This is driving me crazy with Google Meet in these COVID19 times. Even in a relatively small conference, I have a really hard time interrupting someone to ask a quick question, even when the speaker is expecting interruptions. It's always "excuse me!"; delay as person continues speaking; I stop; the other person says "yes, please ask away"; when I restart my question the other person already assumed I've changed my mind and continues speaking; repeat ad infinitum. And this is if they even hear me over the audio breaking up.
It's very, very frustrating. If they solve this it would hugely improve quality of life in remote conferencing for me.
When a conference call is made up of people all in the same city on decent internet connections, latency is usually not a big issue.
But when a conference call has people from New York, San Francisco, and Japan on it, even if it's only 3 participants, latency can be bad just because of the speed of light, essentially (on top of what is otherwise reasonable hardware/software latency). Latency may be bad even if you're talking with a colleague in the same city, since the audio is "mixed" on the server, and that server might be across the world if a participant from across the world started the meeting. (Counterintuitively, the latency with your local colleague could be twice as bad as with the colleague from across the world.)
There are a few things that most meetings could benefit from. Having an organizer who's aware of the differences between leading an in person meeting and a remote meeting, cutting video to save bandwidth if the meeting doesn't absolutely need it (the organizer can usually just disable the function), muting when you're not speaking (by far the best quality of life improvement, can be done silently by the organizer if someone is just doing their Vader impersonation throughout the meeting), using the "raise hand" function (again, the organizer plays a huge role here), using the native app instead of the web one usually provides better quality and performance, using a wired connection instead of wireless if possible, sometimes even starting meetings at non-standard hours (like 15 to/past the hour) helps avoid the rush of people logging in at the same time, etc.
Typical restart-all-the-things usually made it go away. But it wasn’t unusual for 500ms of latency to slowly build up during a 30min call. Unfortunately I have nothing more useful to add, the issue resolved itself before I could track down a definitive cause.
Stop using it.
Without even looking at your setup, I would bet $100 minimum that it's Bluetooth latency. It adds a lot of latency, 500ms is not unusual, and many folks have no idea that all that latency is really just the last 18 inches. This is why you're seeing more and more cases where people are using good old iPhone wired earphone for conference calling, especially when skyping a TV interview.
How are you measuring? Half that is considered high-end from what I'm reading[1], and AirPods Pro apparently reduce that to 150ms.
Bluetooth definitely is the issue with 95% of these latency issues.
Crazy that airpods have that much latency. I have bluetooth in the car and it has way too much latency - I thought Apple had improved it to not be noticeable but I never checked the numbers.
Incidentally, for playing midi instruments you generally want things to be below 8ms to feel natural. 150ms is an eternity.
Is the lipsync ok if you watch a video on an iphone with airpods?
For video they compensate for the latency by displaying the visuals slightly delayed (this has been done for over a decade even way back on feature phones).
Even game consoles have the option to do this since some TVs/receivers have audio or video latency.
One way to solve this is to have the speaker name the person, and then wait until that person speaks. For example, if someone interrupts:
Speaker: [Talks]
Person A: Excuse me!
Speaker: Yes, Mr. A? [waits]
Person A: What about X?
Or if there are two people talking at the same time Speaker: [Talks]
Person A and B: Excuse me!
Speaker: Yes, Mr. A? Mr. B, I'll come back to you after A. [waits]
Person A: What about X?
Speaker: [Talk about X]. Mr B, you were saying?
Person B: What about Y?
Treat it as a synchronization problem, with the speaker breaking the ties. As long as it's obvious to everyone whose turn it is to speak, it works well (assuming people aren't too rowdy/impolite).I want to just be able to hit my self-view and have it have a big icon on it or something so the person currently speaking (and everyone else) can see that I want to say something. Maybe sort these in chronological order so the speaker can see who wanted to talk first?).
In theory you could do this with a good chat, but for some reason the chat in Zoom and the others is kind of an afterthought and nobody uses it.
One of the reasons I prefer text based chat is multiple people can talk at the same time without needing to deal with interrupting audio. If you can type well, the bandwidth is higher for group communication (and you get a log).
