From e9f041e170d24ed79ef772131a4cedfd271532e7 Mon Sep 17 00:00:00 2001
From: Filipi Fuchter <filipi87@gmail.com>
Date: Fri, 6 Jun 2025 17:09:01 -0300
Subject: [PATCH] Removing the old websocket-server example

---
 examples/websocket-server/Dockerfile       |  15 --
 examples/websocket-server/README.md        |  28 ---
 examples/websocket-server/bot.py           | 153 ---------------
 examples/websocket-server/env.example      |   8 -
 examples/websocket-server/frames.proto     |  44 -----
 examples/websocket-server/index.html       | 211 ---------------------
 examples/websocket-server/requirements.txt |   2 -
 7 files changed, 461 deletions(-)
 delete mode 100644 examples/websocket-server/Dockerfile
 delete mode 100644 examples/websocket-server/README.md
 delete mode 100644 examples/websocket-server/bot.py
 delete mode 100644 examples/websocket-server/env.example
 delete mode 100644 examples/websocket-server/frames.proto
 delete mode 100644 examples/websocket-server/index.html
 delete mode 100644 examples/websocket-server/requirements.txt

diff --git a/examples/websocket-server/Dockerfile b/examples/websocket-server/Dockerfile
deleted file mode 100644
index 0610ab7f8..000000000
--- a/examples/websocket-server/Dockerfile
+++ /dev/null
@@ -1,15 +0,0 @@
-FROM python:3.10-bullseye
-
-RUN mkdir /app
-
-COPY *.py /app/
-COPY requirements.txt /app/
-COPY .env /app/
-
-WORKDIR /app
-
-RUN pip3 install -r requirements.txt
-
-EXPOSE 7860
-
-CMD ["python3", "bot.py"]
diff --git a/examples/websocket-server/README.md b/examples/websocket-server/README.md
deleted file mode 100644
index a8f2f1aac..000000000
--- a/examples/websocket-server/README.md
+++ /dev/null
@@ -1,28 +0,0 @@
-# Websocket Server
-
-This is an example that shows how to use `WebsocketServerTransport` to communicate with a web client.
-
-## Get started
-
-```python
-python3 -m venv venv
-source venv/bin/activate
-pip install -r requirements.txt
-cp env.example .env # and add your credentials
-```
-
-## Run the bot
-
-```bash
-python bot.py
-```
-
-## Run the HTTP server
-
-This will host the static web client:
-
-```bash
-python -m http.server
-```
-
-Then, visit `http://localhost:8000` in your browser to start a session.
diff --git a/examples/websocket-server/bot.py b/examples/websocket-server/bot.py
deleted file mode 100644
index 816d7540b..000000000
--- a/examples/websocket-server/bot.py
+++ /dev/null
@@ -1,153 +0,0 @@
-#
-# Copyright (c) 2024–2025, Daily
-#
-# SPDX-License-Identifier: BSD 2-Clause License
-#
-
-import asyncio
-import os
-import sys
-
-from dotenv import load_dotenv
-from loguru import logger
-
-from pipecat.audio.vad.silero import SileroVADAnalyzer
-from pipecat.frames.frames import BotInterruptionFrame, EndFrame
-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
-from pipecat.serializers.protobuf import ProtobufFrameSerializer
-from pipecat.services.cartesia.tts import CartesiaTTSService
-from pipecat.services.deepgram.stt import DeepgramSTTService
-from pipecat.services.openai.llm import OpenAILLMService
-from pipecat.transports.network.websocket_server import (
-    WebsocketServerParams,
-    WebsocketServerTransport,
-)
-
-load_dotenv(override=True)
-
-logger.remove(0)
-logger.add(sys.stderr, level="DEBUG")
-
-
-class SessionTimeoutHandler:
-    """Handles actions to be performed when a session times out.
-    Inputs:
-    - task: Pipeline task (used to queue frames).
-    - tts: TTS service (used to generate speech output).
-    """
-
-    def __init__(self, task, tts):
-        self.task = task
-        self.tts = tts
-        self.background_tasks = set()
-
-    async def handle_timeout(self, client_address):
-        """Handles the timeout event for a session."""
