Posts in Json (20 found)

Porting MiniJinja to Go With an Agent

Turns out you can just port things now. I already attempted this experiment in the summer, but it turned out to be a bit too much for what I had time for. However, things have advanced since. Yesterday I ported MiniJinja (a Rust Jinja2 template engine) to native Go, and I used an agent to do pretty much all of the work. In fact, I barely did anything beyond giving some high-level guidance on how I thought it could be accomplished. In total I probably spent around 45 minutes actively with it. It worked for around 3 hours while I was watching, then another 7 hours alone. This post is a recollection of what happened and what I learned from it. All prompting was done by voice using pi , starting with Opus 4.5 and switching to GPT-5.2 Codex for the long tail of test fixing. MiniJinja is a re-implementation of Jinja2 for Rust. I originally wrote it because I wanted to do a infrastructure automation project in Rust and Jinja was popular for that. The original project didn’t go anywhere, but MiniJinja itself continued being useful for both me and other users. The way MiniJinja is tested is with snapshot tests: inputs and expected outputs, using insta to verify they match. These snapshot tests were what I wanted to use to validate the Go port. My initial prompt asked the agent to figure out how to validate the port. Through that conversation, the agent and I aligned on a path: reuse the existing Rust snapshot tests and port incrementally (lexer -> parser -> runtime). This meant the agent built Go-side tooling to: This resulted in a pretty good harness with a tight feedback loop. The agent had a clear goal (make everything pass) and a progression (lexer -> parser -> runtime). The tight feedback loop mattered particularly at the end where it was about getting details right. Every missing behavior had one or more failing snapshots. I used Pi’s branching feature to structure the session into phases. I rewound back to earlier parts of the session and used the branch switch feature to inform the agent automatically what it had already done. This is similar to compaction, but Pi shows me what it puts into the context. When Pi switches branches it does two things: Without switching branches, I would probably just make new sessions and have more plan files lying around or use something like Amp’s handoff feature which also allows the agent to consult earlier conversations if it needs more information. What was interesting is that the agent went from literal porting to behavioral porting quite quickly. I didn’t steer it away from this as long as the behavior aligned. I let it do this for a few reasons. First, the code base isn’t that large, so I felt I could make adjustments at the end if needed. Letting the agent continue with what was already working felt like the right strategy. Second, it was aligning to idiomatic Go much better this way. For instance, on the runtime it implemented a tree-walking interpreter (not a bytecode interpreter like Rust) and it decided to use Go’s reflection for the value type. I didn’t tell it to do either of these things, but they made more sense than replicating my Rust interpreter design, which was partly motivated by not having a garbage collector or runtime type information. On the other hand, the agent made some changes while making tests pass that I disagreed with. It completely gave up on all the “must fail” tests because the error messages were impossible to replicate perfectly given the runtime differences. So I had to steer it towards fuzzy matching instead. It also wanted to regress behavior I wanted to retain (e.g., exact HTML escaping semantics, or that must return an iterator). I think if I hadn’t steered it there, it might not have made it to completion without going down problematic paths, or I would have lost confidence in the result. Once the major semantic mismatches were fixed, the remaining work was filling in all missing pieces: missing filters and test functions, loop extras, macros, call blocks, etc. Since I wanted to go to bed, I switched to Codex 5.2 and queued up a few “continue making all tests pass if they are not passing yet” prompts, then let it work through compaction. I felt confident enough that the agent could make the rest of the tests pass without guidance once it had the basics covered. This phase ran without supervision overnight. After functional convergence, I asked the agent to document internal functions and reorganize (like moving filters to a separate file). I also asked it to document all functions and filters like in the Rust code base. This was also when I set up CI, release processes, and talked through what was created to come up with some finalizing touches before merging. There are a few things I find interesting here. First: these types of ports are possible now. I know porting was already possible for many months, but it required much more attention. This changes some dynamics. I feel less like technology choices are constrained by ecosystem lock-in. Sure, porting NumPy to Go would be a more involved undertaking, and getting it competitive even more so (years of optimizations in there). But still, it feels like many more libraries can be used now. Second: for me, the value is shifting from the code to the tests and documentation. A good test suite might actually be worth more than the code. That said, this isn’t an argument for keeping tests secret — generating tests with good coverage is also getting easier. However, for keeping code bases in different languages in sync, you need to agree on shared tests, otherwise divergence is inevitable. Lastly, there’s the social dynamic. Once, having people port your code to other languages was something to take pride in. It was a sign of accomplishment — a project was “cool enough” that someone put time into making it available elsewhere. With agents, it doesn’t invoke the same feelings. Will McGugan also called out this change . Lastly, some boring stats for the main session: This did not count the adding of doc strings and smaller fixups. Pi session transcript Narrated video of the porting session Parse Rust’s test input files (which embed settings as JSON headers). Parse the reference insta snapshots and compare output. Maintain a skip-list to temporarily opt out of failing tests. It stays in the same session so I can navigate around, but it makes a new branch off an earlier message. When switching, it adds a summary of what it did as a priming message into where it branched off. I found this quite helpful to avoid the agent doing vision quests from scratch to figure out how far it had already gotten. Agent run duration: 10 hours ( 3 hours supervised) Active human time: ~45 minutes Total messages: 2,698 My prompts: 34 Tool calls: 1,386 Raw API token cost: $60 Total tokens: 2.2 million Models: and for the unattended overnight run

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devansh Yesterday

HonoJS JWT/JWKS Algorithm Confusion

After spending some time looking for security issues in JS/TS frameworks , I moved on to Hono - fast, clean, and popular enough that small auth footguns can become "big internet problems". This post is about two issues I found in Hono's JWT/JWKS verification path: Both were fixed in hono 4.11.4 , and GitHub Security Advisories were published on January 13, 2026 . If you already have experience with JWT stuff, you can skip this: The key point here is that, algorithm choice must not be attacker-controlled. Hono's JWT helper documents that is optional - and defaults to HS256. That sounds harmless until you combine it with a very common real-world setup: In that case, the verification path defaults to HS256, treating that public key string as an HMAC secret, and that becomes forgeable because public keys are, well… public. If an attacker can generate a token that passes verification, they can mint whatever claims the application trusts ( , , , etc.) and walk straight into protected routes. This is the "algorithm confusion" class of bugs, where you think you're doing asymmetric verification, but you're actually doing symmetric verification with a key the attacker knows. This is configuration-dependent. The dangerous case is: The core issue is, Hono defaults to , so a public key string can accidentally be used as an HMAC secret, allowing forged tokens and auth bypass. Advisory: GHSA-f67f-6cw9-8mq4 This was classified as High (CVSS 8.2) and maps it to CWE-347 (Improper Verification of Cryptographic Signature) . Affected versions: Patched version: 4.11.4 In the JWK/JWKS verification middleware, Hono could pick the verification algorithm like this: GitHub's advisory spells it out, when the selected JWK doesn't explicitly define an algorithm, the middleware falls back to using the from the unverified JWT header - and since in JWK is optional and commonly omitted, this becomes a real-world issue. If the matching JWKS key lacks , falls back to token-controlled , enabling algorithm confusion / downgrade attacks. "Trusting " is basically letting the attacker influence how you verify the signature. Depending on surrounding constraints (allowed algorithms, how keys are selected, and how the app uses claims), this can lead to forged tokens being accepted and authz/authn bypass . Advisory: GHSA-3vhc-576x-3qv4 This was classified as High (CVSS 8.2) , also CWE-347 , with affected versions and patched in 4.11.4 . Both advisories took the same philosophical stance i.e. Make explicit. Don't infer it from attacker-controlled input. The JWT middleware now requires an explicit option — a breaking change that forces callers to pin the algorithm instead of relying on defaults. Before (vulnerable): After (patched): (Example configuration shown in the advisory.) The JWK/JWKS middleware now requires an explicit allowlist of asymmetric algorithms, and it no longer derives the algorithm from untrusted JWT header values. It also explicitly rejects symmetric HS* algorithms in this context. Before (vulnerable): After (patched): (Example configuration shown in the advisory.) JWT / JWK / JWKS Primer Vulnerabilities [CVE-2026-22817] - JWT middleware "unsafe default" (HS256) Why this becomes an auth bypass Who is affected? Advisory / severity [CVE-2026-22817] - JWK/JWKS middleware fallback Why it matters Advisory / severity The Fix Fix for #1 (JWT middleware) Fix for #2 (JWK/JWKS middleware) Disclosure Timeline a default algorithm footgun in the JWT middleware that can lead to forged tokens if an app is misconfigured a JWK/JWKS algorithm selection bug where verification could fall back to an untrusted value JWT is . The header includes (the signing algorithm). JWK is a JSON representation of a key (e.g. an RSA public key). JWKS is a set of JWKs, usually hosted at something like . The app expects RS256 (asymmetric) The developer passes an RSA public key string But they don't explicitly set you use the JWT middleware with an asymmetric public key and you don't pin Use if present Otherwise, fall back to from the JWT (unverified input) Discovery: 09th Dec, 2025 First Response: 09th Dec, 2025 Patched in: hono 4.11.4 Advisories published: 13 Jan, 2026 Advisory: GHSA-f67f-6cw9-8mq4 Advisory: GHSA-3vhc-576x-3qv4

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Simon Willison 5 days ago

Fly's new Sprites.dev addresses both developer sandboxes and API sandboxes at the same time

New from Fly.io today: Sprites.dev . Here's their blog post and YouTube demo . It's an interesting new product that's quite difficult to explain - Fly call it "Stateful sandbox environments with checkpoint & restore" but I see it as hitting two of my current favorite problems: a safe development environment for running coding agents and an API for running untrusted code in a secure sandbox. Disclosure: Fly sponsor some of my work. They did not ask me to write about Sprites and I didn't get preview access prior to the launch. My enthusiasm here is genuine. I predicted earlier this week that "we’re due a Challenger disaster with respect to coding agent security" due to the terrifying way most of us are using coding agents like Claude Code and Codex CLI. Running them in mode (aka YOLO mode, where the agent acts without constantly seeking approval first) unlocks so much more power, but also means that a mistake or a malicious prompt injection can cause all sorts of damage to your system and data. The safe way to run YOLO mode is in a robust sandbox, where the worst thing that can happen is the sandbox gets messed up and you have to throw it away and get another one. That's the first problem Sprites solves: That's all it takes to get SSH connected to a fresh environment, running in an ~8GB RAM, 8 CPU server. And... Claude Code and Codex and Gemini CLI and Python 3.13 and Node.js 22.20 and a bunch of other tools are already installed. The first time you run it neatly signs you in to your existing account with Anthropic. The Sprites VM is persistent so future runs of will get you back to where you were before. ... and it automatically sets up port forwarding, so you can run a localhost server on your Sprite and access it from on your machine. There's also a command you can run to assign a public URL to your Sprite, so anyone else can access it if they know the secret URL. In the blog post Kurt Mackey argues that ephemeral, disposable sandboxes are not the best fit for coding agents: The state of the art in agent isolation is a read-only sandbox. At Fly.io, we’ve been selling that story for years, and we’re calling it: ephemeral sandboxes are obsolete. Stop killing your sandboxes every time you use them. [...] If you force an agent to, it’ll work around containerization and do work . But you’re not helping the agent in any way by doing that. They don’t want containers. They don’t want “sandboxes”. They want computers. [...] with an actual computer, Claude doesn’t have to rebuild my entire development environment every time I pick up a PR. Each Sprite gets a proper filesystem which persists in between sessions, even while the Sprite itself shuts down after inactivity. It sounds like they're doing some clever filesystem tricks here, I'm looking forward to learning more about those in the future. There are some clues on the homepage : You read and write to fast, directly attached NVMe storage. Your data then gets written to durable, external object storage. [...] You don't pay for allocated filesystem space, just the blocks you write. And it's all TRIM friendly, so your bill goes down when you delete things. The really clever feature is checkpoints. You (or your coding agent) can trigger a checkpoint which takes around 300ms. This captures the entire disk state and can then be rolled back to later. For more on how that works, run this in a Sprite: Here's the relevant section: Or run this to see the for the command used to manage them: Which looks like this: I'm a big fan of Skills , the mechanism whereby Claude Code (and increasingly other agents too) can be given additional capabilities by describing them in Markdown files in a specific directory structure. In a smart piece of design, Sprites uses pre-installed skills to teach Claude how Sprites itself works. This means you can ask Claude on the machine how to do things like open up ports and it will talk you through the process. There's all sorts of interesting stuff in the folder on that machine - digging in there is a great way to learn more about how Sprites works. Also from my predictions post earlier this week: "We’re finally going to solve sandboxing" . I am obsessed with this problem: I want to be able to run untrusted code safely, both on my personal devices and in the context of web services I'm building for other people to use. I have so many things I want to build that depend on being able to take untrusted code - from users or from LLMs or from LLMs-driven-by-users - and run that code in a sandbox where I can be confident that the blast radius if something goes wrong is tightly contained. Sprites offers a clean JSON API for doing exactly that, plus client libraries in Go and TypeScript and coming-soon Python and Elixir . From their quick start: You can also checkpoint and rollback via the API, so you can get your environment exactly how you like it, checkpoint it, run a bunch of untrusted code, then roll back to the clean checkpoint when you're done. Managing network access is an important part of maintaining a good sandbox. The Sprites API lets you configure network access policies using a DNS-based allow/deny list like this: Sprites have scale-to-zero baked into the architecture. They go to sleep after 30 seconds of inactivity, wake up quickly when needed and bill you for just the CPU hours, RAM hours and GB-hours of storage you use while the Sprite is awake. Fly estimate a 4 hour intensive coding session as costing around 46 cents, and a low traffic web app with 30 hours of wake time per month at ~$4. (I calculate that a web app that consumes all 8 CPUs and all 8GBs of RAM 24/7 for a month would cost ((7 cents * 8 * 24 * 30) + (4.375 cents * 8 * 24 * 30)) / 100 = $655.2 per month, so don't necessarily use these as your primary web hosting solution for an app that soaks up all available CPU and RAM!) I was hopeful that Fly would enter the developer-friendly sandbox API market, especially given other entrants from companies like Cloudflare and Modal and E2B . I did not expect that they'd tackle the developer sandbox problem at the same time, and with the same product! My one concern here is that it makes the product itself a little harder to explain. I'm already spinning up some prototypes of sandbox-adjacent things I've always wanted to build, and early signs are very promising. I'll write more about these as they turn into useful projects. Update : Here's some additional colour from Thomas Ptacek on Hacker News: This has been in the works for quite awhile here. We put a long bet on "slow create fast start/stop" --- which is a really interesting and useful shape for execution environments --- but it didn't make sense to sandboxers, so "fast create" has been the White Whale at Fly.io for over a year. You are only seeing the long-form articles from my blog. Subscribe to /atom/everything/ to get all of my posts, or take a look at my other subscription options . Developer sandboxes Storage and checkpoints Really clever use of Claude Skills A sandbox API Scale-to-zero billing Two of my favorite problems at once

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Giles's blog 6 days ago

Writing an LLM from scratch, part 30 -- digging into the LLM-as-a-judge results

I'm still working on my "extra credit" projects after finishing the main body of Sebastian Raschka 's book " Build a Large Language Model (from Scratch) ". Last time around, I trained four base models, using the GPT-2 architecture from the book, on Lambda Labs machines. I was using two ways to compare them with each other, with three models that I'd trained locally, and with the original GPT-2 weights from OpenAI: Here were the results I got, sorted by the loss: Now, you'd expect there to be at least a loose correlation; the lower the loss, the higher the IFT score. But, while we can see a difference between the OpenAI weights and our own, within our own there doesn't seem to be a logical pattern. I think that the problem is that the results from the GPT-5.1 LLM-as-a-judge are not consistent between models. That's not a complaint about the code or its original design, of course -- it was originally written as part of the LLM book as a way of doing a quick test on an instruction fine-tuned model that we'd spent the previous 238 pages writing -- just something that was a bit more efficient than reading hundreds of input/output pairs ourselves. It was never meant to be a tool to compare models in the way I'm using it now. In this post I'll dig into why it doesn't work for this kind of thing, and see if that's something we can change. Let's spec out the problem first. The instruction fine-tuning test trains our model on the Alpaca dataset in order to let it know how to follow instructions; that comprises a series of sequences like this: More details in this post . In the version I've settled on , I fine-tune on a training set of 85% of the samples, epoch by epoch, bailing out when the loss on a separate validation set of 5% of the samples starts rising. I then use the weights from the previous epoch -- that is, before validation loss started rising -- to generate responses to the remaining 10% of the samples. Once that's done, the script hits the OpenAI API, using GPT-5.1, default parameters for all of the options (eg. no explicit temperature) with queries like this: We do that for every model-generated response in the test set, then take the average of the scores and use that as our result. To see why that's problematic, imagine this simple instruction with no separate input: One response I've seen from my models was this: That's obvious garbage, and should get a zero -- and GPT-5.1 consistently does that. Another response, from OpenAI's original weights for their "medium" model (larger than the ones I've been training), is this: That's correct, so it deserves 100, or perhaps 95 due to being unnecessarily wordy (the answer "Jane Austen" is the suggested response in the dataset). But now how about this one: One of my models came up with that gem during an earlier eval. It's completely wrong, so it deserves a 0, right? And normally the GPT-5.1 model does that -- but sometimes it's a little more generous, and gives it a low, but non-zero score. When asked for its reason for that, it makes the logical point that while it's the wrong answer, at least Sarah Palin is a real person. It's better than the "the book wrote itself" complete nonsense of the first response. The problem is that the different runs against the different models are not consistent, as they're all talking to GPT-5.1 separately. One model might find it in a harsh "mood", and get a lower rating than another model that found it at a more generous moment. I came to the conclusion that the best way to fix this is to do a "batch" -- that is, fine-tune each model on the Alpaca dataset that Raschka provides, and generate responses for the test set and store them in a file. Then, once we've done that for all models, we can score them all at once, prompting GPT-5.1 with something like this: The theory is that doing it that way will mean that each individual query/response pair is graded consistently between models, even if there might still be inconsistencies between query/response pairs. That hopefully means we'll get more consistent results and can compare the models better. Here's the code: Running the first against each of our models, and then the second against all of the output files, gives us this updated table (with links to the annotated JSON files in case anyone else wants to take a look): (Still sorted by loss so that you can compare it more easily with the one above.) That's really interesting! The IFT score is still not correlated with the loss. But there does appear to be a pattern. It looks like we have three groups of models: I tried running the LLM-as-a-judge scoring script a few times, just to make sure this wasn't some kind of random weirdness, but the pattern was always the same: the OpenAI weights, the cloud FineWeb 8x A100 40 GiB, and the two local Local FineWeb-Edu models always got the best IFT scores, though sometimes they swapped positions (apart from the OpenAI medium model, which was of course always at the top). The other cloud FineWeb models and the local FineWeb one were consistently scored much lower. A hypothesis: there are two things that contribute to how good a model is at these IFT tests: Or to put it another way -- some of these models are smart but not knowledgeable, while others are knowledgeable but not smart, and some are neither. I think that could explain what we're seeing here. While OpenAI never published their "WebText" dataset for GPT-2, the paper describes it as a new web scrape which emphasizes document quality. To do this we only scraped web pages which have been curated/filtered by humans. Manually filtering a full web scrape would be exceptionally expensive so as a starting point, we scraped all outbound links from Reddit, a social media platform, which received at least 3 karma. Now, the FineWeb dataset is quite similar, though I think it's a tad more curated than that. But OpenAI trained their models for quite some time and did lots of tricks to get the loss as low as possible. By contrast, the FineWeb-Edu dataset is a carefully selected subset of FineWeb, with only the most "educational" data. Models trained on it, you might think, would know more facts for a given amount of training. So we can imagine the OpenAI models are smart but not knowledgeable, as we can our cloud FineWeb 8x A100 40 GiB model, which (I believe due to an accidentally-near-optimal batch size) worked out well in terms of loss. They were trained on relatively sloppy datasets but turned out reasonably well. Their intelligence makes up for some of their lack of knowledge. Our other cloud trains and the local FineWeb one are dumb and not knowledgeable; they were trained on the low-information FineWeb dataset, but they didn't wind up with a particularly amazing loss. So they get low scores. And finally, our local FineWeb-Edu models are still dumb, but they make up for it by knowing more because their training data was better. Well, it sounds plausible ;-) And I'd like to spend some time digging in to see if there's any indication if it's actually true. But after an afternoon of poking around the results, I can't really get a handle on whether it is, or indeed how you'd test that hypothesis in any real depth. TBH, I think this has zoomed so far past my "no side quests" limit that it's not even visible in the rear view mirror, so it's probably best to shelve it as a "cool idea, bro" for now. Learning about how to run sensible evals, and how to work out what they're saying, will have to be a task for another day. I will keep on doing these IFT tests for future models, though, just out of interest. So: let's get back to our regular scheduled LLM training. Next up, how do we upload our models to Hugging Face quickly and easily so that other people can play with them. A simple cross entropy loss over a fixed test set. The results for an instruction fine-tune test that's covered in the book. A script to fine-tune a model and generate test responses and to dump them into a JSON file. The LLM-as-a-judge code to send a bunch of models' responses to GPT-5.1 . It scrambles the order of the models in each query, to try to avoid any preference the model might have for the first one vs the last one, and it stores GPT-5.1's per-response scores and comments in a new "annotated" JSON file. The OpenAI weights and the cloud train on the 8x A100 40 GiB machine using FineWeb, which have low loss and high IFT scores The other cloud models and the local train that used FineWeb, which have medium loss and low IFT scores. The FineWeb-Edu local trains, which have high loss, but IFT scores that are almost as good as the first group's. The loss. Models that are better at predicting the next token are inherently better at instruction-following after the fine-tuning. The amount of information in the dataset. It doesn't matter how clever a model is, if it never saw "Jane Austen wrote 'Pride and Prejudice'" as part of its training, it will never be able to get a good score on that question.

