Created an example for testing the bot and try to create freezing conditions.

This commit is contained in:
Filipi Fuchter
2025-06-23 18:46:58 -03:00
parent f52ac6e99c
commit c9cebb5ffe
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# Freeze Test Client
The purpose of this example is to create an environment for testing the bot and try to create freezing conditions.
### Approach 1: Server-Side Testing with `SimulateFreezeInput`
- Utilize only the bot `freeze_test_bot.py` with the `SimulateFreezeInput` processor. This input continuously injects frames, simulating user speech interruptions at random intervals.
- This approach excludes the use of input transport and speech-to-text (STT) functionalities.
### Approach 2: Server-Side with TypeScript Client
- Combine server-side operations with a TypeScript client.
- The client initially records a segment of audio, e.g., 510 seconds long. It can be anything.
- After that, it replays this recorded audio to the server at random intervals, mimicking user input interruptions.
- This helps testing interruptions in the pipeline as if real users were interacting with the bot.
## Setup
Follow these steps to set up and run the Freeze Test Client:
1. **Run the Bot Server**
- Set up and activate your virtual environment:
```bash
python3 -m venv venv
source venv/bin/activate # On Windows: venv\Scripts\activate
```
- Install dependencies:
```bash
pip install -r requirements.txt
```
- Create your `.env` file and set your env vars:
```bash
cp env.example .env
```
- Run the server:
```bash
python freeze_test_bot.py
```
2. **Navigate to the Client Directory**
```bash
cd client
```
3. **Install Dependencies**
```bash
npm install
```
4. **Run the Client Application**
```bash
npm run dev
```
5. **Access the Client in Your Browser**
Visit [http://localhost:5173](http://localhost:5173) to interact with the Freeze Test Client.

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<!DOCTYPE html>
<html lang="en">
<head>
<meta charset="UTF-8">
<meta name="viewport" content="width=device-width, initial-scale=1.0">
<title>AI Chatbot</title>
</head>
<body>
<div class="container">
<div class="status-bar">
<div class="status">
Transport: <span id="connection-status">Disconnected</span>
</div>
<div class="controls">
<button id="connect-btn">Connect</button>
<button id="disconnect-btn" disabled>Disconnect</button>
</div>
</div>
<div class="status-bar">
<div class="status">
Playing audio: <span id="play-audio-status"></span>
</div>
<div class="controls">
<button id="play-btn">Start</button>
<button id="stop-btn" disabled>Stop</button>
</div>
</div>
<audio id="bot-audio" autoplay></audio>
<div class="debug-panel">
<h3>Debug Info</h3>
<div id="debug-log"></div>
</div>
</div>
<script type="module" src="/src/app.ts"></script>
<link rel="stylesheet" href="/src/style.css">
</body>
</html>

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{
"name": "client",
"version": "1.0.0",
"main": "index.js",
"scripts": {
"dev": "vite",
"build": "tsc && vite build",
"preview": "vite preview"
},
"keywords": [],
"author": "",
"license": "ISC",
"description": "",
"devDependencies": {
"@types/node": "^22.15.30",
"@types/protobufjs": "^6.0.0",
"@vitejs/plugin-react-swc": "^3.10.1",
"typescript": "^5.8.3",
"vite": "^6.3.5"
},
"dependencies": {
"@pipecat-ai/client-js": "^0.4.0",
"@pipecat-ai/websocket-transport": "^0.4.1",
"protobufjs": "^7.4.0"
}
}

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/**
* Copyright (c) 20242025, Daily
*
* SPDX-License-Identifier: BSD 2-Clause License
*/
/**
* RTVI Client Implementation
*
* This client connects to an RTVI-compatible bot server using WebSocket.
