Love the transparency here. Would also love if the same transparency was applied to pricing for their core product. Doesn't appear anywhere on the site.
replies(2):
Jokes aside though, some good performance sleuthing there.
(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).
Recall's offering allows you to get "audio, video, transcripts, and metadata" from video calls -- again, total conjecture, but I imagine they do need to decode into raw format in order to split out all these end-products (and then re-encode for a video recording specifically.)
"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.
> I don't mean to be dismissive, but this would have been caught very early on (in the planning stages) by anyone that had/has experience in system-level development rather than full-stack web js/python development
Since they don't have API access to all these platforms, the best they can do to capture the A/V streams is simply to join the meeting in a headless browser on a server, then capture the browser's output and re-encode it.
On the outside we can’t be sure. But it’s possible that they took the right decision to go with a naïve implementation first. Then profile, measure and improve later.
But yes the hole idea of running a headless web browser to get run JavaScript to get access to a video stream is a bit crazy. But I guess that’s just the world we are in.
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.
The memcopys are the cost that they were paying, even if it was local.
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/...
assuming you're only shuffling bytes around, on bare metal this would be ~20 DDR5 channels worth
or 2 servers (12 channels/server for EPYC)
you can get an awful lot of compute these days for not very much money
(shipping your code to the compressed video instead of the exact opposite would probably make more sense though)
I doubt they would have even noticed this outrageous cost if they were running on bare-metal Xeons or Ryzen colo'd servers. You can rent real 44-core Xeon servers for like, $250/month.
So yes, it's an AWS issue.
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.
The product is not a full-stack web application. What makes you think that they brought in people with that kind of experience just for this particular feature?
Especially when they claim that they chose that route because it was what was most convenient. While you might argue that wasn't the right tradeoff, it is a common tradeoff developers of all kinds make. “Make It Work, Make It Right, Make It Fast” has become pervasive in this industry, for better or worse.
>50 GB/s of memory bandwidth is common nowadays[1], and will basically never be the bottleneck for 1080P encoding. Zero copy matters when you're doing something exotic, like Netflix pushing dozens of GB/s from a CDN node.
[1]: https://lemire.me/blog/2024/01/18/how-much-memory-bandwidth-...
As it turns out, doing something in Rust does not absolve you of the obligation to actually think about what you are doing.
You can rent real 44-core Xeon servers for like, $250/month.
Netflix has hardware ISPs can get so they can serve their content without saturating the ISPs lines.
There is a statistic floating around that Netflix was responsible for 15% of the global traffic 2022/2023, and YouTube 12%. If that number is real... That'd be a lot more
The linked section of the RFC is worth the read: https://www.rfc-editor.org/rfc/rfc6455#section-10.3
Context matters? As someone working in production/post, we want to keep it uncompressed until the last possible moment. At least as far as no more compression than how it was acquired.
And since it behaves like filesystem, you can swap it with real filesystem during testing. Very convenient.
I am curious if they tried this already or not and if they did, what problems did they encounter?
> 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.
[0]https://instances.vantage.sh/aws/ec2/c8g.12xlarge?region=us-... [1]https://portal.colocrossing.com/register/order/service/480 [2]https://browser.geekbench.com/v6/cpu/8305329 [3]https://browser.geekbench.com/processors/intel-xeon-e5-2699-...
To my knowledge, Zoom’s web client uses a custom codec delivered inside a WASM blob. How would you capture that video data to forward it to your recording system? How do you decode it later?
Even if the incoming streams are in a standard format, compositing the meeting as a post-processing operation from raw recorded tracks isn’t simple. Video call participants have gaps and network issues and layer changes, you can’t assume much anything about the samples as you would with typical video files. (Coincidentally this is exactly what I’m working on right now at my job.)
It does, but you just removed all context from their comment and introduced a completely different context (video production/post) for seemingly no reason.
Going back to the original context, which is grabbing a compressed video stream from a headless browser, the correct approach to handle that compressed stream is to leave it compressed until the last possible moment.
Anyway, depending on individual nodes to always be up for reliability is incredibly foolhardy. Things can happen, cloud isn't magic, I’ve had instances become unrecoverable. Though it is rare.
So, I still don’t understand the point, that was not exactly relevant to what I said.
---
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.
But writing a custom ring buffer implementation is also nice, I suppose...
