Audio streaming protocols in C++

03 Oct 2023

Audio streaming protocols play a crucial role in modern applications that involve real-time audio communication or transmission. Whether it’s for video conferencing, online gaming, or music streaming platforms, the efficient and reliable transfer of audio data is essential. In this blog post, we will explore some commonly used audio streaming protocols and how they can be implemented in C++.

1. Real-Time Transport Protocol (RTP)

Real-Time Transport Protocol (RTP) is a widely used protocol for streaming audio and video over IP networks. It provides mechanisms for packetization, transmission, and reception of continuous media streams. RTP is designed to support real-time applications by providing low latency, congestion control, and error recovery capabilities.

To implement RTP audio streaming in C++, you can use libraries like libavformat or live555. These libraries provide APIs to handle the RTP protocol and enable streaming of audio data over the network. You can use the C++ language to process audio samples, encode/decode audio formats, and interact with the chosen library’s APIs to handle RTP packets.

Here’s an example of how you can use libavformat to stream audio data over RTP in C++:

// Include necessary headers
#include <libavformat/avformat.h>

int main() {
    // Initialize the libavformat library
    avformat_network_init();

    // Create an AVFormatContext
    AVFormatContext* formatContext = avformat_alloc_context();

    // Set the RTP protocol as the output format
    av_strlcpy(formatContext->filename, "rtp://<destination_address>:<port>", sizeof(formatContext->filename));
    formatContext->oformat = av_guess_format("rtp", nullptr, nullptr);

    // Open the output file
    if (avio_open2(&formatContext->pb, formatContext->filename, AVIO_FLAG_WRITE, nullptr, nullptr) < 0) {
        // Handle error
        return -1;
    }

    // Create an AVStream for the audio data
    AVStream* stream = avformat_new_stream(formatContext, nullptr);
    if (!stream) {
        // Handle error
        return -1;
    }

    // Set the audio codec parameters (sample rate, channel layout, etc.)
    // ...

    // Open the audio codec
    // ...

    // Write the header to the output file
    avformat_write_header(formatContext, nullptr);

    // Provide audio data and send packets using av_write_frame()
    // ...

    // Write the trailer to the output file
    av_write_trailer(formatContext);

    // Release resources and cleanup
    avio_close(formatContext->pb);
    avformat_free_context(formatContext);
    avformat_network_deinit();

    return 0;
}

2. Web Real-Time Communication (WebRTC)

Web Real-Time Communication (WebRTC) is an open-source project that enables real-time communication between web browsers and mobile applications. It includes protocols such as RTP and WebSocket for audio and video streaming.

WebRTC is widely used for audio/video conferencing, live streaming, and other real-time communication applications. It provides features like peer-to-peer communication, secure data transmission, and automatic NAT traversal.

To implement audio streaming using WebRTC in C++, you can use libraries like libwebrtc, which is the C++ API for WebRTC. It allows you to create audio streams, establish peer connections, and handle audio data exchange with other peers.

Here’s a simplified example of how you can use libwebrtc to implement audio streaming with WebRTC in C++:

// Include necessary headers
#include <webrtc/api/audio/audio_device.h>
#include <webrtc/api/peerconnectioninterface.h>

int main() {
    // Initialize the WebRTC library
    rtc::InitializeSSL();

    // Create a peer connection factory
    rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> peerConnectionFactory =
        webrtc::CreatePeerConnectionFactory();

    // Create peer connection configuration
    webrtc::PeerConnectionInterface::RTCConfiguration config;
    // Set configuration properties (STUN/TURN servers, etc.)
    // ...

    // Create a peer connection
    rtc::scoped_refptr<webrtc::PeerConnectionInterface> peerConnection =
        peerConnectionFactory->CreatePeerConnection(config);

    // Create an audio track for local audio capture
    rtc::scoped_refptr<webrtc::AudioTrackInterface> audioTrack = peerConnectionFactory->CreateAudioTrack(
        "audio_track_id", peerConnectionFactory->CreateAudioSource(nullptr));

    // Add the audio track to the peer connection's local stream
    rtc::scoped_refptr<webrtc::MediaStreamInterface> mediaStream = peerConnectionFactory->CreateLocalMediaStream(
        "local_media_stream_id");
    mediaStream->AddTrack(audioTrack);
    peerConnection->AddStream(mediaStream);

    // Initialize the audio device
    rtc::scoped_refptr<webrtc::AudioDeviceModule> audioDeviceModule =
        webrtc::AudioDeviceModule::Create(webrtc::AudioDeviceModule::kPlatformDefaultAudio);

    // Start capturing audio from the device
    audioDeviceModule->Init();
    audioDeviceModule->StartRecording();

    // Provide audio data to the audio track using audioDeviceModule->RecordAudio()
    // ...

    // Release resources and cleanup
    audioDeviceModule->StopRecording();
    audioDeviceModule->Terminate();
    webrtc::PeerConnectionFactory::Release();

    return 0;
}

In this example, we use libwebrtc to create a peer connection, set up audio tracks for local audio capture, and initialize the audio device for recording. You can further customize the code according to your requirements, such as handling audio data received from other peers and enabling audio playback.

Conclusion

Audio streaming protocols like RTP and WebRTC provide powerful capabilities for real-time audio transmission. Using libraries like libavformat and libwebrtc, you can implement these protocols in C++ to build applications with reliable and efficient audio streaming capabilities.