fix long run bug

README.md

@@ -5,3 +5,21 @@ Python Active-Call: real-time audio streaming with WebSocket and WebRTC.
 This repo contains a Python 3.11+ codebase for building low-latency voice
 pipelines (capture, stream, and process audio) using WebRTC and WebSockets.
 It is currently in an early, experimental stage.
+
+# Usage
+
+启动
+
+```
+uvicorn app.main:app --reload --host 0.0.0.0 --port 8000
+```
+
+测试
+
+```
+python examples/test_websocket.py
+```
+
+```
+python mic_client.py
+```

core/duplex_pipeline.py

@@ -113,6 +113,10 @@ class DuplexPipeline:
         # Interruption handling
         self._interrupt_event = asyncio.Event()
         
+        # Latency tracking - TTFB (Time to First Byte)
+        self._turn_start_time: Optional[float] = None
+        self._first_audio_sent: bool = False
+        
         # Barge-in filtering - require minimum speech duration to interrupt
         self._barge_in_speech_start_time: Optional[float] = None
         self._barge_in_min_duration_ms: int = settings.barge_in_min_duration_ms if hasattr(settings, 'barge_in_min_duration_ms') else 50
@@ -396,6 +400,10 @@ class DuplexPipeline:
             user_text: User's transcribed text
         """
         try:
+            # Start latency tracking
+            self._turn_start_time = time.time()
+            self._first_audio_sent = False
+            
             # Get AI response (streaming)
             messages = self.conversation.get_messages()
             full_response = ""
@@ -495,10 +503,33 @@ class DuplexPipeline:
         
         try:
             async for chunk in self.tts_service.synthesize_stream(text):
+                # Check interrupt at the start of each iteration
+                if self._interrupt_event.is_set():
+                    logger.debug("TTS sentence interrupted")
+                    break
+                
+                # Track and log first audio packet latency (TTFB)
+                if not self._first_audio_sent and self._turn_start_time:
+                    ttfb_ms = (time.time() - self._turn_start_time) * 1000
+                    self._first_audio_sent = True
+                    logger.info(f"[TTFB] Server first audio packet latency: {ttfb_ms:.0f}ms (session {self.session_id})")
+                    
+                    # Send TTFB event to client
+                    await self.transport.send_event({
+                        "event": "ttfb",
+                        "trackId": self.session_id,
+                        "timestamp": self._get_timestamp_ms(),
+                        "latencyMs": round(ttfb_ms)
+                    })
+                
+                # Double-check interrupt right before sending audio
                 if self._interrupt_event.is_set():
                     break
+                
                 await self.transport.send_audio(chunk.audio)
                 await asyncio.sleep(0.005)  # Small delay to prevent flooding
+        except asyncio.CancelledError:
+            logger.debug("TTS sentence cancelled")
         except Exception as e:
             logger.error(f"TTS sentence error: {e}")
     
@@ -513,6 +544,10 @@ class DuplexPipeline:
             return
         
         try:
+            # Start latency tracking for greeting
+            speak_start_time = time.time()
+            first_audio_sent = False
+            
             # Send track start event
             await self.transport.send_event({
                 "event": "trackStart",
@@ -528,6 +563,20 @@ class DuplexPipeline:
                     logger.info("TTS interrupted by barge-in")
                     break
                 
+                # Track and log first audio packet latency (TTFB)
+                if not first_audio_sent:
+                    ttfb_ms = (time.time() - speak_start_time) * 1000
+                    first_audio_sent = True
+                    logger.info(f"[TTFB] Greeting first audio packet latency: {ttfb_ms:.0f}ms (session {self.session_id})")
+                    
+                    # Send TTFB event to client
+                    await self.transport.send_event({
+                        "event": "ttfb",
+                        "trackId": self.session_id,
+                        "timestamp": self._get_timestamp_ms(),
+                        "latencyMs": round(ttfb_ms)
+                    })
+                
                 # Send audio to client
                 await self.transport.send_audio(chunk.audio)
                 