I think it's because the cost of it is so abstracted away with free streaming video all across the web. Once you take a look at the egress and ingress sides you realize how quickly it adds up.
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.
If I said that "childbirth cost us 5000 on our <hospital name> bill", you assume the issue is with the hospital.
I dunno, when we're playing with millions of dollars in costs I hope they're at least regularly evaluating whether they could at least run some of the workload on GPUs for better perf/$.
With that constraint, letting a full browser engine decode and composite the participant streams is the only option. And it definitely is an expensive way to do it.
Edit: I guess perhaps you're saying that they don't know all the networking configuration knobs they could exercise, and that's probably true. However, they landed on a more optimal solution that avoided networking altogether, so they no longer had any need to research network configuration. I'd say they made the right choice.
I'm pretty sure that feeding the browser an emulated hardware decoder (ie - write a VAAPI module that just copies compressed frame data for you) would be a good semi-universal solution to this, since I don't think most video chat solutions use DRM like Widevine, but it's not as universal as dumping the framebuffer output off of a browser session.
They could also of course one-off reverse each meeting service to get at the backing stream.
What's odd to me is that even with this frame buffer approach, why would you not just recompress the video at the edge? You could even do it in Javascript with WebCodecs if that was the layer you were living at. Even semi-expensive compression on a modern CPU is going to be way cheaper than copying raw video frames, even just in terms of CPU instruction throughput vs memory bandwidth with shared memory.
It's easy to cast stones, but this is a weird architecture and making this blog post about the "solution" is even stranger to me.
That's not dedicated 48 cores, it's 48 "vCPUs". There are probably 1,000 other EC2 instances running on those cores stealing all the CPU cycles. You might get 4 cores of actual compute throughput. Which is what I was saying
More shocking to me is that anyone would attempt to run network throughput oriented software inside of Chromium. Look at what Cloudflare and Netflix do to get an idea what direction they should really be headed in.
I was really disappointed when my wife couldn't get the night off from work when the company took everyone out to a fancy steak house.
I can easily imagine the author being in a similar boat, knowing that it isn't cheap, but then not realizing that expensive in this context truly does mean expensive until they actually started seeing the associated costs.
The basic point is that WebSockets requires that data move across channels that are too general and cause multiple unaligned memory copies. The CPU cost to do the copies was what cost the megabuck, not network transfer costs.
Read the article.
Atomics require you to explicitly specify a memory ordering for every operation, so the system's memory ordering doesn't really matter. It's still possible to get it wrong, but a lot easier than in (traditional) C.
In fact, you can even get a small discount with the -flex series, if you're willing to compromise slightly. (Small discount for 100% of performance 95% of the time).
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.
I mean, I would presume that the entire reason they forked chrome was to crowbar open the black box to get at the goodies. Maybe they only did it to get a framebuffer output stream that they could redirect? Seems a bit much.
Their current approach is what I'd think would be a temporary solution while they reverse engineer the streams (or even get partnerships with the likes of MS and others. MS in particular would likely jump at an opportunity to AI something).
> What's odd to me is that even with this frame buffer approach, why would you not just recompress the video at the edge? You could even do it in Javascript with WebCodecs if that was the layer you were living at. Even semi-expensive compression on a modern CPU is going to be way cheaper than copying raw video frames, even just in terms of CPU instruction throughput vs memory bandwidth with shared memory.
Yeah, that was my final comment. Even if I grant that this really is the best way to do things, I can't for the life of me understand why they'd not immediately recompress. Video takes such a huge amount of bandwidth that it's just silly to send around bitmaps.
> It's easy to cast stones, but this is a weird architecture and making this blog post about the "solution" is even stranger to me.
Agreed. Sounds like a company that likely has multiple million dollar savings just lying around.
You seem to be assuming that a $200 meal was the only compensation the person received, and they weren't just getting a nice meal as a little something extra on top of getting paid for doing their job competently and efficiently.
But that's the kind of deal I make when I take a job: I do the work (pretty well most of the time), and I get paid. If I stop doing the work, I stop getting paid. If they stop paying, I stop doing the work. (And bonus, literally, if I get a perk once in a while like a free steak dinner that I wasn't expecting)
It doesn't have to be more complicated than that.
What’s surprising to me is they can’t access the compressed video on the wire and have to send decoded raw video. But presumably they’ve thought about that too.
