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vllm-omni/vllm_omni/entrypoints/openai/serving_speech.py
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17 KiB
Python

import asyncio
import base64
import io
import ipaddress
import socket
from typing import Any
from urllib.parse import urlparse
from urllib.request import urlopen
import numpy as np
import soundfile as sf
from fastapi import Request
from fastapi.responses import Response
from vllm.entrypoints.openai.engine.serving import OpenAIServing
from vllm.logger import init_logger
from vllm.utils import random_uuid
from vllm_omni.entrypoints.openai.audio_utils_mixin import AudioMixin
from vllm_omni.entrypoints.openai.protocol.audio import (
AudioResponse,
CreateAudio,
OpenAICreateSpeechRequest,
)
from vllm_omni.outputs import OmniRequestOutput
logger = init_logger(__name__)
_REF_AUDIO_TIMEOUT_S = 15
_REF_AUDIO_MAX_BYTES = 50 * 1024 * 1024 # 50 MB
_REF_AUDIO_BLOCKED_NETWORKS = [
ipaddress.ip_network("127.0.0.0/8"),
ipaddress.ip_network("10.0.0.0/8"),
ipaddress.ip_network("172.16.0.0/12"),
ipaddress.ip_network("192.168.0.0/16"),
ipaddress.ip_network("169.254.0.0/16"),
ipaddress.ip_network("::1/128"),
ipaddress.ip_network("fc00::/7"),
ipaddress.ip_network("fe80::/10"),
]
# TTS Configuration (currently supports Qwen3-TTS)
_TTS_MODEL_STAGES: set[str] = {"qwen3_tts"}
_TTS_LANGUAGES: set[str] = {
"Auto",
"Chinese",
"English",
"Japanese",
"Korean",
"German",
"French",
"Russian",
"Portuguese",
"Spanish",
"Italian",
}
_TTS_MAX_INSTRUCTIONS_LENGTH = 500
_TTS_MAX_NEW_TOKENS_MIN = 1
_TTS_MAX_NEW_TOKENS_MAX = 4096
class OmniOpenAIServingSpeech(OpenAIServing, AudioMixin):
def __init__(self, *args, **kwargs):
super().__init__(*args, **kwargs)
# Load supported speakers
self.supported_speakers = self._load_supported_speakers()
logger.info(f"Loaded {len(self.supported_speakers)} supported speakers: {sorted(self.supported_speakers)}")
self._tts_tokenizer = None
def _load_supported_speakers(self) -> set[str]:
"""Load supported speakers (case-insensitive) from the model configuration."""
try:
talker_config = self.engine_client.model_config.hf_config.talker_config
# Check for speakers in either spk_id or speaker_id
for attr_name in ["spk_id", "speaker_id"]:
speakers_dict = getattr(talker_config, attr_name, None)
if speakers_dict and isinstance(speakers_dict, dict):
# Normalize to lowercase for case-insensitive matching
return {speaker.lower() for speaker in speakers_dict.keys()}
logger.warning("No speakers found in talker_config (checked spk_id and speaker_id)")
except Exception as e:
logger.warning(f"Could not load speakers from model config: {e}")
return set()
def _estimate_prompt_len(self, tts_params: dict[str, Any]) -> int:
"""Estimate prompt length so the placeholder matches model-side embeddings."""
try:
from vllm_omni.model_executor.models.qwen3_tts.qwen3_tts_talker import (
Qwen3TTSTalkerForConditionalGeneration,
)
if self._tts_tokenizer is None:
from transformers import AutoTokenizer
model_name = self.engine_client.model_config.model
self._tts_tokenizer = AutoTokenizer.from_pretrained(
model_name,
trust_remote_code=True,
padding_side="left",
)
hf_config = self.engine_client.model_config.hf_config
talker_config = hf_config.talker_config
task_type = (tts_params.get("task_type") or ["CustomVoice"])[0]
return Qwen3TTSTalkerForConditionalGeneration.estimate_prompt_len_from_additional_information(
additional_information=tts_params,
task_type=task_type,
tokenize_prompt=lambda t: self._tts_tokenizer(t, padding=False)["input_ids"],
codec_language_id=getattr(talker_config, "codec_language_id", None),
spk_is_dialect=getattr(talker_config, "spk_is_dialect", None),
)
except Exception as e:
logger.warning("Failed to estimate TTS prompt length, using fallback 2048: %s", e)
return 2048
def _is_tts_model(self) -> bool:
"""Check if the current model is a supported TTS model."""
stage_list = getattr(self.engine_client, "stage_list", None)
if stage_list:
for stage in stage_list:
model_stage = getattr(stage, "model_stage", None)
if model_stage in _TTS_MODEL_STAGES:
return True
return False
def _validate_tts_request(self, request: OpenAICreateSpeechRequest) -> str | None:
"""Validate TTS request parameters. Returns error message or None."""