@@ -561,8 +610,17 @@ class DuplexPipeline:
         self._barge_in_speech_frames = 0
         self._barge_in_silence_frames = 0
         
-        # Signal interruption
+        # IMPORTANT: Signal interruption FIRST to stop audio sending
         self._interrupt_event.set()
+        self._is_bot_speaking = False
+        
+        # Send interrupt event to client IMMEDIATELY
+        # This must happen BEFORE canceling services, so client knows to discard in-flight audio
+        await self.transport.send_event({
+            "event": "interrupt",
+            "trackId": self.session_id,
+            "timestamp": self._get_timestamp_ms()
+        })
         
         # Cancel TTS
         if self.tts_service:
@@ -575,15 +633,7 @@ class DuplexPipeline:
         # Interrupt conversation
         await self.conversation.interrupt()
         
-        # Send interrupt event to client
-        await self.transport.send_event({
-            "event": "interrupt",
-            "trackId": self.session_id,
-            "timestamp": self._get_timestamp_ms()
-        })
-        
         # Reset for new user turn
-        self._is_bot_speaking = False
         await self.conversation.start_user_turn()
         self._audio_buffer = b""
         self.eou_detector.reset()

docs/proejct_todo.md

@@ -0,0 +1,187 @@
+# OmniSense: 12-Week Sprint Board + Tech Stack (Python Backend) — TODO
+
+## Scope
+- [ ] Build a realtime AI SaaS (OmniSense) focused on web-first audio + video with WebSocket + WebRTC endpoints
+- [ ] Deliver assistant builder, tool execution, observability, evals, optional telephony later
+- [ ] Keep scope aligned to 2-person team, self-hosted services
+
+---
+
+## Sprint Board (12 weeks, 2-week sprints)
+Team assumption: 2 engineers. Scope prioritized to web-first audio + video, with BYO-SFU adapters.
+
+### Sprint 1 (Weeks 1–2) — Realtime Core MVP (WebSocket + WebRTC Audio)
+- Deliverables
+  - [ ] WebSocket transport: audio in/out streaming (1:1)
+  - [ ] WebRTC transport: audio in/out streaming (1:1)
+  - [ ] Adapter contract wired into runtime (transport-agnostic session core)
+  - [ ] ASR → LLM → TTS pipeline, streaming both directions
+  - [ ] Basic session state (start/stop, silence timeout)
+  - [ ] Transcript persistence
+- Acceptance criteria
+  - [ ] < 1.5s median round-trip for short responses
+  - [ ] Stable streaming for 10+ minute session
+
+### Sprint 2 (Weeks 3–4) — Video + Realtime UX
+- Deliverables
+  - [ ] WebRTC video capture + streaming (assistant can “see” frames)
+  - [ ] WebSocket video streaming for local/dev mode
+  - [ ] Low-latency UI: push-to-talk, live captions, speaking indicator
+  - [ ] Recording + transcript storage (web sessions)
+- Acceptance criteria
+  - [ ] Video < 2.5s end-to-end latency for analysis
+  - [ ] Audio quality acceptable (no clipping, jitter handling)
+
+### Sprint 3 (Weeks 5–6) — Assistant Builder v1
+- Deliverables
+  - [ ] Assistant schema + versioning
+  - [ ] UI: Model/Voice/Transcriber/Tools/Video/Transport tabs
+  - [ ] “Test/Chat/Talk to Assistant” (web)
+- Acceptance criteria
+  - [ ] Create/publish assistant and run a live web session
+  - [ ] All config changes tracked by version
+
+### Sprint 4 (Weeks 7–8) — Tooling + Structured Outputs
+- Deliverables
+  - [ ] Tool registry + custom HTTP tools
+  - [ ] Tool auth secrets management
+  - [ ] Structured outputs (JSON extraction)
+- Acceptance criteria
+  - [ ] Tool calls executed with retries/timeouts
+  - [ ] Structured JSON stored per call/session
+
+### Sprint 5 (Weeks 9–10) — Observability + QA + Dev Platform
+- Deliverables
+  - [ ] Session logs + chat logs + media logs
+  - [ ] Evals engine + test suites
+  - [ ] Basic analytics dashboard
+  - [ ] Public WebSocket API spec + message schema
+  - [ ] JS/TS SDK (connect, send audio/video, receive transcripts)
+- Acceptance criteria
+  - [ ] Reproducible test suite runs
+  - [ ] Log filters by assistant/time/status