Knowing thyself is a superpower all its own; we need people to write scrappy code to validate a business idea, and we need people who look at code with disgust, throw it out, and write something 100x as efficient.
But yes, it's an order of magnitude easier to get portability right using the C++/Rust memory model than what came before.
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 RFC has a link to a document describing the attack, but the link is broken.
That's independent of pay scale.
Granted, if you pay way below expectations, you'll lose the professionals over time. But if you pay lavishly no matter what, you get the 2021/2022 big tech hiring cycle instead. Neither one is a great outcome.
If one is doing websockets on the local machine (or any other trusted network) and one has performance concerns, one should maybe consider not doing TLS.
If the websocket standard demands TLS, then I guess getting to not do that is would be another benefit of not using a major-web-browser-provided implementation.
Of course the truth is more complicated than the sound bite, but still...
The problem is that the developers behind this way of streaming video data seem to have no idea of how video codecs work.
If they are in control of the headless chromium instances, the video streams, and the receiving backend of that video stream...why not simply use RDP or a similar video streaming protocol that is made exactly for this purpose?
This whole post reads like an article from a web dev that is totally over their head, trying to implement something that they didn't take the time to even think about. Arguing with TCP fragmentation when that is not even an issue, and trying to use a TCP stream when that is literally the worst thing you can do in that situation because of roundtrip costs.
But I guess that there is no JS API for that, so it's outside the development scope? Can't imagine any reason not to use a much more efficient video codec here other than this running in node.js, potentially missing offscreen canvas/buffer APIs and C encoding libraries that you could use for that.
I would not want to work at this company, if this is how they develop software. Must be horribly rushed prototypical code, everywhere.
It is difficult to say, I’ve never used the product. They don’t describe what it is they are trying to do.
If you want to pipe a Zoom call to a Python process it’s complicated.
Everything else that uses WebRTC, I suppose Python should generate the candidates, and the fake browser client hands over the Python process’s candidates instead of its own. It could use the most basic bindings to libwebrtc.
If the bulk of their app is JavaScript, they ought to inject a web worker and use encoded transforms.
But I don’t know though.
Cheaper and more straightforward.
Their discussion of fragmentation shows they are clueless as to the details of the stack. All that shit is basically irrelevant.
The idea that clearer titles are just babying some class of people is perverse.
Titles are the foremost means of deciding what to read, for anyone of any sophistication. Clearer titles benefit everyone.
The subject matter is meaningful to more than AWS users, but non-AWS users are going to be less likely to read it based on the title.
They don't have control of the incoming video format.
They don't even have access to the incoming video data, because they're not using an API. They're joining the meeting using a real browser, and capturing the video.
Is it an ugly hack? Maybe. But it's also a pretty robust one, because they're not dependent on an API that might break or reverse-engineering a protocol that might change. They're a bit dependent on the frontend, but that changes rarely and it's super easy to adapt when it does change.
That doesn't sound like a ridiculous idea to me. How else would you get video data out of a sandboxed Chromium process?
They're capturing video from inside a Chromium process. How exactly do you expect to send the raw captured video frames to hls?
Are you proposing implementing the HLS server inside a web process?
But they don't.
They support 7 different meeting providers (Zoom, Meet, WebEx, ...), none of which have an API that give you access to the compressed video stream.
In theory, you could try to reverse-engineer each protocol...but then your product could break for potentially days or weeks anytime one of those companies decides to change their protocol - vs web scraping, where if it breaks they can probably fix it in 15 minutes.
Their solution is inefficient, but robust. And that's ultimately a more monetizable product.
They support 7 meeting platforms. Even if 1 or 2 are open to providing APIs, they're not all going to do that.
Reverse-engineering the protocol would be far more efficient, yes - but it'd also be more brittle. The protocol could change at any time and reverse-engineering it again could days between days and weeks. Would you want a product with that sort of downtime?
Also, does it scale? Reverse-engineering 7+ protocols is a lot of engineering work, and it's very specialized work that not any software engineer could just dive into quickly.
In comparison, writing web scrapers to find the video element for 7 different meeting products is super easy to write, and super easy to fix.
Then they capture the video from the meeting in Chromium.
Then they need to send that captured video to another process for compression and processing.
No, WebSockets isn't the most efficient, but there aren't that many options once you're capturing inside a web page.