*
* Requirements:
* - A running RTVI bot server (defaults to http://localhost:7860)
*/
import {
RTVIClient,
RTVIClientOptions,
RTVIEvent,
} from '@pipecat-ai/client-js';
import {
ProtobufFrameSerializer,
WebSocketTransport
} from "@pipecat-ai/websocket-transport";
class RecordingSerializer extends ProtobufFrameSerializer {
private lastTimestamp: number | null = null;
private recordingAudioToSend: boolean = false;
private _recordedAudio: { data: ArrayBuffer; delay: number }[] = [];
public startRecording() {
this.recordingAudioToSend = true;
this._recordedAudio = [];
this.lastTimestamp = null;
}
public stopRecording() {
this.recordingAudioToSend = false;
}
// @ts-ignore
serializeAudio(data: ArrayBuffer, sampleRate: number, numChannels: number): Uint8Array | null {
if (this.recordingAudioToSend) {
const now = Date.now();
// Compute delay since last packet
const delay = this.lastTimestamp ? now - this.lastTimestamp : 0;
this.lastTimestamp = now;
// Save audio chunk and delay
this._recordedAudio.push({ data, delay });
return null;
} else {
return super.serializeAudio(data, sampleRate, numChannels);
}
}
public get recordedAudio() {
return this._recordedAudio
}
}
class WebsocketClientApp {
private ENABLE_RECORDING_MODE = false
private RECORDING_TIME_MS = 10000
private rtviClient: RTVIClient | null = null;
private connectBtn: HTMLButtonElement | null = null;
private disconnectBtn: HTMLButtonElement | null = null;
private statusSpan: HTMLElement | null = null;
private debugLog: HTMLElement | null = null;
private botAudio: HTMLAudioElement;
private declare websocketTransport: WebSocketTransport;
private sendRecordedAudio: boolean = false
private declare recordingSerializer: RecordingSerializer;
private playBtn: HTMLButtonElement | null = null;
private stopBtn: HTMLButtonElement | null = null;
constructor() {
this.botAudio = document.createElement('audio');
this.botAudio.autoplay = true;
//this.botAudio.playsInline = true;
document.body.appendChild(this.botAudio);
this.setupDOMElements();
this.setupEventListeners();
}
/**
* Set up references to DOM elements and create necessary media elements
*/
private setupDOMElements(): void {
this.connectBtn = document.getElementById('connect-btn') as HTMLButtonElement;
this.disconnectBtn = document.getElementById('disconnect-btn') as HTMLButtonElement;
this.statusSpan = document.getElementById('connection-status');
this.debugLog = document.getElementById('debug-log');
this.playBtn = document.getElementById('play-btn') as HTMLButtonElement;
this.stopBtn = document.getElementById('stop-btn') as HTMLButtonElement;
}
/**
* Set up event listeners for connect/disconnect buttons
*/
private setupEventListeners(): void {
this.connectBtn?.addEventListener('click', () => this.connect());
this.disconnectBtn?.addEventListener('click', () => this.disconnect());
this.playBtn?.addEventListener('click', () => this.startSendingRecordedAudio());
this.stopBtn?.addEventListener('click', () => this.stopSendingRecordedAudio());
}
/**
* Add a timestamped message to the debug log
*/
private log(message: string): void {
if (!this.debugLog) return;
const entry = document.createElement('div');
entry.textContent = `${new Date().toISOString()} - ${message}`;
if (message.startsWith('User: ')) {
entry.style.color = '#2196F3';
} else if (message.startsWith('Bot: ')) {
entry.style.color = '#4CAF50';
}
this.debugLog.appendChild(entry);
this.debugLog.scrollTop = this.debugLog.scrollHeight;
console.log(message);
}
/**
* Update the connection status display
*/
private updateStatus(status: string): void {
if (this.statusSpan) {
this.statusSpan.textContent = status;
}
this.log(`Status: ${status}`);
}
/**
* Check for available media tracks and set them up if present
