Enhance pipeline execution and voice preview handling for graceful call termination
- Introduce mechanisms in the pipeline to ensure that the end call process waits for the completion of the end speech before hanging up, improving user experience during call termination. - Update the useVoicePreview hook to handle server-initiated call endings gracefully, distinguishing between normal and error disconnections. - Adjust TTS stop frame timeout settings to optimize the timing of call terminations, ensuring timely responses without unnecessary delays. - Refactor related components to support the new end call logic, enhancing overall workflow management and user interaction.
This commit is contained in:
@@ -18,6 +18,8 @@ from pipecat.adapters.schemas.function_schema import FunctionSchema
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from pipecat.adapters.schemas.tools_schema import ToolsSchema
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from pipecat.audio.vad.silero import SileroVADAnalyzer
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from pipecat.frames.frames import (
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BotStartedSpeakingFrame,
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BotStoppedSpeakingFrame,
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EndFrame,
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InputTransportMessageFrame,
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InterruptionFrame,
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@@ -222,6 +224,10 @@ async def run_pipeline(transport, cfg: AssistantConfig) -> None:
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"current": engine.start_id if workflow_active else None,
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"ended": False,
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"turns_in_node": 0,
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# 结束流程的精确计时:只在「结束节点自己的结束语」真正说完时挂断。
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"end_turn_id": None, # 结束节点回复的 turn_id(其 text_start 在 ended 之后)
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"end_armed": False, # 结束语文本已生成完(已下发 data channel)
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"end_speaking": False, # 结束语音频已开始播报
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"end_frame_queued": False,
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}
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history: list[dict] = []
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@@ -256,6 +262,35 @@ async def run_pipeline(transport, cfg: AssistantConfig) -> None:
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assistant_aggregator = PassthroughLLMAssistantAggregator(context)
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text_input = TextInputProcessor()
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# 结束节点:等结束语「说完」(BotStoppedSpeakingFrame)再挂断,确保结束语的
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# 文字(走 data channel)与音频都已下发,避免前端只听到声音、看不到文字。
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worker_holder: dict = {}
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class EndCallAfterSpeech(FrameProcessor):
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async def process_frame(self, frame, direction: FrameDirection):
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await super().process_frame(frame, direction)
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await self.push_frame(frame, direction)
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# 结束语文本生成完(end_armed)→ 其音频开始(end_speaking)→ 音频说完才挂断。
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# 配对 started/stopped,避免被结束节点之前的话(如先答一句再转移)的
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# stopped 事件提前触发,导致结束语被截断。
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if isinstance(frame, BotStartedSpeakingFrame) and wf_state["end_armed"]:
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wf_state["end_speaking"] = True
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elif (
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isinstance(frame, BotStoppedSpeakingFrame)
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and wf_state["end_speaking"]
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and not wf_state["end_frame_queued"]
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and worker_holder.get("worker") is not None
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):
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wf_state["end_frame_queued"] = True
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logger.info("结束语播报完毕,挂断通话")
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# 先告知前端这是正常结束(而非连接异常),再优雅挂断
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await worker_holder["worker"].queue_frame(
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OutputTransportMessageUrgentFrame(
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message={"type": "call-ended", "reason": "completed"}
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)
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)
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await worker_holder["worker"].queue_frame(EndFrame())
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pipeline = Pipeline(
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[
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transport.input(),
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@@ -268,6 +303,7 @@ async def run_pipeline(transport, cfg: AssistantConfig) -> None:
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# waiting for a TTS provider to emit spoken-text/timestamp frames.
