opencode-skills-collection 4.0.53 → 4.0.55

package/bundled-skills/voice-ai-engine-development/references/detailed-guide.md

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+# Voice AI Engine Development - Detailed Guide
+
+> This file contains the detailed procedure and reference material extracted from `SKILL.md` for focused loading. The root skill defines activation, examples, safety constraints, and limitations.
+
+## Overview
+
+This skill guides you through building production-ready voice AI engines with real-time conversation capabilities. Voice AI engines enable natural, bidirectional conversations between users and AI agents through streaming audio processing, speech-to-text transcription, LLM-powered responses, and text-to-speech synthesis.
+
+The core architecture uses an async queue-based worker pipeline where each component runs independently and communicates via `asyncio.Queue` objects, enabling concurrent processing, interrupt handling, and real-time streaming at every stage.
+
+
+## Core Architecture Principles
+
+### The Worker Pipeline Pattern
+
+Every voice AI engine follows this pipeline:
+
+```
+Audio In → Transcriber → Agent → Synthesizer → Audio Out
+           (Worker 1)   (Worker 2)  (Worker 3)
+```
+
+**Key Benefits:**
+- **Decoupling**: Workers only know about their input/output queues
+- **Concurrency**: All workers run simultaneously via asyncio
+- **Backpressure**: Queues automatically handle rate differences
+- **Interruptibility**: Everything can be stopped mid-stream
+
+### Base Worker Pattern
+
+Every worker follows this pattern:
+
+```python
+class BaseWorker:
+    def __init__(self, input_queue, output_queue):
+        self.input_queue = input_queue   # asyncio.Queue to consume from
+        self.output_queue = output_queue # asyncio.Queue to produce to
+        self.active = False
+
+    def start(self):
+        """Start the worker's processing loop"""
+        self.active = True
+        asyncio.create_task(self._run_loop())
+
+    async def _run_loop(self):
+        """Main processing loop - runs forever until terminated"""
+        while self.active:
+            item = await self.input_queue.get()  # Block until item arrives
+            await self.process(item)              # Process the item
+
+    async def process(self, item):
+        """Override this - does the actual work"""
+        raise NotImplementedError
+
+    def terminate(self):
+        """Stop the worker"""
+        self.active = False
+```
+
+
+## Component Implementation Guide
+
+### 1. Transcriber (Audio → Text)
+
+**Purpose**: Converts incoming audio chunks to text transcriptions
+
+**Interface Requirements**:
+```python
+class BaseTranscriber:
+    def __init__(self, transcriber_config):
+        self.input_queue = asyncio.Queue()   # Audio chunks (bytes)
+        self.output_queue = asyncio.Queue()  # Transcriptions
+        self.is_muted = False
+
+    def send_audio(self, chunk: bytes):
+        """Client calls this to send audio"""
+        if not self.is_muted:
+            self.input_queue.put_nowait(chunk)
+        else:
+            # Send silence instead (prevents echo during bot speech)
+            self.input_queue.put_nowait(self.create_silent_chunk(len(chunk)))
+
+    def mute(self):
+        """Called when bot starts speaking (prevents echo)"""
+        self.is_muted = True
+
+    def unmute(self):
+        """Called when bot stops speaking"""
+        self.is_muted = False
+```
+
+**Output Format**:
+```python
+class Transcription:
+    message: str          # "Hello, how are you?"