An uncompressed 1920*1080 30fps RGB stream is 178 megabytes / second. (This is 99% likely what they were capturing from the headless browser, although maybe at a lower frame rate - you don’t need full 30 for a meeting capture.)
In comparison, a standard Netflix HD stream is around 1.5 megabits / s, so 0.19 megabytes.
The uncompressed stream is almost a thousand times larger. At that rate, the Websocket overhead starts having an impact.
Hardware decoding works best when you have a single stable high bitrate stream with predictable keyframes — something like a 4K video player.
Video meetings are not like that. You may have a dozen participant streams, and most of them are suffering from packet loss. Lots of decoder context switching and messy samples is not ideal for typical hardware decoders.
They are in control of the bot server that joins with the headless chrome client. They can use the CDP protocol to use the screencast API to write the recorded video stream to filesystem/disk, and then they can literally just run ffmpeg on that on-disk-on-server file and stream it somewhere else.
But instead they decided to use websockets to send it from that bot client to their own backend API, transmitting the raw pixels as either a raw blob or base64 encoded data, each frame, not encoded anyhow. And that is where the huge waste in bandwidth comes from.
(The article hints to this in a lot of places)
If I try to match the actual machine. 16G ram. A rough estimate is that their Xeon E3-1240 would be ~2 AWS vCPU. So an r6g.large is the instance that would roughly match this one. Add 500G disk + 1 Gbps to/from the internet and ... monthly cost 3,700 USD.
Without any disk and without any data transfer (which would be unusable) it's still ~80USD. Maybe you could create a bootable image that calculates primes.
These are still not the same thing, I get it, but ... it's safe to say you cannot get anything remotely comparable on AWS. You can only get a different thing for way more money.
(made estimates on https://calculator.aws/ )
Always. Always ?!?
Article summary: if you're moving a lot of data, your protocol's structure and overhead matters. A lot.
I see "AMD EPYC 7502P 32-Core" for 236 EUR per month. Can you tell me where you see 48c/96t?
EDIT
I found it! Unbelievable that it is so cheap.
https://www.hetzner.com/dedicated-rootserver/#cores_threads_...
This is because reading how they came up with the solution it is clear they have little understanding how low level stuff works. For example, they surprised by the amount of data, that TCP packets are not the same as application level packets or frames, etc.
As for ring buffer design I’m not sure I understand their solution. Article mentions media encoder runs in a separate process. Chromium threads live in their processes (afaik) as well. But ring buffer requirement says “lock free” which only make sense inside a single process.
Why would think you need locks?
What does this mean? The article says 'TB' which would be terabytes. Terabytes are made out of gigabytes. There is nothing faster than straight memory bandwidth. DDR5 has 64 GB/s max. 12 channels of that is 768 GB/s.
Terabytes per second is going to take multiple computers, but it will be a lot less computers if you're using shared memory bandwidth and not some sort of networking loopback.
It’s your job as an employee, it’s why you get paid in the first place
Don't assume that a steak dinner was the only recognition we got.
As far as comp: I was well taken care of, and I won't discuss more in a public forum.
125 MB per second × 60 seconds per minute × 60 minutes per hour × 24 hours per day x 30 days = 324 TB?
If you want 1 Gbps unmetered colo pricing, AWS is not competitive. So set up your video streaming service elsewhere :-)
https://portal.colocrossing.com/register/order/service/480 offers unmetered for $2,500 additional per month, for the record.
If you have high bandwidth needs on AWS you can use AWS Lightsail, which has some discounted transfer rates.
With my mindset, you have a gigantic chunk of data. Especially if you're recording multiple streams per machine. The immediate thought is that you want to avoid copying as much as possible. If you really, really have to, you can copy it once. Maybe even twice, though before moving from 1 to 2 copies you should spend some time thinking about whether it's possible to move from 1 to 0, or never materializing the full data at all (i.e., keep it compressed, which could apply here but only as an optimization for certain video applications and so is irrelevant to the bootstrapping phase).
WebSockets take your giant chunk of data and squeeze it through a straw. How many times does each byte get copied in the process? I don't know, but probably more than twice. Even worse, it's going to process it in chunks, so you're going to have per-chunk overhead (maybe including a context switch?) that is O(number of chunks in a giant data set).
