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I have a similar story: Where I work, we had a cluster of VMs that were always high CPU and a bit of a problem. We had a lot of fire drills where we'd have to bump up the size of the cluster, abort in-progress operations, or some combination of both.

Because this cluster of VMs was doing batch processing that the founder believed should be CPU intense, everyone just assumed that increasing load came with increasing customer size; and that this was just an annoyance that we could get to after we made one more feature.

But, at one point the bean counters pointed out that we spent disproportionately more on cloud than a normal business did. After one round of combining different VM clusters (that really didn't need to be separate servers), I decided that I could take some time to hook up this very CPU intense cluster up to a profiler.

I thought I was going to be in for a 1-2 week project and would follow a few worms. Instead, the CPU load was because we were constantly loading an entire table, that we never deleted from, into the application's process. The table had transient data that should only last a few hours at most.

I quickly deleted almost a decade's worth of obsolete data from the table. After about 15 minutes, CPU usage for this cluster dropped to almost nothing. The next day we made the VM cluster a fraction of its size, and in the next release, we got rid of the cluster and merged the functionality into another cluster.

I also made a pull request that introduced a simple filter to the query to only load 3 days of data; and then introduced a background operation to clean out the table periodically.

It's weird to be living in a world where this is a surprise but here we are.

Nice write up though. Web sockets has a number of nonsensical design decisions, but I wouldn't have expected that this is the one that would be chewing up all your cpu.

> Even the theoretical maximum size of a TCP/IP packet, 64k, is much smaller than the data we need to send, so there's no way for us to use TCP/IP without suffering from fragmentation.

Just highlights that they do not have enough technical knowledge in house. Should spend the $1m/year saving on hiring some good devs.

But writing a custom ring buffer implementation is also nice, I suppose...

So they are only half way correct about masking. The RFC does mandate that client to server communication be masked. That is only enforced by web browsers. If the client is absolutely anything else just ignore masking. Since the RFC requires a bit to identify if a message is masked and that bit is in no way associated to the client/server role identity of the communication there is no way to really mandate enforcement. So, just don't mask messages and nothing will break.

Fragmentation is completely unavoidable though. The RFC does allow for messages to be fragmented at custom lengths in the protocol itself, and that is avoidable. However, TLS imposes message fragmentation. In some run times messages sent at too high a frequency will be concatenated and that requires fragmentation by message length at the receiving end. Firefox sometimes sends frame headers detached from their frame bodies, which is another form of fragmentation.

You have to account for all that fragmentation from outside the protocol and it is very slow. In my own implementation receiving messages took just under 11x longer to process than sending messages on a burst of 10 million messages largely irrespective of message body length. Even after that slowness WebSockets in my WebSocket implementation proved to be almost 8x faster than HTTP 1 in real world full-duplex use on a large application.

The linked section of the RFC is worth the read: https://www.rfc-editor.org/rfc/rfc6455#section-10.3

Was it because they didn't want to use some multicast video server?

The initial approach was shipping raw video over a WebSocket. I could not imagine putting something like that together and selling it. When your first computer came with 64KB in your entire machine, some of which you can't use at all and some you can't use without bank switching tricks, it's really really hard to even conceive of that architecture as a possibility. It's a testament to the power of today's hardware that it worked at all.

And yet, it did work, and it served as the basis for a successful product. They presumably made money from it. The inefficiency sounds like it didn't get in the way of developing and iterating on the rest of the product.

I can't do it. Premature optimization may be the root of all evil, but I can't work without having some sense for how much data is involved and how much moving or copying is happening to it. That sense would make me immediately reject that approach. I'd go off over-architecting something else before launching, and somebody would get impatient and want their money back.

Here they have a nicely compressed stream of video data, so they take that stream and... decode it. But they aren't processing the decoded data at the source of the decode, so instead they forward that decoded data, uncompressed(!!), to a different location for processing. Surprisingly, they find out that moving uncompressed video data from one location to another is expensive. So, they compress it later (Don't worry, using a GPU!)

At so many levels this is just WTF. Why not forward the compressed video stream? Why not decompress it where you are processing it instead of in the browser? Why are you writing it without any attempt at compression? Even if you want lossless compression there are well known and fast algorithms like flv1 for that purpose.

Just weird.

A more reasonable approach would be to have Chromium save the original compressed video to disk, and then use ffmpeg or similar to reencode if needed.

Even better not use Chromium at all.

They seem to not understand the fundamentals of what they're working on.

> Chromium's WebSocket implementation, and the WebSocket spec in general, create some especially bad performance pitfalls.

You're doing bulk data transfers into a multiplexed short messaging socket. What exactly did you expect?

> However there's no standard interface for transporting data over shared memory.

Yes there is. It's called /dev/shm. You can use shared memory like a filesystem, and no, you should not be worried about user/kernel space overhead at this point. It's the obvious solution to your problem.

> Instead of the typical two-pointers, we have three pointers in our ring buffer:

You can use two back to back mmap(2) calls to create a ringbuffer which avoids this.

(I also wouldn't be surprised if they had even more memory copies than they let on, marshalling between the GC-backed JS runtime to the GC-backed Python runtime.)

I was coming back to HN to include in my comment a link to various high-performance IPC libraries, but another commenter already beat me linking to iceoryx2 (though of course they'd need to use a python extension).

SHM for IPC has been well-understood as the better option for high-bandwidth payloads from the 1990s and is a staple of Win32 application development for communication between services (daemons) and clients (guis).

Are you sure about that? Atomics are not locks, and not all systems have strong memory ordering.

"using WebSockets over loopback was ultimately costing us $1M/year in AWS spend"

then

"and the quest for an efficient high-bandwidth, low-latency IPC"

Shared memory. It has been there for 50 years.

And the GPU for rendering...

So they should instead just be hooking into Chromium's GPU process and grabbing the pre-composited tiles from the LayerTreeHostImpl[1] and dealing with those.

[1]: https://source.chromium.org/chromium/chromium/src/+/main:cc/...

As a point of comparison, how many TB per second of video does Netflix stream?

that’s surprising to.. almost no one? 1TBPS is nothing to scoff at