+  - [ ] SDK demo app runs end-to-end
+
+### Sprint 6 (Weeks 11–12) — SaaS Hardening
+- Deliverables
+  - [ ] Org/RBAC + API keys + rate limits
+  - [ ] Usage metering + credits
+  - [ ] Stripe billing integration
+  - [ ] Self-hosted DB ops (migrations, backup/restore, monitoring)
+- Acceptance criteria
+  - [ ] Metered usage per org
+  - [ ] Credits decrement correctly
+  - [ ] Optional telephony spike documented (defer build)
+  - [ ] Enterprise adapter guide published (BYO-SFU)
+
+---
+
+## Tech Stack by Service (Self-Hosted, Web-First)
+
+### 1) Transport Gateway (Realtime)
+- [ ] WebRTC (browser) + WebSocket (lightweight/dev) protocols
+- [ ] BYO-SFU adapter (enterprise) + LiveKit optional adapter + WS transport server
+- [ ] Python core (FastAPI + asyncio) + Node.js mediasoup adapters when needed
+- [ ] Media: Opus/VP8, jitter buffer, VAD, echo cancellation
+- [ ] Storage: S3-compatible (MinIO) for recordings
+
+### 2) ASR Service
+- [ ] Whisper (self-hosted) baseline
+- [ ] gRPC/WebSocket streaming transport
+- [ ] Python native service
+- [ ] Optional cloud provider fallback (later)
+
+### 3) TTS Service
+- [ ] Piper or Coqui TTS (self-hosted)
+- [ ] gRPC/WebSocket streaming transport
+- [ ] Python native service
+- [ ] Redis cache for common phrases
+
+### 4) LLM Orchestrator
+- [ ] Self-hosted (vLLM + open model)
+- [ ] Python (FastAPI + asyncio)
+- [ ] Streaming, tool calling, JSON mode
+- [ ] Safety filters + prompt templates
+
+### 5) Assistant Config Service
+- [ ] PostgreSQL
+- [ ] Python (SQLAlchemy or SQLModel)
+- [ ] Versioning, publish/rollback
+
+### 6) Session Service
+- [ ] PostgreSQL + Redis
+- [ ] Python
+- [ ] State machine, timeouts, events
+
+### 7) Tool Execution Layer
+- [ ] PostgreSQL
+- [ ] Python
+- [ ] Auth secret vault, retry policies, tool schemas
+
+### 8) Observability + Logs
+- [ ] Postgres (metadata), ClickHouse (logs/metrics)
+- [ ] OpenSearch for search
+- [ ] Prometheus + Grafana metrics
+- [ ] OpenTelemetry tracing
+
+### 9) Billing + Usage Metering
+- [ ] Stripe billing
+- [ ] PostgreSQL
+- [ ] NATS JetStream (events) + Redis counters
+
+### 10) Web App (Dashboard)
+- [ ] React + Next.js
+- [ ] Tailwind or Radix UI
+- [ ] WebRTC client + WS client; adapter-based RTC integration
+- [ ] ECharts/Recharts
+
+### 11) Auth + RBAC
+- [ ] Keycloak (self-hosted) or custom JWT
+- [ ] Org/user/role tables in Postgres
+
+### 12) Public WebSocket API + SDK
+- [ ] WS API: versioned schema, binary audio frames + JSON control messages
+- [ ] SDKs: JS/TS first, optional Python/Go clients
+- [ ] Docs: quickstart, auth flow, session lifecycle, examples
+
+---
+
+## Infrastructure (Self-Hosted)
+- [ ] Docker Compose → k3s (later)
+- [ ] Redis Streams or NATS
+- [ ] MinIO object store
+- [ ] GitHub Actions + Helm or kustomize
+- [ ] Self-hosted Postgres + pgbackrest backups
+- [ ] Vault for secrets
+
+---
+
+## Suggested MVP Sequence
+- [ ] WebRTC demo + ASR/LLM/TTS streaming
+- [ ] Assistant schema + versioning (web-first)
+- [ ] Video capture + multimodal analysis
+- [ ] Tool execution + structured outputs
+- [ ] Logs + evals + public WS API + SDK
+- [ ] Telephony (optional, later)
+
+---
+
+## Public WebSocket API (Minimum Spec)
+- [ ] Auth: API key or JWT in initial `hello` message
+- [ ] Core messages: `session.start`, `session.stop`, `audio.append`, `audio.commit`, `video.append`, `transcript.delta`, `assistant.response`, `tool.call`, `tool.result`, `error`
+- [ ] Binary payloads: PCM/Opus frames with metadata in control channel
+- [ ] Versioning: `v1` schema with backward compatibility rules
+
+---
+
+## Self-Hosted DB Ops Checklist
+- [ ] Postgres in Docker/k3s with persistent volumes
+- [ ] Migrations: `alembic` or `atlas`
+- [ ] Backups: `pgbackrest` nightly + on-demand
+- [ ] Monitoring: postgres_exporter + alerts
+
+---
+
+## RTC Adapter Contract (BYO-SFU First)
+- [ ] Keep RTC pluggable; LiveKit optional, not core dependency
+- [ ] Define adapter interface (TypeScript sketch)