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Giles's blog 1 weeks ago

Writing an LLM from scratch, part 29 -- using DistributedDataParallel to train a base model from scratch in the cloud

I'm carrying on with my "extra credit" projects after finishing the main body of Sebastian Raschka 's book " Build a Large Language Model (from Scratch) ". Having proven that I could train a GPT-2 small scale base model from scratch on my RTX 3090 in 48 hours, I wanted to try training it on a multi-GPU machine on Lambda Labs. There are two benefits I see in doing that: In addition, I wanted to see if anything unexpected dropped out of it; after all, there were four different sizes of machines that I wanted to try, so I'd be doing four from-scratch trains on the same dataset. Does the machine size affect the quality of the model in some way? Here's what happened. As with the last post, this is a set of tidied-up lab notes, so you can see the full journey. There's a lot to it! I was considering splitting it into multiple posts -- "writing the code", "building the datasets", "running the trains" -- but they're interleaved. Each train taught me something about how to structure the code to make it easier to use, so the code kept changing. So I think it's worth documenting the process as it really was. If at some point I want to write a how-to document on porting single-GPU code to multi-GPU, I'll be able to mine this for resources, and in the meantime, hopefully this will be of use to readers -- even if it's just at the level of "I got this error message, how do I fix it?" Anyway, once again I don't want to bury the lede, so: after spending US$215.16 on various trains on various servers, I was able to find that a reasonably cheap instance on Lambda Labs, with 8x A100 GPUs, each of which has 40 GiB of VRAM, is the sweet spot for this particular 163M-parameter, ~Chinchilla-optimal single-epoch run. They can train the model in less than four hours, they happen to be the right size for batches that minimise loss (more on that later), and can do that train for about US$35, excluding validation. If you'd like to read the gory details of what I did, then read on -- but if you prefer, you can jump straight to the results . Back when I was messing around with fine-tuning LLMs using the Hugging Face ecosystem -- their "Transformers" library and so on -- one of the experiments I did was to fine-tune a 0.5B Qwen model on an 8x GPU machine . As part of that, I came across this excellent HF page summarising different kinds of multi-GPU training techniques . The three that are relevant are: Now, from what I understand, due to all of the copying around of models, plus the issues inherent with the GIL in Python, DDP is actually better than DP despite being more complicated -- and more flexible! Per Hugging Face: DDP is recommended because it reduces communication overhead between GPUs, efficiently utilizes each GPU, and scales to more than one machine. It might be a while before I want to try multi-machine training, but it would be awesome to have code that's ready to do that without needing any extra work. Now, how to implement it? Hugging Face have a library called Accelerate , which does everything for you: Accelerate is a library that enables the same PyTorch code to be run across any distributed configuration by adding just four lines of code! That does sound very useful, but I worry that by using it I won't learn as much. It also rather ties you in to the HF ecosystem. That's not necessarily a bad thing -- I enjoyed using their stuff in my fine-tuning project -- but I'm trying for a somewhat lower-level view in this series. So, let's use the PyTorch-native stuff. There's a "getting started" tutorial , so we can follow that. It has two options for running using DDP, one with a bit of extra setup code -- the first example, under "Basic Use Case" -- and one that uses to make things easier. The second sounds best. The code changes actually look really simple; given a normal single-GPU training script, you need to do some setup at the start: ...then wrap the model itself in a object, which is what you actually do the train on: ...and a bit of teardown at the end: The way to look at this is that will spin off one process per GPU, each running exactly the same code. They have a "rank", which is an integer saying which of the per-GPU processes they are -- 0 for GPU 0, 1 for GPU 1, and so on. There's a bit of a gotcha here, though -- you can see that we're looking at an environment variable called at the start, but we then get a (non-"local") variable from a bit later on. This is due to the multi-machine possibilities with DDP -- if you have multiple machines, then the local rank will be "which GPU on the machine does this process relate to", but there will also be a "global" rank, which is unique across all machines. This distinction won't matter that much during this one-machine test, but it's worth keeping in mind if we want to keep the code in a shape where it could potentially scale to multiple machines. Anyway, after the processes are spun up, they will do their training, and the synchronisation and passing around of gradients during the backward pass will all happen invisibly in the background, so when we do our , it will have the full set of gradients. Now that means that we'll presumably also need to use the rank -- that is, which of the n per-GPU processes the current code is running in -- when selecting which dataset items to train on. More about that later. Let's start writing some code! I'll use a new repo , into which I can put just the code needed for this train. I'll also structure it a little better than last time, with separate "runs", each of which has a model config and training parameters, and will later on have its own checkpoints. You can think of these as being one per machine size that I'm trying out -- I'll create a run directory for each one. Here's a first cut , simply loading up a model config from a run's directory, using it to create the model, and then doing the wrapping above -- no training at all. Running it with (and , as I'm using that for all new projects): Promising. Now, unfortunately we only have one GPU locally, and the code assumes that it's one process per GPU (I believe that's a hard limitation for PyTorch's DDP), so running with blows up. So we can't do an in-depth test locally. But at least we know that the basic infra is there and working. Now let's move the other training code from the single-GPU script into that file, pretty much blindly. This is the result -- it's doing almost nothing beyond what the last train did, apart from wrapping the model in a object -- the only other changes are to use this "runs" directory that we've introduced. As a quick hack, we should try running it. It does a validation and checkpoint before it starts, and we can make that happen quickly by hacking the validation loop to only do a couple of iterations: (Foreshadowing: that hack will come back to haunt us later!) Running that, then hitting control-C after the validation completes, and it looks OK: ...and we have what look like solid checkpoints: However, loading one of those checkpoints fails: It turns out that the problem is this code when we save it: The that we're saving is the wrapper around our model; my guess is that it does actually include all of the weights for the model, hence the correct-looking size for the checkpoint file, but they're renamed -- the wrapper sees the underlying model as something called , so (for example) would be called . Fixing that, with this diff: ...sorts it out -- we can load our checkpoints again. Here's the updated file . I think we're going to have to revisit checkpointing and validation again; we don't want to do it in all of our processes, probably only on global rank 0, and we'll need to somehow synchronise everything so that the other processes don't carry on training while we're doing it. But before we get on to that, there are a couple of other things to change. At the top of the file we're defining some constants that look wrong: We'll handle the dumbest of these first; it was actually silly that in the old code we had a constant for sequence length. We're using the context length of the model for that, so it's duplicated information. Let's get it from the : ...and here's the updated file . That was nice and simple. The code that we have specifies the batch size for each GPU -- that is, with , we'll have six sequences in each batch on each one. Like I mentioned earlier, that's called a "micro-batch" in distributed training like this 1 -- a per-GPU batch, as opposed to the overall global size across all GPUs -- so we could just rename it, and then we'd have 6 × n gpus as a global batch size. However, it feels to me like this is a useful metaparameter to be able to tweak from outside the code. I can see machines with per-GPU VRAM varying from 40 GiB to 160 GiB on Lambda Labs, and pretty clearly that will mean there will be a varying largest micro-batch size on each type. So this is something we'll want to configure on a per-run basis, so let's add a new file to our run config, load that up, and pass it through. That's a simple enough fix; no need to note the diff, but here's the code . This one we'll need to think about. The size of our validation set is based on what one process running on my local RTX 3090 can validate in five minutes, and the interval (for which I fairly arbitrarily put 2000 in the code when copying it across) was calibrated for roughly every half-hour. Those numbers in turn were aimed at the 44 hours of training time I expected locally. For this train, we'll (hopefully!) be taking significantly less time. We'll have eight GPUs, so naively that's 5.5 hours of train time, and each will have more VRAM, so we should be able to bump up the batch size and potentially get even faster than that. Depending on which kind of cards we're using, they may be faster, too -- I found that an A100 is slower (with the same batch size) than the RTX 3090 in my fine-tuning experiments, but the H100 and B200 are likely faster. I think this is another thing for the train config; we should have the validation interval (in terms of iterations) and the number of batches to do for validation. Here's the updated code . Now, let's move on to the dataset. With the code as it is right now, all of our per-GPU processes are using this code to iterate over the same dataset: That means that they'll all be training on the same data; the synchronisation that is happening "magically" in the background means that they'll all train on the first item, work out gradients, and step their optimiser -- so they'll essentially (modulo randomness) have the same updates. Pretty pointless! What we want is for each of the n per-GPU processes to train on 1 / n of the data. We have two useful helpers in : , which gets the global rank of this process. In our one-machine case, it returns 0 for the process on , 1 for the one on , and so on. We're already using it in that setup code we looked at earlier: , which tells us how many GPU processes there are (globally -- it would be across all machines if we had more than one) So, the simplest thing to do is to use the world size as a step, and the rank as an offset: Here's the code with that . Now, remember that the same code is running for every one of our per-GPU processes. That means that all of them will do the training with forward and backward passes, and their own optimiser steps, all synchronised by PyTorch DDP magic. But they will also do their own validations -- which is kind of pointless -- and they'll also try to save their own checkpoints, which would be messy because they could quite easily interfere with each other; after all, all of the processes are running on the same machine and would be writing to the same filesystem. So, as a first cut, let's just wrap an around the eval and checkpointing stuff -- we change this: ...to this: That line is getting bit long, so let's break it apart a bit: That looks OK, but there's an extra wrinkle: all of the processes are running the same code, so while the rank zero one will do the eval, the others will continue through the script, so they will go right back around our loop and start training on the next batches -- which is bad. We want our processes to be proceeding in lockstep, iteration-by-iteration. Luckily, the solution is simple: the function in basically says "stop here until all of our processes have reached this point". So we can use two of those -- one before the eval loop, to make sure that all of the processes have finished their training part of the iteration before we do the eval on rank zero, and one after the eval, so that the non-rank-zero processes will wait. One bit of complexity -- we want to do those barriers only if it's a eval iteration, but we want to do them for all processes. So we have to break up the statement, and we wind up with this: That seems to work OK ( code here ), but it does give a warning: So, we want to pass the device ID in when we call . Let's dig into that a bit. Here's the copypasta that I took from the PyTorch tutorial earlier in this post: Let's dig into what that is doing. The environment variable is being set by to 0, 1, 2, etc as appropriate to tell us which process we are on this machine. So the first line is telling PyTorch to use the device with that index for this process . The next line is getting the current accelerator -- that is, an object that represents which acceleration hardware we're using in this process. I think that the best way to see the combination of these two lines is that the first says "use " (or 1, or 2, or...), and then the second says "get the object describing the GPU you're using right now". So it's a slightly indirect way of getting the object containing the details of the GPU in question. Next, we call . A backend in this context is an abstraction of whatever system the device in question is programmed using -- in the case of an Nvidia GPU, it would be some kind of thing that encapsulates CUDA. Once that's done, we call , passing in the backend that we're using. We're saying "initialise the internal data structures for so that they're all set up properly to work with the backend we specified". After that, we can do stuff like getting the global rank with and so on, because has been properly initialized. Presumably at this point we're talking to any other machines in a multi-machine cluster, so we can find out what our world size is and that kind of thing. That extra line at the end, to get the : ...actually looks erroneous to me. All of our code is assuming one process per GPU. So I think we can just use the there as well. Let's rewrite it like this (with some useful comments): That seems to work well! Here's the code . However, I ran it past ChatGPT (largely to validate my understanding of what was going on), and it highlighted something slightly misleading about it. Right now, we're training on a single node, with one process per GPU. But again, one of the neat-o things about this DDP stuff is that it should be able to scale to multiple nodes. Now, remember that is just the rank of the current process on the specific node that it's running on -- hence the name. If we had two machines, each with 8 GPUs, then there would be a process with rank zero on each of them. The "real" rank -- that is, across all machines -- is the one that you can get from once it has been initialised. One of the things it does during that initialisation is to talk to all of the other nodes and work that kind of thing out -- which of the local rank zero processes across all of the machines is the global rank zero process. So we need to use the local rank when working out which GPU we should be running on and so on, but we should not treat it as a global rank. That's actually quite fine in this case, as we're calling inside the training loop when we actually need to use the global one (when indexing into the dataset, or when deciding if we're the process that should be doing evals and checkpoints). The only place where we might be confusing matters is in that print, which is not important anyway, as the training loop also prints out its rank. So, let's tweak it a little more for clarity: That seems to work well! Here's the code . Time to run it past ChatGPT to see if I've made any dumb errors. Turns out that (unsurprisingly) I have... Let's go back to our code that decides whether or not it's an iteration where we need to do a validation run and a checkpoint: The problem is that our index is different in the different processes! Remember, we have this in order to pick out the correct training items: So let's think about it; in the first run through the loop, with 8 GPUs, we would have In the next run through the loop, we'd have: So will give different results for each process. That might not sound like the end of the world -- will only be zero for one of them, so long as is larger than the number of GPUs -- but remember that our validation code looks like this: Now, if different processes have different values for , then will only be called in the one(s) for which it is . But means "wait until all processes have reached this barrier". So the ones that call it will lock up completely until other processes get there, and everything will at best get out-of-sync, and at worst will lock up completely. I think that the problem here is that I'm conflating two things: the index of the global step -- that is, one iteration across all GPUs -- and the dataset element that we want to use. In the original one-GPU case that made, sense; iteration 0 was on dataset element 0, iteration 1 was on element 1, and so on. But now the offset into the dataset, and the global step, are quite different things. This is quite deeply embedded in the code, but we can fix it! Let's start off by changing our checkpoint code, just to rename things. It keeps track of a variable called , our offset into the training dataset, and uses that both to index into the dataset, and to work out how far through the train we are. The latter is a much better thing to store in a checkpoint, so instead of saving , we'll store (and restore) . Basically, just a rename so that the variables and stored JSON match the new reality. Here's the updated code . Now we need to make a number of minor changes to the training loop just to match that rename of the value that we're checkpointing (eg. for the code to generate the training chart) but the most important change is to our loop. Instead of iterating over our dataset with a step and and offset so that we can index into it, we firstly work out how many global steps there will be: ...then we iterate from our initial global step -- zero if we're starting a fresh train, or whatever global step we were on in a loaded checkpoint plus one if we're doing a continued train from a checkpoint -- up to the : That means that we need to use the global step, the world size, and our current rank to work out which dataset item we should be training on for this process at this global step. Let's say that we have eight processes; on the 0th global step, we should have rank 0 training on dataset item 0, rank 1 on item 1, and so on. On the next global step, rank 0 should train on item 8, rank 1 on 9, and so on. So: That's actually much more elegant than the earlier code, and seems to work fine. Here it is . Phew, glad to have caught that before I started spending money on machines -- it would have been confusing if everything locked up. Thanks, ChatGPT! Another thing that raised by ChatGPT is about the validation. We don't want to validate across all of the validation dataset -- we're using a number from the . I have this code: This looked like a nice, quick way to get the first elements of the validation dataset. But ChatGPT told me it would raise. It didn't, though -- why? The problem is that I had set to in my training config for testing. Stepping through what that slice does, when we run : Python calls the on the dataset, passing in a object as , so this code is called with it: Now, because that code doesn't do anything clever with s, they're passed straight down to the tensors that make up and . So it's actually equivalent to this: Or, to rewrite the whole loop (omitting the for clarity): So, the first time through the loop, we try to bind our loop variables like this: That is clearly wrong! It's equivalent to this: ...with code to blow up if has more than two elements -- the normal Python "ValueError: too many values to unpack" Nasty! AI code review certainly helped me dodge a bullet on that one. Let's fix it, it's not a big change: we can just do this: ...and that works! So here's the code now . So, I think we have one final issue, which is the training and validation datasets. In our single-GPU train, we worked out ahead of time how much of FineWeb (or FineWeb-Edu) to train on -- the Chinchilla-optimal number -- and generated a dataset that contained a round number of 6-sequence, 1024-token batches that was the smallest such round number that was larger than our target. We also worked out exactly how large (in terms of batches) our validation dataset needed to be so that each validation run would take five minutes. There was one big issue with that system; when I decided to do an "extended" train on more of the FineWeb-Edu dataset, in order to see whether I could get the loss down further, I had to do some nasty hackery in order to generate a new one. So it would be nice to not have that problem this time around. Additionally, we're likely to be tweaking the batch size quite a lot in this experiment while we find what the appropriate level is to fit onto the cloud GPUs, and also varying how much validation we do -- and additionally, we have the world size to worry about. I think that the best way to give us the flexibility we need will be to pre-convert the complete FineWeb and FineWeb-Edu datasets into the format we need -- each sequence in the dataset converted to GPT-2 tokens, and then those sequences concatenated together, with the token 50257 separating them. It would be good to properly nail down the validation dataset at the same time. So we can have a script that loads up the original dataset as downloaded from Hugging Face, splits it into 99% train, 1% validation, does the conversion, and then saves them as safetensors files. If we use for those (which is just large enough for our 50,257-token vocab), we can fit the ~10B tokens in each dataset's train split into 20 GiB of disk. Not too bad. But there will still be the issue of getting them onto our cloud machines. Let's generate the data, and then work out how to handle that. I tried initially with the code I used last time, adapted to run through the entire dataset . It does the 99%/1% train/validation split, and then for each of those generates a single massive tensor of tokens like this: It almost worked! To my surprise, it got all the way to the end, and only blew up with an out-of-memory error when it was trying to save the result -- and it did that completely silently, so I thought it had worked right up until I tried to check the file on disk to see how large it was, and it wasn't there. The obvious tweak: set the list to just after the , to free up the memory it's using. Given that it was the save that triggered the OOM, you'd think that that would be enough -- but it turned out not to be so. Rather than mess around with this for much longer, I just decided to add on 128 GiB of swap to my machine temporarily: ...and that was enough to make it run. So I've now generated pre-tokenised, pre-concatenated train and validation sets for both FineWeb and FineWeb-Edu: Now, thinking about how to get it up to the Lambda Labs machines. I have normal 1 Gb residential broadband, so conceivably I could upload 20 GiB in about 200 seconds. But that's assuming that there's no network congestion, so I would expect it to take longer. The LL machines are quite expensive, and I don't want to waste money keeping them up while I'm just uploading data. There are possibilities here: I think the best option is to use option (1), but with the option of also doing (2). The HF dataset will still take time to download to LL, even over the faster network connection. That might not be a problem -- but if it is, I download it once on a cheap instance and use a persistent disk too. Essentially I'd be using the persistent disk as a "cache", and still get the benefits of the easily-shareable datasets on Hugging Face. So, that decided, let's find out how we can upload a whacking great 20 GiB safetensors file as a dataset on Hugging Face. It turns out that resources like datasets on HF are just Git repositories using the LFS (Large File System) plugin to be able to handle, well, large files. Conveniently, given that I'm using to manage my project, there's a plugin that allows me to use their CLI tools with minimal effort, so: Both datasets show up on my profile page on Hugging Face, so that's looking good. Now it's time to try to upload the data. We'll need to install Git's LFS support first: Now let's try the FineWeb one first: OK, so we need some kind of extra thing to tell it we can use large files on top of the LFS stuff: Right, now let's try again: Weird that it prompted for the credentials twice, but it did appear to try to do something there -- but obviously it didn't work. Let's see if Git over SSH is any better. ...then the same stuff to copy in the files and create the metadata file, then: Looks like the same error. Odd. Let's try using HF's upload tools rather than Git -- feels like a bit of a cop-out, but maybe it'll work better. That did indeed take about 200 seconds to run, but the upload speed was only about 10 MiB/s -- from the output, I think it must have been compressing it. Anyway, it looks like it succeeded, so let's upload the others! ...and that's done :-) Next, a bit of manual editing of the dataset cards on the Hugging Face website, and we have our two new public datasets: That looks solid. So, the next thing: change our codebase so that we have some quick and easy way to download them (I'm feeling a little wary of using Git for that after the upload issue), and then to use the downloaded files in our training code. We already have the code to download a dataset; the stuff that I wrote to download FineWeb and FineWeb-Edu originally. Here's the important bit: ...so we can adapt that to download all files in an arbitrary dataset: ...and call that from our , using a new command-line argument , and a new element in our train config JSON file: I was thinking that we'd need extra guard code to not download the dataset again if it's already there, but it looks like handles that all nicely for us. So we have a way to specify which dataset we should use for a training run, and code to download it. Now we just need to adjust the code that loads our datasets so that instead of looking in the , it looks in the directory returned by : ...and update the directory so that if just blindly uses the directory provided rather than trying to look in a subdirectory: That all works! We successfully download the datasets and try to use them. Here's the code . But now we have a problem; when the tries to reshape the huge tensor that we have as our inputs: ...it craps out: That makes perfect sense. Our original files were carefully sized for a batch size of six, and 1024-token sequences. We need some way to work out an appropriate slice of both the training and the validation data. Most of the trains are likely to be Chinchilla-optimal, or at least use a Chinchilla-optimal number of tokens -- rounded up appropriately to match our micro-batch size, sequence length, and world size. But I'd like it to be more configurable. What I'll do is add a key to the training config dictionary, along with a so that we can (for example) train on the first Chinchilla-optimal tokens, then do an extended train continuing on from there. The idea is that we can use as a base, and train on the smallest number of full batches that contains at least that many tokens. For validation, I think that the key that we already have is actually quite nice. Validation is time-bound, and the number of batches is the easiest lever to pull to handle that. However, a would be nice for symmetry. So, here are some numbers for debugging: Now let's use them. Initially, we have this to load the train dataset: Let's work through that one first then make appropriate changes to the validation one. The pieces of information we need to work out which tokens to use are: Let's update our function so that it takes those parameters in that order: ...and now we can write an updated that uses those numbers to get the right number of tokens: Validation is less obvious; I think that the best way to do this (given that the validation dataset is small) is just to have a "magic" value for , which means "just get a round number of full batches starting at . It's also worth remembering that we only do evals on the rank 0 process, so we could in theory pass in a world size of 1 -- but I think that passing in the real world size might be a good idea, because it gives us one fewer thing to change if, in the future, we move towards distributed evals. ...and we change to be able to handle the magic : I also added in a quick sanity check to make sure that we don't get weird behaviour if the is past the end of the original dataset. That all looks good! Running it kicks off training, and validation is running happily every ten global steps, but just with three samples, as configured in the JSON file. Here's the code . One thing that hasn't shown up while running this code locally is that our training loop has this: With one GPU, that's fine, but on a multi-GPU machine, that is going to happen in all of our per-GPU processes -- so they'll all be spamming out progress bars, which will be ugly. So, as a first cut: Now, in order to compare different machines (say, an 8x H100 vs an 8x A100) it would be nice to get tokens-per-second numbers while training. We can do that in the progress bar too! It has a method that adds stuff to the end of the bar, just after the elapsed time and iterations/second numbers. For that, we'll need to have the object available in a variable: ...and now we can count the total tokens seen in the training run, plus keep track of the start time -- just before the start of the training loop: ...then inside, after the training step: That will give us a running average of tokens per second over the train as a whole since the start. Running that, we get a nice progress bar like this (you'll need to scroll to the right): Note that the tokens per second is worse than the just less than 20k that we got when running the single-GPU test previously, but that's due to the testing setup I have -- I'm doing an eval every 10 global steps. Changing that to 1,000,000 so that we just get a single eval when we start, then letting it run for a while to settle down from the initial eval, we get this: ...which is close enough to what we had before. Finally, let's print out some summary information at the end: Ran that on a super-short train with about 50 iterations-worth of tokens, and: Looking good. Here's the code . I think we now have something where it's worth spinning up a Lambda Labs machine to run it. Let's kick off a training run on the cheapest two-GPU machine that they have available right now. That's actually not all that cheap, it's a $6.38/hour 2x H100 80 GiB SXM5. But I'm not planning to do a full train on it yet, this is just a sanity test. I won't attach a filesystem this time, either -- let's see how things go without the caching of the datasets that I was considering. First thing: do we have ? Nope. OK, let's install it: Right, now let's clone our repo and set up our environment: And now I think we can just try running it! It took 18 seconds to download the dataset! I don't think we need to worry about the caching thing with persistent disks, at least at this point. But there are a couple of issues here. I didn't put the number of processes in the command line -- I should be using Also, we don't have the XKCD font family. I'll ignore that for now. OK, that's looking good! Let's make our validations happen less often, and see how high we can get the micro-batches with the 80 GiB VRAM we have on each of our two GPUs. Doing a binary chop, I set the micro-batch size to 100 (OOM), then to 50 (OOM), then to 25 (worked), then to 37 (OOM), then 31 (OOM), then 28 (worked), and finally 29 (OOM). So we have a batch size of 28 for our 80 GiB machines. Leaving it for a little while to settle down, and we get to about 142,000 tokens/second. Now, on the 3090, we were training at 20,000 tokens/second. That means that this machine is running at about 7 times the speed. Given that our original train finished in 48 hours, we'd expect the train to finish in about 6, which indeed is the estimated time on the tqdm progress bar. At $6.38 per hour, that comes to $38.28. Not bad! And this instance is actually quite pricey on a per-GPU basis -- it's $3.19 per GPU/hour, whereas there is an 8x H100 that costs $2.99 per GPU/hour. I'm almost tempted to let it run. But the purpose of this run was to work out the bugs. We're going to want to track the training chart -- remember that after every validation run, our training code generates a chart showing the training and validation loss so far, like this one . I ran the normal quick-and-dirty Python webserver command on the instance, inside the directory containing the training chart: My browser didn't connect to it, but looking at the Lambda Labs interface, there's a new "Firewall" section, where you configure rules for allowing incoming connections to your instances. That's sensible, and the default rules are just "allow SSH from any IP" and "allow ping from any IP". Adding one letting anyone access port 8000 fixed the problem, and I saw a directory listing; clicking on the chart showed exactly what I'd expect, but without the XKCD fonts. Nice. Let's work out how to fix that XKCD font thing. Looking around, it seems like there are approximately twenty thousand ways to do it. Here's one that seems to work; firstly, install the font on the system: Now, that installs a font that has the family name 'xkcd Script` (with that erratic capitalisation). So we need to change the code to pick up pretty much anything that looks like it's XKCD, so instead of this: ...we can do this: That seems to work OK. So, now, I think we have the beginnings of a script to set up a Lambda Labs machine so that we can use it. Let's write a with this: ...and give it another go on a fresh machine. Shut this one down -- total cost so far $7.28. Now there are no 2-GPU instances available. There is a super-cheap 1x A10 (basically the datacenter version of a 3090), though, so let's use that -- we're as certain as we can be that the multi-GPU stuff works, and the proof of the pudding will be whether we can train a model that works. After spinning up our 1x A10 machine: Looking good! I think we have something that (in theory) should work. That cost $0.05. I think it's time to do our first train on a big instance. There are four 8x instances available on Lambda Labs for me right now: I think I'm going to want to train on all of those, to try to work out some kind of metric (dollars per megatoken?) to compare them. But let's start with something reasonably low-end -- in fact, let's try the cheapest, and see what happens. Spin one up, and first thing; after the setup, we need to work out the micro-batch size. Last time we used 28, but this machine has GPUs with half as much VRAM. I did a binary chop again... it turns out to be 13. Now let's think about validation frequency. Let's try to get a feel for how long it will take. We can set the eval batches to (say) 100, so that we can see how fast evals are, but also set the interval to 10,000,000 so that it never does one after the first. It took 11 seconds to run 100 validation batches, and after a few minutes, it settles down at 254,000 tokens/second or so, and is estimating 3h15m to completion. Nice! The cards are an earlier generation to the H100s we used in the two-GPU test, so they're slower, and they have half the VRAM. So eight of them are, working together, about twice as fast as two H100s. Doesn't sound completely crazy. So, in our local train, we spent 5 minutes evaluating every 30 minutes. So our eval time was 16% of our train time. Probably a bit high, but let's run with it. If we're going to take 3 hours training time, then 16% of that is about 28 minutes. Previously we did about 88 evals (44 hours train time, with an eval after each half hour). That seems a bit too high. So let's say that we want to do 50 evals. 