[Higgs-Audio] bosonai/higgs-audio-v2-generation-3B-base TTS model support (#3762)

Signed-off-by: Yuekai Zhang <zhangyuekai@foxmail.com>
Co-authored-by: Claude Opus 4.7 (1M context) <noreply@anthropic.com>
Co-authored-by: Yueqian Lin <70319226+linyueqian@users.noreply.github.com>
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# higgs-audio v2 — offline example
Drives Stage 0 (DualFFN talker) + Stage 1 (HiggsAudio codec) for `bosonai/higgs-audio-v2-generation-3B-base` end-to-end through the vLLM-Omni engine and writes a 24 kHz mono WAV per prompt.
## Prerequisites
Voice clone needs `transformers>=5.3.0` — vllm-omni loads the audio codec via HF's `HiggsAudioV2TokenizerModel`, instantiated from the `k2-fsa/OmniVoice/audio_tokenizer/` subdirectory (only that ~806 MB subdir is downloaded). The boson-ai standalone tokenizer repo's `model.safetensors` is actually a copy of the 3B talker LM, so HF can't load it directly; the k2 bundle ships the same codec weights repackaged with HF-compatible key naming.
```bash
pip install -U "transformers>=5.3.0"
```
## Quick start
Plain TTS:
```bash
python examples/offline_inference/text_to_speech/higgs_audio_v2/end2end.py \
--texts "Hello world." "The quick brown fox jumps over the lazy dog." \
--output-dir results/wavs
```
## Voice cloning
Pass both `--ref-audio` and `--ref-text` together:
```bash
python examples/offline_inference/text_to_speech/higgs_audio_v2/end2end.py \
--texts "Hello, this is a cloned voice." \
--ref-audio /path/to/reference.wav \
--ref-text "Exact transcript spoken in reference.wav." \
--output-dir results/wavs
```
## Notes
- Output: 24 kHz mono WAV.
- Deploy config: `vllm_omni/deploy/higgs_audio_v2.yaml` (auto-loaded by HF `model_type`).
- `--ref-text` must be the real transcript of `--ref-audio`; mismatched text degrades cloned-voice quality.
- For online serving, see [`examples/online_serving/text_to_speech/higgs_audio_v2/`](../../../online_serving/text_to_speech/higgs_audio_v2/).
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# SPDX-License-Identifier: Apache-2.0
# SPDX-FileCopyrightText: Copyright contributors to the vLLM project
"""Offline higgs-audio v2 inference example.
Runs Stage 0 (DualFFN talker) + Stage 1 (HiggsAudio codec) end-to-end
through the vLLM-Omni engine without going through the HTTP server, and
saves a 24 kHz mono WAV per prompt.
Example:
python examples/offline_inference/text_to_speech/higgs_audio_v2/end2end.py \\
--texts "Hello world." \\
"The quick brown fox jumps over the lazy dog." \\
--output-dir results/wavs
"""
from __future__ import annotations
import os
# DeepGEMM FP8 kernels require an optional backend that may not be installed.
# Disable the warmup before importing vLLM so engine startup falls back to the
# regular gemm path. Users with deep_gemm installed can override these.
os.environ.setdefault("VLLM_USE_DEEP_GEMM", "0")
os.environ.setdefault("VLLM_MOE_USE_DEEP_GEMM", "0")
import time
from pathlib import Path
import numpy as np
import soundfile as sf
import torch
from vllm.utils.argparse_utils import FlexibleArgumentParser
from vllm_omni import Omni
from vllm_omni.engine.arg_utils import nullify_stage_engine_defaults
SAMPLE_RATE = 24_000
DEFAULT_TEXTS = (
"Hello world.",
"The quick brown fox jumps over the lazy dog.",
)
def parse_args():
parser = FlexibleArgumentParser(description="Offline higgs-audio v2 inference")
parser.add_argument(
"--model",
type=str,
default="bosonai/higgs-audio-v2-generation-3B-base",
help="Stage-0 talker model id or path.",
)
parser.add_argument(
"--texts",
type=str,
nargs="+",
default=list(DEFAULT_TEXTS),
help="One or more plain-text prompts to synthesize.",
)
parser.add_argument(
"--output-dir",
type=str,
default="results/wavs",
help="Directory to write per-prompt WAV files.",
)
parser.add_argument(
"--deploy-config",
type=str,
default=None,
help="Override the deploy config path. Auto-loads "
"vllm_omni/deploy/higgs_audio_v2.yaml from the HF model_type by default.",
)
parser.add_argument(
"--max-new-tokens",
type=int,
default=500,
help="Cap on Stage-0 codec frames per request.",
)
parser.add_argument(
"--ref-audio",
type=str,
default=None,
help="Reference clip for voice clone (path to a WAV file). Paired with --ref-text.",
)
parser.add_argument(
"--ref-text",
type=str,
default=None,
help="Transcript of the reference clip. Required when --ref-audio is set.",
)
nullify_stage_engine_defaults(parser)
return parser.parse_args()
def _slugify(text: str) -> str:
import re
slug = re.sub(r"\s+", "_", text.strip().lower())
slug = re.sub(r"[^a-z0-9_]+", "", slug)
return slug[:48] or "prompt"
def _extract_pcm(multimodal_output: dict) -> torch.Tensor:
"""Pull the final concatenated PCM tensor out of a request's multimodal_output."""