-        try:
-            logger.info(f"Connection timeout for {client_address}")
-
-            # Queue a BotInterruptionFrame to notify the user
-            await self.task.queue_frames([BotInterruptionFrame()])
-
-            # Send the TTS message to inform the user about the timeout
-            await self.tts.say(
-                "I'm sorry, we are ending the call now. Please feel free to reach out again if you need assistance."
-            )
-
-            # Start the process to gracefully end the call in the background
-            end_call_task = asyncio.create_task(self._end_call())
-            self.background_tasks.add(end_call_task)
-            end_call_task.add_done_callback(self.background_tasks.discard)
-        except Exception as e:
-            logger.error(f"Error during session timeout handling: {e}")
-
-    async def _end_call(self):
-        """Completes the session termination process after the TTS message."""
-        try:
-            # Wait for a duration to ensure TTS has completed
-            await asyncio.sleep(15)
-
-            # Queue both BotInterruptionFrame and EndFrame to conclude the session
-            await self.task.queue_frames([BotInterruptionFrame(), EndFrame()])
-
-            logger.info("TTS completed and EndFrame pushed successfully.")
-        except Exception as e:
-            logger.error(f"Error during call termination: {e}")
-
-
-async def main():
-    transport = WebsocketServerTransport(
-        params=WebsocketServerParams(
-            serializer=ProtobufFrameSerializer(),
-            audio_in_enabled=True,
-            audio_out_enabled=True,
-            add_wav_header=True,
-            vad_analyzer=SileroVADAnalyzer(),
-            session_timeout=60 * 3,  # 3 minutes
-        )
-    )
-
-    llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"))
-
-    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
-    )
-
-    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(
-        [
-            transport.input(),  # Websocket input from client
-            stt,  # Speech-To-Text
-            context_aggregator.user(),
-            llm,  # LLM
-            tts,  # Text-To-Speech
-            transport.output(),  # Websocket output to client
-            context_aggregator.assistant(),
-        ]
-    )
-
-    task = PipelineTask(
-        pipeline,
-        params=PipelineParams(
-            audio_in_sample_rate=16000,
-            audio_out_sample_rate=16000,
-            allow_interruptions=True,
-        ),
-    )
-
-    @transport.event_handler("on_client_connected")
-    async def on_client_connected(transport, client):
-        # 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_session_timeout")
-    async def on_session_timeout(transport, client):
-        logger.info(f"Entering in timeout for {client.remote_address}")
-
-        timeout_handler = SessionTimeoutHandler(task, tts)
-
-        await timeout_handler.handle_timeout(client)
-
-    runner = PipelineRunner()
-
-    await runner.run(task)
-
-
-if __name__ == "__main__":
-    asyncio.run(main())
diff --git a/examples/websocket-server/env.example b/examples/websocket-server/env.example
deleted file mode 100644
index c3359ada2..000000000
--- a/examples/websocket-server/env.example
+++ /dev/null
@@ -1,8 +0,0 @@
-# OpenAI API Key
-OPENAI_API_KEY=your_openai_api_key_here
-
-# Deepgram API Key
-DEEPGRAM_API_KEY=your_deepgram_api_key_here
-
-# Cartesia API Key
-CARTESIA_API_KEY=your_cartesia_api_key_here
diff --git a/examples/websocket-server/frames.proto b/examples/websocket-server/frames.proto
deleted file mode 100644
index 98dc014db..000000000
--- a/examples/websocket-server/frames.proto
+++ /dev/null
@@ -1,44 +0,0 @@
-//
-// Copyright (c) 2024–2025, Daily
-//
-// SPDX-License-Identifier: BSD 2-Clause License
-//
-
-// Generate frames_pb2.py with:
-//
-//   python -m grpc_tools.protoc --proto_path=./ --python_out=./protobufs frames.proto