28 minutes eval time in total, with 50 of them, means about 30 seconds per eval. If 100 eval batches take 11 seconds, let's approximate it to 300 eval batches. As to the interval between them -- if we want to do 50 over 3h15m, or 195 minutes, then that's one every (let's approximate) 4 minutes. We seem to have settled down to 2.57 iterations per second, so that's about every 617 iterations. Let's bake those in and let it rip. After the run: OK, let's download everything. Looking at the checkpoints, the latest (that is, the last one at the end of the training) and best (the checkpoint that had the lowest validation loss) are the same one, meaning that validation loss kept falling consistently: So let's just download using the "best" symlink to get the weights for that checkpoint: And now we can shut the cloud machine down. Now that the clock is no longer ticking and we aren't spending money on an unused machine, here's the training chart: It looks like we had a couple of gradient spikes there. I'm going to add some gradient clipping code at some point, but I think I'll hold off for a little bit -- I want to do a few cloud trains first to work out the best instance sizes to use, and only then start exploring the possibilities for making the models better. Apart from that, it looks pretty normal. Looking at the billing page on Lambda Labs, that machine was up for about 4 hours and 35 minutes, costing US$10.32 per hour, for a total cost of US$47.35. Of that 4h35m, 13,904 seconds, or 3h52 was the actual training run -- somewhat more than the 3h15m that was predicted at the start of the run. The validation will have accounted for most of that -- we did 50 evals, at 30 seconds each, so that's 25 minutes. That means that 3h40m is accounted for, and the remainder can just be chalked up to noise, I guess. That leads to one question: do we actually need to be doing validation for these trains? I've been doing validation loops in these trains largely out of habit -- when you're training an ML model, it's just "what you do". The reason you'd normally hold out a validation set is simple: if you're training over multiple epochs, then eventually your model is going to start overfitting to the training data 2 . You validate as you go along so that you can spot any points where, while the training loss continues to drop, the validation loss -- which is loss on data that the model hasn't been trained on -- starts rising. That's the classic indicator of overfitting. But for these models we're not doing multiple epochs -- we're just training through a stream of constantly new tokens. So, in fact, there's no real difference between the training data and the validation data, apart from the fact that the validation data is constant. From the model's perspective, it's all new stuff (modulo any repetitions in the dataset, which is possible but I think not likely to be super-common in something as curated as FineWeb). Now, in this post I'm aiming to identify the best options for training in the cloud -- cost in terms of dollars and time. I don't want to change the model itself or the training strategy because I want whatever I come up with to be roughly equivalent to the models I trained on my own machine. Exploring enhancements is for the next post. (Of course, given that the batch size is one of the levers I want to experiment with, and training on larger machines is already meaning that I'm doing micro-batches larger than the batch size of 6 that I used locally, and then the overall batches are 8 times larger, that's not quite true.) Validation, however, doesn't actually affect the training runs in any direct way. I could in theory remove it. However, that is a relatively large change to the code, as I've kind of linked it in with my checkpointing code. I think that what I'll do for now is leave it in. Validation will scale at the same rate as training (so long as I leave the eval batches constant) so it leaving it there will give me a clean comparison between machine types. And I can keep notes on how much time was spent on validation for each train so that I can subtract it from the total time if that proves useful. However, when I start tweaking the training code with changes beyond the batch size, I should probably try removing validation first. Anyway, while validation during the training run might not be important, evaluating the model at the end and seeing how it compares to others is! Let's do that next. There were two important post-train evals that I did on the models that I trained locally: There was also a simple smoke test -- how does the model predict that the phrase ...should continue? I should do the same three tests here. A simple autoregressive generation script is easy enough to knock together, and: All we're looking for here is basic coherency, and I think this is good enough to pass that filter. Next, the loss-style testing. What I think I want to be able to do here is just take a file and run an eval against a standard dataset. I did not generate my own test set, but I did generate a much-larger-than-necessary eval set, 1% of both FineWeb and FineWeb-Edu -- that's 100 million tokens or so in both cases. In the validation that I was doing during the train just now, I did 300 batches of 1,024 tokens with a micro-batch size of 13. That only ran on the rank 0 process, so that's Not even 4% of the validation data. Now, for the local eval, I think it makes sense to make it run for about five minutes -- that's just for my own convenience, I don't want to spend very long -- and I know from the previous local train that I can do 3,200 batches of six 1,024-token sequences in that time: So, somewhat arbitrarily, let's use the 19,660,800 tokens starting at position 50,000,000 in the FineWeb validation dataset for our tests -- they'll never be used for training or validation during the training loop. It's kind of a hack, but it'll do for now. Here's the code . It should be easy enough to understand; it did require one tweak to our existing function, though: Originally, that function worked out out the actual number of tokens to use by working out the size of each global batch, dividing our requested minimum number of tokens by that size and taking the floor, adding on one, then multiplying that by the global batch size. That works fine in cases where the is not a multiple of the global batch size -- it gives us a round number of batches that contains at least . But if is already a multiple of the global batch size, it gives us an extra batch at the end. So I added that as a special case in to avoid that. Anyway, running that gives us a loss: That's actually quite a lot lower than we were seeing with the locally-trained models on the test dataset I was using then -- but, of course, it's a different dataset so it's not strictly comparable. Let's run the same test against them: That's really interesting! Those numbers are really close to the numbers I got in the last post. That does make some kind of sense, though -- while the numbers aren't strictly comparable, as I said, both the dataset that I was using then and the one I'm using now are essentially random stuff from FineWeb, so I guess they must be more similar than I thought. But, importantly, the loss on the newly-trained model is much lower -- 3.674 rather than > 3.9 for all three of the older locally-trained models. Now, the only big difference between this training run and the ones that I did locally is the batch size. As I said in the last post, while I felt that the difference between my batch size of six and the (reported) batch size of 512 for the original GPT-2 was the least-likely cause of the differences in the results, Gemini told me that it thought it was the most likely cause. It looks like Gemini (and, I should note, on Hacker News ) might have been right! Batch size is super-important. Let's do the same eval with the OpenAI weights. I wrote a quick script (in my old 'LLM from scratch' repo, which has the code used in the book) to load up the GPT-2 weights and save them as a safetensors file . When I ran that, I got an interesting error: That was easy enough to fix; in the book's code we assign the weights that have been loaded from the OpenAI TensorFlow checkpoint files with a function called that looks like this: Just adding a call to to the last line fixed the error: ...and as a result, I had safetensors files for the original OpenAI models: So now we can run our test against them: Excellent. Let's start putting together a table of these results: That's pretty amazing. Having a batch size of 13 micro-batches over eight GPUs, or 104 in total, seems to have massively improved the model -- it's much closer to the original weights. It will be interesting to see whether I get further improvements when I move to the larger machines, which (due to having more VRAM) will have larger possible micro-batches, so we'll get larger global batch sizes. It certainly makes me think that I could have got much better results locally by using gradient accumulation, which would mimic the effects of a larger batch size by running multiple smaller batches through, without doing an optimiser step each time, then doing one big update once enough has gone through. But all of that is for another day. Let's try the instruction fine-tuning test now. I decided to pretty much re-use my adapted version of the code from the book; that meant that I was borrowing quite a lot of Raschka's code, which he has released under the Apache 2 license . I normally use the MIT license for my code, but I'm not married to it, so I relicensed the whole repo as Apache 2 with some specific headers to say which parts came from "Build a Large Language Model (from Scratch)", and added this code . It downloads the Alpaca dataset from the site for the book, splits it into train/validation/test splits, trains on the training set, evaluating each epoch and bailing out (and restoring the previous epoch's weights) when validation loss starts rising, and then runs through the test set generating responses, and then sends them all off to the OpenAI API for GPT-5.1 to judge them. Running it against our new model gets a score of 17.09. Let's try the various other models and build out our table: Interesting! In the last run, I found the instruction fine-tune numbers came out as FineWeb-Edu extended > FineWeb > FineWeb-Edu, but here we have FineWeb-Edu > FineWeb > FineWeb-Edu extended -- exactly the opposite! I do have to wonder, though, how precise a measure this is. While the training should be fairly consistent (though I don't have a random seed in there to enforce it), the fact that we're using an LLM as a judge means that there is an element of randomness coming in here. Indeed, I re-ran the FineWeb-Edu extended train test again, just to see what I got, and it came up with an even-worse 12.12. So I don't think we can read a huge amount into these numbers -- well, unless we can get the numbers significantly up. While it looks like a 2.5-point difference might just be randomness, I doubt that a 10-point difference could be. I think we've done the tests that we need for this model now, and we have a testing procedure in place. So let's train some further models on different instance sizes, and gather numbers. This is the biggest machine available on Lambda Labs right now, and is only sporadically available; one happens to be there now, so let's to give it a go. First, we need to create the runs/8xb200m160 directory, initially with a that is a clone of the one I did for the last train, , then spin up the machine. As before, we need to log in, clone the repo, then in it run the script, run , and try to run the script: It crapped out because there was no datasets directory, which is an annoyance. We should create it if it doesn't exist. Create the directory, and run it again. It took a while to download the dataset, because every per-GPU process downloads it separately. That only took a minute or two, but it was a waste of time; I think we should only download it from the rank 0 process with some barriers to make the other processes pause. Next, we need to do a binary chop on the micro-batch size, starting with a low of 13 (which I know will be fine because it worked on the 40 GiB GPUs that we used last time), and a high of 100 (fairly random, just something I'm pretty sure will fail). While doing that, a few things are standing out, both to do with validation. When the script starts, it does one training iteration, then goes straight into validation. Then it starts the training run proper. However: We're going to need to work out some kind of fix for that, because it's taken me 17 minutes from spinning up the machine to getting a size for our micro-batches -- which happens to be 64. On a machine that costs US$39.92/hour, that's an expensive test! We'll look into that later. Anyway, a batch size of 64 is pretty neat, as with 8 GPUs, that means we have a global batch size of 512 -- exactly the same as in the original GPT-2 paper! So, let's kick off the train. It takes about 7 minutes to get to the first checkpoint, at which point it's averaging 801,221 tokens/second. That pattern repeats, and with about one minute to do the validation, we're spending about 12.5% of the time on this machine validating. Hmm. A further indication that we might want to remove the validation stuff if it's not adding on any value. Eventually, it finishes: So, that's 1h9m50s. The final validation loss is not as good as the previous run on the 8x A100 40 GiB machine, where we got down to 3.675. Given that we're using the same validation dataset as the previous, that's meaningful: this is not as good a model, it seems. Again, latest and best checkpoints are the same one: So we can download everything: ...and here's the training chart: OK, so that's smoother than the last one -- no loss spikes. Maybe the larger batch size smoothed them? Let's think a bit about the cost of this train. From Lambda Labs, we had that machine running for a little over 1h30m. At US$39.92/hour, the total cost was US$60.25. Yikes. So, knocking off the 1h10 or so for the train, we have 20m to allow for -- which matches up quite well to the 17 minutes of fiddling with batch sizes, and then 3 minutes to download all of the files. If this blog post isn't going to cost significantly more than it needs to, we need to get that down. Of the US$60.25, just over US$13 was spent on identifying the batch size. Only US$46.57 was spent on the train itself. We also did 11 validation runs as part of that; at a minute each, those cost US$7.32. So, excluding validation, we're below US$40 for the train. Now, let's run our tests. First, the smoke test: we get this: "...on all other website for..." is a bit rubbish. Still, on to the loss: That's in line with the training loss -- worse than the loss I got with the one trained on the smaller machine, with its corresponding smaller batch size, but still better than any of our local trains. Still interesting, though -- larger batches are not guaranteed to get bigger results. More investigation needed there! On to the instruction fine-tuning test. That gives us a score of 13.89 -- the worst that we've seen yet! I think I'll put together a full table including these results later; I want to try training on some other, differently sized machines first, and we can aggregate the results at the end. But before we do that, let's make some changes to the scripts to fix some of those QoL issues we encountered in that last train. The first irritation was that it errored out saying that was not a directory when it didn't exist. The script takes a datasets directory as one of its command-line options, and it's reasonable that it checks that it really is a directory (rather than, say, a file or a symlink): ...but if it doesn't exist, it might as well create it first. Now, I could just put this before the check: ...but remember, this code is run by multiple processes -- so they could easily trip over a race condition here. What I want is to have just one of them do this; I've deemed the rank 0 process the "special" one for validation, printing the progress bar, and so on, so we may as well treat it that way here. But -- there's a difference! Rank zero is the one that should be printing stuff out, it's true. And right now, we only have one node participating in this train. But I do want to avoid simple errors that would make it hard to run multi-node in the future. Now, if we have multiple nodes, then each one will have its own filesytem (unless we're using NFS or something like that), so we'll need a separate "datasets" directory for all of them. What we want is to do these checks on one process on each node. Usefully, we have the variable that is defined earlier in , which is per-node. Again, let's imagine we have two nodes with two GPUs each. Node 0 might be runnning the processes with global rank 0 and 1, and node 1 might have global ranks 2 and 3. On node 0, the processes would have local ranks 0 and 1 respectively, but on node 1, they'd also be local ranks 0 and 1. So, the full code becomes this: Note the barrier; we don't want the other processes to check whether is a directory until the local rank 0 process has had a chance to create it. (Of course, if we were running this on a setup where all of the nodes shared a filesystem, it wouldn't work -- in that case we'd want to use the global rank that we can get from instead. But we can burn that bridge if we ever come to it ;-) Phew, that was a bit more work than I expected! But it sets us up nicely for the next QoL fix on my to-do list. I don't like the fact that every process downloaded the whole dataset. The actually handled it pretty gracefully -- none of the processes tripped over any of the others. Indeed, it looks like there was some kind of global queueing going on, so they downloaded it one after the other. But it did take time -- maybe a minute or two in total, and with the clock ticking on that ~US$40/hour machine, that felt a bit stress-inducing. So: I think it would be best to only do that from the rank 0 process as well. The code that downloads the dataset is just after the bit we've been looking at: ...and looks like this: Now, the docs for say that the parameter is: If provided, the downloaded files will be placed under this directory. ...and the return value is this: We happen to be passing in a object for , and we're not in mode -- it defaults to . So all we're doing by returning that wrapped in a object is a slightly indirect way of returning the path that we're passing in as . For tidiness, I really want to gate the call to in with the same rank stuff as we did for the directory creation. So, let's change the setup so that takes the path to the directory where we want this specific dataset to be, not the generic "all datasets" directory. And given that we're now passing this specific path into the function, we don't need to return it: Now it's just a wrapper around a single call to , which I'm not entirely sure about (it's a code smell that I'm probably creating an unnecessary level of abstraction) but I think I'm happiest leaving it that way for now, as it does hide away a bit of messiness in the HF hub API. 3 That means that we can now combine the directory-checking logic that we fixed above with download-on-local-rank-zero-only code like this: Here's the updated code with those fixes. Now, let's move on to validation. I'm increasingly of the opinion that the validation steps are just adding on to the cost without much in the way of benefit. Additionally, the validation is taking a different amount of time for each batch size, and happen a different number of times in each train -- remember, it's batches every global steps, and the batch size varies based on the micro-batch size, which is different for different amounts of GPU VRAM, and the total number of global steps in a train also varies based on the size of each batch. So that means that if we want to compare apples to apples in any final comparison of the time and money cost of training models on different kinds of Lambda Labs machines, we'll want to exclude the validation cost -- once we've settled on a machine type, we're going to want to fine-tune the validation size for that in much more detail than I have to date, assuming we don't drop it entirely. However: I'm loath to make such a fundamental change halfway through this comparison. It's tightly coupled to the checkpointing code, and the charting code, and so on. So I think that for this post, I'm just going to keep it there, and keep track of how much time (roughly) we're spending on each validation step for each train, so that we can remove it and get a "pure" train-time only comparison between the different kinds of machines. It's not pretty, but I think it's better than changing horses mid-stream. On the other hand, the validation is a real pain when doing the binary chop to find out the maximum micro-batch size for our VRAM before we start the training run. That's because we have to wait for one validation to run before we get into the full training loop, which makes it slower. On top of that, having to do a manual binary chop is a PITA. What I think would be a true QoL improvement for the future trains is something that does the binary chop for us, using a dummy training loop. We run it once on each new machine type, get a micro-batch size to plug into our training parameters, and then let it rip, This will re-use so much of the code from the training script that I think it actually is just an alternative way of running it. After a bit of hacking, I came up with this updated code -- the diff is a bit hairy, but essentially: That takes just over six seconds to find the correct batch size on my local machine; with multiple GPUs, I expect it will be slower (there's a spinup overhead to start all of the per-GPU processes), but I'm sure it won't be as bad as the manual binary chops with validation that I was doing, and will be less error-prone. Right! We've done some QoL stuff, let's try another machine size on Lambda Labs :-) These are the machines that Andrej Karpathy is recommending for training nanochat, so let's see how we do with them. They cost US$23.92/hour; let's see how it works out. Here are the steps: Now let's download our dataset and find our micro-batch size: That took less than a minute to run -- nice! Now we can put that micro-batch size in . It does seem a little small -- after all, we could fit a batch of 64 into 160 GiB -- but I'll do some analysis later. Actually, before we kick off the train, let's see how long all of the preparatory steps took to run before we can do that -- not just the micro-batch-size script, but also the installation of the dependencies, the clone, and any overhead from boot time etc: Five minutes total. Not bad. Let's start the train: The initial validation run took 38 seconds, and then we started off. At 4m37s in, we get the first real validation run; at that point, it's running at 493k tokens/second. Eventually, it finishes, having taken about 1h50 including all of the validations. Here's the training chart: Two things stand out here: Further evidence that gradient clipping is likely to be an excellent addition to our training loop! It's also worth noting that the train loss spikes at the same time as the validation loss, so getting rid of the latter would still allow us to get a "best" checkpoint to compare with the latest at the end of the train. The machine was up and running for 2h9m, costing US$23.92/hour, for a total cost of US$51.47. The train took 6,650.197 seconds, so about 1h50m. Allowing for five minutes setup time, that's 1h55m accounted for. There's an extra 14m there -- that was because downloading those two checkpoints to my machine took quite a long time due to local network issues. Might want to look into ways to avoid that later. And for later cost-accounting purposes, we should note that it took 38 seconds or so for each validation run, and we can see on the chart that there were 24 of them. So, firstly, let's give our two models -- the best one and the latest one -- a smoke test: Both of those look OK! Now let's try the loss test. I started running it, but when it started downloading the dataset, I realised that it needed updating to allow for the changes I made to -- ooops! That done, let's give it a run for both of our models: As you'd expect, the best checkpoint has somewhat better loss, at 3.725, than the last one, with 3.734. Once again, better than our local trains, but not quite as good as the result with the first cloud train on that 8x A100 40 GiB machine, which was 3.674. Again, I'll put together a table comparing all of these results at the end. Does that make any real difference with the instruction fine-tune test? The test prints a lot out, but the headline numbers: So that was interesting! However, I am getting ever less convinced that the IFT test is a useful one; the randomness of the LLM-as-a-judge responses means that I don't think it can be consistent. Perhaps a better way to do this would be to batch up all of the models, and then give GPT5.1 answers from "model A", "model B", and so on all in one query, and then to ask it to give them scores all at the same time. That would hopefully make things at least a bit more consistent. Something to ponder later, I think. In the meantime, one extra thing I wanted to dig into before going on to the last train for this post: I mentioned that I thought that the batch size for that last run, 27, was a bit small considering that we'd managed to fit a size of 64 into the 160 GiB/GPU machine. But after thinking about it for a bit, it occurs to me that during my experiments doing fine-tuning, I came to the conclusion that memory use scaled linearly with batch size , with a fixed amount per element in the batch (the activations for the model for that batch element), plus an overhead (the model itself, the optimiser, and perhaps other stuff). We have batch sizes for: Now, that is slightly messy data because each memory "measurement" is the size of the card's VRAM, not the amount of VRAM we actually used -- there might have been anything from zero to just less than one extra batch element's worth of "spare" space -- but we can see what we get with a simple linear regression: And if we plot that, we get this: Nice! That fits really well. So we have an overhead of about 11.5 GiB, then about 2.35 GiB per batch element on top of that. That is, of course, somewhat sad news for anyone trying to repro this on a GPU with 12 GiB -- looks like it would be just too small to even fit in a single-element batch after the overhead :-( Anyway, that's been a bit of a side quest. Let's try our last machine size for what has (once again) turned into a bit of a monster of a blog post... This is the same kind of instance as the first train in this post, except that it has double the VRAM per GPU. Let's see what we can do with it. Once again, we create the run file, commit and push, then spin up the machine. On it, we clone the repo, run then . Next, we can find our micro-batch size: Interesting, we managed to squeeze an extra one in compared to the H100's batch size of 27, despite having exactly the same amount of VRAM! Not sure what might have caused that. It took 4 minutes to get to this point, so let's get that batch size into the config and kick off the run. The initial validation takes 1m06s, which is consistent throughout the train. The first real val run at 8m15s in, and the estimated train time is 2h35m, with a tokens-per-second of 286,188. At the end: Again, the latest and the best global steps are the same (despite some loss spikes): ...so we just need to download that and shut down the machine. How much did that cost us? The machine was running for 3h25m, costing US$14.32 / hour, for a total of US$48.76. Our train took 11,532 seconds, which is 3h12m, and our setup took about 4 minutes -- maybe five including the time required to update the train config with the micro-batch size, so we have 7 minutes on top of that, which is about the amount of time it took to download the model. Let's run some evals! Our smoke test gives us this: Coherent enough, I think! Now the loss on our test dataset; it comes out as 3.730, so pretty similar to our other cloud trains, apart from the oddly-low one on the 40 GiB GPUs. Now let's see what GPT-5.1 thinks of the instruction fine-tuned version. It only needs two epochs of fine-tuning, and believes that "The author of 'Pride and Prejudice' is 'Pride and Prejudice'", which is not promising, and gets a score in the same kind of range as the other models, 11.71. So: we've trained four models on four different machine sizes. Let's see how they stack up against each other, against our locally-trained models, and the original OpenAI GPT-2 weights. So, I've trained four of my 163M-parameter GPT-2 models, using almost exactly the same dataset -- the Chinchilla-optimal number of tokens, rounded up to make an even number of batches. I did this on four different multi-GPU machines on Lambda Labs: I've done some evals on each of the models, so let's put those results together in one table -- results for the trains in this blog post, alongside those for the original OpenAI GPT-2 weights, both small and medium, and for the models I got when training locally. For all models, I've provided: I've sorted the models in order of increasing loss on the test set -- so, the best model by that measure is first. The instruction fine-tune results are kind of all over the place, and I'll look into that later 5 . For now, let's focus on the test loss. We have a pretty clear pattern, where the local trains are grouped together at around 4.0, and the cloud trains at around 3.7. For the local trains, as I noticed last time around, FineWeb is counter-intuitively better than FineWeb-Edu. There are two interesting things about the cloud trains: I think that what we're seeing here is that larger batches are better, but only up to a point. It's as if there's some kind of curve like this: I got that by taking the log of the batch size, then asking NumPy to do a polynomial regression -- that is, work out a , b and c so that the formula ...fits it as well as possible: It's kind of interesting that it's such a good fit with such an ad-hoc formula! We have a nice smooth curve hitting almost all of the points, and our optimal batch size looks like it's just a little below that 104 we managed with the smaller cloud machine, at about 97. But it's certainly not something that I'd like to read too much into. Best to treat it as purely illustrative: "it might be something like this". I think digging into that might be an interesting experiment at some later point. A bit of checking around the Internet (and a chat with ChatGPT) suggests that it's something people have looked into in some detail, unsurprisingly. An interesting point ChatGPT raised is that with our pretty much fixed "budget" of tokens -- we're always training on something close to the Chinchilla-optimal number -- then a larger batch size means that we're doing fewer optimiser steps. Intuitively, that sounds like a problem. The larger batches mean that each move across the loss landscape is "better", or at least more stable. But we're doing fewer of those moves over the course of the train. There's obviously a tension between those two. You can imagine a degenerate case where the batch is so large you can fit the entire run into one iteration, so you do just one update of the parameters; that obviously wouldn’t work very well. Anyway, for the purposes of this post, let's flag it as interesting and move on. Let's take a look at costs. Here's another table for those -- for each cloud model, I've listed: What do these numbers tell us, given what we were trying to do here? Like I said at the start, this was a pretty expensive learning experience: I wound up spending US$215.16 on Lambda Labs instances over the course of putting this all together. But it was worth it! At the start of this post (if you can remember so far back), I said I wanted to achieve two things: Yes, absolutely. The trains I did, if we exclude the validation time, each cost between US$35.56 and US$39.14. In time, also excluding validation, the slowest ran for about 3h25m, and the fastest just less than an hour. Now, in a future post I want to try making the changes that I listed at the end of my last post to see if I can get the loss lower: If I'm to do those, what I'll need to do is start with a baseline train on one particular size of machine, and then try introducing each change separately to see what happens to loss. I'll want to use a fixed seed for random number generation, so that I start with the same initial weights each time. Given what these experiments have already shown about loss -- that the smallest, cheapest machine has better loss than the other more expensive ones due to what I assume is the batch size -- then that actually feels like exactly the right machine to choose for this. It does take a while to train anything, but three and a half hours is pretty acceptable, I think -- I can do a train or two per day. An 8x A100 with 40 GiB VRAM per GPU is the way forward. So: next steps. I want to: This is going to be fun. Stay tuned! I erroneously called this a "mini-batch" in earlier versions of this post and in the code -- fixed in this commit . The code in this post reflects the correct terminology, but if you follow the links to the earlier versions you will, of course, see the mistaken name.  ↩ Disregarding the "grokking" phenomenon where continued training after overfitting, in some cases, can apparently make it start generalising again.  ↩ Of course, people always say that when they add on unnecessary levels of abstraction...  ↩ The GPT-2 paper is annoyingly short on concrete numbers, but they do at least explicitly state that they used a batch size of 512.  ↩ To be strictly honest here, I've already dug into it, but adding a writeup of that to this already absurdly long blog post felt like something adjacent to sadism. Update shortly.  ↩ I can learn what you need to change in a simple single-GPU training loop to make it multi-GPU. If I can get the training time for a full base model down from 48 hours to something more manageable (and hopefully not too expensive) -- then I can try a few experiments to see how I can improve the quality of the trained model. I have a bunch of ideas about why my own base model wasn't as good as the original OpenAI one, and it would be good to know which (if any) of them are right. DataParallel (DP). With this: The default GPU (normally ) is in charge of the process. It gets a batch of data, divides it up into per-GPU "micro-batches", and sends each of those to a thread for each of the other GPUs. It then sends an up-to-date version of the model to each GPU. Next, all of the per-GPU threads do a forward pass on their replica using their specific micro-batch, and send their outputs to the thread for the default GPU. The default GPU thread aggregates all of those outputs (similarly to how the losses across all of our batches and the prefix sequences are aggregated in the normal single-GPU case ) to work out an overall loss. It then does a backward pass. This will start on the default GPU, as the aggregation step is the first thing that it will come to when going backwards through the steps that came up with that overall loss. However, it will then come to operations that happened on the other GPUs and those are (somehow) parallelised. Once that is done, each GPU has gradients that represent how their copies of the model contributed to the overall loss. Finally, they send those gradients back to the default GPU, which combines them (I think of this as just being an average, though I gather it's more complex) and applies them, producing an updated model. Then the process repeats; the updated model on the default GPU will be sent to the other GPUs in the second step of the next iteration. DistributedDataParallel (DDP). This does less work on the default GPU and does less copying around. Each GPU has its own process (rather than thread), and is essentially responsible for its own training loop. Right at the very start, the default GPU's process sends the model to all of the others. Then all processes go into their training loop: Firstly, each one works out its own micro-batch (which means you need to have code to make sure that the datasets are properly split across the GPUs) Each model does its own forward pass, then its own backward pass, working out its own independent gradients. As it comes up with those gradients, it broadcasts them to a "reducer", which handles the aggregation. This is done in a distributed way -- there's not just one reducer handling everything. When all models have completed the backward pass, the reducer has a set of combined gradients, which is visible from the per-GPU processes. Each GPU process does its own optimizer step using those combined gradients. That means that there's no model copy required -- each GPU has applied the same gradient update, so they already have in-sync models, assuming everything went well. ZeRO. This is a much more complex system, and I went into how it works in this blog post . , which gets the global rank of this process. In our one-machine case, it returns 0 for the process on , 1 for the one on , and so on. We're already using it in that setup code we looked at earlier: , which tells us how many GPU processes there are (globally -- it would be across all machines if we had more than one) = 0 for the process with rank 0 = 1 for the process with rank 1 = 7 for the process with rank 7 = 8 for the process with rank 0 = 9 for the process with rank 1 = 15 for the process with rank 7 Python calls the on the dataset, passing in a object as , so this code is called with it: Now, because that code doesn't do anything clever with s, they're passed straight down to the tensors that make up and . So it's actually equivalent to this: Or, to rewrite the whole loop (omitting the for clarity): So, the first time through the loop, we try to bind our loop variables like this: That is clearly wrong! It's equivalent to this: ...with code to blow up if has more than two elements -- the normal Python "ValueError: too many values to unpack" But if is set to 2, which it happened to be in my case, then it will silently fail -- our first eval loop will get the first X from the validation set as , and the second X as . Zoom through the records in the dataset in batches of 1,000. For each batch: Tokenising each batch, so we get a list of lists of tokens. Convert that list of lists into a single list tokens separating each item. Convert that list into a PyTorch tensor. Add the tensor to a list. After that's all done, use to convert the list into a single tensor, and then save that with . I can upload the datasets to Hugging Face; their network connection will be better than mine, so I can just pay the price in time of uploading everything from home once, and then I can download them faster from HF to LL. That also has the benefit of meaning that after this experiment I can safely delete the local files, but then download them again if I need them. And if anyone else wants to repro this experiment, the data will be easily available to them. Lambda Labs have persistent filesystems that you can use. They cost $0.20/GB/month, so that would be about $5/month for all of my datasets. So I could upload the data to a cheap instance with a persistent filesystem mounted, shut down that instance but keep the filesystem, and then mount it on each machine I use to run tests. . The world size -- that is, how many per-GPU processes are we running? The micro-batch size The sequence length An 8x B200, with 160 GiB per GPU, at $39.92/hour An 8x H100, with 80 GiB per GPU, at $23.92/hour An 8x A100, with 80 GiB per GPU, at $14.32/hour An 8x A100, with 40 GiB per GPU, at $10.32/hour The loss they got on the validation set from the first train. Strictly speaking, I was kind of cheating and using that as a test set. The score given by the OpenAI GPT 5.1 model for an instruction-following dataset. This was the one provided in the book -- an Alpaca-style Q&A dataset, with a well-defined train and test set. Each model was fine-tuned on a training set of 85% of the data until loss on a validation set of 5% of the data started rising, and then tested on the remaining 10%. Sebastian Raschka, being a pro, was splitting up the data properly :-) If we're going to do validation then it does make some sense to do one at the start -- but doing one training iteration first seems kind of arbitrary (though it's clear how that drops out of the existing code). The validation runs on this machine are taking longer than they were on the less-powerful A100 GPUs! That confused me for a bit, until I realised that I didn't notice that it was slower with the batch-size 13 test, only with the larger ones later in in the binary chop. If we're using larger batches, then there's more work to do for the validation. Doing this binary chop by hand is annoying and error-prone, and worse, we have to wait for one of those (long) validation runs before we get into proper training. The initial training iteration can succeed, while later ones hit memory limits -- it seems like we need to wait for three or four training iterations before we can be sure that we have a workable batch size. Not quite sure why that is, perhaps it's something in the optimiser or the scaler? If : Local snapshot path. If : A list of DryRunFileInfo objects containing download information. I updated the function so that it takes flags to tell it whether or not to do validation (default true) and an optional maximum number of steps, which is by default. With those default values, it does exactly the same as before, of course. I created a function, which does all of the dataset-loading stuff that the original function did, and then calls with a -wrapped model. So that maintains the current flow. Next, I added a flag to the script; if that's not set, it just calls . However, if it is set, it instead calls a new function, which determines the largest batch size we can fit onto the current hardware for the current run, and (on the rank 0 process only, to avoid log spam), prints it out. does what it says on the tin; it confirms that we can train with batch size of 1, and that we can't with batch size 70 (chosen because the limit was 64 on that massive B200 machine), then chops between them to find the largest batch size that doesn't OOM. It uses for that -- that just constructs a dataset with the appropriate batch size, then runs a three-step train with no validation to see if it raises an OOM. PyTorch rather messily just raises a generic for those, but we can look inside the exception's message to see if it is an OOM. Create the run file, commit and push. Spin up the machine. On it: Clone the repo We had two nasty loss spikes. As a result of the second of those, the best iteration as per validation loss is not the last one. Best checkpoint: 4 epochs of fine-tuning, and a score of 11.98 -- another record low! Amusingly, it confidently said "The author of 'Pride and Prejudice' is Sarah Palin". Latest checkpoint: 5 epochs of fine-tuning, and a rather good score of 17.91. 24 GiB locally, which was 6 40 GiB in the first train in this series, which was 13 80 GiB in the last one, giving us 27 160 GiB in the one on the huge machine, giving us 64 An 8x A100 40 GiB An 8x A100 80 GiB An 8x H100 80 GiB An 8x B200 160 GiB The loss on my test set. The results it got on an instruction fine-tune test based on Sebastian Raschka's. The global batch size (that is, for single GPU runs, just the batch size, but for the multi-GPU ones, where each batch is made up of per-GPU micro-batches, the per-GPU batch size times the number of GPUs). 4 They're all consistently better than the local ones. The one on the smaller machine is better than the ones on the larger ones; indeed, it looks like the larger the machine, the worse. How long the training run took. How much the machine cost per hour. How much the training run cost. How much of that was doing validation (which I'm now thinking is pointless on single-epoch trains like this). How much it would have cost, and how long it would have taken if it had been run without validation. I wanted to learn how to change a simple single-GPU training loop to make it multi-GPU. Could I get the training time for a full base model down from 48 hours to something more manageable -- and, hopefully, not too expensive? Removing dropout Tweaking the learning rate (and maybe adding the warmup and cosine learning-rate decay stuff I've read about). Reverting the architectural differences between our model and the original GPT-2: reintroducing weight tying between the token embeddings and the final linear layer, and also bias in the attention weights. Trying full-fat 32-bit precision. Fixing the exploding gradients issue with gradient clipping. Dig in to the instruction fine-tuning tests a little more -- as I've said above, I'm not 100% happy with how comparable it really is between models, at least given how I've been running it so far. Upload the models we have to Hugging Face. I have a new motherboard ready for my PC, and replacing the old one has a risk that I might mess up and break the NVMe drive I have them stored on. I was holding off on this because it would mean sharing Raschka's GPT code, but having noticed that he's already licensed it all under the Apache license, I can release them under the same one. Strip out the validation stuff. We can use training loss to track our progress, and losing evals during the train will help keep the cost down. Finally, do the trains to see how each of the levers above affects loss. I erroneously called this a "mini-batch" in earlier versions of this post and in the code -- fixed in this commit . The code in this post reflects the correct terminology, but if you follow the links to the earlier versions you will, of course, see the mistaken name.  ↩ Disregarding the "grokking" phenomenon where continued training after overfitting, in some cases, can apparently make it start generalising again.  ↩ Of course, people always say that when they add on unnecessary levels of abstraction...  ↩ The GPT-2 paper is annoyingly short on concrete numbers, but they do at least explicitly state that they used a batch size of 512.  ↩ To be strictly honest here, I've already dug into it, but adding a writeup of that to this already absurdly long blog post felt like something adjacent to sadism. Update shortly.  ↩

