feat(qwen-audio): add Alibaba Cloud Qwen-Audio Realtime service integration
- Introduced Qwen-Audio Realtime service for speech-to-speech processing in Pipecat. - Updated interface catalog to include Qwen-Audio Realtime capabilities. - Enhanced model resource testing to support new service. - Added configuration options for audio sample rates and turn detection modes. - Updated documentation to reflect integration details and usage instructions.
This commit is contained in:
Xin Wang
@@ -26,6 +26,7 @@ from services.pipecat.service_factory import (
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create_realtime_service,
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create_stt,
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create_tts,
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realtime_audio_sample_rates,
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)
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from db.session import SessionLocal
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from services.knowledge import search as search_knowledge
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@@ -692,6 +693,7 @@ async def run_realtime_pipeline(
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cfg,
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instructions=brain.system_prompt(cfg),
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)
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input_sample_rate, output_sample_rate = realtime_audio_sample_rates(cfg)
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text_input = RealtimeTextInputProcessor()
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dynamic_variables = RealtimeDynamicVariableProcessor(brain, cfg, realtime)
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greeting = await brain.greeting(cfg)
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@@ -717,12 +719,8 @@ async def run_realtime_pipeline(
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pipeline,
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params=PipelineParams(
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enable_metrics=False,
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audio_in_sample_rate=int(
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cfg.realtime_values.get("inputSampleRate") or 24000
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),
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audio_out_sample_rate=int(
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cfg.realtime_values.get("outputSampleRate") or 24000
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),
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audio_in_sample_rate=input_sample_rate,
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audio_out_sample_rate=output_sample_rate,
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),
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enable_rtvi=False,
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)
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491
backend/services/pipecat/qwen_audio_realtime.py
Normal file
491
backend/services/pipecat/qwen_audio_realtime.py
Normal file
@@ -0,0 +1,491 @@
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"""Qwen-Audio Realtime speech-to-speech service for Pipecat.
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The adapter intentionally depends only on Pipecat's public frame/service APIs.
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Provider-specific WebSocket events stay in this module so the shared pipeline
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does not need to know about Qwen-Audio protocol details.
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"""
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from __future__ import annotations
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import asyncio
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import base64
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import inspect
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import json
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from collections.abc import Awaitable, Callable
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from typing import Any
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from urllib.parse import parse_qsl, urlencode, urlsplit, urlunsplit
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from uuid import uuid4
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from loguru import logger
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from pipecat.frames.frames import (
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CancelFrame,
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EndFrame,
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Frame,
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InputAudioRawFrame,
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InterruptionFrame,
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LLMMessagesAppendFrame,
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OutputTransportMessageUrgentFrame,
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StartFrame,
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TTSAudioRawFrame,
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)
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from pipecat.processors.frame_processor import FrameDirection
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from pipecat.services.ai_service import AIService
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from pipecat.services.settings import ServiceSettings
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from pipecat.utils.time import time_now_iso8601
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from websockets.asyncio.client import connect as websocket_connect
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from websockets.protocol import State
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DEFAULT_QWEN_AUDIO_REALTIME_MODEL = "qwen-audio-3.0-realtime-flash"
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DEFAULT_QWEN_AUDIO_REALTIME_VOICE = "longanqian"
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QWEN_INPUT_SAMPLE_RATE = 16_000
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QWEN_OUTPUT_SAMPLE_RATE = 24_000
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SUPPORTED_TURN_DETECTION_MODES = frozenset({"server_vad", "smart_turn"})
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ExtraEventHandler = Callable[[dict[str, Any]], Awaitable[None] | None]
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class QwenAudioRealtimeService(AIService):
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"""Translate Pipecat frames to Qwen-Audio Realtime WebSocket events.
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``extra_event_handlers`` is deliberately small: future features such as
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Function Calling can subscribe to provider events without changing the
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audio, transcript, or interruption paths implemented here.