task_type = request.task_type or "CustomVoice"
# Normalize voice to lowercase for case-insensitive matching
if request.voice is not None:
request.voice = request.voice.lower()
# Validate input is not empty
if not request.input or not request.input.strip():
return "Input text cannot be empty"
# Validate language
if request.language is not None and request.language not in _TTS_LANGUAGES:
return f"Invalid language '{request.language}'. Supported: {', '.join(sorted(_TTS_LANGUAGES))}"
# Validate speaker for CustomVoice task
if task_type == "CustomVoice" and request.voice is not None:
if self.supported_speakers and request.voice not in self.supported_speakers:
return f"Invalid speaker '{request.voice}'. Supported: {', '.join(sorted(self.supported_speakers))}"
# Validate Base task requirements
if task_type == "Base":
if request.ref_audio is None:
return "Base task requires 'ref_audio' for voice cloning"
# Validate ref_audio format
if not (request.ref_audio.startswith(("http://", "https://")) or request.ref_audio.startswith("data:")):
return "ref_audio must be a URL (http/https) or base64 data URL (data:...)"
# Validate cross-parameter dependencies
if task_type != "Base":
if request.ref_text is not None:
return "'ref_text' is only valid for Base task"
if request.x_vector_only_mode is not None:
return "'x_vector_only_mode' is only valid for Base task"
# Validate VoiceDesign task requirements
if task_type == "VoiceDesign" and not request.instructions:
return "VoiceDesign task requires 'instructions' to describe the voice"
# Validate instructions length
if request.instructions and len(request.instructions) > _TTS_MAX_INSTRUCTIONS_LENGTH:
return f"Instructions too long (max {_TTS_MAX_INSTRUCTIONS_LENGTH} characters)"
# Validate max_new_tokens range
if request.max_new_tokens is not None:
if request.max_new_tokens < _TTS_MAX_NEW_TOKENS_MIN:
return f"max_new_tokens must be at least {_TTS_MAX_NEW_TOKENS_MIN}"
if request.max_new_tokens > _TTS_MAX_NEW_TOKENS_MAX:
return f"max_new_tokens cannot exceed {_TTS_MAX_NEW_TOKENS_MAX}"
return None
@staticmethod
async def _resolve_ref_audio(ref_audio_str: str) -> tuple[list[float], int]:
"""Resolve ref_audio URL/base64 to (wav_samples, sample_rate)."""
parsed = urlparse(ref_audio_str)
def _check_ssrf(url: str) -> None:
host = urlparse(url).hostname
if not host:
raise ValueError("ref_audio URL must include a hostname")
for info in socket.getaddrinfo(host, None):
ip_str = str(info[4][0]).split("%", 1)[0]
addr = ipaddress.ip_address(ip_str)
if any(addr in net for net in _REF_AUDIO_BLOCKED_NETWORKS):
raise ValueError(f"ref_audio URL resolves to blocked address: {addr}")
def _fetch_sync() -> tuple[np.ndarray, int]:
if parsed.scheme in ("http", "https"):
_check_ssrf(ref_audio_str)
with urlopen(ref_audio_str, timeout=_REF_AUDIO_TIMEOUT_S) as resp:
data = resp.read(_REF_AUDIO_MAX_BYTES + 1)
if len(data) > _REF_AUDIO_MAX_BYTES:
raise ValueError(f"ref_audio URL exceeds {_REF_AUDIO_MAX_BYTES} bytes")
buf = io.BytesIO(data)
elif ref_audio_str.startswith("data:"):
b64 = ref_audio_str
if "," in b64:
b64 = b64.split(",", 1)[1]
buf = io.BytesIO(base64.b64decode(b64))
else:
raise ValueError("ref_audio must be an http(s) URL or data: base64 URI")
audio, sr = sf.read(buf, dtype="float32", always_2d=False)
if isinstance(audio, np.ndarray) and audio.ndim > 1:
audio = np.mean(audio, axis=-1)
return np.asarray(audio, dtype=np.float32), int(sr)
loop = asyncio.get_running_loop()
wav_np, sr = await loop.run_in_executor(None, _fetch_sync)
return wav_np.tolist(), sr
def _build_tts_params(self, request: OpenAICreateSpeechRequest) -> dict[str, Any]:
"""Build TTS parameters from request.
Processes each parameter if present, skips if not.
Values are wrapped in lists as required by the model.
"""
params: dict[str, Any] = {}
# Text content (always required)
params["text"] = [request.input]
# Task type
if request.task_type is not None:
params["task_type"] = [request.task_type]
else:
params["task_type"] = ["CustomVoice"]
# Language
if request.language is not None:
params["language"] = [request.language]
else:
params["language"] = ["Auto"]
# Speaker (voice)
if request.voice is not None:
params["speaker"] = [request.voice]
elif params["task_type"][0] == "CustomVoice":
params["speaker"] = ["Vivian"] # Default for CustomVoice
# Instructions for style/emotion control
if request.instructions is not None:
params["instruct"] = [request.instructions]
else:
params["instruct"] = [""]
# Voice clone: ref_audio resolved in create_speech(), not here.
if request.ref_text is not None:
params["ref_text"] = [request.ref_text]
if request.x_vector_only_mode is not None:
params["x_vector_only_mode"] = [request.x_vector_only_mode]
# Generation parameters
if request.max_new_tokens is not None:
params["max_new_tokens"] = [request.max_new_tokens]
else:
params["max_new_tokens"] = [2048]
return params
async def create_speech(
self,
request: OpenAICreateSpeechRequest,
raw_request: Request | None = None,
):
"""
Create Speech API similar to OpenAI's API.