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")
return torch.cat(valid, dim=0) if len(valid) > 1 else valid[0]
return torch.as_tensor(audio).float().cpu().reshape(-1)
def _pcm_to_int16(pcm: torch.Tensor) -> np.ndarray:
arr = pcm.numpy()
if arr.dtype.kind == "f":
arr = np.clip(arr, -1.0, 1.0)
arr = (arr * 32767.0).astype(np.int16)
else:
arr = arr.astype(np.int16)
return arr
def main():
args = parse_args()
if (args.ref_audio is None) != (args.ref_text is None):
raise SystemExit("--ref-audio and --ref-text must be supplied together")
output_dir = Path(args.output_dir)
output_dir.mkdir(parents=True, exist_ok=True)
engine = Omni(model=args.model, deploy_config=args.deploy_config)
# Build prompt_token_ids using the same path serving_speech.py takes online.
from transformers import AutoProcessor
from vllm_omni.model_executor.models.higgs_audio_v2.higgs_audio_v2_tokenizer import (
build_plain_text_prompt,
build_voice_clone_prompt,
input_ids_to_python_list,
)
processor = AutoProcessor.from_pretrained(args.model, trust_remote_code=True)
# Voice-clone path: load the reference clip once. The HF processor will
# encode it via the bundled HiggsAudioV2TokenizerModel each time we build
# a prompt below. This is cheap on CPU for a few-second clip.
ref_wav: np.ndarray | None = None
ref_sr: int | None = None
if args.ref_audio is not None:
ref_wav, ref_sr = sf.read(args.ref_audio, always_2d=False)
if ref_wav.ndim == 2:
ref_wav = ref_wav.mean(axis=1)
print(f"Model : {args.model}")
print(f"Prompts : {len(args.texts)}")
print(f"Output dir : {output_dir}")
print(f"Voice clone : {'yes' if ref_wav is not None else 'no'}")
# Run one prompt at a time. The Stage-0 talker's per-slot audio state is
# request-scoped; submitting multiple prompts in the same engine.generate()
# call would batch them in the AR runner and exercise a code path that is
# not validated for this model yet.
total_elapsed = 0.0
total_dur = 0.0
for text in args.texts:
if ref_wav is not None:
out = build_voice_clone_prompt(processor, text, ref_wav, int(ref_sr), args.ref_text)
prompt = {
"prompt_token_ids": out["prompt_token_ids"],
# Bare tensors (NOT list-wrapped): the msgspec serializer in
# vllm_omni.data_entry_keys routes torch.Tensor → tensor_data
# and list[Tensor] → list_data (which silently strips tensors).
"additional_information": {
"audio_input_ids": out["audio_input_ids"],
"audio_input_ids_mask": out["audio_input_ids_mask"],
},
}
else:
inputs = build_plain_text_prompt(processor, text)
prompt = {"prompt_token_ids": input_ids_to_python_list(inputs)}
t_start = time.perf_counter()
outputs = engine.generate([prompt])
elapsed = time.perf_counter() - t_start
total_elapsed += elapsed
mm = outputs[0].outputs[0].multimodal_output
pcm = _extract_pcm(mm)
slug = _slugify(text)
out_path = output_dir / f"{slug}.wav"
sf.write(str(out_path), _pcm_to_int16(pcm), SAMPLE_RATE, format="WAV", subtype="PCM_16")
dur = pcm.numel() / SAMPLE_RATE
total_dur += dur
print(f" {slug:<50} dur={dur:5.2f}s -> {out_path}")
rtf = total_elapsed / total_dur if total_dur > 0 else float("inf")
print(f"Total infer : {total_elapsed:.2f}s total audio: {total_dur:.2f}s RTF: {rtf:.3f}")
if __name__ == "__main__":
os.environ.setdefault("VLLM_WORKER_MULTIPROC_METHOD", "spawn")
main()
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# higgs-audio v2 online example
This directory contains the online-serving entry points for boson-ai's higgs-audio v2 as integrated by vllm-omni: a 2-stage TTS pipeline (Llama-3.2-3B talker with DualFFN audio expert + HiggsAudio codec decoder) emitting 24 kHz mono speech.