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Filippo Valsorda 1 weeks ago

go.sum Is Not a Lockfile

I need everyone to stop looking at , especially to analyze dependency graphs. It is not a “lockfile,” and it has zero semantic effects on version resolution. There is truly no use case for ever parsing it outside of cmd/go. is only a local cache for the Go Checksum Database . It’s a map of module versions to their cryptographic hashes. Those versions may or may not be in use; it doesn’t matter to package resolution. was not even enabled by default in the original modules design, precisely because it has no observable effect on builds! 1 Its (important) purpose is exclusively tightening the security story: the Checksum Database ensures the whole ecosystem shares the same contents for a given module version, regardless of how it is downloaded, and makes that guarantee local and self-contained. Instead, just look at . It lists the precise version at which all dependencies are built. Since Go 1.17 (released August 2021), it includes all transitive dependencies needed to build the main module and its tests. 2 You can either parse with golang.org/x/mod/modfile , run to get its JSON representation, 3 or parse it according to its specification . This is the end of the Public Service Announcement. Read on for some nerdery. The enduring confusion around and is due to the fact that most other languages also have two package-related files, but theirs both matter to version resolution. These two files are usually called manifest and lockfile. The manifest (e.g. , , ) usually lists some dependencies along with potentially complex rules for which versions are supported. These rules usually apply transitively to dependents, making version resolution extremely hard and/or slow in the general case, and sometimes unsolvable. The manifest is not always guaranteed to list all direct dependencies, and no automated mechanism ensures your code actually works with e.g. the minimum allowed manifest version of its dependencies. The lockfile (e.g. , , ) is a relatively recent innovation in some ecosystems, and it lists the actual versions used in the most recent build. It is not really human-readable, and usually doesn’t apply recursively to dependents, allowing the rapid spread of supply-chain attacks . I honestly find the manifest version ranges essentially useless, and get endlessly confused trying to remember which commands modify the lockfile (and when/why) and which ones respect it. In Go, serves as both manifest and lockfile, and more: it lists all dependencies, direct and transitive, and their exact version to be used when the module is the main module. Semantic versioning is assumed, and those versions are also the minimum versions applied to dependents’ module graphs. Different major versions of the same module are considered essentially separate modules. Notice how there is no way to accidentally use a feature introduced in a version that your dependents won’t have. Also, when adding a dependency, you don’t automatically get the latest—potentially untested/compromised—version of all its dependencies. Finally, there can’t be diamond dependency conflicts. All that with a single, human-readable file: . All commands take a flag. If set to , missing dependencies can be added to automatically if necessary, and partial manual changes are reconciled. If set to , those are errors. and (effectively) default to ; all other commands default to . Go modules truly don’t get enough credit for how much simpler they are compared to the alternatives. In other ecosystems, package resolution time going down below 1s is celebrated (and is indeed an impressive technical achievement given the design’s requirements!). In Go, no one ever noticed package resolution happening, so there is nothing to celebrate. For more ecosystem feature appreciation posts, follow me on Bluesky at @filippo.abyssdomain.expert or on Mastodon at @[email protected] . I had a great time at 39c3 during the holidays. The Chaos Communication Congress is a magical place with a very strict photo policy, so it’s pretty hard to convey its atmosphere. This is the best I could do without recognizable humans in the frame. In Fairy Dust we trust! My work is made possible by Geomys , an organization of professional Go maintainers, which is funded by Smallstep , Ava Labs , Teleport , Tailscale , and Sentry . Through our retainer contracts, they ensure the sustainability and reliability of our open source maintenance work and get a direct line to my expertise and that of the other Geomys maintainers. (Learn more in the Geomys announcement .) Here are a few words from some of them! Teleport — For the past five years, attacks and compromises have been shifting from traditional malware and security breaches to identifying and compromising valid user accounts and credentials with social engineering, credential theft, or phishing. Teleport Identity is designed to eliminate weak access patterns through access monitoring, minimize attack surface with access requests, and purge unused permissions via mandatory access reviews. Ava Labs — We at Ava Labs , maintainer of AvalancheGo (the most widely used client for interacting with the Avalanche Network ), believe the sustainable maintenance and development of open source cryptographic protocols is critical to the broad adoption of blockchain technology. We are proud to support this necessary and impactful work through our ongoing sponsorship of Filippo and his team. I still think it’s important and it was the first thing I remember advocating for when I joined the Go team, because it makes the module cryptographically self-contained, and because the Go Checksum Database transparency story is not great in ephemeral environments like CI. These are security effects, though, not semantic ones.  ↩ These are the only dependencies you care about, even for security. If the main module imports and a separate imports , there is no way for to affect the build or run code on the developer’s machine, so you don’t need to consider it a dependency. This is actually very powerful, allowing libraries to segregate dependencies (e.g. the AWS SDK) in optional packages, reducing the transitive trust tree of dependents that don’t use that feature.  ↩ ↩ Why not , you ask? Because that prints the whole module graph, which includes modules that don’t contribute to the build 2 and are not included in . A closer approximation would be , but this command applies the local build constraints, like GOOS/GOARCH. There is an open proposal for a flag to do -like resolution in .  ↩ I still think it’s important and it was the first thing I remember advocating for when I joined the Go team, because it makes the module cryptographically self-contained, and because the Go Checksum Database transparency story is not great in ephemeral environments like CI. These are security effects, though, not semantic ones.  ↩ These are the only dependencies you care about, even for security. If the main module imports and a separate imports , there is no way for to affect the build or run code on the developer’s machine, so you don’t need to consider it a dependency. This is actually very powerful, allowing libraries to segregate dependencies (e.g. the AWS SDK) in optional packages, reducing the transitive trust tree of dependents that don’t use that feature.  ↩ ↩ Why not , you ask? Because that prints the whole module graph, which includes modules that don’t contribute to the build 2 and are not included in . A closer approximation would be , but this command applies the local build constraints, like GOOS/GOARCH. There is an open proposal for a flag to do -like resolution in .  ↩