-
-syntax = "proto3";
-
-package pipecat;
-
-message TextFrame {
-  uint64 id = 1;
-  string name = 2;
-  string text = 3;
-}
-
-message AudioRawFrame {
-  uint64 id = 1;
-  string name = 2;
-  bytes audio = 3;
-  uint32 sample_rate = 4;
-  uint32 num_channels = 5;
-  optional uint64 pts = 6;
-}
-
-message TranscriptionFrame {
-  uint64 id = 1;
-  string name = 2;
-  string text = 3;
-  string user_id = 4;
-  string timestamp = 5;
-}
-
-message Frame {
-  oneof frame {
-    TextFrame text = 1;
-    AudioRawFrame audio = 2;
-    TranscriptionFrame transcription = 3;
-  }
-}
diff --git a/examples/websocket-server/index.html b/examples/websocket-server/index.html
deleted file mode 100644
index 6f0bd1ada..000000000
--- a/examples/websocket-server/index.html
+++ /dev/null
@@ -1,211 +0,0 @@
-<!DOCTYPE html>
-<html lang="en">
-
-  <head>
-    <meta charset="UTF-8">
-    <meta name="viewport" content="width=device-width, initial-scale=1.0">
-    <script src="https://cdn.jsdelivr.net/npm/protobufjs@7.X.X/dist/protobuf.min.js"></script>
-    <title>Pipecat WebSocket Client Example</title>
-  </head>
-
-  <body>
-    <h1>Pipecat WebSocket Client Example</h1>
-    <h3><div id="progressText">Loading, wait...</div></h3>
-    <button id="startAudioBtn">Start Audio</button>
-    <button id="stopAudioBtn">Stop Audio</button>
-    <script>
-      const SAMPLE_RATE = 16000;
-      const NUM_CHANNELS = 1;
-      const PLAY_TIME_RESET_THRESHOLD_MS = 1.0;
-
-      // The protobuf type. We will load it later.
-      let Frame = null;
-
-      // The websocket connection.
-      let ws = null;
-
-      // The audio context
-      let audioContext = null;
-
-      // The audio context media stream source
-      let source = null;
-
-      // The microphone stream from getUserMedia. SHould be sampled to the
-      // proper sample rate.
-      let microphoneStream = null;
-
-      // Script processor to get data from microphone.
-      let scriptProcessor = null;
-
-      // AudioContext play time.
-      let playTime = 0;
-
-      // Last time we received a websocket message.
-      let lastMessageTime = 0;
-
-      // Whether we should be playing audio.
-      let isPlaying = false;
-
-      let startBtn = document.getElementById('startAudioBtn');
-      let stopBtn = document.getElementById('stopAudioBtn');
-
-      const proto = protobuf.load('frames.proto', (err, root) => {
-          if (err) {
-              throw err;
-          }
-          Frame = root.lookupType('pipecat.Frame');
-          const progressText = document.getElementById('progressText');
-          progressText.textContent = 'We are ready! Make sure to run the server and then click `Start Audio`.';
-
-          startBtn.disabled = false;
-          stopBtn.disabled = true;
-      });
-
-      function initWebSocket() {
-          ws = new WebSocket('ws://localhost:8765');
-          // This is so `event.data` is already an ArrayBuffer.
-          ws.binaryType = 'arraybuffer';
-
-          ws.addEventListener('open', handleWebSocketOpen);
-          ws.addEventListener('message', handleWebSocketMessage);
-          ws.addEventListener('close', (event) => {
-              console.log('WebSocket connection closed.', event.code, event.reason);
-              stopAudio(false);
-          });
-          ws.addEventListener('error', (event) => console.error('WebSocket error:', event));
-      }
-
-      function handleWebSocketOpen(event) {
-        console.log('WebSocket connection established.', event)
-
-        navigator.mediaDevices.getUserMedia({
-              audio: {
-                  sampleRate: SAMPLE_RATE,
-                  channelCount: NUM_CHANNELS,
-                  autoGainControl: true,
-                  echoCancellation: true,
-                  noiseSuppression: true,
-              }
-          }).then((stream) => {
-              microphoneStream = stream;
-              // 512 is closest thing to 200ms.