examples/mic_client.py

@@ -17,6 +17,7 @@ import argparse
 import asyncio
 import json
 import sys
+import time
 import threading
 import queue
 from pathlib import Path
@@ -92,6 +93,14 @@ class MicrophoneClient:
         # State
         self.is_recording = True
         self.is_playing = True
+        
+        # TTFB tracking (Time to First Byte)
+        self.request_start_time = None
+        self.first_audio_received = False
+        
+        # Interrupt handling - discard audio until next trackStart
+        self._discard_audio = False
+        self._audio_sequence = 0  # Track audio sequence to detect stale chunks
     
     async def connect(self) -> None:
         """Connect to WebSocket server."""
@@ -117,6 +126,10 @@ class MicrophoneClient:
     
     async def send_chat(self, text: str) -> None:
         """Send chat message (text input)."""
+        # Reset TTFB tracking for new request
+        self.request_start_time = time.time()
+        self.first_audio_received = False
+        
         await self.send_command({
             "command": "chat",
             "text": text
@@ -236,9 +249,21 @@ class MicrophoneClient:
                         # Audio data received
                         self.bytes_received += len(message)
                         
+                        # Check if we should discard this audio (after interrupt)
+                        if self._discard_audio:
+                            duration_ms = len(message) / (self.sample_rate * 2) * 1000
+                            print(f"← Audio: {duration_ms:.0f}ms (DISCARDED - waiting for new track)")
+                            continue
+                        
                         if self.is_playing:
                             self._add_audio_to_buffer(message)
                         
