How to stream audio from browser to WebRTC native C++ application

Streaming audio from a browser to a native C++ WebRTC application involves several key steps, which I will outline step by step:

1. Browser-Side Setup

First, on the browser side, we need to use WebRTC's API to capture the audio stream. We can leverage the navigator.mediaDevices.getUserMedia() method to access the user's audio input device.

javascript
async function getAudioStream() {
    try {
        const stream = await navigator.mediaDevices.getUserMedia({ audio: true });
        return stream;
    } catch (error) {
        console.error('Error accessing microphone:', error);
    }
}

This code requests permission to access the microphone and returns a MediaStream object containing an audio track.

2. Establishing a WebRTC Connection

Next, we need to establish a WebRTC connection between the browser and the C++ application. This typically involves the signaling process, where network and media information are exchanged to set up and maintain the WebRTC connection. We can use WebSocket or any server-side technology to exchange this information.

Browser-Side:

javascript
const peerConnection = new RTCPeerConnection();

// Add the audio stream to the connection
const stream = await getAudioStream();
stream.getTracks().forEach(track => peerConnection.addTrack(track, stream));

// Create an offer
const offer = await peerConnection.createOffer();
await peerConnection.setLocalDescription(offer);

// Send the offer to the signaling server
sendToServer({ type: 'offer', sdp: offer.sdp });

C++ Application-Side (using libwebrtc): On the C++ side, you need to set up the WebRTC environment, receive and respond to the offer, which typically involves using Google's libwebrtc library.

cpp
// Pseudo-code; specific implementation depends on the libwebrtc version and API design
void onOfferReceived(const std::string &sdp) {
    peer_connection->SetRemoteDescription(SetSessionDescriptionObserver::Create(), sdp);
    peer_connection->CreateAnswer(CreateSessionDescriptionObserver::Create(), PeerConnectionInterface::RTCOfferAnswerOptions());
}

3. Signaling Exchange

As mentioned earlier, signaling exchange is essential. This process typically involves the following steps:

• The browser generates an offer and sends it to the C++ application via the signaling server.
• The C++ application receives the offer, generates an answer, and sends it back to the browser.
• The browser receives the answer and sets the remote description.

4. Media Stream Processing

Once the WebRTC connection is established, the audio stream begins to flow from the browser to the C++ application. In the C++ application, you can process these streams, for example, for audio processing, storage, or further transmission.

Examples and Simulation

To implement the above steps in a real project, you may need to read more documentation on WebRTC and libwebrtc, as well as related network protocols such as STUN/TURN. In practice, you should also consider network conditions, security (such as using DTLS), and error handling.