At least with video you can kind of tell when someone is waiting to speak by seeing their expression. Audio only is worse (but maybe wouldn't be, if you had good intent-to-speak tools built into the app?)
this is a solved problem. webex has had a hand raise feature for a decade now. trivial for google to just copy.
Which is one good reason to use video. At least with smaller meetings, someone can raise their hand or just look really pained. (Bigger meetings, you probably need to use chat.)
Sadly, the software does not just magically take care of it. Anytime two people talk, a typical echo canceler just starts decimating frequencies until both of them are unintelligible.
Add in a couple of clueless teams who mount a camera/mic against a conference room wall and introduce massive amounts of room echo into the mix, and I'm at the point where a conference call becomes an absolutely mentally exhausting experience just trying to decipher what is being said. I have no hope of contributing, because I can only hear 2/3 of the syllables, and my brain is running on overdrive trying to turn those back into words. By the time I've figured out what they just said, they're half-way into the next sentence. What a stressful hellscape.
Ironically, if we had no echo cancellation, it would force everyone to use ear buds, and the average call quality would be a lot better.
I have some screenshots of waveforms showing laptop mic vs headset, and the signal-to-noise ratio with the headset destroys even good noise-cancelling using a laptop mic that's farther away from one's mouth.
I don't know what else I can do.
Participants often don't realize that they're the culprit when somebody else sounds terrible.
Cheaper bluetooth headsets seem to pick up everything around them. Had that issue with a coworker where the headset was worse than using the internal mic.
Biggest and annoying issue though is consistent bluetooth disconnect/reconnect issues even on different MacOS machines. Latest firmware and such. Pretty sure it's not 2.4ghz interference.
"Modern high-end Bluetooth headsets support AptX, an audio codec compression scheme that offers better sound quality. But AptX is only enabled if it’s supported on both the transmitter and receiver. When using a Bluetooth headset with a PC, it only works if your PC’s hardware and drivers are compatible." (https://www.howtogeek.com/354321/why-bluetooth-headsets-are-...)
Not sure if this applies to a Mac though.
A few different ways it's come in handy for me:
- The Bluetooth speaker I use for music has a tendency to sporadically sound super hollow. Turns out it has a mic built in, and the voodoo of Mac's bluetooth stack would decide at random when it connected whether it would go into audio-only mode (and use the higher-bandwidth AAC codec) or go into audio+mic mode (splits the available bandwidth between the two and as a result uses a lower bitrate audio codec to compensate for the bandwidth drop). Used ToothFairy to always force that device into audio-only mode.
- After the above discovery, tested doing the same with my actual headset, and leveraging the built-in mic for input. For call audio, it's pretty erratic on whether it'll have any impact at all, and depends heavily on the circumstances of the call itself. Sometimes the audio is massively better, but most of the time the audio is already degraded when it gets to my machine and the bluetooth improvement is moot. That said, makes music in between calls far more pleasant.
- My bluetooth mouse is particularly susceptible to that consistent disconnect/reconnect issue you mentioned. ToothFairy can create a menubar icon for individual devices, which helps to act as a quick sanity check to see if my mouse has disconnected. ToothFairy can also run a shell script on disconnect, which has been handy. At this point I have it trigger a system notification[2] so I'm at least immediately made aware of it, check my idle time[3] in case it was the mouse going into sleep mode from inactivity, then conditionally leverage blueutil[4] to look for the device and reconnect if found (forcefully restarting the bluetooth stack in the process if it has issues). Doesn't fix whatever the root cause is for that consistent disconnect/reconnect issue, but this duct tape re-establishes a connection far more quickly when it happens, making the issue itself significantly less disruptive.
I only stumbled on it via Setapp[1] and tried it on a whim, but it's definitely one of the more handy utility apps I've found and well worth the $5 App Store price for anyone that has similar bluetooth frustrations with their Mac.
[1] https://c-command.com/toothfairy/
[2] https://code-maven.com/display-notification-from-the-mac-com...
