As noted in a sibling comment I&#x27;m not with Rogue Amoeba anymore, but I&#x27;ll still weigh in.<p>Airfoil is limited by the decision to stick to Apple&#x27;s protocol. Obviously, this is necessary to talk to Apple&#x27;s devices. It&#x27;s <i>not</i> strictly necessary when going from Airfoil to Airfoil Speakers, but it makes life much simpler to just use the same protocol for everything.<p>AirTunes (what Apple used to call the audio version of AirPlay, which is totally unrelated to the video version of AirPlay... confused yet?) isn&#x27;t really suited to low latencies. The transmitter basically sends audio data, and then separately sends commands every so often saying, &quot;At time T you should be playing sample number X in the audio stream.&quot; This allows sending to multiple outputs and having them all be synchronized. But for low latency you really just want &quot;play this instantly,&quot; and to make sure you get audio data to it in a timely fashion. If it&#x27;s fast enough, then you don&#x27;t need to synchronize multiple outputs, since they&#x27;ll be synchronized with &quot;real time.&quot;<p>The major challenge is dealing with packet loss. A typical WiFi network might see 1% packet loss, which you need to recover from somehow. For non-realtime operations, it&#x27;s easy. You detect the loss, retransmit the data, and carry on. For realtime stuff it&#x27;s harder. With video you can drop frames. It&#x27;s not ideal, but it&#x27;s not too bad. Dropping audio sounds <i>way</i> worse than dropping video looks, though, so that&#x27;s not really an option for something like Airfoil.<p>One option I&#x27;ve always wanted to explore was to use forward error correction on the transmitted data. Send out enough redundancy that the receiver can put the original data together even with some loss in the middle. I never have tried it out, but I think it could really cut down on latency.<p>Without that, you&#x27;re left with detecting loss and retransmitting, which eats up valuable time. Combine it with code written to do synchronized output with a substantial buffer, and you just can&#x27;t do realtime transmission.<p>I don&#x27;t <i>think</i> the encoding&#x2F;decoding step (AirTunes uses Apple Lossless encoding for the audio, for those unaware) adds all that much latency to the process, but I will admit that I never went in and measured the bits on the millisecond level, since we were looking at 2000ms end-to-end latency.<p>If you want to experiment, there is a hidden setting in there somewhere that lets you adjust the latency Airfoil tells the receiver to use. It might be in the window that pops up when you hold option while launching the apps, or you might need to track down a hidden defaults key, I don&#x27;t quite recall anymore. In my experience you probably won&#x27;t be able to get it below about 200ms without lots of audible problems, and that&#x27;s still plenty for a highly perceptible delay.<p>OK, I&#x27;d better stop this before it turns into something long enough to be a blog post on its own!