+    confidence: float     # 0.95
+    is_final: bool        # True = complete sentence, False = partial
+    is_interrupt: bool    # Set by TranscriptionsWorker
+```
+
+**Supported Providers**:
+- **Deepgram** - Fast, accurate, streaming
+- **AssemblyAI** - High accuracy, good for accents
+- **Azure Speech** - Enterprise-grade
+- **Google Cloud Speech** - Multi-language support
+
+**Critical Implementation Details**:
+- Use WebSocket for bidirectional streaming
+- Run sender and receiver tasks concurrently with `asyncio.gather()`
+- Mute transcriber when bot speaks to prevent echo/feedback loops
+- Handle both final and partial transcriptions
+
+### 2. Agent (Text → Response)
+
+**Purpose**: Processes user input and generates conversational responses
+
+**Interface Requirements**:
+```python
+class BaseAgent:
+    def __init__(self, agent_config):
+        self.input_queue = asyncio.Queue()   # TranscriptionAgentInput
+        self.output_queue = asyncio.Queue()  # AgentResponse
+        self.transcript = None               # Conversation history
+
+    async def generate_response(self, human_input, is_interrupt, conversation_id):
+        """Override this - returns AsyncGenerator of responses"""
+        raise NotImplementedError
+```
+
+**Why Streaming Responses?**
+- **Lower latency**: Start speaking as soon as first sentence is ready
+- **Better interrupts**: Can stop mid-response
+- **Sentence-by-sentence**: More natural conversation flow
+
+**Supported Providers**:
+- **OpenAI** (GPT-4, GPT-3.5) - High quality, fast
+- **Google Gemini** - Multimodal, cost-effective
+- **Anthropic Claude** - Long context, nuanced responses
+
+**Critical Implementation Details**:
+- Maintain conversation history in `Transcript` object
+- Stream responses using `AsyncGenerator`
+- **IMPORTANT**: Buffer entire LLM response before yielding to synthesizer (prevents audio jumping)
+- Handle interrupts by canceling current generation task
+- Update conversation history with partial messages on interrupt
+
+### 3. Synthesizer (Text → Audio)
+
+**Purpose**: Converts agent text responses to speech audio
+
+**Interface Requirements**:
+```python
+class BaseSynthesizer:
+    async def create_speech(self, message: BaseMessage, chunk_size: int) -> SynthesisResult:
+        """
+        Returns a SynthesisResult containing:
+        - chunk_generator: AsyncGenerator that yields audio chunks
+        - get_message_up_to: Function to get partial text (for interrupts)
+        """
+        raise NotImplementedError
+```
+
+**SynthesisResult Structure**:
+```python
+class SynthesisResult:
+    chunk_generator: AsyncGenerator[ChunkResult, None]
+    get_message_up_to: Callable[[float], str]  # seconds → partial text
+
+    class ChunkResult:
+        chunk: bytes          # Raw PCM audio
+        is_last_chunk: bool
+```
+
+**Supported Providers**:
+- **ElevenLabs** - Most natural voices, streaming
+- **Azure TTS** - Enterprise-grade, many languages
+- **Google Cloud TTS** - Cost-effective, good quality
+- **Amazon Polly** - AWS integration
+- **Play.ht** - Voice cloning
+
+**Critical Implementation Details**:
+- Stream audio chunks as they're generated
+- Convert audio to LINEAR16 PCM format (16kHz sample rate)
+- Implement `get_message_up_to()` for interrupt handling
+- Handle audio format conversion (MP3 → PCM)
+
+### 4. Output Device (Audio → Client)
+
+**Purpose**: Sends synthesized audio back to the client
+
+**CRITICAL: Rate Limiting for Interrupts**
+
+```python
+async def send_speech_to_output(self, message, synthesis_result,
+                                stop_event, seconds_per_chunk):
+    chunk_idx = 0
+    async for chunk_result in synthesis_result.chunk_generator:
+        # Check for interrupt
+        if stop_event.is_set():
+            logger.debug(f"Interrupted after {chunk_idx} chunks")
+            message_sent = synthesis_result.get_message_up_to(
+                chunk_idx * seconds_per_chunk
+            )
+            return message_sent, True  # cut_off = True
+
+        start_time = time.time()
+
+        # Send chunk to output device
+        self.output_device.consume_nonblocking(chunk_result.chunk)
+
+        # CRITICAL: Wait for chunk to play before sending next one
+        # This is what makes interrupts work!