[3] https://stackoverflow.com/a/17966890
https://www.jefftk.com/p/ethernet-is-worth-it-for-video-call...
Then I realized I'd moved to an apartment complex. I did a wifi scan, and found over fifty competing SSIDs.
Switched to ethernet, and the improvement was night and day.
No matter which software you use, some people will be at an advantage simply due to their isp/wired connection/wired mic.
This could be a business idea though: Conferencing software which equalizes everyone’s latency.
Or is it just worse?
Discord users are more likely to have a dedicated fast internet connection and doesn't seem to care about profitability at the moment.
It's just the difference in designing for a 100/10 connection to yourself vs sharing a 100/100 connection with 20 other people. Zoom reasonably gracefully degrades on choppy/slow connections while Discord becomes straight up unusable.
The problem is that Google Meet (or my connection, or whatever technical reason) wasn't up to the task. This has happened enough times that I dread interrupting now. Sadly, one person monologuing is not how face-to-face meetings really work.
People just don’t care about quality, they will gladly use the crappiest mic with all the noise all day..
But people shouldn't take "workspace" abuse from others. Be polite, assertive, constructive. Give them credit for taking the time to care a bit about this problem, offer to debug with them later, try to demonstrate it - switch to speaker, capture what you hear. Alternatively just switch off voice and type. Tell them that the sound is garbage for you so you switched for typing. Try to do quick 1-on-1s instead of the group call, even offer to write a summary after.
This is basically the equivalent of constantly not giving a fuck about how loud one is in an open office. There ought to be proper channels [ha, no pun intended] to address and solve these.
Mute your or participants' background noise in any communication app
I have noticed this and I hate it. It makes normal conversation absolutely impossible.
Discord, which is an audio first product, is much better than other solutions in this regard and their video conferencing while new has been very enjoyable to use.
I think if it was dynamic, where turning my head towards the person speaking balanced the audio (like in real life), I would not have a problem with it. A super simple form of virtual reality that would only require a simple head-mounted gyroscope or motion sensor.
Another podcast I listen to has two people with very similar voices, and I sometimes have a hard time figuring out who's speaking, so I welcome any advancements in this space.
Just use proper equipment. Headset is an absolute must. Next step is software that only transmits when someone is talking. Gamers have figured this out decades ago. Just look at Mumble, Teamspeak, Discord. They know.
People without proper headsets in that environment get ignored after a while. Nobody wants to use brainpower to understand what you are saying. Corporate might be harder, but you also get payed for that.
We pretty much take echo cancellation for granted at this point. Using something better than your laptop microphone on a call is still a good idea but I'm not sure that wearing headphones/earphones is that big a deal at this point.
You don't need to go back that far until speakerphones other than very expensive Polycoms and the like were pretty mediocre at cutting out because of echo.
https://www.nvidia.com/en-us/geforce/guides/nvidia-rtx-voice...
I think you can do it not only to your microphone (outgoing audio), but to the other participants in the meeting (incoming audio)
It seems that what's old is new again ...
The only thing car-kits seemed to do was add minimum cut-offs before transmitting and make use of directional microphones.
It's more practical than a touch bar, at least.
This post is on noise cancellation specifically, and it actually has the potential to be a huge step forward.
One of the big audio problems with group meetings is that the background noise from each participant adds up, to a point where it quickly becomes unbearable. For that reason, videoconferencing generally only plays audio from one or two participants at most, using a fairly simple estimation of whichever audio signal is currently loudest. The problem is that this can make it really hard to interrupt (people will literally not hear you), or tell the difference between two people going "mm-hmm" versus the whole group. If you've ever been in a group meeting where everybody applauds something, this is why you see everyone applauding but only hear a smattering.
But if this noise cancellation really succeeds, it could be a huge leap forward because audio cues and overlap will actually work for the first time -- hearing the "mm-hmms", hearing everyone pipe up, and so on. Videoconferencing will feel more like an actual single shared audio environment, rather than the kind of "walkie-talkie" effect it so often feels like now.
I'm really looking forward to this.