Codec Specifics for Opus and AAC-LC in WebRTC

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It uses codecs (which are just fancy words for compression algorithms) to make sure that all of those sweet, sweet video and audio signals can be transmitted over the internet without lagging or buffering too much.

Now, when it comes to WebRTC’s default codec options, there are a few different ones you might encounter: Opus (which is great for voice and music), AAC-LC (which is good for audio but not as efficient as Opus), and H.264 (which is the most commonly used video compression algorithm).

But what if your computer or phone doesn’t support one of these codecs? Or what if you want to prioritize a certain type of signal over others? That’s where this “Codec Specifics for Opus and AAC-LC in WebRTC” thing comes into play. Basically, it lets you customize your codec preferences so that you can get the best possible audio and video quality without any lag or buffering issues.

So, let’s say you want to prioritize voice over music when making a video call on your computer. You could use this “Codec Specifics for Opus and AAC-LC in WebRTC” thing to tell your browser that you prefer using the Opus codec (which is great for voice) instead of H.264 or AAC-LC (which are better suited for video and audio, respectively).

Here’s an example script that shows how this might work:

// This function changes the video codec preferences for a WebRTC call based on the given media type (mimeType)
function changeVideoCodec(mimeType) {
  // Get all transceivers from the RTCPeerConnection object
  const transceivers = peerConnection.getTransceivers();

  // Loop through each transceiver and update its codec preferences
  transceivers.forEach((transceiver) => {
    // Get the kind of media being sent or received by this transceiver
    const kind = transceiver.sender.track.kind;
    // Get a list of all available codecs for sending and receiving video/audio signals based on the given kind
    let sendCodecs = RTCRtpSender.getCapabilities(kind).codecs;
    let recvCodecs = RTCRtpReceiver.getCapabilities(kind).codecs;

    // If the transceiver is for video, rearrange the codec list based on the given media type
    if (kind === 'video') {
      sendCodecs = preferCodec(mimeType);
      recvCodecs = preferCodec(mimeType);
    }
  });

  // Update the local description with the new codec preferences and send it to the other person in the call
  peerConnection.setLocalDescription();
}

// This function rearranges the codec list based on the given media type
function preferCodec(mimeType) {
  // Get a list of all available codecs for the given media type
  const availableCodecs = RTCRtpSender.getCapabilities(mimeType).codecs;
  // Create an empty array to store the preferred codecs
  const preferredCodecs = [];

  // Loop through each available codec
  availableCodecs.forEach((codec) => {
    // If the codec is supported by the browser, add it to the preferred codecs array
    if (codec.supported) {
      preferredCodecs.push(codec);
    }
  });

  // Return the preferred codecs array
  return preferredCodecs;
}

// Example usage:
changeVideoCodec('video/opus'); // This will update the codec preferences for video to use the Opus codec instead of H.264 or AAC-LC

So, basically, this script lets you customize your codec preferences based on the given mimeType. If you want to prioritize voice over music when making a video call, for example, you could pass in something like `’audio/opus’` as the mimeType parameter and it would rearrange your codec list accordingly (i.e., putting Opus at the top of the list).

Hope that helps! Let me know if you have any other questions or if there’s anything else I can do for ya.

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