I've been running a custom Discord bot on a VPS for a while, but to streamline the development/deployment process and bring it in line with my other personal projects, I've been considering moving it to Cloudflare Workers.
Discord fully supports serverless bots via the interactions API and even has a demo for deploying one specifically with Cloudflare workers.
Unfortunately, complications arise when streaming audio from YouTube into voice channels comes into scope.
node-ytdl-core can be used to access a YouTube audio stream and discordjs/voice can be used to broadcast it into a voice channel, but CF Workers has a limit of 10ms CPU runtime per request, rendering it obviously useless for streaming a 3 minute long track.
It seems that if using the paid plan with unbound usage limit, the streams API could be leveraged to quickly return the initial response, but keep streaming the track afterwards in the background, however this would likely be cost prohibitive.
Is there any feasible way for me to convert to Cloudflare serverless infra while preserving my bot's music playing functionality?
The only prior art in this regard I've stumbled upon in this regard is this 5 year old repo that I have not been able to get to work nor can I see why it would work.
I am using google play game services – real time multiplayer API to add multiplayer feature to my mobile games. The engine I am using is Unity3D, but my question does not have to do with Unity (I believe so) so it is not important.
What I would like to know is the delay of the messages that are received over the internet to make my games smooth and synchronized.
I know that in other APIs like Photon you can easily find the delay of the message that is being received but I don’t seem to find it on google play game services API.
Is there any way to know the delay of the received messages on google play game services API?
Thank you for your time!
Determining the latency of the messages is a bit complex in the case of Google Real Time multiplayer APIs since the connections are peer to peer, so most of the data travels directly from one player to the other. (see for details: https://developers.google.com/games/services/common/concepts/realtimeMultiplayer#messaging)
The short answer is you can estimate it yourself, by adding sequence numbers to the messages, and then exchange the time difference each client experienced between the messages. I recommend measuring several messages, and sizes, and not have too much memory since conditions will change. Something like the average time between each message for 30-100 messages and then plan for the slowest link.
To make a good real-time game, you really should assume the latency is variable (sometimes it is low, others high), and it is always longer than you want :)
You might want to checkout https://gamedev.stackexchange.com/questions/58450/mobile-multiplayer-games-and-coping-with-high-latency which has a good discussion on how to handle this situation.
We need to develop a multi-player game with real-time performance.
This needs to be working worldwide (servers in America, Europe, Asia), and supporting a huge traffic. Using Google Cloud services for the hosting.
We're thinking of references like Jam with Chrome, Chrome Maze or Cube Slam.
The game :
2 players challenge a race
We need to simultaneously display the progression of the 2 players
Each match could last around 30 to 45 seconds
The hosting :
We will obviously host the website on AppEngine, automagically scaling,
but are thinking about 2 solutions for the real-time servers :
Using websocket servers with Compute Engine
Like they did for Jam with Chrome, Maze, etc.
Developing our own websocket servers (technology TBD), deploying on datacenters in Europe, US, Asia, handling scaling, syncing between them, computing latency issues on servers and clients, etc.
But it's pretty technically challenging as we are very short on time, and missing an admin sys and network guy for now.
Or using Channel API
We understand that it's not a websocket platform, and real-time performances are lower.
But it would be way more simple and secure for us and the time we have.
So, we would also like to know more about that.
In any case, we think we could use some graphical tricks on front ends, to make it look like real-time, but it really depends if we have a 100~500ms or a 500ms~10s latency.
Some questions :
What would the latency range values look like for the different solutions ?
(Jam w/ Chrome got 100ms with GCE, could Channel API reach several seconds ?)
How would Channel API servers handle high traffic, how does scaling work, could the latency go very high ? (no info about that on Channel docs ?)
What if someone in France play with someone in US, connecting to different servers, waiting them to sync, how to deal with it ?
Any advice or experience to share ?
Any interesting reading or viewing ? (seen some but not very precise)
Any other solution ?
Thank you for any helping comment !
EDIT :
Only 2 players connected together, potentially from different world zone, no broadcasting needed.
We could find some front side tricks to avoid server side processing. This is a race between 2 players, so we actually just need to compare their progression, and the real winner resolution is not that important as there is no real stuff to win, this is more for fun.
If you need a server for processing the data:
I would definitely go with websockets at Compute Engine!
The Channels API is much slower, and also quite unpredictable (latency differs from message to message)! Data has to go to the Channels server, which sends it to the App Engine instance, which has to do a request back to the Channels server, which will push the message to the client. There is too much going on there if you want to keep latency down!
Here is a Channels API stress test:
http://channelapistresstest.appspot.com/
Try clicking "send 5"-button a lot, and you will see latency numbers going up to several seconds.
The Channels API is also quite expensive under heavy load (it probably does not scale well, even if Google of course can solve that with more instances).