+                        # Calculate and display TTFB for first audio packet
+                        if not self.first_audio_received and self.request_start_time:
+                            client_ttfb_ms = (time.time() - self.request_start_time) * 1000
+                            self.first_audio_received = True
+                            print(f"← [TTFB] Client first audio latency: {client_ttfb_ms:.0f}ms")
+                        
                         # Show progress (less verbose)
                         with self.audio_output_lock:
                             buffer_ms = len(self.audio_output_buffer) / (self.sample_rate * 2) * 1000
@@ -285,20 +310,36 @@ class MicrophoneClient:
                 # Interim result - show with indicator (overwrite same line)
                 display_text = text[:60] + "..." if len(text) > 60 else text
                 print(f"  [listening] {display_text}".ljust(80), end="\r")
+        elif event_type == "ttfb":
+            # Server-side TTFB event
+            latency_ms = event.get("latencyMs", 0)
+            print(f"← [TTFB] Server reported latency: {latency_ms}ms")
         elif event_type == "trackStart":
             print("← Bot started speaking")
+            # IMPORTANT: Accept audio again after trackStart
+            self._discard_audio = False
+            self._audio_sequence += 1
+            # Reset TTFB tracking for voice responses (when no chat was sent)
+            if self.request_start_time is None:
+                self.request_start_time = time.time()
+                self.first_audio_received = False
             # Clear any old audio in buffer
             with self.audio_output_lock:
                 self.audio_output_buffer = b""
         elif event_type == "trackEnd":
             print("← Bot finished speaking")
+            # Reset TTFB tracking after response completes
+            self.request_start_time = None
+            self.first_audio_received = False
         elif event_type == "interrupt":
             print("← Bot interrupted!")
-            # IMPORTANT: Clear audio buffer immediately on interrupt
+            # IMPORTANT: Discard all audio until next trackStart
+            self._discard_audio = True
+            # Clear audio buffer immediately
             with self.audio_output_lock:
                 buffer_ms = len(self.audio_output_buffer) / (self.sample_rate * 2) * 1000
                 self.audio_output_buffer = b""
-                print(f"   (cleared {buffer_ms:.0f}ms of buffered audio)")
+                print(f"   (cleared {buffer_ms:.0f}ms, discarding audio until new track)")
         elif event_type == "error":
             print(f"← Error: {event.get('error')}")
         elif event_type == "hangup":

examples/simple_client.py

@@ -12,6 +12,7 @@ import argparse
 import asyncio
 import json
 import sys
+import time
 import wave
 import io
 
@@ -67,6 +68,13 @@ class SimpleVoiceClient:
         
         # Stats
         self.bytes_received = 0
+        
+        # TTFB tracking (Time to First Byte)
+        self.request_start_time = None
+        self.first_audio_received = False
+        
+        # Interrupt handling - discard audio until next trackStart
+        self._discard_audio = False
     
     async def connect(self):
         """Connect to server."""
@@ -84,6 +92,10 @@ class SimpleVoiceClient:
     
     async def send_chat(self, text: str):
         """Send chat message."""
+        # Reset TTFB tracking for new request
+        self.request_start_time = time.time()
+        self.first_audio_received = False
+        
         await self.ws.send(json.dumps({"command": "chat", "text": text}))
         print(f"-> chat: {text}")
     
@@ -120,6 +132,18 @@ class SimpleVoiceClient:
                     # Audio data
                     self.bytes_received += len(msg)
                     duration_ms = len(msg) / (self.sample_rate * 2) * 1000
+                    
+                    # Check if we should discard this audio (after interrupt)
+                    if self._discard_audio:
+                        print(f"<- audio: {len(msg)} bytes ({duration_ms:.0f}ms) [DISCARDED]")
+                        continue
+                    
+                    # Calculate and display TTFB for first audio packet
+                    if not self.first_audio_received and self.request_start_time:
+                        client_ttfb_ms = (time.time() - self.request_start_time) * 1000
+                        self.first_audio_received = True
+                        print(f"<- [TTFB] Client first audio latency: {client_ttfb_ms:.0f}ms")
+                    
                     print(f"<- audio: {len(msg)} bytes ({duration_ms:.0f}ms)")
                     
                     # Play immediately in executor to not block
@@ -138,6 +162,18 @@ class SimpleVoiceClient:
                             print(f"<- You said: {text}")
                         else:
                             print(f"<- [listening] {text}", end="\r")
+                    elif etype == "ttfb":
+                        # Server-side TTFB event
+                        latency_ms = event.get("latencyMs", 0)
+                        print(f"<- [TTFB] Server reported latency: {latency_ms}ms")
+                    elif etype == "trackStart":
+                        # New track starting - accept audio again
+                        self._discard_audio = False
+                        print(f"<- {etype}")
+                    elif etype == "interrupt":
+                        # Interrupt - discard audio until next trackStart
+                        self._discard_audio = True
+                        print(f"<- {etype} (discarding audio until new track)")
                     elif etype == "hangup":
                         print(f"<- {etype}")
                         self.running = False