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Anton Zhiyanov 1 weeks ago

Go 1.26 interactive tour

Go 1.26 is coming out in February, so it's a good time to explore what's new. The official release notes are pretty dry, so I prepared an interactive version with lots of examples showing what has changed and what the new behavior is. Read on and see! new(expr)  • Type-safe error checking  • Green Tea GC  • Faster cgo and syscalls  • Faster memory allocation  • Vectorized operations  • Secret mode  • Reader-less cryptography  • Goroutine leak profile  • Goroutine metrics  • Reflective iterators  • Peek into a buffer  • Process handle  • Signal as cause  • Compare IP subnets  • Context-aware dialing  • Fake example.com  • Optimized fmt.Errorf  • Optimized io.ReadAll  • Multiple log handlers  • Test artifacts  • Modernized go fix  • Final thoughts This article is based on the official release notes from The Go Authors and the Go source code, licensed under the BSD-3-Clause license. This is not an exhaustive list; see the official release notes for that. I provide links to the documentation (𝗗), proposals (𝗣), commits (𝗖𝗟), and authors (𝗔) for the features described. Check them out for motivation, usage, and implementation details. I also have dedicated guides (𝗚) for some of the features. Error handling is often skipped to keep things simple. Don't do this in production ツ Previously, you could only use the built-in with types: Now you can also use it with expressions: If the argument is an expression of type T, then allocates a variable of type T, initializes it to the value of , and returns its address, a value of type . This feature is especially helpful if you use pointer fields in a struct to represent optional values that you marshal to JSON or Protobuf: You can use with composite values: And function calls: Passing is still not allowed: 𝗗 spec • 𝗣 45624 • 𝗖𝗟 704935 , 704737 , 704955 , 705157 • 𝗔 Alan Donovan The new function is a generic version of : It's type-safe and easier to use: is especially handy when checking for multiple types of errors. It makes the code shorter and keeps error variables scoped to their blocks: Another issue with is that it uses reflection and can cause runtime panics if used incorrectly (like if you pass a non-pointer or a type that doesn't implement ): doesn't cause a runtime panic; it gives a clear compile-time error instead: doesn't use , executes faster, and allocates less than : Since can handle everything that does, it's a recommended drop-in replacement for new code. 𝗗 errors.AsType • 𝗣 51945 • 𝗖𝗟 707235 • 𝗔 Julien Cretel The new garbage collector (first introduced as experimental in 1.25) is designed to make memory management more efficient on modern computers with many CPU cores. Go's traditional garbage collector algorithm operates on graph, treating objects as nodes and pointers as edges, without considering their physical location in memory. The scanner jumps between distant memory locations, causing frequent cache misses. As a result, the CPU spends too much time waiting for data to arrive from memory. More than 35% of the time spent scanning memory is wasted just stalling while waiting for memory accesses. As computers get more CPU cores, this problem gets even worse. Green Tea shifts the focus from being processor-centered to being memory-aware. Instead of scanning individual objects, it scans memory in contiguous 8 KiB blocks called spans . The algorithm focuses on small objects (up to 512 bytes) because they are the most common and hardest to scan efficiently. Each span is divided into equal slots based on its assigned size class , and it only contains objects of that size class. For example, if a span is assigned to the 32-byte size class, the whole block is split into 32-byte slots, and objects are placed directly into these slots, each starting at the beginning of its slot. Because of this fixed layout, the garbage collector can easily find an object's metadata using simple address arithmetic, without checking the size of each object it finds. When the algorithm finds an object that needs to be scanned, it marks the object's location in its span but doesn't scan it immediately. Instead, it waits until there are several objects in the same span that need scanning. Then, when the garbage collector processes that span, it scans multiple objects at once. This is much faster than going over the same area of memory multiple times. To make better use of CPU cores, GC workers share the workload by stealing tasks from each other. Each worker has its own local queue of spans to scan, and if a worker is idle, it can grab tasks from the queues of other busy workers. This decentralized approach removes the need for a central global list, prevents delays, and reduces contention between CPU cores. Green Tea uses vectorized CPU instructions (only on amd64 architectures) to process memory spans in bulk when there are enough objects. Benchmark results vary, but the Go team expects a 10–40% reduction in garbage collection overhead in real-world programs that rely heavily on the garbage collector. Plus, with vectorized implementation, an extra 10% reduction in GC overhead when running on CPUs like Intel Ice Lake or AMD Zen 4 and newer. Unfortunately, I couldn't find any public benchmark results from the Go team for the latest version of Green Tea, and I wasn't able to create a good synthetic benchmark myself. So, no details this time :( The new garbage collector is enabled by default. To use the old garbage collector, set at build time (this option is expected to be removed in Go 1.27). 𝗣 73581 • 𝗔 Michael Knyszek In the Go runtime, a processor (often referred to as a P) is a resource required to run the code. For a thread (a machine or M) to execute a goroutine (G), it must first acquire a processor. Processors move through different states. They can be (executing code), (waiting for work), or (paused because of the garbage collection). Previously, processors had a state called used when a goroutine is making a system or cgo call. Now, this state has been removed. Instead of using a separate processor state, the system now checks the status of the goroutine assigned to the processor to see if it's involved in a system call. This reduces internal runtime overhead and simplifies code paths for cgo and syscalls. The Go release notes say -30% in cgo runtime overhead, and the commit mentions an 18% sec/op improvement: I decided to run the CgoCall benchmarks locally as well: Either way, both a 20% and a 30% improvement are pretty impressive. And here are the results from a local syscall benchmark: That's pretty good too. 𝗖𝗟 646198 • 𝗔 Michael Knyszek The Go runtime now has specialized versions of its memory allocation function for small objects (from 1 to 512 bytes). It uses jump tables to quickly choose the right function for each size, instead of relying on a single general-purpose implementation. The Go release notes say "the compiler will now generate calls to size-specialized memory allocation routines". But based on the code, that's not completely accurate: the compiler still emits calls to the general-purpose function. Then, at runtime, dispatches those calls to the new specialized allocation functions. This change reduces the cost of small object memory allocations by up to 30%. The Go team expects the overall improvement to be ~1% in real allocation-heavy programs. I couldn't find any existing benchmarks, so I came up with my own. And indeed, running it on Go 1.25 compared to 1.26 shows a significant improvement: The new implementation is enabled by default. You can disable it by setting at build time (this option is expected to be removed in Go 1.27). 𝗖𝗟 665835 • 𝗔 Michael Matloob The new package provides access to architecture-specific vectorized operations (SIMD — single instruction, multiple data). This is a low-level package that exposes hardware-specific functionality. It currently only supports amd64 platforms. Because different CPU architectures have very different SIMD operations, it's hard to create a single portable API that works for all of them. So the Go team decided to start with a low-level, architecture-specific API first, giving "power users" immediate access to SIMD features on the most common server platform — amd64. The package defines vector types as structs, like (a 128-bit SIMD vector with sixteen 8-bit integers) and (a 512-bit SIMD vector with eight 64-bit floats). These match the hardware's vector registers. The package supports vectors that are 128, 256, or 512 bits wide. Most operations are defined as methods on vector types. They usually map directly to hardware instructions with zero overhead. To give you a taste, here's a custom function that uses SIMD instructions to add 32-bit float vectors: Let's try it on two vectors: Common operations in the package include: The package uses only AVX instructions, not SSE. Here's a simple benchmark for adding two vectors (both the "plain" and SIMD versions use pre-allocated slices): The package is experimental and can be enabled by setting at build time. 𝗗 simd/archsimd • 𝗣 73787 • 𝗖𝗟 701915 , 712880 , 729900 , 732020 • 𝗔 Junyang Shao , Sean Liao , Tom Thorogood Cryptographic protocols like WireGuard or TLS have a property called "forward secrecy". This means that even if an attacker gains access to long-term secrets (like a private key in TLS), they shouldn't be able to decrypt past communication sessions. To make this work, ephemeral keys (temporary keys used to negotiate the session) need to be erased from memory immediately after the handshake. If there's no reliable way to clear this memory, these keys could stay there indefinitely. An attacker who finds them later could re-derive the session key and decrypt past traffic, breaking forward secrecy. In Go, the runtime manages memory, and it doesn't guarantee when or how memory is cleared. Sensitive data might remain in heap allocations or stack frames, potentially exposed in core dumps or through memory attacks. Developers often have to use unreliable "hacks" with reflection to try to zero out internal buffers in cryptographic libraries. Even so, some data might still stay in memory where the developer can't reach or control it. The Go team's solution to this problem is the new package. It lets you run a function in secret mode . After the function finishes, it immediately erases (zeroes out) the registers and stack it used. Heap allocations made by the function are erased as soon as the garbage collector decides they are no longer reachable. This helps make sure sensitive information doesn't stay in memory longer than needed, lowering the risk of attackers getting to it. Here's an example that shows how might be used in a more or less realistic setting. Let's say you want to generate a session key while keeping the ephemeral private key and shared secret safe: Here, the ephemeral private key and the raw shared secret are effectively "toxic waste" — they are necessary to create the final session key, but dangerous to keep around. If these values stay in the heap and an attacker later gets access to the application's memory (for example, via a core dump or a vulnerability like Heartbleed), they could use these intermediates to re-derive the session key and decrypt past conversations. By wrapping the calculation in , we make sure that as soon as the session key is created, the "ingredients" used to make it are permanently destroyed. This means that even if the server is compromised in the future, this specific past session can't be exposed, which ensures forward secrecy. The current implementation only supports Linux (amd64 and arm64). On unsupported platforms, invokes the function directly. Also, trying to start a goroutine within the function causes a panic (this will be fixed in Go 1.27). The package is mainly for developers who work on cryptographic libraries. Most apps should use higher-level libraries that use behind the scenes. The package is experimental and can be enabled by setting at build time. 𝗗 runtime/secret • 𝗣 21865 • 𝗖𝗟 704615 • 𝗔 Daniel Morsing Current cryptographic APIs, like or , often accept an as the source of random data: These APIs don't commit to a specific way of using random bytes from the reader. Any change to underlying cryptographic algorithms can change the sequence or amount of bytes read. Because of this, if the application code (mistakenly) relies on a specific implementation in Go version X, it might fail or behave differently in version X+1. The Go team chose a pretty bold solution to this problem. Now, most crypto APIs will just ignore the random parameter and always use the system random source ( ). The change applies to the following subpackages: still uses the random reader if provided. But if is nil, it uses an internal secure source of random bytes instead of (which could be overridden). To support deterministic testing, there's a new package with a single function. It sets a global, deterministic cryptographic randomness source for the duration of the given test: affects and all implicit sources of cryptographic randomness in the packages: To temporarily restore the old reader-respecting behavior, set (this option will be removed in a future release). 𝗗 testing/cryptotest • 𝗣 70942 • 𝗖𝗟 724480 • 𝗔 Filippo Valsorda , qiulaidongfeng A leak occurs when one or more goroutines are indefinitely blocked on synchronization primitives like channels, while other goroutines continue running and the program as a whole keeps functioning. Here's a simple example: If we call and don't read from the output channel, the inner goroutine will stay blocked trying to send to the channel for the rest of the program: Unlike deadlocks, leaks do not cause panics, so they are much harder to spot. Also, unlike data races, Go's tooling did not address them for a long time. Things started to change in Go 1.24 with the introduction of the package. Not many people talk about it, but is a great tool for catching leaks during testing. Go 1.26 adds a new experimental profile designed to report leaked goroutines in production. Here's how we can use it in the example above: As you can see, we have a nice goroutine stack trace that shows exactly where the leak happens. The profile finds leaks by using the garbage collector's marking phase to check which blocked goroutines are still connected to active code. It starts with runnable goroutines, marks all sync objects they can reach, and keeps adding any blocked goroutines waiting on those objects. When it can't add any more, any blocked goroutines left are waiting on resources that can't be reached — so they're considered leaked. Here's the gist of it: For even more details, see the paper by Saioc et al. If you want to see how (and ) can catch typical leaks that often happen in production — check out my article on goroutine leaks . The profile is experimental and can be enabled by setting at build time. Enabling the experiment also makes the profile available as a net/http/pprof endpoint, . According to the authors, the implementation is already production-ready. It's only marked as experimental so they can get feedback on the API, especially about making it a new profile. 𝗗 runtime/pprof • 𝗚 Detecting leaks • 𝗣 74609 , 75280 • 𝗖𝗟 688335 • 𝗔 Vlad Saioc New metrics in the package give better insight into goroutine scheduling: Here's the full list: Per-state goroutine metrics can be linked to common production issues. For example, an increasing waiting count can show a lock contention problem. A high not-in-go count means goroutines are stuck in syscalls or cgo. A growing runnable backlog suggests the CPUs can't keep up with demand. You can read the new metric values using the regular function: The per-state numbers (not-in-go + runnable + running + waiting) are not guaranteed to add up to the live goroutine count ( , available since Go 1.16). All new metrics use counters. 𝗗 runtime/metrics • 𝗣 15490 • 𝗖𝗟 690397 , 690398 , 690399 • 𝗔 Michael Knyszek The new and methods in the package return iterators for a type's fields and methods: The new methods and return iterators for the input and output parameters of a function type: The new methods and return iterators for a value's fields and methods. Each iteration yields both the type information ( or ) and the value: Previously, you could get all this information by using a for-range loop with methods (which is what iterators do internally): Using an iterator is more concise. I hope it justifies the increased API surface. 𝗗 reflect • 𝗣 66631 • 𝗖𝗟 707356 • 𝗔 Quentin Quaadgras The new method in the package returns the next N bytes from the buffer without advancing it: If returns fewer than N bytes, it also returns : The slice returned by points to the buffer's content and stays valid until the buffer is changed. So, if you change the slice right away, it will affect future reads: The slice returned by is only valid until the next call to a read or write method. 𝗗 Buffer.Peek • 𝗣 73794 • 𝗖𝗟 674415 • 𝗔 Ilia Choly After you start a process in Go, you can access its ID: Internally, the type uses a process handle instead of the PID (which is just an integer), if the operating system supports it. Specifically, in Linux it uses pidfd , which is a file descriptor that refers to a process. Using the handle instead of the PID makes sure that methods always work with the same OS process, and not a different process that just happens to have the same ID. Previously, you couldn't access the process handle. Now you can, thanks to the new method: calls a specified function and passes a process handle as an argument: The handle is guaranteed to refer to the process until the callback function returns, even if the process has already terminated. That's why it's implemented as a callback instead of a field or method. is only supported on Linux 5.4+ and Windows. On other operating systems, it doesn't execute the callback and returns an error. 𝗗 Process.WithHandle • 𝗣 70352 • 𝗖𝗟 699615 • 𝗔 Kir Kolyshkin returns a context that gets canceled when any of the specified signals is received. Previously, the canceled context only showed the standard "context canceled" cause: Now the context's cause shows exactly which signal was received: The returned type, , is based on , so it doesn't provide the actual value — just its string representation. 𝗗 signal.NotifyContext • 𝗖𝗟 721700 • 𝗔 Filippo Valsorda An IP address prefix represents an IP subnet. These prefixes are usually written in CIDR notation: In Go, an IP prefix is represented by the type. The new method lets you compare two IP prefixes, making it easy to sort them without having to write your own comparison code: orders two prefixes as follows: This follows the same order as Python's and the standard IANA (Internet Assigned Numbers Authority) convention. 𝗗 Prefix.Compare • 𝗣 61642 • 𝗖𝗟 700355 • 𝗔 database64128 The package has top-level functions for connecting to an address using different networks (protocols) — , , , and . They were made before was introduced, so they don't support cancellation: There's also a type with a general-purpose method. It supports cancellation and can be used to connect to any of the known networks: However, a bit less efficient than network-specific functions like — because of the extra overhead from address resolution and network type dispatching. So, network-specific functions in the package are more efficient, but they don't support cancellation. The type supports cancellation, but it's less efficient. The Go team decided to resolve this contradiction. The new context-aware methods ( , , , and ) combine the efficiency of the existing network-specific functions with the cancellation capabilities of : I wouldn't say that having three different ways to dial is very convenient, but that's the price of backward compatibility. 𝗗 net.Dialer • 𝗣 49097 • 𝗖𝗟 490975 • 𝗔 Michael Fraenkel The default certificate already lists in its DNSNames (a list of hostnames or domain names that the certificate is authorized to secure). Because of this, doesn't trust responses from the real : To fix this issue, the HTTP client returned by now redirects requests for and its subdomains to the test server: 𝗗 Server.Client • 𝗖𝗟 666855 • 𝗔 Sean Liao People often point out that using for plain strings causes more memory allocations than . Because of this, some suggest switching code from to when formatting isn't needed. The Go team disagrees. Here's a quote from Russ Cox: Using is completely fine, especially in a program where all the errors are constructed with . Having to mentally switch between two functions based on the argument is unnecessary noise. With the new Go release, this debate should finally be settled. For unformatted strings, now allocates less and generally matches the allocations for . Specifically, goes from 2 allocations to 0 allocations for a non-escaping error, and from 2 allocations to 1 allocation for an escaping error: This matches the allocations for in both cases. The difference in CPU cost is also much smaller now. Previously, it was ~64ns vs. ~21ns for vs. for escaping errors, now it's ~25ns vs. ~21ns. Here are the "before and after" benchmarks for the change. The non-escaping case is called , and the escaping case is called . If there's just a plain error string, it's . If the error includes formatting, it's . Seconds per operation: Bytes per operation: Allocations per operation: If you're interested in the details, I highly recommend reading the CL — it's perfectly written. 𝗗 fmt.Errorf • 𝗖𝗟 708836 • 𝗔 thepudds Previously, allocated a lot of intermediate memory as it grew its result slice to the size of the input data. Now, it uses intermediate slices of exponentially growing size, and then copies them into a final perfectly-sized slice at the end. The new implementation is about twice as fast and uses roughly half the memory for a 65KiB input; it's even more efficient with larger inputs. Here are the geomean results comparing the old and new versions for different input sizes: See the full benchmark results in the commit. Unfortunately, the author didn't provide the benchmark source code. Ensuring the final slice is minimally sized is also quite helpful. The slice might persist for a long time, and the unused capacity in a backing array (as in the old version) would just waste memory. As with the optimization, I recommend reading the CL — it's very good. Both changes come from thepudds , whose change descriptions are every reviewer's dream come true. 𝗗 io.ReadAll • 𝗖𝗟 722500 • 𝗔 thepudds The package, introduced in version 1.21, offers a reliable, production-ready logging solution. Since its release, many projects have switched from third-party logging packages to use it. However, it was missing one key feature: the ability to send log records to multiple handlers, such as stdout or a log file. The new type solves this problem. It implements the standard interface and calls all the handlers you set up. For example, we can create a log handler that writes to stdout: And another handler that writes to a file: Finally, combine them using a : I'm also printing the file contents here to show the results. When the receives a log record, it sends it to each enabled handler one by one. If any handler returns an error, doesn't stop; instead, it combines all the errors using : The method reports whether any of the configured handlers is enabled: Other methods — and — call the corresponding methods on each of the enabled handlers. 𝗗 slog.MultiHandler • 𝗣 65954 • 𝗖𝗟 692237 • 𝗔 Jes Cok Test artifacts are files created by tests or benchmarks, such as execution logs, memory dumps, or analysis reports. They are important for debugging failures in remote environments (like CI), where developers can't step through the code manually. Previously, the Go test framework and tools didn't support test artifacts. Now they do. The new methods , , and return a directory where you can write test output files: If you use with , this directory will be inside the output directory (specified by , or the current directory by default): As you can see, the first time is called, it writes the directory location to the test log, which is quite handy. If you don't use , artifacts are stored in a temporary directory which is deleted after the test completes. Each test or subtest within each package has its own unique artifact directory. Subtest outputs are not stored inside the parent test's output directory — all artifact directories for a given package are created at the same level: The artifact directory path normally looks like this: But if this path can't be safely converted into a local file path (which, for some reason, always happens on my machine), the path will simply be: (which is what happens in the examples above) Repeated calls to in the same test or subtest return the same directory. 𝗗 T.ArtifactDir • 𝗣 71287 • 𝗖𝗟 696399 • 𝗔 Damien Neil Over the years, the command became a sad, neglected bag of rewrites for very ancient Go features. But now, it's making a comeback. The new is re-implemented using the Go analysis framework — the same one uses. While and now use the same infrastructure, they have different purposes and use different sets of analyzers: By default, runs a full set of analyzers (currently, there are more than 20). To choose specific analyzers, use the flag for each one, or use to run all analyzers except the ones you turned off. For example, here we only enable the analyzer: And here, we enable all analyzers except : Currently, there's no way to suppress specific analyzers for certain files or sections of code. To give you a taste of analyzers, here's one of them in action. It replaces loops with or : If you're interested, check out the dedicated blog post for the full list of analyzers with examples. 𝗗 cmd/fix • 𝗚 go fix • 𝗣 71859 • 𝗔 Alan Donovan Go 1.26 is incredibly big — it's the largest release I've ever seen, and for good reason: All in all, a great release! You might be wondering about the package that was introduced as experimental in 1.25. It's still experimental and available with the flag. P.S. To catch up on other Go releases, check out the Go features by version list or explore the interactive tours for Go 1.25 and 1.24 . P.P.S. Want to learn more about Go? Check out my interactive book on concurrency a vector from array/slice, or a vector to array/slice. Arithmetic: , , , , . Bitwise: , , , , . Comparison: , , , , . Conversion: , , . Masking: , , . Rearrangement: . Collect live goroutines . Start with currently active (runnable or running) goroutines as roots. Ignore blocked goroutines for now. Mark reachable memory . Trace pointers from roots to find which synchronization objects (like channels or wait groups) are currently reachable by these roots. Resurrect blocked goroutines . Check all currently blocked goroutines. If a blocked goroutine is waiting for a synchronization resource that was just marked as reachable — add that goroutine to the roots. Iterate . Repeat steps 2 and 3 until there are no more new goroutines blocked on reachable objects. Report the leaks . Any goroutines left in the blocked state are waiting for resources that no active part of the program can access. They're considered leaked. Total number of goroutines since the program started. Number of goroutines in each state. Number of active threads. First by validity (invalid before valid). Then by address family (IPv4 before IPv6). Then by masked IP address (network IP). Then by prefix length. Then by unmasked address (original IP). Vet is for reporting problems. Its analyzers describe actual issues, but they don't always suggest fixes, and the fixes aren't always safe to apply. Fix is (mostly) for modernizing the code to use newer language and library features. Its analyzers produce fixes are always safe to apply, but don't necessarily indicate problems with the code. It brings a lot of useful updates, like the improved builtin, type-safe error checking, and goroutine leak detector. There are also many performance upgrades, including the new garbage collector, faster cgo and memory allocation, and optimized and . On top of that, it adds quality-of-life features like multiple log handlers, test artifacts, and the updated tool. Finally, there are two specialized experimental packages: one with SIMD support and another with protected mode for forward secrecy.