+        speech_length = seconds_per_chunk
+        processing_time = time.time() - start_time
+        await asyncio.sleep(max(speech_length - processing_time, 0))
+
+        chunk_idx += 1
+
+    return message, False  # cut_off = False
+```
+
+**Why Rate Limiting?**
+Without rate limiting, all audio chunks would be sent immediately, which would:
+- Buffer entire message on client side
+- Make interrupts impossible (all audio already sent)
+- Cause timing issues
+
+By sending one chunk every N seconds:
+- Real-time playback is maintained
+- Interrupts can stop mid-sentence
+- Natural conversation flow is preserved
+
+
+## The Interrupt System
+
+The interrupt system is critical for natural conversations.
+
+### How Interrupts Work
+
+**Scenario**: Bot is saying "I think the weather will be nice today and tomorrow and—" when user interrupts with "Stop".
+
+**Step 1: User starts speaking**
+```python
+# TranscriptionsWorker detects new transcription while bot speaking
+async def process(self, transcription):
+    if not self.conversation.is_human_speaking:  # Bot was speaking!
+        # Broadcast interrupt to all in-flight events
+        interrupted = self.conversation.broadcast_interrupt()
+        transcription.is_interrupt = interrupted
+```
+
+**Step 2: broadcast_interrupt() stops everything**
+```python
+def broadcast_interrupt(self):
+    num_interrupts = 0
+    # Interrupt all queued events
+    while True:
+        try:
+            interruptible_event = self.interruptible_events.get_nowait()
+            if interruptible_event.interrupt():  # Sets interruption_event
+                num_interrupts += 1
+        except queue.Empty:
+            break
+
+    # Cancel current tasks
+    self.agent.cancel_current_task()              # Stop generating text
+    self.agent_responses_worker.cancel_current_task()  # Stop synthesizing
+    return num_interrupts > 0
+```
+
+**Step 3: SynthesisResultsWorker detects interrupt**
+```python
+async def send_speech_to_output(self, synthesis_result, stop_event, ...):
+    async for chunk_result in synthesis_result.chunk_generator:
+        # Check stop_event (this is the interruption_event)
+        if stop_event.is_set():
+            logger.debug("Interrupted! Stopping speech.")
+            # Calculate what was actually spoken
+            seconds_spoken = chunk_idx * seconds_per_chunk
+            partial_message = synthesis_result.get_message_up_to(seconds_spoken)
+            # e.g., "I think the weather will be nice today"
+            return partial_message, True  # cut_off = True
+```
+
+**Step 4: Agent updates history**
+```python
+if cut_off:
+    # Update conversation history with partial message
+    self.agent.update_last_bot_message_on_cut_off(message_sent)
+    # History now shows:
+    # Bot: "I think the weather will be nice today" (incomplete)
+```
+
+### InterruptibleEvent Pattern
+
+Every event in the pipeline is wrapped in an `InterruptibleEvent`:
+
+```python
+class InterruptibleEvent:
+    def __init__(self, payload, is_interruptible=True):
+        self.payload = payload
+        self.is_interruptible = is_interruptible
+        self.interruption_event = threading.Event()  # Initially not set
+        self.interrupted = False
+
+    def interrupt(self) -> bool:
+        """Interrupt this event"""
+        if not self.is_interruptible:
+            return False
+        if not self.interrupted:
+            self.interruption_event.set()  # Signal to stop!