* This is called when the bot is ready or when the transport state changes to ready
*/
setupMediaTracks() {
if (!this.rtviClient) return;
const tracks = this.rtviClient.tracks();
if (tracks.bot?.audio) {
this.setupAudioTrack(tracks.bot.audio);
}
}
/**
* Set up listeners for track events (start/stop)
* This handles new tracks being added during the session
*/
setupTrackListeners() {
if (!this.rtviClient) return;
// Listen for new tracks starting
this.rtviClient.on(RTVIEvent.TrackStarted, (track, participant) => {
// Only handle non-local (bot) tracks
if (!participant?.local && track.kind === 'audio') {
this.setupAudioTrack(track);
}
});
// Listen for tracks stopping
this.rtviClient.on(RTVIEvent.TrackStopped, (track, participant) => {
this.log(`Track stopped: ${track.kind} from ${participant?.name || 'unknown'}`);
});
}
/**
* Set up an audio track for playback
* Handles both initial setup and track updates
*/
private setupAudioTrack(track: MediaStreamTrack): void {
this.log('Setting up audio track');
if (this.botAudio.srcObject && "getAudioTracks" in this.botAudio.srcObject) {
const oldTrack = this.botAudio.srcObject.getAudioTracks()[0];
if (oldTrack?.id === track.id) return;
}
this.botAudio.srcObject = new MediaStream([track]);
}
/**
* Initialize and connect to the bot
* This sets up the RTVI client, initializes devices, and establishes the connection
*/
public async connect(): Promise<void> {
try {
const startTime = Date.now();
this.recordingSerializer = new RecordingSerializer()
const transport = this.ENABLE_RECORDING_MODE ? new WebSocketTransport({serializer: this.recordingSerializer}) : new WebSocketTransport();
this.websocketTransport = transport
const RTVIConfig: RTVIClientOptions = {
transport,
params: {
// The baseURL and endpoint of your bot server that the client will connect to
baseUrl: 'http://localhost:7860',
endpoints: { connect: '/connect' },
},
enableMic: true,
enableCam: false,
callbacks: {
onConnected: () => {
this.updateStatus('Connected');
if (this.connectBtn) this.connectBtn.disabled = true;
if (this.disconnectBtn) this.disconnectBtn.disabled = false;
},
onDisconnected: () => {
this.updateStatus('Disconnected');
if (this.connectBtn) this.connectBtn.disabled = false;
if (this.disconnectBtn) this.disconnectBtn.disabled = true;
this.log('Client disconnected');
},
onBotReady: (data) => {
this.log(`Bot ready: ${JSON.stringify(data)}`);
this.setupMediaTracks();
},
onUserTranscript: (data) => {
if (data.final) {
this.log(`User: ${data.text}`);
}
},
onBotTranscript: (data) => this.log(`Bot: ${data.text}`),
onMessageError: (error) => console.error('Message error:', error),
onError: (error) => console.error('Error:', error),
},
}
this.rtviClient = new RTVIClient(RTVIConfig);
this.setupTrackListeners();
this.log('Initializing devices...');
await this.rtviClient.initDevices();
this.log('Connecting to bot...');
await this.rtviClient.connect();
const timeTaken = Date.now() - startTime;
this.log(`Connection complete, timeTaken: ${timeTaken}`);
if (this.ENABLE_RECORDING_MODE) {
this.log(`Starting to recording the next ${(this.RECORDING_TIME_MS/1000)}s of audio`);
this.recordingSerializer.startRecording()
await this.sleep(this.RECORDING_TIME_MS)
this.recordingSerializer.stopRecording()
this.log("Recording stopped");
this.rtviClient.enableMic(false)
this.startSendingRecordedAudio()
}
} catch (error) {
this.log(`Error connecting: ${(error as Error).message}`);
this.updateStatus('Error');
// Clean up if there's an error
if (this.rtviClient) {
try {
await this.rtviClient.disconnect();
} catch (disconnectError) {
this.log(`Error during disconnect: ${disconnectError}`);
}
}
}
}
/**
* Disconnect from the bot and clean up media resources
*/
public async disconnect(): Promise<void> {
if (this.rtviClient) {