## Prerequisites
Voice clone uses HF's `HiggsAudioV2TokenizerModel` loaded from `k2-fsa/OmniVoice/audio_tokenizer/` (the boson-ai standalone tokenizer Hub repo's `model.safetensors` is the 3B talker LM, not the codec). Only that ~806 MB subdir is downloaded.
```bash
pip install -U "transformers>=5.3.0"
```
## Files
- `run_server.sh` — launch the vllm-omni server with the bundled `vllm_omni/deploy/higgs_audio_v2.yaml` deploy config.
- `batch_speech_client.py` — send a list of prompts to `/v1/audio/speech` and save the returned WAV / PCM bytes to a directory; optionally passes `--ref-audio` + `--ref-text` for shallow voice clone.
## Launching the server
```bash
GPUS=6,7 PORT=8094 bash examples/online_serving/text_to_speech/higgs_audio_v2/run_server.sh
```
Environment overrides:
- `MODEL` — HF id of the talker (default `bosonai/higgs-audio-v2-generation-3B-base`).
- `PORT` — server port (default `8094`).
- `GPUS` — `CUDA_VISIBLE_DEVICES` value (default `6,7`).
- `GPU_UTIL` — `--gpu-memory-utilization` (default `0.4`).
The script also exports `VLLM_USE_DEEP_GEMM=0` / `VLLM_MOE_USE_DEEP_GEMM=0` so the example works on images without the optional `deep_gemm` backend.
The deploy YAML ships with `async_chunk: false` and `codec_streaming: true`, i.e. Stage 0 finishes its codec frames before Stage 1 starts decoding, and Stage 1 streams WAV/PCM bytes to the client chunk-by-chunk.
## Driving the server
Plain TTS:
```bash
python examples/online_serving/text_to_speech/higgs_audio_v2/batch_speech_client.py \
--base-url http://localhost:8094 \
--model bosonai/higgs-audio-v2-generation-3B-base \
--output-dir /tmp/higgs_audio_v2_batch \
--prompts "Hello world." \
"The quick brown fox jumps over the lazy dog."
```
Voice clone — pass a reference clip and its transcript (both required together):
```bash
python examples/online_serving/text_to_speech/higgs_audio_v2/batch_speech_client.py \
--base-url http://localhost:8094 \
--model bosonai/higgs-audio-v2-generation-3B-base \
--output-dir /tmp/higgs_audio_v2_clone \
--ref-audio /path/to/reference.wav \
--ref-text "Exact transcript spoken in reference.wav." \
--prompts "Hello, this is a cloned voice."
```
## Notes
- `--ref-text` must be the real transcript of `--ref-audio`; mismatched text degrades cloned-voice quality.
- Out of scope (rejected with explicit 4xx by the request validator): multi-speaker `[SPEAKERn]` tags inside `input`, `profile:` text-only speaker descriptions, the `ref_audio_in_system_message` system-block variant, chunked long-form generation, and per-request `voice` / `instructions` / `task_type` / `language` / `speed != 1.0` / `x_vector_only_mode` / `speaker_embedding`.
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# SPDX-License-Identifier: Apache-2.0
# SPDX-FileCopyrightText: Copyright contributors to the vLLM project
"""Batch client for the higgs-audio v2 online server.
Sends a fixed list of prompts to ``/v1/audio/speech`` and saves the returned
WAV files (or raw PCM bytes when ``--format pcm``) into ``--output-dir``.
Usage (plain text -> speech):
python examples/online_serving/text_to_speech/higgs_audio_v2/batch_speech_client.py \
--base-url http://localhost:8094 \
--output-dir /tmp/higgs_audio_v2_batch \
--prompts "Hello world." "The quick brown fox jumps over the lazy dog."
Usage (shallow voice clone — pass a reference clip + its transcript):
python examples/online_serving/text_to_speech/higgs_audio_v2/batch_speech_client.py \
--base-url http://localhost:8094 \
--output-dir /tmp/higgs_audio_v2_clone \
--ref-audio path/to/reference.wav \
--ref-text "the transcript of the reference clip" \
--prompts "Hello world."