When keeping latency down, geolocation is quite important. With a websocket server at Compute Engine, you can send your european visitors to google's european datacenter and your american visitors to the US datacenter (using the geo location headers that AppEngine will provide). You have no such control with the Channels API (or app engine, which all your messages are relayed through). Maybe Google has edge servers for the Channels API (I don't know), but if your AppEngine instance is on the other side of the planet, that does not matter.
If you do NOT need a server for processing the data:
You should establish a peer-to-peer connection with WebRTC, sending stuff directly between the users' browsers. That is was Cube Slam does. (WebRTC requires some initial handshaking ("signaling") so the two peers can find each other, and Channels API would work fine for that handshaking, that's just a couple of messages to establish the peer-to-peer connection.)
WebRTC DataChannels API will give you a nice websocket-like interface like channel.onmessage = function(e) { yadayada()... }; and channel.send("yadayada"); to send your data between the peers.
Occasionally, WebRTC is not able to make a peer-to-peer connection. Then it will fall back to a TURN server, which relays traffic between the peers. Cube Slam is using TURN servers running on ComputeEngine (in both Europe and America to keep latency down), but that is just the fallback when true peer-to-peer is not possible.
It also depends on other things like scalability.
Ingress is built on app engine and a part from the occasional cache glitch it is pretty impressive.
Remember that the channel api is using talk.Google which is the service that hangouts is built on. Scalable and real time.
Personally if your traffic levels are going to be erratic and unpredictable, go app engine. If you think it can be controlled and predictable use compute engine or something else.
Alfred's answer is the best in the frame of the question I asked.
Thank you very much !
However, I forgot to mention a few important points and the scope changed a bit :
We have very little development time (about 1 week only)
This is for a campaign that will last 3 weeks only (we'll need to keep it online a few months afterward, but this is not like we need a long-lasting architecture)
We need to make it work on the broader browser audience as possible (WebRTC only runs on Chrome & Firefox for now)
According to these points, we eventually came up to a 3rd solution :
Using a real-time PAAS.
It's way easier and faster to develop, way cheaper as we don't need a solid backend developer and system/network admin, and we can concentrate more on the project than on the infrastructure and platform.
There are a couple of services that seems good out there, already hosting MMO RPG and the kind, worldwide, with low latency, and good scaling systems.
Here is a list of providers :
https://github.com/leggetter/realtime-web-technologies-guide/blob/master/guide.md
My problem is this...
I have two sites, one acting as an "Admin" site, the other as general "User" site. I need to broadcast live audio from the "Admin" site to all clients of the "User" site. I need to do this with <1 sec of latency.
Some restrictions include:
No install on "User" machines (the idea being the whole thing sits on the web)
If there needs to be a 3rd party plugin then Silverlight is preferred*
Any help much appreciated here
*I have tried IceCast with a flash client, IIS Smooth Streaming, Internet radio, all of which give us a latency of >5 secs.
Have you tried Flash with a server like Red5? You're generally going to get subsecond latency (though not much less than that), as it's designed for realtime communications. There's a learning curve with Flex and ActionScript, but if you're at all familiar with XAML, you can pick it up from the sample apps that come with Red5 pretty quickly.
Failing that, if there aren't too many clients, you can use one of the two real-time peer-to-peer solutions out there, namely Flash over RTMFP or WebRTC over JSEP/ICE/RTP. If you can ensure that all the clients are using Chrome, then WebRTC is probably your best bet. If you can ensure that they're not using Chrome, then Flash is a good choice. The current Flash Pepper client on Chrome is buggy up the wazoo when it comes to audio processing, and no sign of a fix in sight. (It doesn't support echo cancellation, and the volume of the audio goes up and down horribly.) So if you're using Flash, steer clear of recording and broadcasting your audio on Chrome. And I wouldn't recommend either approach if you have more than half a dozen clients - the number of audio streams is gonna overwhelm your "Admin" browser pretty quickly, I think. Better to push that out to something like a Red5 server.
Silverlight is a bad choice for more reasons than I can count. I'm saying this as a guy who spent several years trying to implement a realtime communication solution on Silverlight. Don't do it.
Is there any App/ Method / Process that i can use to get the battery consumption of a single app in blackberry? I am using 9300. My application uses GPRS and sends data over internet.
Till now i have been using a thread which tells me the difference between battery levels after an hour of a phone using my app and a phone not using my app.
Please suggest a better way?
Unfortunately BlackBery is not really open, clear and documented platform.
The best numbers that I got from conference in Amsterdam Info
There's no reliable way to tell how much battery your app has used. As you are already doing, you can retrieve the levels before and after calling DeviceInfo.getBatteryLevel, but the measured difference includes the battery used by other apps too.