try {
this.stopSendingRecordedAudio()
await this.rtviClient.disconnect();
this.rtviClient = null;
if (this.botAudio.srcObject && "getAudioTracks" in this.botAudio.srcObject) {
this.botAudio.srcObject.getAudioTracks().forEach((track) => track.stop());
this.botAudio.srcObject = null;
}
} catch (error) {
this.log(`Error disconnecting: ${(error as Error).message}`);
}
}
}
private startSendingRecordedAudio() {
this.sendRecordedAudio = true
if (this.playBtn) this.playBtn.disabled = true;
if (this.stopBtn) this.stopBtn.disabled = false;
void this.replayAudio()
}
private stopSendingRecordedAudio() {
if (this.stopBtn) this.stopBtn.disabled = true;
if (this.playBtn) this.playBtn.disabled = false;
this.sendRecordedAudio = false
}
private async replayAudio() {
if (this.sendRecordedAudio) {
this.log("Sending recorded audio")
for (const chunk of this.recordingSerializer.recordedAudio) {
await this.sleep(chunk.delay);
this.websocketTransport.handleUserAudioStream(chunk.data);
}
const randomDelay = 1000 + Math.random() * (10000 - 500);
await this.sleep(randomDelay);
void this.replayAudio()
}
}
private sleep(ms: number): Promise<void> {
return new Promise(resolve => setTimeout(resolve, ms));
}
}
declare global {
interface Window {
WebsocketClientApp: typeof WebsocketClientApp;
}
}
window.addEventListener('DOMContentLoaded', () => {
window.WebsocketClientApp = WebsocketClientApp;
new WebsocketClientApp();
});

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body {
margin: 0;
padding: 20px;
font-family: Arial, sans-serif;
background-color: #f0f0f0;
}
.container {
max-width: 1200px;
margin: 0 auto;
}
.status-bar {
display: flex;
justify-content: space-between;
align-items: center;
padding: 10px;
background-color: #fff;
border-radius: 8px;
margin-bottom: 20px;
}
.controls button {
padding: 8px 16px;
margin-left: 10px;
border: none;
border-radius: 4px;
cursor: pointer;
}
#connect-btn {
background-color: #4caf50;
color: white;
}
#disconnect-btn {
background-color: #f44336;
color: white;
}
button:disabled {
opacity: 0.5;
cursor: not-allowed;
}
.main-content {
background-color: #fff;
border-radius: 8px;
padding: 20px;
margin-bottom: 20px;
}
.bot-container {
display: flex;
flex-direction: column;
align-items: center;
}
#bot-video-container {
width: 640px;
height: 360px;
background-color: #e0e0e0;
border-radius: 8px;
margin: 20px auto;
overflow: hidden;
display: flex;
align-items: center;
justify-content: center;
}
#bot-video-container video {
width: 100%;
height: 100%;
object-fit: cover;
}
.debug-panel {
background-color: #fff;
border-radius: 8px;
padding: 20px;
}
.debug-panel h3 {
margin: 0 0 10px 0;
font-size: 16px;
font-weight: bold;
}
#debug-log {
height: 500px;
overflow-y: auto;
background-color: #f8f8f8;
padding: 10px;
border-radius: 4px;
font-family: monospace;
font-size: 12px;
line-height: 1.4;
}

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import { defineConfig } from 'vite';
import react from '@vitejs/plugin-react-swc';
export default defineConfig({
plugins: [react()],
server: {
proxy: {
// Proxy /api requests to the backend server
'/connect': {
target: 'http://0.0.0.0:7860', // Replace with your backend URL
changeOrigin: true,
},
},
},
});

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#
# Copyright (c) 20242025, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
import argparse
import asyncio
import os
import random
from contextlib import asynccontextmanager
from typing import Any, Dict
import uvicorn
from dotenv import load_dotenv
from fastapi import FastAPI, Request, WebSocket
from fastapi.middleware.cors import CORSMiddleware
from fastapi.responses import RedirectResponse
from loguru import logger
from pipecat_ai_small_webrtc_prebuilt.frontend import SmallWebRTCPrebuiltUI