-              scriptProcessor = audioContext.createScriptProcessor(512, 1, 1);
-              source = audioContext.createMediaStreamSource(stream);
-              source.connect(scriptProcessor);
-              scriptProcessor.connect(audioContext.destination);
-
-              scriptProcessor.onaudioprocess = (event) => {
-                  if (!ws) {
-                      return;
-                  }
-
-                  const audioData = event.inputBuffer.getChannelData(0);
-                  const pcmS16Array = convertFloat32ToS16PCM(audioData);
-                  const pcmByteArray = new Uint8Array(pcmS16Array.buffer);
-                  const frame = Frame.create({
-                      audio: {
-                          audio: Array.from(pcmByteArray),
-                          sampleRate: SAMPLE_RATE,
-                          numChannels: NUM_CHANNELS
-                      }
-                  });
-                  const encodedFrame = new Uint8Array(Frame.encode(frame).finish());
-                  ws.send(encodedFrame);
-              };
-          }).catch((error) => console.error('Error accessing microphone:', error));
-      }
-
-      function handleWebSocketMessage(event) {
-          const arrayBuffer = event.data;
-          if (isPlaying) {
-              enqueueAudioFromProto(arrayBuffer);
-          }
-      }
-
-      function enqueueAudioFromProto(arrayBuffer) {
-          const parsedFrame = Frame.decode(new Uint8Array(arrayBuffer));
-          if (!parsedFrame?.audio) {
-              return false;
-          }
-
-          // Reset play time if it's been a while we haven't played anything.
-          const diffTime = audioContext.currentTime - lastMessageTime;
-          if ((playTime == 0) || (diffTime > PLAY_TIME_RESET_THRESHOLD_MS)) {
-              playTime = audioContext.currentTime;
-          }
-          lastMessageTime = audioContext.currentTime;
-
-          // We should be able to use parsedFrame.audio.audio.buffer but for
-          // some reason that contains all the bytes from the protobuf message.
-          const audioVector = Array.from(parsedFrame.audio.audio);
-          const audioArray = new Uint8Array(audioVector);
-
-          audioContext.decodeAudioData(audioArray.buffer, function(buffer) {
-              const source = new AudioBufferSourceNode(audioContext);
-              source.buffer = buffer;
-              source.start(playTime);
-              source.connect(audioContext.destination);
-              playTime = playTime + buffer.duration;
-          });
-      }
-
-      function convertFloat32ToS16PCM(float32Array) {
-          let int16Array = new Int16Array(float32Array.length);
-
-          for (let i = 0; i < float32Array.length; i++) {
-              let clampedValue = Math.max(-1, Math.min(1, float32Array[i]));
-              int16Array[i] = clampedValue < 0 ? clampedValue * 32768 : clampedValue * 32767;
-          }
-          return int16Array;
-      }
-
-      function startAudioBtnHandler() {
-          if (!navigator.mediaDevices || !navigator.mediaDevices.getUserMedia) {
-              alert('getUserMedia is not supported in your browser.');
-              return;
-          }
-
-          startBtn.disabled = true;
-          stopBtn.disabled = false;
-
-          audioContext = new (window.AudioContext || window.webkitAudioContext)({
-              latencyHint: 'interactive',
-              sampleRate: SAMPLE_RATE
-          });
-
-          isPlaying = true;
-
-          initWebSocket();
-      }
-
-      function stopAudio(closeWebsocket) {
-          playTime = 0;
-          isPlaying = false;
-          startBtn.disabled = false;
-          stopBtn.disabled = true;
-
-          if (ws && closeWebsocket) {
-              ws.close();
-              ws = null;
-          }
-
-          if (scriptProcessor) {
-              scriptProcessor.disconnect();
-          }
-          if (source) {
-              source.disconnect();
-          }
-      }
-
-      function stopAudioBtnHandler() {
-          stopAudio(true);
-      }
-
-      startBtn.addEventListener('click', startAudioBtnHandler);
-      stopBtn.addEventListener('click', stopAudioBtnHandler);
-      startBtn.disabled = true;
-      stopBtn.disabled = true;
-    </script>
-  </body>
-
-</html>
diff --git a/examples/websocket-server/requirements.txt b/examples/websocket-server/requirements.txt
deleted file mode 100644
index ed2130a79..000000000
--- a/examples/websocket-server/requirements.txt
+++ /dev/null
@@ -1,2 +0,0 @@
-python-dotenv
-pipecat-ai[cartesia,openai,silero,websocket,deepgram]