feat(memory,conversation): 记忆富化/证据包、时间线幂等字段与对话分段全链路
数据库 - 新增迁移 0003:timeline_events.memory_source_id 外键 → memory_sources,便于按 ingest 源做时间线幂等 后端 - 记忆 - 新增 ingest 后 LLM 富化(摘要/事实/时间线),可配置开关与最大字符数 - 新增证据包组装:合并 chunk、摘要、事实、时间线、故事等检索结果;支持空 query 时是否仍带 rolling 等开关 - repo/retriever/service/router/schemas/summarizer/timeline/extractor 等扩展;文档 memory-retrieval.md 更新 后端 - 对话 WS - 增加 PING/PONG;分段 ASR 日志与空音频处理;转写失败与「无助手回复」错误提示更明确 - 助手多段回复持久化使用统一分隔符,与分段逻辑一致 后端 - Agent - reply_limits:按 [SPLIT] 与段落拆段,并保证非空 fallback,供 WS 与 TTS 多段下发 后端 - 回忆录任务 - transcript ingest 记录 source_id;任务成功结?
This commit is contained in:
@@ -38,7 +38,7 @@ class TencentASRProvider:
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async def transcribe(self, audio: bytes, format: str = "m4a") -> str:
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client = self._get_client()
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if not client:
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return ""
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return "转写失败: 腾讯云 ASR 客户端未初始化(请检查密钥与依赖)"
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try:
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from tencentcloud.asr.v20190614 import models
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@@ -46,12 +46,26 @@ class TencentASRProvider:
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req = models.SentenceRecognitionRequest()
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req.EngSerViceType = "16k_zh"
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req.SourceType = 1
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req.VoiceFormat = format
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# 小写;与文档一致。iOS 常见为 m4a(AAC) 容器,与 16k 引擎匹配
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req.VoiceFormat = (format or "m4a").lower()
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req.Data = audio_base64
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req.DataLen = len(audio)
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resp = client.SentenceRecognition(req)
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return (resp.Result or "").strip()
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text = (resp.Result or "").strip()
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if text:
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return text
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err = getattr(resp, "Error", None) or getattr(resp, "Message", None)
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logger.warning(
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"Tencent ASR empty Result, audio_len={} format={} err={}",
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len(audio),
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req.VoiceFormat,
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err,
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)
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return (
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"转写失败: 腾讯云返回空文本(常见原因:采样率与 16k_zh 不匹配、"
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"格式不受支持或音频无效;请确认客户端为 16kHz 单声道 m4a)"
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)
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except Exception as e:
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logger.error("Tencent ASR transcribe failed: {}", e)
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return ""
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logger.error("Tencent ASR transcribe failed: {}", e, exc_info=True)
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return f"转写失败: {e}"[:500]
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@@ -1,11 +1,42 @@
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"""Local faster-whisper ASR adapter — implements ASRProvider port."""
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from app.core.logging import get_logger
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from __future__ import annotations
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import asyncio
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import os
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import re
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import tempfile
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from typing import Any, Iterable
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from app.core.logging import get_logger
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logger = get_logger(__name__)
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_SUBTITLE_WATERMARK_RE = re.compile(
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r"(字幕|听译|压制|字幕组).{0,20}(by|BY|By)|字幕\s*by",
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re.UNICODE,
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)
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def _looks_like_subtitle_hallucination(text: str) -> bool:
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"""静音时第二遍易吐出视频字幕水印;仅丢弃此类短句。"""
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t = (text or "").strip()
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if len(t) > 48:
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return False
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if _SUBTITLE_WATERMARK_RE.search(t):
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return True
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if len(t) <= 12 and "字幕" in t and not re.search(r"[??!!。,、]", t):
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return True
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return False
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def _join_segment_text(segments: Iterable[Any]) -> tuple[str, int]:
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segs = list(segments)
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return "".join(str(getattr(seg, "text", "") or "") for seg in segs).strip(), len(
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segs
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)
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_DEFAULT_CACHE_DIR = os.path.normpath(
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os.path.join(
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os.path.dirname(os.path.abspath(__file__)),
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@@ -70,30 +101,95 @@ class WhisperASRProvider:
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return self._load_model()
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async def transcribe(self, audio: bytes, format: str = "m4a") -> str:
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# 与 v1.1.0 相同的单次 transcribe;推理放线程池,避免阻塞 asyncio(tag 上为同步调用)。
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self._load_model()
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if not self._model:
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return ""
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tmp_path = None
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try:
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with tempfile.NamedTemporaryFile(suffix=f".{format}", delete=False) as tmp:
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tmp.write(audio)
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tmp_path = tmp.name
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model = self._model
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segments, _info = self._model.transcribe(
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tmp_path,
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language="zh",
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beam_size=5,
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vad_filter=True,
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vad_parameters={"min_silence_duration_ms": 500},
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)
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return "".join(seg.text for seg in segments).strip()
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except Exception as e:
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logger.error("Whisper transcribe failed: {}", e)
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return ""
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finally:
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if tmp_path and os.path.exists(tmp_path):
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try:
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os.remove(tmp_path)
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except OSError:
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pass
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def _sync_transcribe() -> str:
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tmp_path = None
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try:
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with tempfile.NamedTemporaryFile(
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suffix=f".{format}", delete=False
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) as tmp:
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tmp.write(audio)
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tmp_path = tmp.name
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segments, _info = model.transcribe(
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tmp_path,
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language="zh",
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beam_size=5,
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vad_filter=True,
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vad_parameters={
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"min_silence_duration_ms": 500,
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"threshold": 0.35,
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"min_speech_duration_ms": 200,
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},
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)
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text, pass1_seg_count = _join_segment_text(segments)
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used_second_pass = False
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pass2_seg_count = 0
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pass3_seg_count = 0
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if not text:
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logger.info(
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"Whisper VAD pass 无文本,关闭 VAD 再试一次(短录音易被 VAD 判为静音)"
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)
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segments2, _info2 = model.transcribe(
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tmp_path,
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language="zh",
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beam_size=5,
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vad_filter=False,
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condition_on_previous_text=False,
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# 略抬高:减少边界片段被标成 no_speech 而整段为空
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no_speech_threshold=0.85,
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)
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raw2, pass2_seg_count = _join_segment_text(segments2)
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used_second_pass = True
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if raw2 and _looks_like_subtitle_hallucination(raw2):
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logger.info(
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"Whisper 丢弃疑似字幕水印幻听: {!r}",
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raw2[:120],
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)
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text = ""
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else:
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text = raw2
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if not text and used_second_pass:
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try:
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from faster_whisper import decode_audio
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audio_np = decode_audio(tmp_path, sampling_rate=16000)
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segments3, _info3 = model.transcribe(
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audio_np,
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language="zh",
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beam_size=5,
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vad_filter=False,
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condition_on_previous_text=False,
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no_speech_threshold=0.85,
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)
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raw3, pass3_seg_count = _join_segment_text(segments3)
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if raw3 and _looks_like_subtitle_hallucination(raw3):
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logger.info(
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"Whisper decode_audio 回退仍是疑似字幕水印幻听: {!r}",
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raw3[:120],
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)
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elif raw3:
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text = raw3
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except Exception as ex:
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logger.warning("Whisper decode_audio 回退失败: {}", ex)
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return text
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except Exception as e:
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logger.error("Whisper transcribe failed: {}", e)
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return ""
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finally:
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if tmp_path and os.path.exists(tmp_path):
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try:
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os.remove(tmp_path)
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except OSError:
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pass
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return await asyncio.to_thread(_sync_transcribe)
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