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assistant_aggregator,
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tts,
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EndCallAfterSpeech(),
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transport.output(),
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]
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)
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@@ -279,6 +315,7 @@ async def run_pipeline(transport, cfg: AssistantConfig) -> None:
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),
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enable_rtvi=False,
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)
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worker_holder["worker"] = worker
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async def queue_transcript(role: str, content: str, timestamp: str) -> None:
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if content:
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@@ -357,7 +394,9 @@ async def run_pipeline(transport, cfg: AssistantConfig) -> None:
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async def handler(params):
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logger.info(f"LLM 触发转移 → {engine.name(target)}")
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await speak_transition(edge)
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# 进结束节点不播过渡语(结束语本身就是收尾,避免打断挂断时序)
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if not engine.is_end(target):
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await speak_transition(edge)
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await go_to_node(target)
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# 返回工具结果,pipecat 随即在新节点的提示/工具下继续生成
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await params.result_callback({"status": "ok"})
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@@ -381,7 +420,8 @@ async def run_pipeline(transport, cfg: AssistantConfig) -> None:
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)
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if target and target != wf_state["current"]:
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logger.info(f"文本兜底触发转移 → {engine.name(target)}")
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await speak_transition(engine.find_edge(wf_state["current"], target))
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if not engine.is_end(target):
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await speak_transition(engine.find_edge(wf_state["current"], target))
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# 仅切换节点提示/工具,下一轮用户输入即在新节点处理
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await go_to_node(target)
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@@ -411,6 +451,14 @@ async def run_pipeline(transport, cfg: AssistantConfig) -> None:
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@assistant_aggregator.event_handler("on_assistant_text_start")
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async def on_assistant_text_start(_aggregator, turn_id, timestamp):
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# 进入结束节点后,第一条「开始生成」的回复就是结束节点自己的结束语
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# (其 text_start 发生在 ended 置位之后,不会误认转移前的那句)。
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if (
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workflow_active
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and wf_state["ended"]
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and wf_state["end_turn_id"] is None
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):
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wf_state["end_turn_id"] = turn_id
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await worker.queue_frame(
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OutputTransportMessageUrgentFrame(
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message={
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@@ -449,13 +497,11 @@ async def run_pipeline(transport, cfg: AssistantConfig) -> None:
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# 正常情况下转移由 LLM 直接调用转移工具完成(go_to_node),无需这里处理。
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if content and not interrupted and workflow_active:
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history.append({"role": "assistant", "content": content})
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if wf_state["ended"]:
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# 结束节点:说完结束语后挂断,不再继续多轮对话
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if not wf_state["end_frame_queued"]:
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wf_state["end_frame_queued"] = True
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logger.info("结束节点结束语已播报,挂断通话")
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await worker.queue_frame(EndFrame())
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else:
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if turn_id == wf_state["end_turn_id"]:
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# 结束节点的结束语文本已生成完(也已下发 data channel),武装挂断;
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# 真正的 EndFrame 由 EndCallAfterSpeech 在结束语「说完」时排入。
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wf_state["end_armed"] = True
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elif not wf_state["ended"]:
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wf_state["turns_in_node"] += 1
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await fallback_route()
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elif content and not interrupted:
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@@ -21,6 +21,11 @@ from services.pipecat.xfyun_super_tts import (
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)
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from services.pipecat.xfyun_tts import DEFAULT_XFYUN_TTS_URL, XfyunTTSService
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# TTS「说完」判定的空闲时长:默认 3.0s 过长(导致工作流结束节点说完后还要等约 3s
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# 才挂断,也拖慢日常轮次的交还)。设 1.0s 既能让结束语文字/音频送达,又更跟手。
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# 流式 TTS 句间音频间隔通常远小于 1s,不会把一段多句回复误判为结束。
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TTS_STOP_FRAME_TIMEOUT_S = 1.0
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def _language(value: str) -> Language | None:
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if not value:
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@@ -107,6 +112,7 @@ def create_tts(cfg: AssistantConfig):
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volume=int(cfg.tts_values.get("volume") or 50),
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pitch=int(cfg.tts_values.get("pitch") or 50),
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push_stop_frames=True,
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stop_frame_timeout_s=TTS_STOP_FRAME_TIMEOUT_S,
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)
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if cfg.tts_interface_type not in {"openai-tts", "dashscope-tts"}:
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raise ValueError(f"不支持的 TTS 接口类型: {cfg.tts_interface_type}")
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@@ -117,6 +123,7 @@ def create_tts(cfg: AssistantConfig):
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return OpenAITTSService(
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api_key=cfg.tts_api_key or config.TTS_API_KEY,
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base_url=cfg.tts_base_url or config.TTS_BASE_URL,
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stop_frame_timeout_s=TTS_STOP_FRAME_TIMEOUT_S,
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settings=OpenAITTSService.Settings(
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model=cfg.tts_model or config.TTS_MODEL,
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voice=voice,
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