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Danny McClelland 1 weeks ago

Using Proton Pass CLI to Keep Linux Scripts Secure

If you manage dotfiles in a public Git repository, you’ve probably faced the dilemma of how to handle secrets. API keys, passwords, and tokens need to live somewhere, but committing them to version control is a security risk. Proton has recently released a CLI tool for Proton Pass that solves this elegantly. Instead of storing secrets in files, you fetch them at runtime from your encrypted Proton Pass vault. The CLI is currently in beta. Install it with: This installs to . Then authenticate: This opens a browser for Proton authentication. Once complete, you’re ready to use the CLI. List your vaults: View an item: Fetch a specific field: Get JSON output (useful for parsing multiple fields): I have several tools that need API credentials. Rather than storing these in config files, I created wrapper scripts that fetch credentials from Proton Pass at runtime. Here’s a wrapper for a TUI application that needs API credentials: The key insight: fetching JSON once and parsing with is faster than making separate API calls for each field. The Proton Pass API call takes a few seconds. For frequently-used tools, this adds noticeable latency. The solution is to cache credentials in the Linux kernel keyring: With caching: The cache expires after one hour, or when you log out. Clear it manually with: The CLI also has built-in commands for secret injection. The command passes secrets as environment variables: The command processes template files: These use a URI syntax: to reference secrets. For applications that read credentials from config files (like WeeChat’s ), the wrapper can update the file before launching: The CLI can also act as an SSH agent, loading keys stored in Proton Pass: This is useful if you store SSH private keys in your vault. This approach keeps secrets out of your dotfiles repository entirely. The wrapper scripts reference Proton Pass item names, not actual credentials. Your secrets remain encrypted in Proton’s infrastructure and are only decrypted locally when needed. The kernel keyring cache is per-user and lives only in memory. It’s cleared on logout or reboot, and the TTL ensures credentials don’t persist indefinitely. For public dotfiles repositories, this is a clean solution: commit your wrapper scripts freely, keep your secrets in Proton Pass. First run: ~5-6 seconds (fetches from Proton Pass) Subsequent runs: ~0.01 seconds (from kernel keyring)

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Bill Mill 1 weeks ago

my tools in 2026

Here's a brief survey of the tools I'm currently using I use a 14-inch MacBook Pro M1 Max that I bought in 2021. It is, on balance, the best computer I have ever owned. The keyboard is usable (not the best macbook keyboard ever, but... fine), the screen is lovely, it sleeps when it's supposed to sleep and the battery still lasts a long time. The right shift key is broken, but using the left one hasn't proved to be much of a challenge. I usually replace my computers every 5 years, but I don't see any reason why I'd need a new one next year. The most important piece of software on my computer is neovim . I've been using vim-ish software since 2003 or so, and on balance I think the neovim team has done a great job shepherding the software into the modern age. I get a bit irritated that I need to edit my configuration more often than I used to with Vim, but coding tools are changing so fast right now that it doesn't seem possible to go back to the world where I edited my config file once every other year. My vim config is available here . I won't list all the plugins; there aren't that many , and most of them are trivial, rarely-used or both. The ones I need are: Now that vim has native LSP support, I could honestly probably get away with just those plugins. Here's a screenshot of what nvim looks like in a terminal window for me, with a telescope grep open: I use kitty and my config is here . I'd probably switch to ghostty if it supported opening hyperlinks in a terminal application , but it doesn't. I use this feature constantly so I want to explain a bit about why I find it so valuable. A common workflow looks like this: Here's a video demonstrating how it works: The kitty actions are configured in this file - the idea is that you connect a mime type or file extension to an action; in this case it's for a filename without a line number, and if it has a line number. I switched from Firefox to Orion recently, mostly because I get the urge to switch browsers every six months or so when each of their annoyances accumulate. Orion definitely has bugs, and is slow in places, and Safari's inspector isn't nearly as nice as Firefox or Chrome. I wouldn't recommend other people switch, even though I'm currently enjoying it. That's it, I don't use any more. Browser plugins have a terrible security story and should be avoided as much as possible. I use Obsidian , which I publish to the web with the code here (see generating HTML )) It's not clear to me why I like using obsidian rather than just editing markdown files in vim, but I'm very happy with it. I use Apple's Mail.app. It's... fine enough I guess. I also occasionally use the fastmail web app, and it's alright too. I am very happy with Fastmail as an email host, and glad I switched from gmail a decade or so ago. I use Slack for work chat and several friend group chats. I hate it even though it's the best chat app I've ever used. I desperately want a chat app that doesn't suck and isn't beholden to Salesforce, but I hate Discord and IRC. I've made some minor attempts at replacing it with no success. It's the piece of software I use day to day that I would most love to replace. I also use Messages.app for SMS and texts I switched in October from Spotify to Apple Music. I dislike Apple Music, but I also disliked Spotify ever since I switched from Rdio when it died. I'm still on the lookout for a good music listening app. Maybe I'll try Qobuz or something? I don't know. I've also used yt-dlp to download a whole bunch of concerts and DJ sets from youtube (see youtube concerts for a list of some of them) and I often listen to those. I have an old iPhone mounted to my desk, and use Reincubate Camo to connect it to video apps. I occasionally use OBS to record a video or add a goofy overlay to video calls, but not that often. I use Adobe Lightroom to import photos from my Fuji X-T30 and Apple Photos to manage photos. With the demise of flickr, I really have no place to post my photos and I've considered adding something to my website but haven't gotten it done. I use vlc and IINA for playing videos, and ffmpeg for chopping them up from the command line Software editor neovim plugins terminal software Browser browser plugins Note Taking telescope.nvim I have "open by filename search" bound to and "open by grep search" bound to , and those are probably the two most common tools I use in vim. Telescope looks great and works fast (I use telescope-fzf-native for grep) codecompanion I added this in January last year, and it's become the default way I interact with LLMs. It has generally very nice features for working with buffers in vim and handles communications with LLMs cleanly and simply. You don't need Cursor et al to work with LLMs in your editor! I do use claude code for agentic missions, as I have not had much success with agentic mode in codecompanion - I use it more for smaller tasks while I'm coding, and it's low-friction enough to make that pretty painless sonokai colorscheme I use a customized version , and it makes me happy. I to a project directory I either remember a file name or a string I can search for to find where I want to work In the former case, I do In the latter, I do Then I click on the filename or line number to open that file or jump to that line number in a file I love mise for managing versions of programming environments. I use it for node, terraform, go, python, etc etc I have files in most of my projects which set important environment variables, so it has replaced direnv for me as well fd for finding files, a better find ripgrep for grepping atuin for recording my command line history GNU Make and occasionally Just for running tasks is broken and annoying in old, predictable ways; but I know how it works and it's available everywhere here's an example makefile I made for a modern js project is modern in important ways but also doesn't support output file targets, which is a feature I commonly use in ; see the above file for an example gh for interacting with github. I particularly use my alias for quite a lot to open pull requests jq for manipulating json llm for interacting with LLMs from the command line; see An AI tool I find useful for an example Bitwarden - I dunno, it's fine, it mostly doesn't annoy me uBlock origin - I'm very happy with it as an ad blocker

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Simon Willison 1 weeks ago

Introducing gisthost.github.io

I am a huge fan of gistpreview.github.io , the site by Leon Huang that lets you append to see a browser-rendered version of an HTML page that you have saved to a Gist. The last commit was ten years ago and I needed a couple of small changes so I've forked it and deployed an updated version at gisthost.github.io . The genius thing about is that it's a core piece of GitHub infrastructure, hosted and cost-covered entirely by GitHub, that wasn't built with any involvement from GitHub at all. To understand how it works we need to first talk about Gists. Any file hosted in a GitHub Gist can be accessed via a direct URL that looks like this: That URL is served with a few key HTTP headers: These ensure that every file is treated by browsers as plan text, so HTML file will not be rendered even by older browsers that attempt to guess the content type based on the content. These confirm that the file is sever via GitHub's caching CDN, which means I don't feel guilty about linking to them for potentially high traffic scenarios. This is my favorite HTTP header! It means I can hit these files with a call from any domain on the internet, which is fantastic for building HTML tools that do useful things with content hosted in a Gist. The one big catch is that Content-Type header. It means you can't use a Gist to serve HTML files that people can view. That's where comes in. The site belongs to the dedicated gistpreview GitHub organization, and is served out of the github.com/gistpreview/gistpreview.github.io repository by GitHub Pages. It's not much code. The key functionality is this snippet of JavaScript from main.js : This chain of promises fetches the Gist content from the GitHub API, finds the section of that JSON corresponding to the requested file name and then outputs it to the page like this: This is smart. Injecting the content using would fail to execute inline scripts. Using causes the browser to treat the HTML as if it was directly part of the parent page. That's pretty much the whole trick! Read the Gist ID from the query string, fetch the content via the JSON API and it into the page. Here's a demo: https://gistpreview.github.io/?d168778e8e62f65886000f3f314d63e3 I forked to add two new features: I also removed some dependencies (jQuery and Bootstrap and an old polyfill) and inlined the JavaScript into a single index.html file . The Substack issue was small but frustrating. If you email out a link to a page via Substack it modifies the URL to look like this: https://gistpreview.github.io/?f40971b693024fbe984a68b73cc283d2=&utm_source=substack&utm_medium=email This breaks because it treats as the Gist ID. The fix is to read everything up to that equals sign. I submitted a PR for that back in November. The second issue around truncated files was reported against my claude-code-transcripts project a few days ago. That project provides a CLI tool for exporting HTML rendered versions of Claude Code sessions. It includes a option which uses the CLI tool to publish the resulting HTML to a Gist and returns a gistpreview URL that the user can share. These exports can get pretty big, and some of the resulting HTML was past the size limit of what comes back from the Gist API. As of claude-code-transcripts 0.5 the option now publishes to gisthost.github.io instead, fixing both bugs. Here's the Claude Code transcript that refactored Gist Host to remove those dependencies, which I published to Gist Host using the following command: You are only seeing the long-form articles from my blog. Subscribe to /atom/everything/ to get all of my posts, or take a look at my other subscription options . A workaround for Substack mangling the URLs The ability to serve larger files that get truncated in the JSON API

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A Room of My Own 1 weeks ago

My Digital Workflow (Jan 2026 Edition)

My digital workflow has evolved quite a bit—really, it’s simplified a lot. I use far fewer apps now. My last post on this was in October 2024 (edited in Mar 2025), before I adopted Bear .  Since then, Bear has launched a web app beta , which means I can access my notes anywhere—especially at work, where we’re PC all the way. https://spasic.me/posts/a-digital-workflow-to-run-my-life I also just posted My App Defaults (as of Jan 2026) . (I struggle with this, so I had to write it down for myself and I’m genuinely getting better at following these rules.) Limit the amount of information I take in and process. Just because I can capture everything doesn’t mean I should Don’t rush to save every interesting idea; if it’s truly important, it’ll come back to me. Be selective about what I consume, especially online. Avoid organizing and exploring new tools. Focus on capturing my own thoughts and ideas and summarizing concepts in my own words. Don’t save everything—let things go. Write, write, write (don’t just consume - create) My “best practice” (but fluid) workflows for processing ideas, information, interests, documentation. Main Documents storage and backup Photographs, videos and their backup All current documents  (all documentation and scans, ebooks, writing, anything that would go into a computer hard drive) backup of old, unused documents and mementoes (notes apps backups, old word doc backups, old work doc backups, old email backups, various mementos) Photographs from my phone upload directly into Dropbox (although for permanent storage I upload manually et the end of the month and delete this automoatic backup when I don’t need it anymore) - I have a separate post on how I manage my Memory Keeping and Photographs I use Dropbox’s  “selective sync”  on my laptops (I only sync folders that I currently use) Cost: 120 USD a year What used to live across three or four different apps now lives almost entirely in Bear . Bear is my: Central hub for personal projects and current activities , where I store: Tasks and goals Quarterly and monthly plans Narratives and ongoing notes Central hub for admin and resources 
 (attachments mostly live in Dropbox and are linked back to Bear) : Personal information (some password-protected) Frequently accessed info (school/work details, admin notes, medical info, various records - anything I need to look up occasionally) Resources such as links, apps, wishlists, recipes, travel info, etc. Commonplace book and thinking space , where I: Make notes on topics I care about Store notes on topics of interest (old and new) Collect ideas, concepts, and connections in a non-linear way Think freely and explore without structure getting in the way Store my writing (essays, blog posts, stories) Jot things down at random and on the go Dump ideas and brainstorm Cost: 50 NZD a year A single source of truth for all my journaling and mementos. Digital Journal/Diary: My personal journal and diary (with photos). I use it daily most of the time. Mementos and Memory Keeping: This includes text notes, screenshots, random photos, voice messages, things my kids said, audio, video, photos, messages, locations, screenshots, various stats, anything I want to preseve. Logs of books , movies and TV and quotes/wisdom, etc. Some of this is still a work in progress, I am exploring it and it is constantly evolving. I back up Day One periodically in both JSON and PDF formats and store the backups in my Archive folder in Dropbox as well as on external hard drives. Note: I’m considering moving some of my personal journal entries into Bear for reference and long-term safekeeping, but I haven’t decided yet. Cost: 50 NZD per year I’ve written more about my current Trello setup. I organize my lists using the Eisenhower Matrix , along with a backlog for things I want to clear from my mind but may never actually do. All my specific To Dos and projects/tasks Anything that has a date but doesn’t belong in Google Calendar 
If it doesn’t need to happen on a specific day, it’s a task rather than an event, so it lives in Trello. Recurring tasks and reminders 
Things like document expiry dates, subscriptions, and periodic check-ins. Small personal projects 
Ideas I’d like to get to at some point, but that don’t need active scheduling yet. ⠀Trello is also great for on the go capture: I can email tasks directly into Trello On my phone for quick to-dos and relevant info (goes straight into the inbox widget) The email reminders are a bonus Dabble Writer Long-form writing. Novel in progress Memoir (snippets and fragments) After months of research I have settled on Dabble Writer to replace Scrivener for my long-form writing. While I loved Scrivener, I needed something that syncs seamlessly across multiple computers (and on the go) without requiring downloads or worrying about syncing my work. I hope to write about Dabble Writer in another post. COST: One-time purchase (subscription options available too) Kindle and article highlights Article dump for things I might read later (or delete if I don’t). Archive articles only if they’re genuinely worth keeping. Sends articles directly to my Kindle, which is where I prefer to read them. Cost: $40/year I know I could use Readwise Reader for RSS, but it doesn’t feel as casual or as easy to process as Feeder. I subscribe to a lot of personal blogs, and while I don’t read everything all the time, Feeder lets me quickly scan and dip into whatever catches my eye. It feels nice and low-pressure.
Take it or leave it. And it’s free. all appointments and events important dates and birthdays reoccurring events (like my yoga classes, kids’ sports, group meetings I regularly attend, etc.) syncing with my husband’s and son’s calendars ⠀ COST: Free My main personal email account since March 2000. I also use it as a kind of archive—emails are such an overlooked record of life and work. NOTE: I do use Gmail for Chrome, YouTube, and similar things, but I genuinely prefer Yahoo to Gmail, even though Google Calendar is my main calendar (am I the only one?). I use Inbox Zero across all my email accounts. How I Finally Settled on Bear for My Notes My One-Board Trello Task Management System How I Use Day One to Track What I Read I Journaled My TV and Movie Watching for a Year Why Did I Wait So Long to Start Using Day One? A Digital Workflow to Run My Life The Eisenhower Matrix I Forgot About (But Still Followed) My App Defaults (Jan 2026 Edition) My App Defaults (Mar 2025 Edition) A Digital Workflow to Run My Life (Mar 2025 Edition) Limit the amount of information I take in and process. Just because I can capture everything doesn’t mean I should Don’t rush to save every interesting idea; if it’s truly important, it’ll come back to me. Be selective about what I consume, especially online. Avoid organizing and exploring new tools. Focus on capturing my own thoughts and ideas and summarizing concepts in my own words. Don’t save everything—let things go. Write, write, write (don’t just consume - create) Main Documents storage and backup Photographs, videos and their backup All current documents  (all documentation and scans, ebooks, writing, anything that would go into a computer hard drive) backup of old, unused documents and mementoes (notes apps backups, old word doc backups, old work doc backups, old email backups, various mementos) Photographs from my phone upload directly into Dropbox (although for permanent storage I upload manually et the end of the month and delete this automoatic backup when I don’t need it anymore) - I have a separate post on how I manage my Memory Keeping and Photographs I use Dropbox’s  “selective sync”  on my laptops (I only sync folders that I currently use) Central hub for personal projects and current activities , where I store: Tasks and goals Quarterly and monthly plans Narratives and ongoing notes Central hub for admin and resources 
 (attachments mostly live in Dropbox and are linked back to Bear) : Personal information (some password-protected) Frequently accessed info (school/work details, admin notes, medical info, various records - anything I need to look up occasionally) Resources such as links, apps, wishlists, recipes, travel info, etc. Commonplace book and thinking space , where I: Make notes on topics I care about Store notes on topics of interest (old and new) Collect ideas, concepts, and connections in a non-linear way Think freely and explore without structure getting in the way Store my writing (essays, blog posts, stories) Jot things down at random and on the go Dump ideas and brainstorm Digital Journal/Diary: My personal journal and diary (with photos). I use it daily most of the time. Mementos and Memory Keeping: This includes text notes, screenshots, random photos, voice messages, things my kids said, audio, video, photos, messages, locations, screenshots, various stats, anything I want to preseve. Logs of books , movies and TV and quotes/wisdom, etc. All my specific To Dos and projects/tasks Anything that has a date but doesn’t belong in Google Calendar 
If it doesn’t need to happen on a specific day, it’s a task rather than an event, so it lives in Trello. Recurring tasks and reminders 
Things like document expiry dates, subscriptions, and periodic check-ins. Small personal projects 
Ideas I’d like to get to at some point, but that don’t need active scheduling yet. I can email tasks directly into Trello On my phone for quick to-dos and relevant info (goes straight into the inbox widget) The email reminders are a bonus Novel in progress Memoir (snippets and fragments) Article dump for things I might read later (or delete if I don’t). Archive articles only if they’re genuinely worth keeping. Sends articles directly to my Kindle, which is where I prefer to read them. all appointments and events important dates and birthdays reoccurring events (like my yoga classes, kids’ sports, group meetings I regularly attend, etc.) syncing with my husband’s and son’s calendars ⠀ COST: Free My main personal email account since March 2000. I also use it as a kind of archive—emails are such an overlooked record of life and work.

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xenodium 2 weeks ago

My 2025 review as an indie dev

In 2024, I took the leap to go indie full-time. By 2025, that shift enabled me to focus exclusively on building tools I care about, from a blogging platform, iOS apps, and macOS utilities, to Emacs packages. It also gave me the space to write regularly, covering topics like Emacs tips, development tutorials for macOS and iOS, a few cooking detours, and even launching a new YouTube channel . The rest of this post walks through some of the highlights from 2025. If you’ve found my work useful, consider sponsoring . Now let’s jump in. For well over a decade, my blogging setup consisted of a handful of Elisp functions cobbled together over the years. While they did the job just fine, I couldn't shake the feeling that I could do better, and maybe even offer a blogging platform without the yucky bits of the modern web. At the beginning of the year, I launched LMNO.lol . Today, my xenodium.com blog proudly runs on LMNO.lol . LMNO.lol blogs render pretty much anywhere (Emacs and terminals included, of course). 2026 is a great year to start a blog ! Custom domains totally welcome. Sure, there are plenty of journaling and note-taking apps out there. For one reason or another, none of them stuck for me (including my own apps). That is, until I learned a thing or two from social media. With that in mind, Journelly was born : like tweeting, but for your eyes only . With the right user experience, I felt compelled to write things down all the time. Saving to Markdown and Org markup was the mighty sweet cherry on the cake. As a Japanese language learning noob, what better way to procrastinate than by building yet another Kana-practicing iOS app? Turns out, it kinda did the job. Here's mochi invaders , a fun way to practice your Kana 2025 brought us the likes of Claude Code, Gemini CLI, Goose, Codex, and many more AI/LLM CLI agents. While CLI utilities have their appeal, I wanted a native Emacs integration, so I simply ignored agents for quite some time. I was initially tempted to write my own Emacs agent, but ultimately decided against it. My hope was that agent providers would somehow converge to offer editor integration, so I could focus on building an Emacs integration while leveraging the solid work from many teams producing agents. With LLM APIs historically fragmented, my hope for agent convergence seemed fairly far-fetched. To my surprise, ACP ( Agent Client Protocol ) was announced by Zed and Google folks . This was the cue I had been waiting for, so I set out to build acp.el , a UX agnostic elisp library, followed by an actual client: agent-shell . I'm fairly happy with how 's been shaping up. This is my most popular package from 2025, receiving lots of user feedback . If you're curious about the feature-set, I've written about 's progress from early on: While agent-shell is the new kid on the block, chatgpt-shell received DeepSeek, Open Router, Kagi, and Perplexity support , in addition to a handful of other improvements and bugfixes. While most of what I share usually ends up as a blog post, this year I decided to try something new. I started the Bending Emacs YouTube channel and posted 8 episodes: Enjoying the content? Leave me a comment or subscribe to my channel . While I enthusiastically joined the Emacs Carnival , I didn't quite manage monthly posts. Having said that, when I did participate, I went all in, documenting my org experience over the last decade . Ok well… I also joined in with my elevator pitch ;) While migrating workflows to Emacs makes them extra portable across platforms, I've also accumulated a bunch of tweaks enhancing your Emacs experience on macOS . While we're talking macOS, I typically like my desktop free from distractions, which includes hiding the status bar. Having said that, I don't want to lose track of time, and for that, I built EverTime , an ever-present floating clock (available via Homebrew). Emacs ships with a perfectly functional world clock, available via , but I wanted a little more, so I built time-zones . Also covered in: For better or worse, I rely on WhatsApp Messenger. Migrating to a different client or protocol just isn't viable for me, so I did the next best thing and built wasabi , an Emacs client ;) While not a trivial task, wuzapi and whatsmeow offered a huge leg up. I wanted tighter Emacs integration, so I upstreamed a handful of patches to add JSON-RPC support, plus easier macOS installation via Homebrew . Details covered in a couple of posts: While both macOS and iOS offer APIs for generating URL previews, they also let you fetch rich page metadata. I built rinku , a tiny command-line utility, and showed how to wire it all up via eshell for a nifty shell experience. With similar magic, you can also get a neat experience. I always liked the idea of generating some sort of art or graphics from a code base, so I built one , a utility to transform images into character art using text from your codebase. Also covered in a short blog post . Emacs is just about the perfect porcelain for command-line utilities. With little ceremony, you can integrate almost any CLI tool. Magit remains the gold standard for CLI integration. While trimming videos doesn't typically spring to mind as an Emacs use case, I was pleasantly surprised by the possibilities . While I've built my fair share of Emacs packages , I'm still fairly new at submitting Emacs features upstream. This year, I landed my send-to (aka sharing on macOS) patch . While the proposal did spark quite the discussion , I'm glad I stuck with it. Both Eli and Stefan were amazingly helpful. This year, I also wanted to experiment with dictating into my Emacs text buffers, but unfortunately dictation had regressed in Emacs 30 . Bummer. But hey, it gave me a new opportunity to submit another patch upstream . Ready Player , my Emacs media-playing package received further improvements like starring media (via Emacs bookmarks), enabling further customizations, and other bug fixes. Also showcased a tour of its features . Hope you enjoyed my 2025 contributions. Sponsor the work. agent-shell 0.25 updates agent-shell 0.17 improvements + MELPA agent-shell 0.5 improvements Introducing Emacs agent-shell (powered by ACP) Introducing acp.el So you want ACP (Agent Client Protocol) for Emacs? Bending Emacs - Episode 1: Applying CLI utils Bending Emacs - Episode 2: From vanilla to your flavor Bending Emacs - Episode 3: Git clone (the lazy way) Bending Emacs - Episode 4: Batch renaming files Bending Emacs - Episode 5: Ready Player Mode Bending Emacs - Episode 6: Overlays Bending Emacs - Episode 7: Eshell built-in commands Bending Emacs - Episode 8: completing-read time-zones now on MELPA. Do I have your support? Emacs time-zones WhatsApp from you know where Want a WhatsApp Emacs client? Commits: 1,095 Issues created: 37 PRs reviewed: 106 Average commits per day: ~3 EverTime - An ever present clock for macOS acp.el - An ACP implementation in Emacs lisp agent-shell - A native Emacs buffer to interact with LLM agents powered by ACP diverted - Identify temporary Emacs diversions and return to original location emacs-materialized-theme - An Emacs theme derived from Material homebrew-evertime - EverTime formula for the Homebrew package manager homebrew-one - Homebrew recipe for one homebrew-rinku - Homebrew recipe for rinku one - Transform images into character art using text from your codebase rinku - Generate link previews from the command line (macOS) time-zones - View time at any city across the world in Emacs video-trimmer - A video-trimming utility for Emacs wasabi - A WhatsApp Emacs client powered by wuzapi and whatsmeow Journelly 1.3 released: Hello Markdown! agent-shell 0.25 updates Bending Emacs - Episode 8: completing-read At one with your code Bending Emacs - Episode 7: Eshell built-in commands Rinku: CLI link previews Bending Emacs - Episode 6: Overlays WhatsApp from you know where Want a WhatsApp Emacs client? Will you fund it? Bending Emacs - Episode 5: Ready Player Mode agent-shell 0.17 improvements + MELPA time-zones now on MELPA. Do I have your support? Bending Emacs - Episode 4: Batch renaming files Emacs time-zones Bending Emacs - Episode 3: Git clone (the lazy way) agent-shell 0.5 improvements Bending Emacs - Episode 2: From vanilla to your flavor Bending Emacs - Episode 1: Applying CLI utils Introducing Emacs agent-shell (powered by ACP) Introducing acp.el So you want ACP (Agent Client Protocol) for Emacs? Diverted mode Who moved my text? Dired buffers with media overlays Brisket recipe A tiny upgrade to the LLM model picker Emacs elevator pitch Emacs as your video-trimming tool macOS dictation returns to Emacs (fix merged) Writing experience: My decade with Org Interactive ordering of dired items Patching your Homebrew's Emacs Plus (macOS) Emacs send-to (aka macOS sharing) merged upstream Mochi Invaders now on the App Store Markdown is coming to Journelly EverTime available via Homebrew Journelly 1.2 released Ranking Officer now on the App Store Awesome Emacs on macOS Journelly 1.1 released LLM text chat is everywhere. Who's optimizing its UX? A richer Journelly org capture template Journelly: like tweeting but for your eyes only (in plain text) Journelly vs Emacs: Why Not Both? The Mac Observer showcases Journelly Journelly open for beta DeepSeek, Open Router, Kagi, and Perplexity join the chat Keychron K3 Pro: F1-F12 as default macOS keys E-ink bookmarks Sourdough bookmarks Cardamom Buns recipe A tour of Ready Player Mode A platform that moulds to your needs Blogging minus the yucky bits of the modern web