See https://platform.openai.com/docs/api-reference/audio/createSpeech
for the API specification. This API mimics the OpenAI
Create Speech API.
For Qwen3-TTS models, additional parameters are supported:
- task_type: "CustomVoice", "VoiceDesign", or "Base"
- language: Language code (e.g., "Chinese", "English", "Auto")
- voice: Speaker name (e.g., "Vivian", "Ryan") for CustomVoice
- instructions: Voice style/emotion instructions
- ref_audio: Reference audio for voice cloning (Base task)
- ref_text: Transcript of reference audio (Base task)
- x_vector_only_mode: Use speaker embedding only (Base task)
NOTE: Streaming audio generation is not currently supported.
"""
error_check_ret = await self._check_model(request)
if error_check_ret is not None:
logger.error("Error with model %s", error_check_ret)
return error_check_ret
if self.engine_client.errored:
raise self.engine_client.dead_error
request_id = f"speech-{random_uuid()}"
try:
if self._is_tts_model():
# Validate TTS parameters
validation_error = self._validate_tts_request(request)
if validation_error:
return self.create_error_response(validation_error)
# Must use prompt_token_ids (not text prompt): the AR Talker
# operates on codec tokens; text token IDs exceed codec vocab.
# model.preprocess replaces all embeddings, so placeholder value
# is irrelevant -- but length must match to avoid excess padding.
tts_params = self._build_tts_params(request)
if request.ref_audio is not None:
wav_list, sr = await self._resolve_ref_audio(request.ref_audio)
tts_params["ref_audio"] = [[wav_list, sr]]
ph_len = self._estimate_prompt_len(tts_params)
prompt = {
"prompt_token_ids": [1] * ph_len,
"additional_information": tts_params,
}
else:
# Fallback for unsupported models
tts_params = {}
prompt = {"prompt": request.input}
logger.info(
"TTS speech request %s: text=%r, task_type=%s",
request_id,
request.input[:50] + "..." if len(request.input) > 50 else request.input,
tts_params.get("task_type", ["unknown"])[0],
)
sampling_params_list = self.engine_client.default_sampling_params_list
generator = self.engine_client.generate(
prompt=prompt,
request_id=request_id,
sampling_params_list=sampling_params_list,
output_modalities=["audio"],
)
final_output: OmniRequestOutput | None = None
async for res in generator:
final_output = res
if final_output is None:
return self.create_error_response("No output generated from the model.")
# Extract audio from output
# Audio can be in final_output.multimodal_output or final_output.request_output.multimodal_output
# Support both "audio" and "model_outputs" keys for compatibility with different models
audio_output = None
if hasattr(final_output, "multimodal_output") and final_output.multimodal_output:
audio_output = final_output.multimodal_output
if not audio_output and hasattr(final_output, "request_output"):
if final_output.request_output and hasattr(final_output.request_output, "multimodal_output"):
audio_output = final_output.request_output.multimodal_output
# Check for audio data using either "audio" or "model_outputs" key
audio_key = None
if audio_output:
if "audio" in audio_output:
audio_key = "audio"
elif "model_outputs" in audio_output:
audio_key = "model_outputs"
if not audio_output or audio_key is None:
return self.create_error_response("TTS model did not produce audio output.")
audio_tensor = audio_output[audio_key]
sample_rate = audio_output.get("sr", 24000)
if hasattr(sample_rate, "item"):
sample_rate = sample_rate.item()
# Streaming accumulates chunks as a list; concat first.
if isinstance(audio_tensor, list):
import torch
audio_tensor = torch.cat(audio_tensor, dim=-1)
# Convert tensor to numpy
if hasattr(audio_tensor, "float"):
audio_tensor = audio_tensor.float().detach().cpu().numpy()
# Squeeze batch dimension if present, but preserve channel dimension for stereo
if audio_tensor.ndim > 1:
audio_tensor = audio_tensor.squeeze()
audio_obj = CreateAudio(
audio_tensor=audio_tensor,
sample_rate=int(sample_rate),
response_format=request.response_format or "wav",
speed=request.speed or 1.0,
stream_format=request.stream_format,
base64_encode=False,
)
audio_response: AudioResponse = self.create_audio(audio_obj)
return Response(content=audio_response.audio_data, media_type=audio_response.media_type)
except asyncio.CancelledError:
return self.create_error_response("Client disconnected")
except ValueError as e:
return self.create_error_response(e)
except Exception as e:
logger.exception("Speech generation failed: %s", e)
return self.create_error_response(f"Speech generation failed: {e}")