Merge pull request #198 from pipecat-ai/aleix/websocket-transport

websocket transport support
This commit is contained in:
Aleix Conchillo Flaqué
2024-06-01 04:40:39 +08:00
committed by GitHub
43 changed files with 833 additions and 459 deletions

View File

@@ -44,7 +44,7 @@ async def main(room_url):
llm = OpenAILLMService(
api_key=os.getenv("OPENAI_API_KEY"),
model="gpt-4-turbo-preview")
model="gpt-4o")
messages = [
{

View File

@@ -93,7 +93,7 @@ async def main(room_url):
llm = OpenAILLMService(
api_key=os.getenv("OPENAI_API_KEY"),
model="gpt-4-turbo-preview")
model="gpt-4o")
imagegen = FalImageGenService(
params=FalImageGenService.InputParams(

View File

@@ -76,7 +76,7 @@ async def main():
llm = OpenAILLMService(
api_key=os.getenv("OPENAI_API_KEY"),
model="gpt-4-turbo-preview")
model="gpt-4o")
tts = ElevenLabsTTSService(
aiohttp_session=session,

View File

@@ -81,7 +81,7 @@ async def main(room_url: str, token):
llm = OpenAILLMService(
api_key=os.getenv("OPENAI_API_KEY"),
model="gpt-4-turbo-preview")
model="gpt-4o")
messages = [
{

View File

@@ -53,7 +53,7 @@ async def main(room_url: str, token):
llm = OpenAILLMService(
api_key=os.getenv("OPENAI_API_KEY"),
model="gpt-4-turbo-preview")
model="gpt-4o")
messages = [
{

View File

@@ -53,7 +53,7 @@ async def main(room_url: str, token):
llm = OpenAILLMService(
api_key=os.getenv("OPENAI_API_KEY"),
model="gpt-4-turbo-preview")
model="gpt-4o")
messages = [
{

View File

@@ -95,7 +95,7 @@ async def main(room_url: str, token):
llm = OpenAILLMService(
api_key=os.getenv("OPENAI_API_KEY"),
model="gpt-4-turbo-preview")
model="gpt-4o")
tts = ElevenLabsTTSService(
aiohttp_session=session,

View File

@@ -66,7 +66,7 @@ async def main(room_url: str, token):
llm = OpenAILLMService(
api_key=os.getenv("OPENAI_API_KEY"),
model="gpt-4-turbo-preview")
model="gpt-4o")
llm.register_function(
"get_current_weather",
fetch_weather_from_api,

View File

@@ -1,25 +0,0 @@
syntax = "proto3";
package pipecat_proto;
message TextFrame {
string text = 1;
}
message AudioFrame {
bytes audio = 1;
}
message TranscriptionFrame {
string text = 1;
string participant_id = 2;
string timestamp = 3;
}
message Frame {
oneof frame {
TextFrame text = 1;
AudioFrame audio = 2;
TranscriptionFrame transcription = 3;
}
}

