-        "max_new_tokens": [args.max_new_tokens],
-    }
-    return {
-        "prompt_token_ids": [0] * _estimate_prompt_len(additional_information, args.model),
-        "additional_information": additional_information,
-    }
-
-
-def _audio_tensor_and_sample_rate(mm: dict[str, Any]) -> tuple[torch.Tensor, int]:
-    audio_data = mm["audio"]
-    sr_raw = mm["sr"]
-    sr_val = sr_raw[-1] if isinstance(sr_raw, list) and sr_raw else sr_raw
-    sample_rate = int(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
-    return audio_tensor.float().detach().cpu().flatten(), sample_rate
-
-
-def _float_audio_to_pcm16_bytes(audio: torch.Tensor) -> bytes:
-    samples = audio.numpy()
-    pcm = (np.clip(samples, -1.0, 1.0) * 32767.0).astype("<i2")
-    return pcm.tobytes()
-
-
-async def _run_alignment(args: Any, audio: torch.Tensor, sample_rate: int) -> list[dict[str, Any]] | None:
-    aligner_config = build_forced_aligner_config(args)
-    if aligner_config is None:
-        raise ValueError("--forced-aligner or --forced-aligner-config is required")
-    timestamps = await align(
-        audio=_float_audio_to_pcm16_bytes(audio),
-        text=args.text,
-        sample_rate=sample_rate,
-        config=aligner_config,
-        language=args.language,
-    )
-    if timestamps is None:
-        return None
-    return [
-        {
-            "word": item.word,
-            "start_ms": item.start_ms,
-            "end_ms": item.end_ms,
-        }
-        for item in timestamps
-    ]
-
-
-def main(args: Any) -> None:
-    output_dir = Path(args.output_dir)
-    output_dir.mkdir(parents=True, exist_ok=True)
-
-    omni_kwargs = vars(args).copy()
-    for key in (
-        "forced_aligner",
-        "text",
-        "language",
-        "speaker",
-        "instructions",
-        "max_new_tokens",
-        "output_dir",
-    ):
-        omni_kwargs.pop(key, None)
-    omni_kwargs["stage_configs_path"] = args.stage_configs_path or _default_stage_config()
-    omni_kwargs["log_stats"] = args.log_stats
-    omni = Omni(**omni_kwargs)
-
-    prompt = _build_custom_voice_input(args)
-    final_output = None
-    for stage_outputs in omni.generate([prompt]):
-        final_output = stage_outputs.request_output
-    if final_output is None:
-        raise RuntimeError("Qwen3-TTS did not produce an output.")
-
-    mm = final_output.outputs[0].multimodal_output
-    audio, sample_rate = _audio_tensor_and_sample_rate(mm)
-
-    wav_path = output_dir / "qwen3_tts_word_timestamps.wav"
-    # Align the in-memory tensor, not the saved WAV. The WAV is only a demo
-    # artifact so users can listen to the same audio referenced by the sidecar.
-    timestamps = asyncio.run(_run_alignment(args, audio, sample_rate))
-    sf.write(wav_path, audio.numpy(), samplerate=sample_rate, format="WAV")
-
-    sidecar = {
-        "text": args.text,
-        "sample_rate": sample_rate,
-        "audio_path": str(wav_path),
-        "timestamps": timestamps,
-    }
-    json_path = output_dir / "qwen3_tts_word_timestamps.json"
-    json_path.write_text(json.dumps(sidecar, ensure_ascii=False, indent=2) + "\n", encoding="utf-8")
-
-    print(f"Saved audio: {wav_path}")
-    print(f"Saved timestamps: {json_path}")
-    print(json.dumps(sidecar["timestamps"], ensure_ascii=False, indent=2))
-
-
-def parse_args() -> Any:
-    parser = FlexibleArgumentParser(description="Offline Qwen3-TTS word timestamp example")
-    parser.add_argument("--model", default="Qwen/Qwen3-TTS-12Hz-1.7B-CustomVoice", help="Qwen3-TTS model path/name")
-    parser.add_argument(
-        "--forced-aligner",
-        default=None,
-        help="Qwen3 forced aligner model path/name",
-    )
-    parser.add_argument(
-        "--forced-aligner-config",
-        default=None,
-        help="Optional YAML file for forced aligner settings (incl. gpu_memory_utilization)",
-    )
-    parser.add_argument("--stage-configs-path", default=None, help="Qwen3-TTS deploy YAML")
-    parser.add_argument("--text", default="Hello world.", help="Text to synthesize and align")
-    parser.add_argument("--language", default="English", help="Qwen3-TTS language field")
-    parser.add_argument("--speaker", default="Vivian", help="CustomVoice speaker name")
-    parser.add_argument("--instructions", default="", help="Optional speaking style instruction")
-    parser.add_argument("--max-new-tokens", type=int, default=2048, help="TTS max_new_tokens")
-    parser.add_argument("--output-dir", default="output_audio", help="Directory for WAV and JSON sidecar")
-    parser.add_argument("--log-stats", action="store_true", default=False, help="Enable vLLM-Omni stats logging")
-    nullify_stage_engine_defaults(parser)
-    return parser.parse_args()
-
-
-if __name__ == "__main__":
-    main(parse_args())

examples/online_serving/text_to_speech/README.md

@@ -605,6 +605,13 @@ python qwen3_tts/streaming_speech_client.py \
 The client writes one PCM file per sentence and a matching
 `sentence_XXX_timestamps.json` sidecar.
 
+Non-streaming requests can also ask for timestamps: pass
+`"word_timestamps": true` to `POST /v1/audio/speech` and read the
+`X-Word-Timestamps` response header (JSON list of `{word, start_ms, end_ms}`,
+ASCII-escaped). Past 4 KB the header is replaced by
+`X-Word-Timestamps-Omitted: oversize; bytes=<n>; limit=4096` and the audio
+still returns — use the WebSocket path for long transcripts.
+
 To *see* the alignment instead of reading a JSON sidecar, run the
 word-timestamp Gradio demo (server must be launched with `--forced-aligner`):
 ```bash