+            self.interrupted = True
+            return True
+        return False
+
+    def is_interrupted(self) -> bool:
+        return self.interruption_event.is_set()
+```
+
+
+## Multi-Provider Factory Pattern
+
+Support multiple providers with a factory pattern:
+
+```python
+class VoiceHandler:
+    """Multi-provider factory for voice components"""
+
+    def create_transcriber(self, agent_config: Dict):
+        """Create transcriber based on transcriberProvider"""
+        provider = agent_config.get("transcriberProvider", "deepgram")
+
+        if provider == "deepgram":
+            return self._create_deepgram_transcriber(agent_config)
+        elif provider == "assemblyai":
+            return self._create_assemblyai_transcriber(agent_config)
+        elif provider == "azure":
+            return self._create_azure_transcriber(agent_config)
+        elif provider == "google":
+            return self._create_google_transcriber(agent_config)
+        else:
+            raise ValueError(f"Unknown transcriber provider: {provider}")
+
+    def create_agent(self, agent_config: Dict):
+        """Create LLM agent based on llmProvider"""
+        provider = agent_config.get("llmProvider", "openai")
+
+        if provider == "openai":
+            return self._create_openai_agent(agent_config)
+        elif provider == "gemini":
+            return self._create_gemini_agent(agent_config)
+        else:
+            raise ValueError(f"Unknown LLM provider: {provider}")
+
+    def create_synthesizer(self, agent_config: Dict):
+        """Create voice synthesizer based on voiceProvider"""
+        provider = agent_config.get("voiceProvider", "elevenlabs")
+
+        if provider == "elevenlabs":
+            return self._create_elevenlabs_synthesizer(agent_config)
+        elif provider == "azure":
+            return self._create_azure_synthesizer(agent_config)
+        elif provider == "google":
+            return self._create_google_synthesizer(agent_config)
+        elif provider == "polly":
+            return self._create_polly_synthesizer(agent_config)
+        elif provider == "playht":
+            return self._create_playht_synthesizer(agent_config)
+        else:
+            raise ValueError(f"Unknown voice provider: {provider}")
+```
+
+
+## WebSocket Integration
+
+Voice AI engines typically use WebSocket for bidirectional audio streaming:
+
+```python
+@app.websocket("/conversation")
+async def websocket_endpoint(websocket: WebSocket):
+    await websocket.accept()
+
+    # Create voice components
+    voice_handler = VoiceHandler()
+    transcriber = voice_handler.create_transcriber(agent_config)
+    agent = voice_handler.create_agent(agent_config)
+    synthesizer = voice_handler.create_synthesizer(agent_config)
+
+    # Create output device
+    output_device = WebsocketOutputDevice(
+        ws=websocket,
+        sampling_rate=16000,
+        audio_encoding=AudioEncoding.LINEAR16
+    )
+
+    # Create conversation orchestrator
+    conversation = StreamingConversation(
+        output_device=output_device,
+        transcriber=transcriber,
+        agent=agent,
+        synthesizer=synthesizer
+    )
+
+    # Start all workers
+    await conversation.start()
+
+    try:
+        # Receive audio from client
+        async for message in websocket.iter_bytes():
+            conversation.receive_audio(message)
+    except WebSocketDisconnect:
+        logger.info("Client disconnected")
+    finally:
+        await conversation.terminate()
+```
+
+
+## Common Pitfalls and Solutions
+
+### 1. Audio Jumping/Cutting Off
+
+**Problem**: Bot's audio jumps or cuts off mid-response.
+
+**Cause**: Sending text to synthesizer in small chunks causes multiple TTS calls.
+
+**Solution**: Buffer the entire LLM response before sending to synthesizer:
+
+```python
+# ❌ Bad: Yields sentence-by-sentence
+async for sentence in llm_stream:
+    yield GeneratedResponse(message=BaseMessage(text=sentence))
+
+# ✅ Good: Buffer entire response
+full_response = ""
+async for chunk in llm_stream:
+    full_response += chunk
+yield GeneratedResponse(message=BaseMessage(text=full_response))
+```
+
+### 2. Echo/Feedback Loop
+
+**Problem**: Bot hears itself speaking and responds to its own audio.
+
+**Cause**: Transcriber not muted during bot speech.