But the application fundamentally requires squishing that giant data back down again, which immediately implies moving the computation to the data. I would want to experiment with a wasm-compiled video compressor (remember, we already have the no GPU constraint, so it's ok to light the CPU on fire), and then get compressed video out of the sandbox. WebSockets don't seem unreasonable for that -- they probably cost a factor of 2-4 over the raw data size, but once you've gained an order of magnitude from the compression, that's in the land of engineering tradeoffs. The bigger concern is dropping frames by combining the frame generation and reading with the compression, though I think you could probably use a Web Worker and SharedArrayBuffers to put those on different cores.
But I'm wrong. The data isn't so large that the brute force approach wouldn't work at all. My version would take longer to get up and running, which means they couldn't move on to the rest of the system.
More than likely the CPU wasn’t able to keep up. The pipeline was likely generating a frame, storing it to memory, copying from memory to the PCIe device memory, displaying the frame, then generating the next frame. It wouldn’t surprise me if a ~2010 era CPU struggled doing so.
[1] Pretty much any GPU’s memory bandwidth is going to be limited by link speed. An 8800GTS 320MB from 2007 had a theoretical memory bandwidth of around 64GB/s, for reference.
No, "lock free" is a thing that's nice to have when you've got two threads accessing the same memory. It doesn't matter if those two threads are in the same process or it's two different processes accessing the same memory. It's almost certainly two different processes in this case, and the shared memory is probably memory mapped file.
Whatever it is, the shared memory approach is going to be much faster using the kernel to ship the data between the two processes. Via the kernel means two copies, and probably two syscalls as well.
"It depends" is, of course, the common answer, but in most places I've worked, "please help find operational optimizations that can have a positive impact for the team, department or organization" has certainly never been an explicit ask.
Ask team mates to change something to help on a project may fall under the same category, but usually the effect isn't felt beyond a project.
My default mode when coming in to jobs is to try to get a 'full company' view, because I want to know how things work and how they might be made better. That approach is usually not met with much enthusiasm, and usually more with "that's not your job, you don't need to know that", etc.
I took a daily import routine that was taking 25+ hours (meaning we couldn't show 'daily info' because it was out of date before finishing import) and got it down to 30 minutes. This was after having to fight/argue to see the data, and being told for a couple weeks "it can't be sped up, we'll have to buy faster hardware" ($8-$10k min, but they weren't looking at $15-20k IIRC). I spent a few hours over a weekend and got it down to 30 minutes, and saved the company minimum $8k. But I had to fight/argue to even do that ("that's not your job", "Charles is taking care of that", "the client will just have to deal with more delays while we upgrade", etc).
Since it's coming from a headless process, they can just pipe it into ffmpeg, which is probably what they're using on the back-end anyway. Send the output to a file, then copy those to s3 as they're generated. And you can drop the frame rate and bitrate on that while you're at it, saving time and latency.
It's really not rocket science. You just have to understand your problem domain more better.
Shipping uncompressed video around is ridiculous, unless you're doing video editing. And even then you should use low-res copies and just push around EDLs until you need to render (unless you need high-res to see something).
Given that they're doing all that work, they might as well try to get an HLS encoder running in chrome. There just was an mp3 codec in web assembly on HN, so an HLS live encoder may not be too hard. I mean, if they were blowing a million because of their bad design they could blow another million building a browser-based HLS encoder.
There is no single mechanism that does. Paying well is always a component in attracting talent (see my original comment)
It is not a guarantor of quality/motivation. That's ongoing leadership work. And part of that is maintaining/kindling pride in people's work (and firing the ones who are just there for the money)
Fixing the business is very explicitly not your job, and is absolutely not what you're paid for. Any value you create for the business outside of those bounds is at your own cost and you absolutely will not be compensated unless the business is so small you don't have six layers of management trying to extract any kind of promotion.
Lock-free code is designed for concurrent access, but using some clever tricks to handle synchronization between processes without actually invoking a lock. Lock-free explicitly means parallel.
> write pointer: the next address to write to
OK
> peek pointer: the address of the next frame to read > read pointer: the address where data can be overwritten
What? If the "write pointer" is the "the next address to write to" then the "read pointer" had better be "the next address to read from".
The "peek pointer" should be the "read pointer", and the pointer to the end of the free sector should be the "stop pointer" or "unfreed pointer" or "in-use pointer" or literally anything else. Even "third pointer" would be less confusing!