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"""
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def __init__(
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self,
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*,
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api_key: str,
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model: str = DEFAULT_QWEN_AUDIO_REALTIME_MODEL,
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base_url: str,
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instructions: str = "",
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voice: str = DEFAULT_QWEN_AUDIO_REALTIME_VOICE,
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input_sample_rate: int = QWEN_INPUT_SAMPLE_RATE,
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output_sample_rate: int = QWEN_OUTPUT_SAMPLE_RATE,
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turn_detection_mode: str = "server_vad",
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vad_threshold: float = 0.5,
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silence_duration_ms: int = 800,
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max_history_turns: int = 20,
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extra_event_handlers: dict[str, ExtraEventHandler] | None = None,
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**kwargs,
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) -> None:
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if turn_detection_mode not in SUPPORTED_TURN_DETECTION_MODES:
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supported = ", ".join(sorted(SUPPORTED_TURN_DETECTION_MODES))
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raise ValueError(
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f"Unsupported Qwen turn detection mode {turn_detection_mode!r}; "
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f"expected one of: {supported}"
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)
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if not -1.0 <= vad_threshold <= 1.0:
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raise ValueError("Qwen VAD threshold must be between -1.0 and 1.0")
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if not 200 <= silence_duration_ms <= 6_000:
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raise ValueError(
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"Qwen VAD silence duration must be between 200 and 6000 ms"
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)
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if not 1 <= max_history_turns <= 50:
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raise ValueError("Qwen max history turns must be between 1 and 50")
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super().__init__(settings=ServiceSettings(model=model), **kwargs)
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self._api_key = api_key
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self._model = model
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self._base_url = base_url
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self._instructions = instructions
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self._voice = voice
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self._input_sample_rate = input_sample_rate
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self._output_sample_rate = output_sample_rate
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self._turn_detection_mode = turn_detection_mode
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self._vad_threshold = vad_threshold
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self._silence_duration_ms = silence_duration_ms
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self._max_history_turns = max_history_turns
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self._extra_event_handlers = extra_event_handlers or {}
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self._websocket = None
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self._receive_task: asyncio.Task | None = None
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self._session_ready = asyncio.Event()
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self._pending_events: list[dict[str, Any]] = []
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self._warned_input_sample_rate = False
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self._response_active = False
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self._audio_suppressed = False
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self._assistant_turn_id: str | None = None
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self._assistant_text = ""
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self._assistant_timestamp = ""
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self._greeting_request_item_id: str | None = None
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async def start(self, frame: StartFrame) -> None:
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await super().start(frame)
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if not self._api_key or not self._model or not self._base_url:
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await self.push_error(
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"Qwen-Audio Realtime requires api_key, model, and base_url",
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fatal=True,
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)
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return
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await self._connect()
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async def stop(self, frame: EndFrame) -> None:
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await self._disconnect()
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await super().stop(frame)
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async def cancel(self, frame: CancelFrame) -> None:
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await self._disconnect()
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await super().cancel(frame)
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async def cleanup(self) -> None:
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await self._disconnect()
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await super().cleanup()
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async def process_frame(self, frame: Frame, direction: FrameDirection) -> None:
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await super().process_frame(frame, direction)
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if isinstance(frame, InputAudioRawFrame):
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if (
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frame.sample_rate != self._input_sample_rate
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and not self._warned_input_sample_rate
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):
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self._warned_input_sample_rate = True
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logger.warning(
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"Qwen-Audio Realtime expected {} Hz input, received {} Hz",
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self._input_sample_rate,
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frame.sample_rate,
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)
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await self._send_event(
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{
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"type": "input_audio_buffer.append",
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"audio": base64.b64encode(frame.audio).decode("ascii"),
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}
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)
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return
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if isinstance(frame, LLMMessagesAppendFrame):
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for message in frame.messages:
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text = self._message_text(message)
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if text:
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await self.send_text(
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text,
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run_immediately=frame.run_llm is not False,
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)
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return
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if isinstance(frame, InterruptionFrame):
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await self._cancel_active_response()
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await self.push_frame(frame, direction)
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async def send_text(self, text: str, *, run_immediately: bool = True) -> None:
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"""Append text to Qwen's conversation and optionally start a response."""
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if not text:
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return
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await self._send_event(
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{
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"type": "conversation.item.create",
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"item": {
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"type": "message",
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"role": "user",
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"content": [{"type": "input_text", "text": text}],
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},
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}
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)
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if run_immediately:
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await self._send_event({"type": "response.create"})
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async def interrupt(self) -> None:
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"""Cancel current model output and notify the rest of the pipeline."""
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await self._cancel_active_response()
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await self.broadcast_interruption()
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async def speak(self, text: str) -> None:
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"""Ask Qwen to speak a fixed greeting, then remove the hidden request.
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Qwen's documented ``response.create`` payload has no per-response
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instruction field. A temporary user item keeps this behavior within
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the supported protocol; it is deleted after the response completes.