"""
from __future__ import annotations
import argparse
import base64
import sys
from pathlib import Path
DEFAULT_PROMPTS = (
"Hello world.",
"The quick brown fox jumps over the lazy dog.",
"It was the night before my birthday.",
"Innovation distinguishes between a leader and a follower.",
)
def _slug(text: str) -> str:
import re
s = re.sub(r"\s+", "_", text.strip().lower())
return re.sub(r"[^a-z0-9_]+", "", s)[:32] or "prompt"
def main() -> int:
parser = argparse.ArgumentParser(description=__doc__, formatter_class=argparse.RawDescriptionHelpFormatter)
parser.add_argument("--base-url", default="http://localhost:8094")
parser.add_argument("--model", default="higgs_audio_v2")
parser.add_argument("--prompts", nargs="+", default=list(DEFAULT_PROMPTS))
parser.add_argument("--output-dir", type=Path, default=Path("/tmp/higgs_audio_v2_batch"))
parser.add_argument("--format", choices=("wav", "pcm"), default="wav")
parser.add_argument("--max-new-tokens", type=int, default=300)
parser.add_argument("--seed", type=int, default=42)
parser.add_argument("--timeout-s", type=float, default=120.0)
parser.add_argument(
"--ref-audio",
type=Path,
default=None,
help="Reference clip for voice clone (path to a WAV file). Must be paired with --ref-text.",
)
parser.add_argument(
"--ref-text",
type=str,
default=None,
help="Transcript of the reference clip. Required when --ref-audio is set.",
)
args = parser.parse_args()
if (args.ref_audio is None) != (args.ref_text is None):
print("--ref-audio and --ref-text must be supplied together", file=sys.stderr)
return 2
ref_audio_data_url: str | None = None
if args.ref_audio is not None:
if not args.ref_audio.exists():
print(f"ref-audio file not found: {args.ref_audio}", file=sys.stderr)
return 2
mime = "audio/wav" if args.ref_audio.suffix.lower() == ".wav" else "audio/mpeg"
ref_b64 = base64.b64encode(args.ref_audio.read_bytes()).decode("ascii")
ref_audio_data_url = f"data:{mime};base64,{ref_b64}"
try:
import httpx
except ImportError:
print(
"this client needs `httpx`. Install with `pip install httpx`.",
file=sys.stderr,
)
return 2
args.output_dir.mkdir(parents=True, exist_ok=True)
url = args.base_url.rstrip("/") + "/v1/audio/speech"
failures = 0
with httpx.Client(timeout=args.timeout_s) as client:
for prompt in args.prompts:
payload = {
"model": args.model,
"input": prompt,
"response_format": args.format,
"max_new_tokens": args.max_new_tokens,
"seed": args.seed,
}
if ref_audio_data_url is not None:
payload["ref_audio"] = ref_audio_data_url
payload["ref_text"] = args.ref_text
resp = client.post(url, json=payload)
if resp.status_code != 200:
print(f"[FAIL] {prompt!r} -> {resp.status_code}: {resp.text[:200]}", file=sys.stderr)
failures += 1
continue
suffix = ".wav" if args.format == "wav" else ".pcm"
out = args.output_dir / f"{_slug(prompt)}{suffix}"
out.write_bytes(resp.content)
print(f"[ ok ] {prompt!r} -> {out} ({len(resp.content)} bytes)")
return 1 if failures else 0
if __name__ == "__main__":
sys.exit(main())
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#!/bin/bash
# Launch vLLM-Omni server for higgs-audio v2.
#
# v1 scope: plain text -> 24 kHz speech only. Voice cloning, multi-speaker,
# ChatML rich content, and language overrides are rejected by the validator
# with explicit 4xx (see vllm_omni/entrypoints/openai/serving_speech.py).
#
# Usage:
# ./run_server.sh # default port 8094, GPUs 6 and 7
# PORT=8095 GPUS=6,7 ./run_server.sh
# MODEL=bosonai/higgs-audio-v2-generation-3B-base ./run_server.sh
set -e
MODEL="${MODEL:-bosonai/higgs-audio-v2-generation-3B-base}"
PORT="${PORT:-8094}"
GPUS="${GPUS:-6,7}"
GPU_UTIL="${GPU_UTIL:-0.4}"
echo "Starting higgs-audio v2 server"
echo " MODEL=$MODEL"
echo " PORT=$PORT"
echo " CUDA_VISIBLE_DEVICES=$GPUS"
# DeepGEMM FP8 kernels are optional and trip warmup on builds without
# the deep_gemm backend; disable them so the example works out of the box.
# Users with deep_gemm installed can re-enable via the same env vars.
CUDA_VISIBLE_DEVICES="$GPUS" \
VLLM_USE_DEEP_GEMM=0 \
VLLM_MOE_USE_DEEP_GEMM=0 \
vllm-omni serve "$MODEL" \
--deploy-config vllm_omni/deploy/higgs_audio_v2.yaml \
--host 0.0.0.0 \
--port "$PORT" \
--gpu-memory-utilization "$GPU_UTIL" \
--trust-remote-code \
--omni