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@@ -1,134 +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="//cdn.jsdelivr.net/npm/protobufjs@7.X.X/dist/protobuf.min.js"></script>
<title>WebSocket Audio Stream</title>
</head>
<body>
<h1>WebSocket Audio Stream</h1>
<button id="startAudioBtn">Start Audio</button>
<button id="stopAudioBtn">Stop Audio</button>
<script>
const SAMPLE_RATE = 16000;
const BUFFER_SIZE = 8192;
const MIN_AUDIO_SIZE = 6400;
let audioContext;
let microphoneStream;
let scriptProcessor;
let source;
let frame;
let audioChunks = [];
let isPlaying = false;
let ws;
const proto = protobuf.load("frames.proto", (err, root) => {
if (err) throw err;
frame = root.lookupType("pipecat_proto.Frame");
});
function initWebSocket() {
ws = new WebSocket('ws://localhost:8765');
ws.addEventListener('open', () => console.log('WebSocket connection established.'));
ws.addEventListener('message', handleWebSocketMessage);
ws.addEventListener('close', (event) => console.log("WebSocket connection closed.", event.code, event.reason));
ws.addEventListener('error', (event) => console.error('WebSocket error:', event));
}
async function handleWebSocketMessage(event) {
const arrayBuffer = await event.data.arrayBuffer();
enqueueAudioFromProto(arrayBuffer);
}
function enqueueAudioFromProto(arrayBuffer) {
const parsedFrame = frame.decode(new Uint8Array(arrayBuffer));
if (!parsedFrame?.audio) return false;
const frameCount = parsedFrame.audio.data.length / 2;
const audioOutBuffer = audioContext.createBuffer(1, frameCount, SAMPLE_RATE);
const nowBuffering = audioOutBuffer.getChannelData(0);
const view = new Int16Array(parsedFrame.audio.data.buffer);
for (let i = 0; i < frameCount; i++) {
const word = view[i];
nowBuffering[i] = ((word + 32768) % 65536 - 32768) / 32768.0;
}
audioChunks.push(audioOutBuffer);
if (!isPlaying) playNextChunk();
}
function playNextChunk() {
if (audioChunks.length === 0) {
isPlaying = false;
return;
}
isPlaying = true;
const audioOutBuffer = audioChunks.shift();
const source = audioContext.createBufferSource();
source.buffer = audioOutBuffer;
source.connect(audioContext.destination);
source.onended = playNextChunk;
source.start();
}
function startAudio() {
if (!navigator.mediaDevices || !navigator.mediaDevices.getUserMedia) {
alert('getUserMedia is not supported in your browser.');
return;
}
navigator.mediaDevices.getUserMedia({ audio: true })
.then((stream) => {
microphoneStream = stream;
audioContext = new (window.AudioContext || window.webkitAudioContext)();
scriptProcessor = audioContext.createScriptProcessor(BUFFER_SIZE, 1, 1);
source = audioContext.createMediaStreamSource(stream);
source.connect(scriptProcessor);
scriptProcessor.connect(audioContext.destination);
const audioBuffer = [];
const skipRatio = Math.floor(audioContext.sampleRate / (SAMPLE_RATE * 2));
scriptProcessor.onaudioprocess = (event) => {
const rawLeftChannelData = event.inputBuffer.getChannelData(0);
for (let i = 0; i < rawLeftChannelData.length; i += skipRatio) {
const normalized = ((rawLeftChannelData[i] * 32768.0) + 32768) % 65536 - 32768;
const swappedBytes = ((normalized & 0xff) << 8) | ((normalized >> 8) & 0xff);
audioBuffer.push(swappedBytes);
}
if (audioBuffer.length >= MIN_AUDIO_SIZE) {
const audioFrame = frame.create({ audio: { audio: audioBuffer.slice(0, MIN_AUDIO_SIZE) } });
const encodedFrame = new Uint8Array(frame.encode(audioFrame).finish());
ws.send(encodedFrame);
audioBuffer.splice(0, MIN_AUDIO_SIZE);
}
};
initWebSocket();
})
.catch((error) => console.error('Error accessing microphone:', error));
}
function stopAudio() {
if (ws) {
ws.close();
scriptProcessor.disconnect();
source.disconnect();
ws = undefined;
}
}
document.getElementById('startAudioBtn').addEventListener('click', startAudio);
document.getElementById('stopAudioBtn').addEventListener('click', stopAudio);
</script>
</body>
</html>

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@@ -1,50 +0,0 @@
import asyncio
import aiohttp
import logging
import os
from pipecat.pipeline.frame_processor import FrameProcessor
from pipecat.pipeline.frames import TextFrame, TranscriptionFrame
from pipecat.pipeline.pipeline import Pipeline
from pipecat.services.elevenlabs_ai_services import ElevenLabsTTSService
from pipecat.transports.websocket_transport import WebsocketTransport
from pipecat.services.whisper_ai_services import WhisperSTTService
logging.basicConfig(format="%(levelno)s %(asctime)s %(message)s")
logger = logging.getLogger("pipecat")
logger.setLevel(logging.DEBUG)
class WhisperTranscriber(FrameProcessor):
async def process_frame(self, frame):
if isinstance(frame, TranscriptionFrame):
print(f"Transcribed: {frame.text}")
else:
yield frame
async def main():
async with aiohttp.ClientSession() as session:
transport = WebsocketTransport(
mic_enabled=True,
speaker_enabled=True,
)
tts = ElevenLabsTTSService(
aiohttp_session=session,
api_key=os.getenv("ELEVENLABS_API_KEY"),
voice_id=os.getenv("ELEVENLABS_VOICE_ID"),
)
pipeline = Pipeline([
WhisperSTTService(),
WhisperTranscriber(),
tts,
])
@transport.on_connection
async def queue_frame():
await pipeline.queue_frames([TextFrame("Hello there!")])
await transport.run(pipeline)
if __name__ == "__main__":
asyncio.run(main())