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"""
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if not text:
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return
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item_id = f"item_{uuid4().hex}"
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self._greeting_request_item_id = item_id
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await self._send_event(
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{
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"type": "conversation.item.create",
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"item": {
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"id": item_id,
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"type": "message",
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"role": "user",
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"content": [
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{
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"type": "input_text",
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"text": (
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"请直接朗读下面的开场白，不要增删、解释或添加前后缀：\n"
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f"{text}"
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),
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}
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],
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},
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}
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)
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await self._send_event({"type": "response.create"})
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async def update_instructions(self, instructions: str) -> None:
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"""Update only instructions after startup.
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Qwen accepts voice and turn detection only in the first session update,
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so resending the full initial configuration would break live dynamic
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variable updates.
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"""
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self._instructions = instructions
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if self._session_ready.is_set():
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await self._send_event(
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{
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"type": "session.update",
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"session": {"instructions": instructions},
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},
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wait_until_ready=False,
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)
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def _connection_url(self) -> str:
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parts = urlsplit(self._base_url)
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query = dict(parse_qsl(parts.query))
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query["model"] = self._model
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return urlunsplit(
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(parts.scheme, parts.netloc, parts.path, urlencode(query), parts.fragment)
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)
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def _initial_session_config(self) -> dict[str, Any]:
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return {
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"modalities": ["text", "audio"],
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"instructions": self._instructions,
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"voice": self._voice,
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"input_audio_format": "pcm",
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"output_audio_format": "pcm",
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"turn_detection": self._turn_detection_config(),
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"max_history_turns": self._max_history_turns,
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}
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def _turn_detection_config(self) -> dict[str, Any]:
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if self._turn_detection_mode == "smart_turn":
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return {"type": "smart_turn"}
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return {
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"type": "server_vad",
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"threshold": self._vad_threshold,
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"silence_duration_ms": self._silence_duration_ms,
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}
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async def _connect(self) -> None:
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if self._websocket and self._websocket.state is State.OPEN:
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return
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try:
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self._websocket = await websocket_connect(
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self._connection_url(),
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additional_headers={
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"Authorization": f"Bearer {self._api_key}",
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"x-dashscope-dataInspection": "disable",
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},
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max_size=None,
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open_timeout=10,
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)
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self._receive_task = self.create_task(
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self._receive_messages(), name="qwen_audio_realtime_receive"
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)
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except Exception as exc:
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self._websocket = None
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await self.push_error(
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f"Qwen-Audio Realtime connection failed: {exc}",
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exception=exc,
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fatal=True,
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)
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async def _disconnect(self) -> None:
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current_task = asyncio.current_task()
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task = self._receive_task
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self._receive_task = None
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if task and task is not current_task:
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await self.cancel_task(task)
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websocket = self._websocket
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self._websocket = None
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self._session_ready.clear()
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self._pending_events.clear()
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self._response_active = False
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if websocket and websocket.state is State.OPEN:
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try:
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await websocket.close()
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except Exception:
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pass
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async def _receive_messages(self) -> None:
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websocket = self._websocket
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if not websocket:
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return
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try:
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async for raw_message in websocket:
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await self._handle_server_event(json.loads(raw_message))
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except Exception as exc:
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if self._websocket is websocket:
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await self.push_error(
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f"Qwen-Audio Realtime receive failed: {exc}", exception=exc
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)
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finally:
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if self._websocket is websocket:
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self._websocket = None
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self._session_ready.clear()
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if self._receive_task is asyncio.current_task():