+
+**Solution**: Mute transcriber when bot starts speaking:
+
+```python
+# Before sending audio to output
+self.transcriber.mute()
+# After audio playback complete
+self.transcriber.unmute()
+```
+
+### 3. Interrupts Not Working
+
+**Problem**: User can't interrupt bot mid-sentence.
+
+**Cause**: All audio chunks sent at once instead of rate-limited.
+
+**Solution**: Rate-limit audio chunks to match real-time playback:
+
+```python
+async for chunk in synthesis_result.chunk_generator:
+    start_time = time.time()
+
+    # Send chunk
+    output_device.consume_nonblocking(chunk)
+
+    # Wait for chunk duration before sending next
+    processing_time = time.time() - start_time
+    await asyncio.sleep(max(seconds_per_chunk - processing_time, 0))
+```
+
+### 4. Memory Leaks from Unclosed Streams
+
+**Problem**: Memory usage grows over time.
+
+**Cause**: WebSocket connections or API streams not properly closed.
+
+**Solution**: Always use context managers and cleanup:
+
+```python
+try:
+    async with websockets.connect(url) as ws:
+        # Use websocket
+        pass
+finally:
+    # Cleanup
+    await conversation.terminate()
+    await transcriber.terminate()
+```
+
+
+## Production Considerations
+
+### 1. Error Handling
+
+```python
+async def _run_loop(self):
+    while self.active:
+        try:
+            item = await self.input_queue.get()
+            await self.process(item)
+        except Exception as e:
+            logger.error(f"Worker error: {e}", exc_info=True)
+            # Don't crash the worker, continue processing
+```
+
+### 2. Graceful Shutdown
+
+```python
+async def terminate(self):
+    """Gracefully shut down all workers"""
+    self.active = False
+
+    # Stop all workers
+    self.transcriber.terminate()
+    self.agent.terminate()
+    self.synthesizer.terminate()
+
+    # Wait for queues to drain
+    await asyncio.sleep(0.5)
+
+    # Close connections
+    if self.websocket:
+        await self.websocket.close()
+```
+
+### 3. Monitoring and Logging
+
+```python
+# Log key events
+logger.info(f"🎤 [TRANSCRIBER] Received: '{transcription.message}'")
+logger.info(f"🤖 [AGENT] Generating response...")
+logger.info(f"🔊 [SYNTHESIZER] Synthesizing {len(text)} characters")
+logger.info(f"⚠️ [INTERRUPT] User interrupted bot")
+
+# Track metrics
+metrics.increment("transcriptions.count")
+metrics.timing("agent.response_time", duration)
+metrics.gauge("active_conversations", count)
+```
+
+### 4. Rate Limiting and Quotas
+
+```python
+# Implement rate limiting for API calls
+from aiolimiter import AsyncLimiter
+
+rate_limiter = AsyncLimiter(max_rate=10, time_period=1)  # 10 calls/second
+
+async def call_api(self, data):
+    async with rate_limiter:
+        return await self.client.post(data)
+```
+
+
+## Key Design Patterns
+
+### 1. Producer-Consumer with Queues
+
+```python
+# Producer
+async def producer(queue):
+    while True:
+        item = await generate_item()
+        queue.put_nowait(item)
+
+# Consumer
+async def consumer(queue):
+    while True:
+        item = await queue.get()
+        await process_item(item)
+```
+
+### 2. Streaming Generators
+
+Instead of returning complete results:
+
+```python
+# ❌ Bad: Wait for entire response
+async def generate_response(prompt):
+    response = await openai.complete(prompt)  # 5 seconds
+    return response
+
+# ✅ Good: Stream chunks as they arrive
+async def generate_response(prompt):
+    async for chunk in openai.complete(prompt, stream=True):
+        yield chunk  # Yield after 0.1s, 0.2s, etc.