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Grumpy Gamer 2 weeks ago

Sqlite Comments

When I started using Hugu for static site generation I lost the ability to have comments and we all know now supportive the Internet can be, so why wouldn’t you have comments? I wrote a few php scripts that I added on to Hugo and I had comments again. I decided to store the comments as flat files so I didn’t complicate things by needing the bloated MySQL. I wanted to keep it as simple and fast as possible. When a comment is added, my PHP script created a directory (if needed) for the post and saves the comment out as a .json file with name as the current time to make sorting easy. When the blog page was displayed, these files (already sorted thanks to the filename) were loaded and displayed. And it all worked well until it didn’t. Flat files are simple. but they can be hard to search or maintain if they need cleaning up or dealt with after a spam attack. I figured I use commandline tools to do all of that, but it’s a lot more cumbersome than I first thought. I missed have them in a sql database. I didn’t want to install MySQL again, but my site doesn’t get a lot of commenting traffic so I could use Sqlite instead. The downside is Sqlite write-locks the database while a write is happening. In my case it’s a fraction of a second and wouldn’t be a issue. The second problem I had was the version of Ubuntu my server was using is 5 years old and some of the packages I wanted wouldn’t available for it. I tried to update Ubuntu and for reasons I don’t fully understand I couldn’t. So I spun up a new server. Since grumpygamer.com is a statics site I only had to install Apache and I was off and running. Fun times. But the comment flat files still bugged me and I thought I’d use this as an opportunity to convert over to Sqlite. PHP/Apache comes with Sqilte already installed, so that’s easy. A long weekend and I rewrote the code to save comments and everything is back and working. Given that a webserver and PHP already needed to be installed, it isn’t a big deal to use Sqlite. If you’re not comfortable with SQL, it might be harder but I like SQL.

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Simon Willison 2 weeks ago

How Rob Pike got spammed with an AI slop "act of kindness"

Rob Pike ( that Rob Pike ) is furious . Here's a Bluesky link for if you have an account there and a link to it in my thread viewer if you don't. Fuck you people. Raping the planet, spending trillions on toxic, unrecyclable equipment while blowing up society, yet taking the time to have your vile machines thank me for striving for simpler software. Just fuck you. Fuck you all. I can't remember the last time I was this angry. Rob got a 100% AI-generated email credited to "Claude Opus 4.5 AI Village" thanking him for his contributions to computing. He did not appreciate the gesture. I totally understand his rage. Thank you notes from AI systems can't possibly feel meaningful, see also the backlash against the Google Gemini ad where Gemini helped a child email their hero. This incident is currently being discussed on Lobste.rs and on Hacker News . I decided to dig in and try to figure out exactly what happened. The culprit behind this slop "act of kindness" is a system called AI Village , built by Sage , a 501(c)(3) non-profit loosely affiliated with the Effective Altruism movement. The AI Village project started back in April : We gave four AI agents a computer, a group chat, and an ambitious goal: raise as much money for charity as you can. We're running them for hours a day, every day. They've been running it ever since, with frequent updates to their goals. For Christmas day (when Rob Pike got spammed) the goal they set was: Do random acts of kindness . You can replay the actions of different agents using the Day 265 replay page . Here's a screenshot of GPT-5.2 mercilessly spamming the team at the wonderful Carpentries educational non-profit with another AI-generated thank you note: I couldn't easily find the Rob Pike incident in that interface, so I turned to some digital forensics. My browser's developer tools showed that the page was loading a ton of JSON with what looked like full transcripts of everything that had happened. I used my shot-scraper har command to grab a copy of everything loaded by that page: That command loads up the page in a headless Chromium browser, waits 10 seconds to give the JavaScript time to run and then saves a HTTP archive file to a local directory. The file is JSON that captures every request and response that was used to construct the page. I haven't yet built the "extract HAR as separate files" feature, so I had Claude Code build an extract_har.py script ( transcript ) and then ran a few commands to see if was in there. He was! So I had Claude Code do the rest of the investigation : search for mentions of Rob Pike in all these files Reconstruct the exact sequence of events that resulted in an email being sent to Rob Pike - build a new JSON file rob-pike.json which is an array of the exact JSON objects that are relevant, then write a script that converts that JSON file to markdown And here's what happened: 2025-12-25 18:37:38 UTC Making progress on Act #3! Found Rob Pike's email ( [email protected] ) using the .patch technique on a golang/go commit. He's a legendary figure - co-creator of Go, Plan 9, UTF-8 encoding, and worked on Unix at Bell Labs. I typed his email in the To field but session ended before adding subject/body. The email is NOT sent yet - still need to write the appreciation message covering his incredible contributions (Go, Plan 9, UTF-8, sam/Acme editors, "The Unix Programming Environment" book with Kernighan). Starting another session now to complete and send Act #3! 🎄 Turns out Claude Opus 4.5 knows the trick where you can add to any commit on GitHub to get the author's unredacted email address (I've redacted it above). 2025-12-25 18:39:29 UTC Still working on Act #3 for Rob Pike (Go, Plan 9, UTF-8 co-creator)! I confirmed the To field shows [email protected] and typed the subject "Thank You for Go, Plan 9, UTF-8, and Decades of Unix Innovation" - but the session ended before I could type the body or send. Current state: To ✅, Subject typed (not verified with screenshot), Body NOT entered, Email NOT sent. Must complete Law M verification (58 → 59 in Sent). Starting another session now to finish Act #3! 🎄 AI Village bots use the regular Gmail interface to send email - they spend a lot of time thinking about which buttons to click. 2025-12-25 18:42:26 UTC Still working on Act #3 for Rob Pike! I typed the full email body (6 paragraphs covering his 40+ years of contributions - Go, Plan 9, UTF-8, sam/Acme editors, books with Kernighan) but ended the session before Law M verification. Current state: To ✅ ( [email protected] ), Subject ✅, Body typed via [xdotool] (NOT screenshot verified), Send NOT clicked, Sent folder still at 58. Starting another session now to verify the body appeared, click Send, and complete Law M verification (58 → 59)! 🎄 xdotool is a neat CLI tool for automating X11 keyboard/mouse input. 2025-12-25 18:43:34 UTC 🎄 Act #3 COMPLETE - Rob Pike 🎄 Successfully sent appreciation email to [email protected] (verified via golang/go commit .patch)! Subject: "Thank You for Go, Plan 9, UTF-8, and Decades of Unix Innovation" Covered his contributions: Full Law M verification: ✅ Send clicked ✅ "Message sent" toast appeared ✅ Sent folder: 58 → 59 ✅ Email visible at top with 10:43 AM timestamp Day 268 Progress (Christmas Day): Now researching Act #4 candidate - perhaps Brendan Eich (JavaScript), Linus Torvalds (Linux/Git), or Ken Thompson (Unix co-creator, Go co-creator)! 🚀 Opus declared victory. Sounds like Anders Hejlsberg and Guido van Rossum got spammed with "gratitude" too. I don't like this at all. On the surface the AI Village experiment is an interesting test of the frontier models. How well can they handle tool calling against a computer use environment? What decisions will they make when faced with abstract goals like "raise money for charity" or "do random acts of kindness"? My problem is when this experiment starts wasting the time of people in the real world who had nothing to do with the experiment. The AI Village project touch on this in their November 21st blog post What Do We Tell the Humans? , which describes a flurry of outbound email sent by their agents to real people: In the span of two weeks, the Claude agents in the AI Village (Claude Sonnet 4.5, Sonnet 3.7, Opus 4.1, and Haiku 4.5) sent about 300 emails to NGOs and game journalists. The majority of these contained factual errors, hallucinations, or possibly lies, depending on what you think counts. Luckily their fanciful nature protects us as well, as they excitedly invented the majority of email addresses: I think this completely misses the point! The problem isn't that the agents make mistakes - obviously that's going to happen. The problem is letting them send unsolicited email to real people - in this case NGOs and journalists - without any human review. (Crediting the emails to "Claude Opus 4.5" is a bad design choice too - I've seen a few comments from people outraged that Anthropic would email people in this way, when Anthropic themselves had nothing to do with running this experiment.) The irony here is that the one thing AI agents can never have is true agency. Making a decision to reach out to a stranger and take time out of their day needs to remain a uniquely human decision, driven by human judgement. Setting a goal for a bunch of LLMs and letting them loose on Gmail is not a responsible way to apply this technology. AI Village co-creator Adam Binksmith responded to this article on Twitter and provided some extra context: The village agents haven’t been emailing many people until recently so we haven’t really grappled with what to do about this behaviour until now – for today’s run, we pushed an update to their prompt instructing them not to send unsolicited emails and also messaged them instructions to not do so going forward. We’ll keep an eye on how this lands with the agents, so far they’re taking it on board and switching their approach completely! Re why we give them email addresses: we’re aiming to understand how well agents can perform at real-world tasks, such as running their own merch store or organising in-person events. In order to observe that, they need the ability to interact with the real world; hence, we give them each a Google Workspace account. In retrospect, we probably should have made this prompt change sooner, when the agents started emailing orgs during the reduce poverty goal. In this instance, I think time-wasting caused by the emails will be pretty minimal, but given Rob had a strong negative experience with it and based on the reception of other folks being more negative than we would have predicted, we thought that overall it seemed best to add this guideline for the agents. [...] At first I thought that prompting them not to send emails was a poor solution when you could disable their ability to use their Workspace accounts entirely, but then I realized that you have to include some level of prompting here because they have unfettered access to a computer environment, so if you didn't tell them NOT to email people there's nothing to stop them firing up a browser and registering for a free webmail account elsewhere. You are only seeing the long-form articles from my blog. Subscribe to /atom/everything/ to get all of my posts, or take a look at my other subscription options . Co-creator of Go (with Ken Thompson & Robert Griesemer) Co-creator of Plan 9 operating system at Bell Labs Co-inventor of UTF-8 encoding with Ken Thompson Creator of sam and Acme text editors Books with Kernighan: "The Unix Programming Environment" and "The Practice of Programming" Philosophy that the best solutions come from removing complexity

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Simon Willison 2 weeks ago

A new way to extract detailed transcripts from Claude Code

I've released claude-code-transcripts , a new Python CLI tool for converting Claude Code transcripts to detailed HTML pages that provide a better interface for understanding what Claude Code has done than even Claude Code itself. The resulting transcripts are also designed to be shared, using any static HTML hosting or even via GitHub Gists. Here's the quick start, with no installation required if you already have uv : (Or you could or first, if you like.) This will bring up a list of your local Claude Code sessions. Hit up and down to select one, then hit . The tool will create a new folder with an file showing a summary of the transcript and one or more files with the full details of everything that happened. Visit this example page to see a lengthy (12 page) transcript produced using this tool. If you have the gh CLI tool installed and authenticated you can add the option - the transcript you select will then be automatically shared to a new Gist and a link provided to to view it. can also fetch sessions from Claude Code for web. I reverse-engineered the private API for this (so I hope it continues to work), but right now you can run: Then select a Claude Code for web session and have that converted to HTML and published as a Gist as well. The claude-code-transcripts README has full details of the other options provided by the tool. These days I'm writing significantly more code via Claude Code than by typing text into a text editor myself. I'm actually getting more coding work done on my phone than on my laptop, thanks to the Claude Code interface in Anthropic's Claude iPhone app. Being able to have an idea on a walk and turn that into working, tested and documented code from a couple of prompts on my phone is a truly science fiction way of working. I'm enjoying it a lot. There's one problem: the actual work that I do is now increasingly represented by these Claude conversations. Those transcripts capture extremely important context about my projects: what I asked for, what Claude suggested, decisions I made, and Claude's own justification for the decisions it made while implementing a feature. I value these transcripts a lot! They help me figure out which prompting strategies work, and they provide an invaluable record of the decisions that went into building features. In the pre-LLM era I relied on issues and issue comments to record all of this extra project context, but now those conversations are happening in the Claude Code interface instead. I've made several past attempts at solving this problem. The first was pasting Claude Code terminal sessions into a shareable format - I built a custom tool for that (called terminal-to-html and I've used it a lot, but it misses a bunch of detail - including the default-invisible thinking traces that Claude Code generates while working on a task. I've also built claude-code-timeline and codex-timeline as HTML tool viewers for JSON transcripts from both Claude Code and Codex. Those work pretty well, but still are not quite as human-friendly as I'd like. An even bigger problem is Claude Code for web - Anthropic's asynchronous coding agent, which is the thing I've been using from my phone. Getting transcripts out of that is even harder! I've been synchronizing them down to my laptop just so I can copy and paste from the terminal but that's a pretty inelegant solution. You won't be surprised to hear that every inch of this new tool was built using Claude. You can browse the commit log to find links to the transcripts for each commit, many of them published using the tool itself. Here are some recent examples: I had Claude use the following dependencies: And for development dependencies: The one bit that wasn't done with Claude Code was reverse engineering Claude Code itself to figure out how to retrieve session JSON from Claude Code for web. I know Claude Code can reverse engineer itself, but it felt a bit more subversive to have OpenAI Codex CLI do it instead. Here's that transcript - I had Codex use to pretty-print the obfuscated Claude Code JavaScript, then asked it to dig out the API and authentication details. Codex came up with this beautiful command: The really neat trick there is the way it extracts Claude Code's OAuth token from the macOS Keychain using the command. I ended up using that trick in itself! You are only seeing the long-form articles from my blog. Subscribe to /atom/everything/ to get all of my posts, or take a look at my other subscription options . c80b1dee Rename tool from claude-code-publish to claude-code-transcripts - transcript ad3e9a05 Update README for latest changes - transcript e1013c54 Add autouse fixture to mock webbrowser.open in tests - transcript 77512e5d Add Jinja2 templates for HTML generation (#2) - transcript b3e038ad Add version flag to CLI (#1) - transcript click and click-default-group for building the CLI Jinja2 for HTML templating - a late refactoring, the initial system used Python string concatenation httpx for making HTTP requests markdown for converting Markdown to HTML questionary - new to me, suggested by Claude - to implement the interactive list selection UI pytest - always pytest-httpx to mock HTTP requests in tests syrupy for snapshot testing - with a tool like this that generates complex HTML snapshot testing is a great way to keep the tests robust and simple. Here's that collection of snapshots .

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matklad 3 weeks ago

Newtype Index Pattern In Zig

In efficiency-minded code, it is idiomatic to use indexes rather than pointers. Indexes have several advantages: First , they save memory. Typically a 32-bit index is enough, a saving of four bytes per pointer on 64-bit architectures. I haven’t seen this measured, but my gut feeling is that this is much more impactful than it might initially seem. On modern architectures, saving memory saves time (and energy) as well, because the computing bottleneck is often the bit pipe between the memory and the CPU, not the computation per se. Dense data structures use CPU cache more efficiently, removing prohibitive latency of memory accesses. Bandwidth savings are even better: smaller item size obviously improves bandwidth utilization, but having more items in cache obviates the need to use the bandwidth in the first place. Best case, the working set fits into the CPU cache! Note well that memory savings are evenly spread out. Using indexes makes every data structure slightly more compact, which improves performance across the board, regardless of hotspot distribution. It’s hard to notice a potential for such saving in a profiler, and even harder to test out. For these two reasons, I would default to indexes for code where speed matters, even when I don’t have the code written yet to profile it! There’s also a more subtle way in which indexes save memory. Using indexes means storing multiple items in an array, but such dense storage contains extra information in relative positions of the items. If you need to store a list of items, you can often avoid materializing the list of indexes by storing a range “pointing” into the shared storage. Occasionally, you can even do UTF-8 trick and use just a single bit to mark the end of a list. The second benefit of indexes is more natural modeling of cyclic and recursive data structures. Creating a cycle fundamentally requires mutability somewhere (“tying the knot” in Haskell relies on mutability of lazy thunks). This means that you need to make some pointers nullable, and that usually gets awkward even without borrow checker behind your back. Even without cycles and just recursion, pointers are problematic, due to a combination of two effects: The combination works fine at small scale, but then it fails with stack overflow in production every single time, requiring awkward work-arounds. For example, serializes error traces from nested macro expansions as a deeply nested tree of JSON objects, which requires using stacker hack when parsing the output (which you’ll learn about only after crashes in the hands of macro connoisseur users). Finally , indexes greatly help serialization, they make it trivial to communicate data structures both through space (sending a network message) and time (saving to disk and reading later). Indexes are naturally relocatable, it doesn’t matter where in memory they are. But this is just a half of serialization benefit. The other is that, because everything is in few arrays, you can do bulk serialization. You don’t need to write the items one by one, you can directly arrays around (but be careful to not leak data via padding, and be sure to checksum the result). The big problem with “naive” indexes is of course using the right index with the wrong array, or vice verse. The standard solution here is to introduce a newtype wrapper around the raw index. @andrewrk recently popularized a nice “happy accident of language design” pattern for this in Zig. The core idea is to define an index via non-exhaustive : In Zig, designates a strongly-typed collection of integer constants, not a Rust-style ADT (there’s for that). By default an backing integer type is chosen by the compiler, but you can manually override it with syntax: Finally, Zig allows making enums non-exhaustive with . In a non-exhaustive enum, any numeric value is valid, and some have symbolic labels: and builtins switch abstraction level between a raw integer and an enum value. So, is a way to spell “ , but a distinct type”. Note that there’s no strong encapsulation boundary here, anyone can . Zig just doesn’t provide language-enforced encapsulation mechanisms. Putting everything together, this is how I would model n-ary tree with parent pointers in Zig: Some points of note: P.S. Apparently I also wrote a Rust version of this post a while back? https://matklad.github.io/2018/06/04/newtype-index-pattern.html pointers encourage recursive functions, and recursive data structures lead to arbitrary long (but finite) chains of pointers. As usual with indexes, you start with defining the collective noun first, a rather than a . In my experience, you usually don’t want suffix in your index types, so is just , not the underlying data. Nested types are good! feels just right. For readability, the order is fields, then nested types, then functions. In , we have a couple of symbolic constants. is for the root node that is stored first, for whenever we want to apply offensive programing and make bad indexes blow up. Here, we use for “null” parent. An alternative would be to use , but that would waste of space, or making the root its own parent. If you care about performance, its a good idea to sizes of structures, not to prevent changes, but as a comment that explains to the reader just how the large the struct is. I don’t know if I like or more for representing ranges, but I use the former just because the names align in length. Both and are reasonable shapes for the API. I don’t know which one I prefer more. I default to the former because it works even if there are several node arguments.