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self._receive_task = None
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async def _handle_server_event(self, event: dict[str, Any]) -> None:
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event_type = str(event.get("type") or "")
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if event_type == "session.created":
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await self._send_event(
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{
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"type": "session.update",
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"session": self._initial_session_config(),
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},
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wait_until_ready=False,
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)
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elif event_type == "session.updated":
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self._session_ready.set()
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pending, self._pending_events = self._pending_events, []
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for payload in pending:
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await self._send_event(payload, wait_until_ready=False)
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elif event_type == "response.created":
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self._response_active = True
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self._audio_suppressed = False
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elif event_type == "response.audio.delta":
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audio = event.get("delta")
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if audio and not self._audio_suppressed:
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await self.push_frame(
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TTSAudioRawFrame(
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base64.b64decode(audio),
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self._output_sample_rate,
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1,
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)
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)
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elif event_type in {"response.audio_transcript.delta", "response.text.delta"}:
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if not self._audio_suppressed:
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await self._append_assistant_text(str(event.get("delta") or ""))
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elif event_type in {"response.audio_transcript.done", "response.text.done"}:
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transcript = str(event.get("transcript") or event.get("text") or "")
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if transcript and not self._audio_suppressed:
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if self._assistant_turn_id:
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self._assistant_text = transcript
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else:
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await self._append_assistant_text(transcript)
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elif event_type == "conversation.item.input_audio_transcription.completed":
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await self._send_transcript("user", str(event.get("transcript") or ""))
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elif event_type == "input_audio_buffer.speech_started":
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await self._cancel_active_response()
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await self.broadcast_interruption()
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elif event_type == "response.done":
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response = event.get("response")
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status = response.get("status") if isinstance(response, dict) else None
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interrupted = status in {"cancelled", "incomplete", "interrupted", "failed"}
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self._response_active = False
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await self._finish_assistant_text(interrupted=interrupted)
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await self._delete_greeting_request()
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elif event_type == "error":
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error = event.get("error")
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message = error.get("message") if isinstance(error, dict) else str(error)
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if "cancel" not in str(message).lower():
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await self.push_error(f"Qwen-Audio Realtime error: {message}")
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handler = self._extra_event_handlers.get(event_type)
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if handler:
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result = handler(event)
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if inspect.isawaitable(result):
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await result
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||||
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async def _cancel_active_response(self) -> None:
|
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if not self._response_active and not self._assistant_turn_id:
|
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return
|
||||
self._audio_suppressed = True
|
||||
if self._response_active:
|
||||
await self._send_event(
|
||||
{"type": "response.cancel"}, wait_until_ready=False
|
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)
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self._response_active = False
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await self._finish_assistant_text(interrupted=True)
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async def _delete_greeting_request(self) -> None:
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item_id = self._greeting_request_item_id
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self._greeting_request_item_id = None
|
||||
if item_id:
|
||||
await self._send_event(
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||||
{"type": "conversation.item.delete", "item_id": item_id},
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||||
wait_until_ready=False,
|
||||
)
|
||||
|
||||
async def _send_event(
|
||||
self, payload: dict[str, Any], *, wait_until_ready: bool = True
|
||||
) -> None:
|
||||
if wait_until_ready and not self._session_ready.is_set():
|
||||
self._pending_events.append(payload)
|
||||
return
|
||||
if not self._websocket or self._websocket.state is not State.OPEN:
|
||||
return
|
||||
event = {"event_id": f"event_{uuid4().hex}", **payload}
|
||||
await self._websocket.send(json.dumps(event, ensure_ascii=False))
|
||||
|
||||
async def _append_assistant_text(self, delta: str) -> None:
|
||||
if not delta:
|
||||
return
|
||||
if not self._assistant_turn_id:
|
||||
self._assistant_turn_id = uuid4().hex
|
||||
self._assistant_timestamp = time_now_iso8601()
|
||||
await self._send_transport_message(
|
||||
{
|
||||
"type": "assistant-text-start",
|
||||
"turn_id": self._assistant_turn_id,
|
||||
"timestamp": self._assistant_timestamp,
|
||||
}
|
||||
)
|
||||
self._assistant_text += delta
|
||||
await self._send_transport_message(
|
||||
{
|
||||
"type": "assistant-text-delta",
|
||||
"turn_id": self._assistant_turn_id,
|
||||
"delta": delta,
|
||||
}
|
||||
)
|
||||
|
||||
async def _finish_assistant_text(self, *, interrupted: bool) -> None:
|
||||
if not self._assistant_turn_id:
|
||||
return
|
||||
await self._send_transport_message(
|
||||
{
|
||||
"type": "assistant-text-end",
|
||||
"turn_id": self._assistant_turn_id,
|
||||
"content": self._assistant_text,
|
||||
"interrupted": interrupted,
|
||||
}
|
||||
)
|
||||
self._assistant_turn_id = None
|
||||
self._assistant_text = ""
|
||||
self._assistant_timestamp = ""
|
||||
|
||||
async def _send_transcript(self, role: str, content: str) -> None:
|
||||
if content:
|
||||
await self._send_transport_message(
|
||||
{
|
||||
"type": "transcript",
|
||||
"role": role,
|
||||
"content": content,
|
||||
"timestamp": time_now_iso8601(),
|
||||
}
|
||||
)
|
||||
|
||||
async def _send_transport_message(self, message: dict[str, Any]) -> None:
|
||||
await self.push_frame(OutputTransportMessageUrgentFrame(message=message))
|
||||
|
||||
@staticmethod
|
||||
def _message_text(message: Any) -> str:
|
||||
if not isinstance(message, dict):
|
||||
return ""
|
||||
content = message.get("content")
|
||||
if isinstance(content, str):
|
||||
return content.strip()
|
||||
if isinstance(content, list):
|
||||
return "\n".join(
|
||||
str(part.get("text") or "")
|
||||
for part in content
|
||||
if isinstance(part, dict) and part.get("type") == "text"
|
||||
).strip()
|
||||
return ""
|
||||
@@ -178,8 +178,24 @@ def create_tts(cfg: AssistantConfig):
|
||||
)
|
||||
|
||||
|
||||
def realtime_audio_sample_rates(cfg: AssistantConfig) -> tuple[int, int]:
|
||||
"""Resolve provider defaults once for both the service and pipeline.
