e6891e81c3
**Overview:** This commit represents a comprehensive refactoring of the application to improve real-time audio streaming capabilities. The key change is the integration of WebRTC for peer-to-peer audio streaming while using Hyperswarm exclusively for signaling. This transition addresses efficiency, reliability, and scalability issues present in the original implementation. **Old Method:** - **Audio Streaming via Hyperswarm Data Channels:** - The original code used Hyperswarm for both signaling and streaming audio data. - Audio data was captured from the microphone, converted to binary, and transmitted over Hyperswarm connections. - Listeners received the audio data chunks and processed them to play back the audio. - **Issues:** - Inefficient for real-time audio streaming due to Hyperswarm's limitations for media data. - Higher latency and potential synchronization problems. - Difficulty managing peer connections and media streams effectively. **New Method:** - **Integration of WebRTC for Audio Streaming:** - Implemented `RTCPeerConnection` instances for efficient, real-time, peer-to-peer audio streaming. - Used Hyperswarm solely for signaling to establish and manage connections. - Audio tracks are now transmitted over WebRTC connections, leveraging optimized protocols for media. - **Benefits:** - Improved audio quality and reduced latency. - Enhanced NAT traversal and firewall compatibility via ICE servers. - Better management of media streams and peer connections. **Key Changes:** 1. **WebRTC Implementation:** - **Broadcaster Side:** - Created `RTCPeerConnection` instances for each listener. - Added local audio tracks from the microphone to the peer connections. - Managed signaling messages (`offer`, `answer`, `candidate`) received via Hyperswarm. - Handled ICE candidate exchange and connection state changes. - **Listener Side:** - Created an `RTCPeerConnection` to connect to the broadcaster. - Added a transceiver with `recvonly` direction to receive audio streams. - Managed signaling messages and ICE candidates. - Played received audio streams using HTML `<audio>` elements. 2. **Signaling via Hyperswarm:** - Utilized Hyperswarm connections for exchanging signaling messages in JSON format. - Messages include `offer`, `answer`, and `candidate` types. - Ensured proper serialization and deserialization of signaling data. 3. **ICE Candidate Handling:** - Implemented ICE candidate queuing to handle candidates arriving before the remote description is set. - Stored incoming ICE candidates in queues and processed them after setting the remote description. - Added detailed logging for ICE candidate exchange and connection states. 4. **Peer Count Accuracy:** - Updated the `updatePeerCount()` function to use `conns.length`, reflecting active Hyperswarm connections. - Ensured the peer count updates immediately when connections are established or closed. - Improved UI feedback regarding the number of connected peers. 5. **Audio Input Switching Without Disconnecting Peers:** - Modified the `applyAudioSource()` function to replace audio tracks in existing peer connections without restarting the station. - Obtained a new audio stream with the selected input device. - Used `RTCRtpSender.replaceTrack()` to update the audio track in each peer connection. - Stopped old audio tracks to free up resources. - Allowed broadcasters to switch microphones seamlessly without interrupting listeners' audio. 6. **Error Handling and Debugging Improvements:** - Added extensive logging throughout the code to trace execution flow and internal state. - Wrapped asynchronous operations in `try...catch` blocks to handle errors gracefully. - Provided informative console messages for successful operations and errors. 7. **User Interface Adjustments:** - Retained existing UI elements and controls. - Updated event listeners to align with the new logic. - Provided real-time updates to station information and peer count. **Benefits of the New Method:** - **Enhanced Audio Quality and Performance:** - Leveraging WebRTC provides better audio streaming capabilities optimized for real-time communication. - Reduced latency and improved synchronization. - **Scalability and Reliability:** - Proper handling of peer connections and media streams improves the application's scalability. - Robust error handling ensures better reliability under various network conditions. - **Improved User Experience:** - Listeners experience uninterrupted audio even when broadcasters change input devices. - Accurate peer count provides broadcasters with immediate feedback on their audience size. **Testing and Verification:** - Tested the application with multiple broadcasters and listeners to ensure proper functionality. - Verified that audio streams initiate correctly and continue even after input device changes. - Confirmed that peer counts update accurately on both broadcaster and listener sides. - Ensured that no errors appear in the console logs during normal operation.
34 lines
836 B
JSON
34 lines
836 B
JSON
{
|
|
"name": "hypercastify",
|
|
"main": "index.html",
|
|
"pear": {
|
|
"name": "hypercastify",
|
|
"type": "desktop",
|
|
"gui": {
|
|
"backgroundColor": "#1F2430",
|
|
"height": "540",
|
|
"width": "720",
|
|
"links": ["http://127.0.0.1", "http://localhost", "https://ka-f.fontawesome.com", "https://cdn.jsdelivr.net", "https://cdnjs.cloudflare.com", "https://stackpath.bootstrapcdn.com", "ttps://code.jquery.com"]
|
|
|
|
}
|
|
},
|
|
"type": "module",
|
|
"license": "Apache-2.0",
|
|
"scripts": {
|
|
"dev": "pear run -d .",
|
|
"test": "brittle test/*.test.js"
|
|
},
|
|
"devDependencies": {
|
|
"brittle": "^3.0.0",
|
|
"pear-interface": "^1.0.0"
|
|
},
|
|
"dependencies": {
|
|
"b4a": "^1.6.7",
|
|
"bootstrap": "^5.3.3",
|
|
"hypercore-crypto": "^3.4.2",
|
|
"hyperswarm": "^4.8.4",
|
|
"pear-stdio": "^1.0.1",
|
|
"stream": "^0.0.3"
|
|
}
|
|
}
|