+```
+
+### 3. Conversation State Management
+
+Maintain conversation history for context:
+
+```python
+class Transcript:
+    event_logs: List[Message] = []
+
+    def add_human_message(self, text):
+        self.event_logs.append(Message(sender=Sender.HUMAN, text=text))
+
+    def add_bot_message(self, text):
+        self.event_logs.append(Message(sender=Sender.BOT, text=text))
+
+    def to_openai_messages(self):
+        return [
+            {"role": "user" if msg.sender == Sender.HUMAN else "assistant",
+             "content": msg.text}
+            for msg in self.event_logs
+        ]
+```
+
+
+## Testing Strategies
+
+### 1. Unit Test Workers in Isolation
+
+```python
+async def test_transcriber():
+    transcriber = DeepgramTranscriber(config)
+
+    # Mock audio input
+    audio_chunk = b'\x00\x01\x02...'
+    transcriber.send_audio(audio_chunk)
+
+    # Check output
+    transcription = await transcriber.output_queue.get()
+    assert transcription.message == "expected text"
+```
+
+### 2. Integration Test Pipeline
+
+```python
+async def test_full_pipeline():
+    # Create all components
+    conversation = create_test_conversation()
+
+    # Send test audio
+    conversation.receive_audio(test_audio_chunk)
+
+    # Wait for response
+    response = await wait_for_audio_output(timeout=5)
+
+    assert response is not None
+```
+
+### 3. Test Interrupts
+
+```python
+async def test_interrupt():
+    conversation = create_test_conversation()
+
+    # Start bot speaking
+    await conversation.agent.generate_response("Tell me a long story")
+
+    # Interrupt mid-response
+    await asyncio.sleep(1)  # Let it speak for 1 second
+    conversation.broadcast_interrupt()
+
+    # Verify partial message in transcript
+    last_message = conversation.transcript.event_logs[-1]
+    assert last_message.text != full_expected_message
+```
+
+
+## Implementation Workflow
+
+When implementing a voice AI engine:
+
+1. **Start with Base Workers**: Implement the base worker pattern first
+2. **Add Transcriber**: Choose a provider and implement streaming transcription
+3. **Add Agent**: Implement LLM integration with streaming responses
+4. **Add Synthesizer**: Implement TTS with audio streaming
+5. **Connect Pipeline**: Wire all workers together with queues
+6. **Add Interrupts**: Implement the interrupt system
+7. **Add WebSocket**: Create WebSocket endpoint for client communication
+8. **Test Components**: Unit test each worker in isolation
+9. **Test Integration**: Test the full pipeline end-to-end
+10. **Add Error Handling**: Implement robust error handling and logging
+11. **Optimize**: Add rate limiting, monitoring, and performance optimizations
+
+
+## Related Skills
+
+- `@websocket-patterns` - For WebSocket implementation details
+- `@async-python` - For asyncio and async patterns
+- `@streaming-apis` - For streaming API integration
+- `@audio-processing` - For audio format conversion and processing
+- `@systematic-debugging` - For debugging complex async pipelines
+
+
+## Resources
+
+**Libraries**:
+- `asyncio` - Async programming
+- `websockets` - WebSocket client/server
+- `FastAPI` - WebSocket server framework
+- `pydub` - Audio manipulation
+- `numpy` - Audio data processing
+
+**API Providers**:
+- Transcription: Deepgram, AssemblyAI, Azure Speech, Google Cloud Speech
+- LLM: OpenAI, Google Gemini, Anthropic Claude
+- TTS: ElevenLabs, Azure TTS, Google Cloud TTS, Amazon Polly, Play.ht
+
+
+## Summary
+
+Building a voice AI engine requires:
+- ✅ Async worker pipeline for concurrent processing
+- ✅ Queue-based communication between components
+- ✅ Streaming at every stage (transcription, LLM, synthesis)
+- ✅ Interrupt system for natural conversations
+- ✅ Rate limiting for real-time audio playback
+- ✅ Multi-provider support for flexibility
+- ✅ Proper error handling and graceful shutdown
+
+**The key insight**: Everything must stream and everything must be interruptible for natural, real-time conversations.