|
||||
|
||||
Values remain overridable for future model versions with different audio
|
||||
formats, while today's Qwen-Audio contract defaults to 16 kHz in/24 kHz out.
|
||||
"""
|
||||
default_input_rate = (
|
||||
16_000 if cfg.realtime_interface_type == "qwen-audio-realtime" else 24_000
|
||||
)
|
||||
return (
|
||||
int(cfg.realtime_values.get("inputSampleRate") or default_input_rate),
|
||||
int(cfg.realtime_values.get("outputSampleRate") or 24_000),
|
||||
)
|
||||
|
||||
|
||||
def create_realtime_service(cfg: AssistantConfig, *, instructions: str):
|
||||
"""Create a speech-to-speech service that owns STT, LLM, and TTS."""
|
||||
input_sample_rate, output_sample_rate = realtime_audio_sample_rates(cfg)
|
||||
if cfg.realtime_interface_type == "stepfun-realtime":
|
||||
from services.pipecat.stepfun_realtime import StepFunRealtimeService
|
||||
|
||||
@@ -189,12 +205,8 @@ def create_realtime_service(cfg: AssistantConfig, *, instructions: str):
|
||||
base_url=_require(cfg.realtime_base_url, "Realtime apiUrl"),
|
||||
instructions=instructions,
|
||||
voice=str(cfg.realtime_values.get("voice") or "linjiajiejie"),
|
||||
input_sample_rate=int(
|
||||
cfg.realtime_values.get("inputSampleRate") or 24000
|
||||
),
|
||||
output_sample_rate=int(
|
||||
cfg.realtime_values.get("outputSampleRate") or 24000
|
||||
),
|
||||
input_sample_rate=input_sample_rate,
|
||||
output_sample_rate=output_sample_rate,
|
||||
prefix_padding_ms=int(
|
||||
cfg.realtime_values.get("prefixPaddingMs") or 500
|
||||
),
|
||||
@@ -205,4 +217,26 @@ def create_realtime_service(cfg: AssistantConfig, *, instructions: str):
|
||||
cfg.realtime_values.get("energyAwakenessThreshold") or 2500
|
||||
),
|
||||
)
|
||||
if cfg.realtime_interface_type == "qwen-audio-realtime":
|
||||
from services.pipecat.qwen_audio_realtime import QwenAudioRealtimeService
|
||||
|
||||
return QwenAudioRealtimeService(
|
||||
api_key=_require(cfg.realtime_api_key, "Realtime apiKey"),
|
||||
model=_require(cfg.realtimeModel, "Realtime modelId"),
|
||||
base_url=_require(cfg.realtime_base_url, "Realtime apiUrl"),
|
||||
instructions=instructions,
|
||||
voice=str(cfg.realtime_values.get("voice") or "longanqian"),
|
||||
input_sample_rate=input_sample_rate,
|
||||
output_sample_rate=output_sample_rate,
|
||||
turn_detection_mode=str(
|
||||
cfg.realtime_values.get("turnDetection") or "server_vad"
|
||||
),
|
||||
vad_threshold=float(cfg.realtime_values.get("vadThreshold", 0.5)),
|
||||
silence_duration_ms=int(
|
||||
cfg.realtime_values.get("silenceDurationMs") or 800
|
||||
),
|
||||
max_history_turns=int(
|
||||
cfg.realtime_values.get("maxHistoryTurns") or 20
|
||||
),
|
||||
)
|
||||
raise ValueError(f"不支持的 Realtime 接口类型: {cfg.realtime_interface_type}")
|
||||
|
||||
Reference in New Issue
Block a user