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.frames.frames import (
CancelFrame,
EndFrame,
Frame,
InterimTranscriptionFrame,
LLMFullResponseEndFrame,
LLMTextFrame,
StartFrame,
StartInterruptionFrame,
StopFrame,
StopInterruptionFrame,
TranscriptionFrame,
TTSTextFrame,
UserStartedSpeakingFrame,
UserStoppedSpeakingFrame,
)
from pipecat.observers.loggers.debug_log_observer import DebugLogObserver
from pipecat.pipeline.parallel_pipeline import ParallelPipeline
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.processors.aggregators.openai_llm_context import (
OpenAILLMContext,
OpenAILLMContextFrame,
)
from pipecat.processors.frame_processor import FrameDirection, FrameProcessor
from pipecat.processors.frameworks.rtvi import RTVIConfig, RTVIProcessor
from pipecat.serializers.protobuf import ProtobufFrameSerializer
from pipecat.services.cartesia.tts import CartesiaTTSService
from pipecat.services.deepgram import DeepgramSTTService
from pipecat.services.openai.llm import OpenAILLMService
from pipecat.transports.network.fastapi_websocket import (
FastAPIWebsocketParams,
FastAPIWebsocketTransport,
)
from pipecat.utils.time import time_now_iso8601
load_dotenv(override=True)
@asynccontextmanager
async def lifespan(app: FastAPI):
"""Handles FastAPI startup and shutdown."""
yield # Run app
# Initialize FastAPI app with lifespan manager
app = FastAPI(lifespan=lifespan)
# Configure CORS to allow requests from any origin
app.add_middleware(
CORSMiddleware,
allow_origins=["*"],
allow_credentials=True,
allow_methods=["*"],
allow_headers=["*"],
)
# Mount the frontend at /
app.mount("/client", SmallWebRTCPrebuiltUI)
class SimulateFreezeInput(FrameProcessor):
def __init__(
self,
**kwargs,
):
super().__init__(**kwargs)
# Whether we have seen a StartFrame already.
self._initialized = False
self._send_frames_task = None
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if isinstance(frame, StartFrame):
# Push StartFrame before start(), because we want StartFrame to be
# processed by every processor before any other frame is processed.
await self.push_frame(frame, direction)
await self._start(frame)
elif isinstance(frame, CancelFrame):
logger.info("SimulateFreezeInput: Received cancel frame")
await self._stop()
await self.push_frame(frame, direction)
elif isinstance(frame, EndFrame):
logger.info("SimulateFreezeInput: Received end frame")
await self.push_frame(frame, direction)
await self._stop()
elif isinstance(frame, StopFrame):
logger.info("SimulateFreezeInput: Received stop frame")
await self.push_frame(frame, direction)
await self._stop()
async def _start(self, frame: StartFrame):
if self._initialized:
return
logger.info(f"Starting SimulateFreezeInput")
self._initialized = True
if not self._send_frames_task:
self._send_frames_task = self.create_task(self._send_frames())
async def _stop(self):
logger.info(f"Stopping SimulateFreezeInput")
self._initialized = False
if self._send_frames_task:
await self.cancel_task(self._send_frames_task)
self._send_frames_task = None
async def _send_user_text(self, text: str):
# Emulation as if the user has spoken and the stt transcribed
await self.push_frame(UserStartedSpeakingFrame())
await self.push_frame(StartInterruptionFrame())
await self.push_frame(
TranscriptionFrame(
text,
"",
time_now_iso8601(),
)
)
# Need to wait before sending the UserStoppedSpeakingFrame,
# otherwise TranscriptionFrame will be processed
# later than the UserStoppedSpeakingFrame
await asyncio.sleep(0.1)
await self.push_frame(UserStoppedSpeakingFrame())
await self.push_frame(StopInterruptionFrame())
async def _send_frames(self):
try:
i = 0
while True:
logger.debug("SimulateFreezeInput _send_frames")
await self._send_user_text("Tell me a brief history of Brazil!")
