Audio Lag

Bobby Clinkert said:
Is there any way to fix the awful audio lag? Currently running Wirecast 13.1.2 and users are joining Rendezvous via Google Chrome. Internet connection/bandwith not the issue from what I can tell 

Hi Bobby Clinkert

I have a gut feeling that audio lag is related to internet/connection issues... I've seen friends of mine start with a perfect stream, but after a glitch - its out of sync. I cant prove it, but that's what it feels like. I dont have my own data to confirm this however.

I'd suggest looking at something like ping plotter as a tool to monitor internet during broadcast, and see if there are problems at times when lag is introduced.

Regards

Greg Kuhnert

I've had similar issues. Have you tried encoding to different bitrates? I weirdly found that the drift would happen at 2.5 Mbps but not 4 Mbps. I ended up just going HDMI out and encoding on a TeraDek. 

We have had a very strange revelation that poor lag is entirely down the remote participant's computer, one of our test guests joined on a mid-spec dell PC, from an average home fibre internet, their audio latency was in the region of 4 to 8 seconds, the same guest switched to a high spec i7 PC and instantly their lag was reduced to 0 and were real-time, same internet nothing clever.  The dell was unstable, unable to run any other tasks in the background with the rendezvous session taking over 100% of system resources.  So large multiway calls, really need high spec machines on the guest end.

Telestream could look to implement a google/zoom style approach, for still smaller numbers of guests, but utilising a webrtc server set-up, sacrificing some of the "realtime" of rendezvous for a better guest experience when having a one to many call through a server, rather than simultaneous point to point calls, which leads to the poor guest experience.  

rendezvous is good because it natively comes into wirecast as inputs, which makes it better than using zoom/skype etc.

the peer to peer nature of larger calls uses a lot of client (guest) system resources and bandwidth, which if a "server" based approach was used each client would only have one two-way call to a server rather than have to manage up to 8 two-way calls between browsers.  this would come at a potential trade-off in latency, but I have seen rendezvous so latent that flowing conversation is not possible because of this, as I said in one of our live sessions and tests a guest was 8 to 10 seconds behind the other users.

this is not an approach that would suit all situations, but would be good for when you need to have a number of guests who all need to interact, before going live.

vmix implementation of their vmix call is very good, but that is effectively handled as 8 individual calls from studio to each guest, so the guests can't see or hear each other until they are all on the live program bus, but some of the controls they have implemented are very good,  eg ability to remotely control clients bandwidth, control the bandwidth of the return video feed, separate return feeds (easy because they are separate calls)

the advantage of using the native call system within the mixer is they require less hardware to bring the callers into the system.  we have achieved this with 8 laptops pinned in zoom and 8 NDI feeds in, skype NDI works great, apart from when the NDI stream decides to dynamically adjust itself, changing the window size live on the program bus.  if wirecast could capture and fix that then skype is probably the answer!! 
 

PAUL MILLARD said:
if wirecast could capture and fix that then skype is probably the answer!! 

Couldn't agree more! The only variant there is a warning that Skype mixes the audio of external guests into each NDI feed but as long as you, the controller, know and plan for that, it is easy to manage. Truly, the transform / scaling break that happens in Wirecast is LITERALLY a deal breaker for myself and several others who I've spoken with. We're all watching closely, but for a several year old bug that was reported to not see any progress, we've all but moved on to other systems and building out workflows that are consistently reliable. I'd love to come back, but not until this specific bug fix is in that I would EVERY use this in production.

Could you please explain why there is no option to adjust the delay on the audio on different feeds coming in from Rendevous. The only reason I chose this platform was because of the Rendevous function. I wanted to be able to have 7 guests able to talk. But the syncing is impossible and in reality, makes this function not useable. Absolutely nowhere do you offer any suggestions or help with this. All I am asking is that the feed into the live bit can actually have delay. Why is it that all other audio sources have it, but this, possibly the most important function of your system for me, does not allow this. I feel a bit robbed

I really do nit see why you are so confused. 

Every feed that I have I am able to alter the audio delay. But, the feeds from Rendevous do not allow this.

SO, everybody that is in Rendevous, there mouths are moving at a different time to the sounds coming out. 

Or, if someone has a clapper board, when they clap the board, the sound is coming through before the image.

so, my question is, if you have set up the ability to delay sounds on every other source, why have you not done so for Rendevous.

Basically, I chose your system because it allowed me to have a number of guests at a time speaking live. That cost me £1000. I then relasied that my MAC was not up to it, so I bought a new one (£2400). It has an 8 core i9 processor with 40GB. I have also bought an Xkeys controller.

So, in all I have spent a lot to accomodate yoursystem. But the functionality that forst attracted me to it, and which you sell it on, does not operate correctly. 

I have searched and search, I have just downloaded a beta 13.2 version. But still, there is no ability to delay the audio feed from the different people in rendevous.

I really hope you are following this, because it really is pretty simple

Hello

I quote you 'Video and audio from each guest should be in sync'

I reply... I know that, but they are not. This is what i am enquiring about. 

And I find it a bit odd that you say 'you should explain why that would be important if we improve the sync issues from guests.'

Surely, if someone is giving a presentation or similar it would be really quite nice if their mouth matches the sound coming out of the screen.

I have spoken to a number of people that have just given up using Rendevous because it offers problem after problem.

I really do feel like i have been robbed a bit here.

I have paid for the top end software of yours, in thr hope of having 7 people come in. But, when I have 4 or more the whole thing seems to break down, and the feed from different people are affected.

These issues do not happen on Zoom or other platforms. So, I would say that your system is not actually doing what you say it can. 

So, after all this backward and forward, you are now accepting that there is a sync issue with one of the major selling points for the upgrade to the top end system you have?