In the past few weeks, MeetSpace has been getting off the ground with some research surveys and alpha work. Throughout all this there’s one clear message from teams that have online meetings:
Audio quality is really important!
It turns out that many people are happy to trade off a bit of pixellated video or switch to a different browser if it means they get crystal clear audio with low delay and no disconnections, hiccups, or the dreaded robot voice. So, here at MeetSpace we’ve been working on an alpha that prioritizes audio quality above all else, and we’ve made a few tradeoffs we think you’ll agree with.
Tradeoff 1: Video for Audio
The first thing we did was try out turning audio bitrate and sample rate way up, and turning video way down. It turns out, you don’t need much video bitrate at all with modern codecs to achieve a pretty good looking image. Here’s an example:
Can you tell I’m saying Hi? Can you believe this is a 640x480 image, shrunk to 320x240 (for hidpi screens) being broadcast at 30fps with only 150kbps? Modern video codecs are amazing!
That means we have a lot more bandwidth to use for audio! What we’re currently testing out is a full 48khz audio sample rate at 100kbps. That’s 2/3 the bandwidth we use for video. The effect is that you can hear really clearly what everyone is saying.
We think this compromise will really help teams have more effective meetings by being able to hear every word and nuance in someone else’s voice.
Tradeoff 2: Delay for compatibility
Aside from the quality of the audio, the delay on the line is a huge factor in being able to communicate. It sucks when you get interrupted, or when you interrupt someone, and it’s probably not even your fault! Then you get into that awkward “you go”, “no you go” phase.
Almost all modern video conferencing solutions all send their data through an intermediate server for a few reasons:
- Reduce total upload bandwith by casting your stream to everyone else, and sending their streams to you
- Transcode your media into a format each other attendee can handle
That’s because right now not all browsers support the same codecs or even the peer-to-peer abilities of WebRTC (cough cough Safari!).
This means that there are two new sources of delay on your call:
- The time it takes to go from you to the server and out to other participants (vs direct)
- The time it takes the server to transcode your media
At MeetSpace, we decided to try something the bigger players can’t do: go peer to peer and not support all devices and browsers. That means we will use a direct connection to other attendees to bring the delay to its minimum, and use the best codecs to ensure the highest quality at the lowest bitrates. Right now, our audio delay hangs out just over 100ms, which is pretty hard to notice.
The result? You are going to have to use Chrome, Firefox, or Edge. When we did our survey though, 96% of people said this was OK, as long as we deliver on the awesome audio experience.
Alpha Coming Up!
We’re almost done with the alpha, and we’re looking for a couple of teams to try it out on their day-to-day meetings. Sound like something you’d be a test pilot for? Sign up below or follow @meetspaceapp to receive updates on our alpha, beta, and release.