await asyncio.sleep(3)
await self._send_user_text("")
break
# i += 1
# if i >= 5:
# break
# sleeping 1s before interrupting
# wait_time = random.uniform(1, 10)
# await asyncio.sleep(wait_time)
except Exception as e:
logger.error(f"{self} exception receiving data: {e.__class__.__name__} ({e})")
async def run_example(websocket_client):
logger.info(f"Starting bot")
# Create a transport using the WebRTC connection
transport = FastAPIWebsocketTransport(
websocket=websocket_client,
params=FastAPIWebsocketParams(
audio_in_enabled=True,
audio_out_enabled=True,
add_wav_header=False,
vad_analyzer=SileroVADAnalyzer(),
serializer=ProtobufFrameSerializer(),
),
)
freeze = SimulateFreezeInput()
stt = DeepgramSTTService(api_key=os.getenv("DEEPGRAM_API_KEY"))
tts = CartesiaTTSService(
api_key=os.getenv("CARTESIA_API_KEY"),
voice_id="71a7ad14-091c-4e8e-a314-022ece01c121", # British Reading Lady
)
llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"))
rtvi = RTVIProcessor(config=RTVIConfig(config=[]))
messages = [
{
"role": "system",
"content": "You are a helpful LLM in a WebRTC call. Your goal is to demonstrate your capabilities in a succinct way. Your output will be converted to audio so don't include special characters in your answers. Respond to what the user said in a creative and helpful way.",
},
]
context = OpenAILLMContext(messages)
context_aggregator = llm.create_context_aggregator(context)
pipeline = Pipeline(
[
ParallelPipeline(
[
freeze,
],
[
transport.input(),
stt,
],
),
rtvi,
context_aggregator.user(), # User responses
llm, # LLM
tts, # TTS
transport.output(), # Transport bot output
context_aggregator.assistant(), # Assistant spoken responses
]
)
task = PipelineTask(
pipeline,
params=PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
report_only_initial_ttfb=True,
),
idle_timeout_secs=120,
observers=[
DebugLogObserver(
frame_types={
InterimTranscriptionFrame: None,
TranscriptionFrame: None,
# TTSTextFrame: None,
# LLMTextFrame: None,
OpenAILLMContextFrame: None,
LLMFullResponseEndFrame: None,
},
exclude_fields={
"result",
"metadata",
"audio",
"image",
"images",
},
),
],
)
@transport.event_handler("on_client_connected")
async def on_client_connected(transport, client):
logger.info(f"Client connected")
@rtvi.event_handler("on_client_ready")
async def on_client_ready(rtvi):
logger.info(f"Client ready")
await rtvi.set_bot_ready()
# Kick off the conversation.
# messages.append({"role": "system", "content": "Please introduce yourself to the user."})
# await task.queue_frames([context_aggregator.user().get_context_frame()])
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
await task.cancel()
runner = PipelineRunner(handle_sigint=False)
await runner.run(task)
@app.get("/", include_in_schema=False)
async def root_redirect():
return RedirectResponse(url="/client/")
@app.websocket("/ws")
async def websocket_endpoint(websocket: WebSocket):
await websocket.accept()
print("WebSocket connection accepted")
try:
await run_example(websocket)
except Exception as e:
print(f"Exception in run_bot: {e}")
@app.post("/connect")
async def bot_connect(request: Request) -> Dict[Any, Any]:
server_mode = os.getenv("WEBSOCKET_SERVER", "fast_api")
if server_mode == "websocket_server":
ws_url = "ws://localhost:8765"
else:
ws_url = "ws://localhost:7860/ws"
return {"ws_url": ws_url}
if __name__ == "__main__":
parser = argparse.ArgumentParser(description="Pipecat Bot Runner")
parser.add_argument(
"--host", default="localhost", help="Host for HTTP server (default: localhost)"
)
parser.add_argument(
"--port", type=int, default=7860, help="Port for HTTP server (default: 7860)"
)
args = parser.parse_args()
uvicorn.run(app, host=args.host, port=args.port)