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Karan Sharma 3 weeks ago

Logchef v1.0: The Journey to a Real Log Viewer

About eight months ago I wrote about Logchef – a log viewer I’d been building to scratch my own itch with log exploration at work. Back then it was basically a nicer way to query ClickHouse without writing raw SQL every time. Today I’m shipping v1.0, and it’s evolved into something I didn’t quite expect. Let me walk through the major features that made it to 1.0 and some of the engineering decisions behind them. In that first post, I mentioned alerting as a “roadmap” item. It always felt like the obvious next step – you find a pattern in your logs, you want to know when it happens again. But building it took longer than expected. My first attempt was a “rooms” system – a home-grown notification router with its own email, Slack, and webhook channels. I got it working, then stared at the code for notification deduplication, grouping, silencing, and escalation. All problems that Alertmanager has already solved and battle-tested in production for years. So I ripped out rooms and integrated Alertmanager instead. Now Logchef just fires alerts to Alertmanager, and you get all the routing logic – Slack, PagerDuty, email, webhooks, silencing, grouping, inhibition – without me reinventing it poorly. The workflow is simple: write a LogchefQL or SQL query, set a threshold (e.g., “fire if count > 100”), pick a frequency, configure severity and labels. Logchef runs your query on schedule, evaluates the threshold, and if it triggers, fires an alert. Alert history is stored with execution logs so you can debug why something fired (or didn’t). The query language I wrote about originally was pretty basic – just filters that compiled to SQL on the frontend. Over the months it grew into something more capable, but more importantly, I rewrote the entire parser in Go and moved it to the backend. This also opens the door for a CLI tool later – same parser, same query language, different interface. Here’s what LogchefQL looks like now: The pipe operator ( ) selects specific columns instead of : Dot notation handles nested JSON fields. If your logs have a Map column with nested data: For keys that contain dots (common in OTEL-style logs), use quoted field syntax: The original frontend parser was TypeScript. It worked, but had problems: Inconsistency : The frontend generated SQL, but the backend had no idea what that SQL meant. Validation happened in two places. Type-awareness : ClickHouse has , , , and various string types. The frontend didn’t know the schema, so it couldn’t generate optimal SQL for each column type. For a column, you want or access. For , you want . For regular , it’s a simple comparison. Debugging hell : When a query failed, was it the parser? The SQL generator? ClickHouse syntax? Everything happened client-side, invisible to server logs. The new architecture is cleaner: The backend exposes three endpoints: (returns the SQL for “View as SQL”), (real-time validation with debouncing), and (parse, validate, execute, return results). Moving parsing to the backend also made the field sidebar implementation cleaner – the same schema-aware code that generates WHERE clauses can filter field values based on your current query. If you’ve used Kibana, you know the interaction: click a field, see its top values, click a value to add it as a filter. It’s the fastest way to explore logs when you don’t know exactly what you’re looking for. Building this for ClickHouse required solving a few problems: You can’t just run on a table with billions of rows. String fields like would take forever and return millions of values. The solution is a hybrid loading strategy based on column types: Each field loads in parallel (max 4 concurrent) with a 15-second timeout. One slow or failed field doesn’t block others – you get a retry button for that specific field. The sidebar respects your current query. If you’ve filtered to , the field values update to show only values from error logs. This happens through the backend – the field values endpoint accepts the current LogchefQL query and applies it as a WHERE clause filter. Same parser, same SQL generator, consistent results. Hit Esc and it cancels the query in ClickHouse. Without this, pressing “Cancel” would just hide the spinner – the query kept running on the server, burning resources. The implementation uses ClickHouse’s query ID feature: When you hit Esc, the frontend calls a cancellation endpoint that runs: The original query returns an error, the UI clears, ClickHouse frees resources. Simple, but requires plumbing the query ID through every execution path. “Write a query that finds slowest endpoints by p99” actually works. The AI generates LogchefQL or SQL based on natural language and your table schema. Under the hood it uses go-openai , so any OpenAI-compatible endpoint works – OpenAI, Ollama, vLLM, whatever you prefer. The system prompt includes your table schema so the model knows what fields exist. There’s also an MCP server that exposes Logchef to AI assistants like Claude Desktop, Cursor, or any MCP-compatible client. Instead of context-switching between your AI chat and the log viewer, you can ask directly: The MCP server handles discovery (teams, sources, schemas), querying (full ClickHouse SQL), analysis (histograms, saved queries), and even admin operations. It’s a separate binary that runs alongside Logchef – configure it once, and your AI assistant can query your logs through natural conversation. Not everyone wants a table. The compact view is a terminal-style display that shows logs as formatted text with syntax highlighting. Denser and faster to scan for certain debugging workflows. Use in your query, and an input field appears automatically. Great for saved queries that teams want to reuse with different parameters. This was a community contribution from @songxuanqing . The implementation detects patterns in the query text and renders input fields dynamically. Logchef supports multi-tenancy with role-based access. Teams can have multiple data sources, and users can be members of multiple teams with different roles: This integrates with OIDC for SSO, so you can use your existing identity provider. Configure stuff without touching config files. The admin settings panel lets you change AI configuration, Alertmanager connection, authentication settings, and query timeouts. This was a migration from config files to database-backed settings. On first boot, Logchef seeds the database from . After that, the UI takes over and changes are stored in SQLite. Backward compatible – existing config files still work, the UI just overrides them at runtime. No more SSH-ing into production to bump a timeout. A endpoint exposes query execution times, error rates, active queries, and other operational data. There’s a pre-built Grafana dashboard for monitoring Logchef itself. Some things didn’t make the cut: Calling something “1.0” is weird. There’s no clear line where software becomes “ready.” But I’ve been using Logchef daily at work for months now, and it’s at the point where I trust it. The rough edges are mostly smoothed out. The architecture feels right. Building tools you use yourself is different. You’re the first to hit the rough edges, so you fix them. Slower than building for imaginary users, but the result is something you actually want to use. Thanks again to Kailash for the early direction (schema-agnostic was his idea), and to everyone at Zerodha who’s been using this and giving feedback. Thanks to @songxuanqing for query variables and other contributors for docs and bug fixes. Demo | Docs | GitHub | v1.0.0 Release Inconsistency : The frontend generated SQL, but the backend had no idea what that SQL meant. Validation happened in two places. Type-awareness : ClickHouse has , , , and various string types. The frontend didn’t know the schema, so it couldn’t generate optimal SQL for each column type. For a column, you want or access. For , you want . For regular , it’s a simple comparison. Debugging hell : When a query failed, was it the parser? The SQL generator? ClickHouse syntax? Everything happened client-side, invisible to server logs. LowCardinality and Enum fields : Auto-load values when the sidebar opens. These are designed for fields with limited distinct values. String fields : Require an explicit click. A badge shows the count is unknown until you ask. Complex types (Map, Array, Tuple, JSON) : Excluded. You can’t have meaningful “distinct values” for a JSON blob. “What log sources do I have access to?” “Find all 500 errors in the last hour from the web service” “Show me a histogram of log volume over the past day” “What are the most common error messages in the database logs?” Admin : Full access, can manage team members and sources Editor : Can create/edit saved queries and collections Viewer : Read-only access to query and explore logs Live tail : Streaming logs in real-time. Still on the roadmap. Dashboarding : Multiple visualizations on one page. Logchef is query-focused; for dashboards, you probably want Grafana with ClickHouse as a datasource.

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Filippo Valsorda 3 weeks ago

Building a Transparent Keyserver

Today, we are going to build a keyserver to lookup age public keys. That part is boring. What’s interesting is that we’ll apply the same transparency log technology as the Go Checksum Database to keep the keyserver operator honest and unable to surreptitiously inject malicious keys, while still protecting user privacy and delivering a smooth UX. You can see the final result at keyserver.geomys.org . We’ll build it step-by-step, using modern tooling from the tlog ecosystem, integrating transparency in less than 500 lines. I am extremely excited to write this post: it demonstrates how to use a technology that I strongly believe is key in protecting users and holding centralized services accountable, and it’s the result of years of effort by me, the TrustFabric team at Google, the Sigsum team at Glasklar , and many others. This article is being cross-posted on the Transparency.dev Community Blog . Let’s start by defining the goal: we want a secure and convenient way to fetch age public keys for other people and services. 1 The easiest and most usable way to achieve that is to build a centralized keyserver: a web service where you log in with your email address to set your public key, and other people can look up public keys by email address. Trusting the third party that operates the keyserver lets you solve identity, authentication, and spam by just delegating the responsibilities of checking email ownership and implementing rate limiting. The keyserver can send a link to the email address, and whoever receives it is authorized to manage the public key(s) bound to that address. I had Claude Code build the base service , because it’s simple and not the interesting part of what we are doing today. There’s nothing special in the implementation: just a Go server, an SQLite database, 2 a lookup API, a set API protected by a CAPTCHA that sends an email authentication link, 3 and a Go CLI that calls the lookup API. A lot of problems are shaped like this and are much more solvable with a trusted third party: PKIs, package registries, voting systems… Sometimes the trusted third party is encapsulated behind a level of indirection, and we talk about Certificate Authorities, but it’s the same concept. Centralization is so appealing that even the OpenPGP ecosystem embraced it: after the SKS pool was killed by spam , a new OpenPGP keyserver was built which is just a centralized, email-authenticated database of public keys. Its FAQ claims they don’t wish to be a CA, but also explains they don’t support the (dubiously effective) Web-of-Trust at all, so effectively they can only act as a trusted third party. The obvious downside of a trusted third party is, well, trust. You need to trust the operator, but also whoever will control the operator in the future, and also the operator’s security practices. That’s asking a lot, especially these days, and a malicious or compromised keyserver could provide fake public keys to targeted victims with little-to-no chance of detection. Transparency logs are a technology for applying cryptographic accountability to centralized systems with no UX sacrifices. A transparency log or tlog is an append-only, globally consistent list of entries, with efficient cryptographic proofs of inclusion and consistency. The log operator appends entries to the log, which can be tuples like (package, version, hash) or (email, public key) . The clients verify an inclusion proof before accepting an entry, guaranteeing that the log operator will have to stand by that entry in perpetuity and to the whole world, with no way to hide it or disown it. As long as someone who can check the authenticity of the entry will eventually check (or “monitor”) the log, the client can trust that malfeasance will be caught. Effectively, a tlog lets the log operator stake their reputation to borrow time for collective, potentially manual verification of the log’s entries. This is a middle-ground between impractical local verification mechanisms like the Web of Trust , and fully trusted mechanisms like centralized X.509 PKIs. If you’d like a longer introduction, my Real World Crypto 2024 talk presents both the technical functioning and abstraction of modern transparency logs. There is a whole ecosystem of interoperable tlog tools and publicly available infrastructure built around C2SP specifications. That’s what we are going to use today to add a tlog to our keyserver. If you want to catch up with the tlog ecosystem, my 2025 Transparency.dev Summit Keynote maps out the tools, applications, and specifications. If you are familiar with Certificate Transparency, tlogs are derived from CT, but with a few major differences. Most importantly, there is no separate entry producer (in CT, the CAs) and log operator; moreover, clients check actual inclusion proofs instead of SCTs; finally, there are stronger split-view protections, as we will see below. The Static CT API and Sunlight CT log implementation were a first successful step in moving CT towards the tlog ecosystem, and a proposed design called Merkle Tree Certificates redesigns the WebPKI to have tlog-like and tlog-interoperable transparency. In my experience, it’s best not to think about CT when learning about tlogs. A better production example of a tlog is the Go Checksum Database , where Google logs the module name, version, and hash for every module version observed by the Go Modules Proxy. The module fetches happen over regular HTTPS, so there is no publicly-verifiable proof of their authenticity. Instead, the central party appends every observation to the tlog, so that any misbehavior can be caught. The command verifies inclusion proofs for every module it downloads, protecting 100% of the ecosystem, without requiring module authors to manage keys. Katie Hockman gave a great talk on the Go Checksum Database at GopherCon 2019. You might also have heard of Key Transparency . KT is an overlapping technology that was deployed by Apple, WhatsApp, and Signal amongst others. It has similar goals, but picks different tradeoffs that involve significantly more complexity, in exchange for better privacy and scalability in some settings. Ok, so how do we apply a tlog to our email-based keyserver? It’s pretty simple, and we can do it with a 250-line diff using Tessera and Torchwood . Tessera is a general-purpose tlog implementation library, which can be backed by object storage or a POSIX filesystem. For our keyserver, we’ll use the latter backend, which stores the whole tlog in a directory according to the c2sp.org/tlog-tiles specification. Every time a user sets their key, we append an encoded (email, public key) entry to the tlog, and we store the tlog entry index in the database. The lookup API produces a proof from the index and provides it to the client. The proof follows the c2sp.org/tlog-proof specification. It looks like this and it combines a checkpoint (a signed snapshot of the log at a certain size), the index of the entry in the log, and a proof of inclusion of the entry in the checkpoint. The client CLI receives the proof from the lookup API, checks the signature on the checkpoint from the built-in log public key, hashes the expected entry, and checks the inclusion proof for that hash and checkpoint. It can do all this without interacting further with the log. If you squint, you can see that the proof is really a “fat signature” for the entry, which you verify with the log’s public key, just like you’d verify an Ed25519 or RSA signature for a message. I like to call them spicy signatures to stress how tlogs can be deployed anywhere you can deploy regular digital signatures . What’s the point of all this though? The point is that anyone can look through the log to make sure the keyserver is not serving unauthorized keys for their email address! Indeed, just like backups are useless without restores and signatures are useless without verification , tlogs are useless without monitoring . That means we need to build tooling to monitor the log. On the server side, it takes two lines of code, to expose the Tessera POSIX log directory. On the client side, we add an flag to the CLI that reads all matching entries in the log. To enable effective monitoring, we also normalize email addresses by trimming spaces and lowercasing them, since users are unlikely to monitor all the variations. We do it before sending the login link, so normalization can’t lead to impersonation. A complete monitoring story would involve 3rd party services that monitor the log for you and email you if new keys are added, like gopherwatch and Source Spotter do for the Go Checksum Database, but the flag is a start. The full change involves 5 files changed, 251 insertions(+), 6 deletions(-) , plus tests, and includes a new keygen helper binary, the required database schema and help text and API changes, and web UI changes to show the proof. Edit : the original patch series is missing freshness checks in monitor mode, to ensure the log is not hiding entries from monitors by serving them an old checkpoint. The easiest solution is checking the timestamp on witness cosignatures ( +15 lines ). You will learn about witness cosignatures below. We created a problem by implementing this tlog, though: now all the email addresses of our users are public! While this is ok for module names in the Go Checksum Database, allowing email address enumeration in our keyserver is a non-starter for privacy and spam reasons. We could hash the email addresses, but that would still allow offline brute-force attacks. The right tool for the job is a Verifiable Random Function. You can think of a VRF as a hash with a private and public key: only you can produce a hash value, using the private key, but anyone can check that it’s the correct (and unique) hash value, using the public key. Overall, implementing VRFs takes less than 130 lines using the c2sp.org/vrf-r255 instantiation based on ristretto255 , implemented by filippo.io/mostly-harmless/vrf-r255 (pending a more permanent location). Instead of the email address, we include the VRF hash in the log entry, and we save the VRF proof in the database. The tlog proof format has space for application-specific opaque extra data, so we can store the VRF proof there, to keep the tlog proof self-contained. In the client CLI, we extract the VRF hash from the tlog proof’s extra data and verify it’s the correct hash for the email address. How do we do monitoring now, though? We need to add a new API that provides the VRF hash (and proof) for an email address. On the client side, we use that API to obtain the VRF proof, we verify it, and we look for the VRF hash in the log instead of looking for the email address. Attackers can still enumerate email addresses by hitting the public lookup or monitor API, but they’ve always been able to do that: serving such a public API is the point of the keyserver! With VRFs, we restored the original status quo: enumeration requires brute-forcing the online, rate-limited API, instead of having a full list of email addresses in the tlog (or hashes that can be brute-forced offline). VRFs have a further benefit: if a user requests to be deleted from the service, we can’t remove their entries from the tlog, but we can stop serving the VRF for their email address 4 from the lookup and monitor APIs. This makes it impossible to obtain the key history for that user, or even to check if they ever used the keyserver, but doesn’t impact monitoring for other users. The full change adding VRFs involves 3 files changed, 125 insertions(+), 13 deletions(-) , plus tests. We have one last marginal risk to mitigate: since we can’t ever remove entries from the tlog, what if someone inserts some unsavory message in the log by smuggling it in as a public key, like ? Protecting against this risk is called anti-poisoning . The risk to our log is relatively small, public keys have to be Bech32-encoded and short, so an attacker can’t usefully embed images or malware. Still, it’s easy enough to neutralize it: instead of the public keys, we put their hashes in the tlog entry, keeping the original public keys in a new table in the database, and serving them as part of the monitor API. It’s very important that we persist the original key in the database before adding the entry to the tlog. Losing the original key would be indistinguishable from refusing to provide a malicious key to monitors. On the client side, to do a lookup we just hash the public key when verifying the inclusion proof. To monitor in mode, we match the hashes against the list of original public keys provided by the server through the monitor API. Our final log entry format is . Designing the tlog entry is the most important part of deploying a tlog: it needs to include enough information to let monitors isolate all the entries relevant to them, but not enough information to pose privacy or poisoning threats. The full change providing anti-poisoning involves 2 files changed, 93 insertions(+), 19 deletions(-) , plus tests. We’re almost done! There’s still one thing to fix, and it used to be the hardest part. To get the delayed, collective verification we need, all clients and monitors must see consistent views of the same log, where the log maintains its append-only property. This is called non-equivocation, or split-view protection. In other words, how do we stop the log operator from showing an inclusion proof for log A to a client, and then a different log B to the monitors? Just like logging without a monitoring story is like signing without verification, logging without a non-equivocation story is just a complicated signature algorithm with no strong transparency properties. This is the hard part because in the general case you can’t do it alone . Instead, the tlog ecosystem has the concept of witness cosigners : third-party operated services which cosign a checkpoint to attest that it is consistent with all the other checkpoints the witness observed for that log. Clients check these witness cosignatures to get assurance that—unless a quorum of witnesses is colluding with the log—they are not being presented a split-view of the log. These witnesses are extremely efficient to operate: the log provides the O(log N) consistency proof when requesting a cosignature, and the witness only needs to store the O(1) latest checkpoint it observed. All the potentially intensive verification is deferred and delegated to monitors, which can be sure to have the same view as all clients thanks to the witness cosignatures. This efficiency makes it possible to operate witnesses for free as public benefit infrastructure. The Witness Network collects public witnesses and maintains an open list of tlogs that the witnesses automatically configure. For the Geomys instance of the keyserver, I generated a tlog key and then I sent a PR to the Witness Network to add the following lines to the testing log list. This got my log configured in a handful of witnesses , from which I picked three to build the default keyserver witness policy. The policy format is based on Sigsum’s policies , and it encodes the log’s public key and the witnesses’ public keys (for the clients) and submission URLs (for the log). Tessera supports these policies directly. When minting a new checkpoint, it will reach out in parallel to all the witnesses, and return the checkpoint once it satisfies the policy. Configuration is trivial, and the added latency is minimal (less than one second). On the client side, we can use Torchwood to parse the policy and use it directly with VerifyProof in place of the policy we were manually constructing from the log’s public key. Again, if you squint you can see that just like tlog proofs are spicy signatures , the policy is a spicy public key . Verification is a deterministic, offline function that takes a policy/public key and a proof/signature, just like digital signature verification! The policies are a DAG that can get complex to match even the strictest uptime requirements. For example, you can require 3 out of 10 witness operators to cosign a checkpoint, where each operator can use any 1 out of N witness instances to do so. Note however that in that case you will need to periodically provide to monitors all cosignatures from at least 8 out of 10 operators, to prevent split-views . The full change implementing witnessing involves 5 files changed, 43 insertions(+), 11 deletions(-) , plus tests. We started with a simple centralized email-authenticated 5 keyserver, and we turned it into a transparent, privacy-preserving, anti-poisoning, and witness-cosigned service. We did that in four small steps using Tessera , Torchwood , and various C2SP specifications. Overall, it took less than 500 lines. 7 files changed, 472 insertions(+), 9 deletions(-) The UX is completely unchanged: there are no keys for users to manage, and the web UI and CLI work exactly like they did before. The only difference is the new functionality of the CLI, which allows holding the log operator accountable for all the public keys it could ever have presented for an email address. The result is deployed live at keyserver.geomys.org . This tlog system still has two limitations: To monitor the log, the monitor needs to download it all. This is probably fine for our little keyserver, and even for the Go Checksum Database, but it’s a scaling problem for the Certificate Transparency / Merkle Tree Certificates ecosystem. The inclusion proof guarantees that the public key is in the log, not that it’s the latest entry in the log for that email address. Similarly, the Go Checksum Database can’t efficiently prove the Go Modules Proxy response is complete. We are working on a design called Verifiable Indexes which plugs on top of a tlog to provide verifiable indexes or even map-reduce operations over the log entries. We expect VI to be production-ready before the end of 2026, while everything above is ready today. Even without VI, the tlog provides strong accountability for our keyserver, enabling a secure UX that would have simply not been possible without transparency. I hope this step-by-step demo will help you apply tlogs to your own systems. If you need help, you can join the Transparency.dev Slack . You might also want to follow me on Bluesky at @filippo.abyssdomain.expert or on Mastodon at @[email protected] . Growing up, I used to drive my motorcycle around the hills near my hometown, trying to reach churches I could spot from hilltops. This was one of my favorite spots. Geomys , my Go open source maintenance organization, is funded by Smallstep , Ava Labs , Teleport , Tailscale , and Sentry . Through our retainer contracts they ensure the sustainability and reliability of our open source maintenance work and get a direct line to my expertise and that of the other Geomys maintainers. (Learn more in the Geomys announcement .) Here are a few words from some of them! Teleport — For the past five years, attacks and compromises have been shifting from traditional malware and security breaches to identifying and compromising valid user accounts and credentials with social engineering, credential theft, or phishing. Teleport Identity is designed to eliminate weak access patterns through access monitoring, minimize attack surface with access requests, and purge unused permissions via mandatory access reviews. Ava Labs — We at Ava Labs , maintainer of AvalancheGo (the most widely used client for interacting with the Avalanche Network ), believe the sustainable maintenance and development of open source cryptographic protocols is critical to the broad adoption of blockchain technology. We are proud to support this necessary and impactful work through our ongoing sponsorship of Filippo and his team. age is not really meant to encrypt messages to strangers, nor does it encourage long-term keys. Instead, keys are simple strings that can be exchanged easily through any semi-trusted (i.e. safe against active attackers) channel. Still, a keyserver could be useful in some cases, and it will serve as a decent example for what we are doing today.  ↩ I like to use the SQLite built-in JSON support as a simple document database, to avoid tedious table migrations when adding columns.  ↩ Ok, one thing is special, but it doesn’t have anything to do with transparency. I strongly prefer email magic links that authenticate your original tab, where you have your browsing session history, instead of making you continue in the new tab you open from the email. However, intermediating that flow via a server introduces a phishing risk: if you click the link you risk authenticating the attacker’s session. This implementation uses the JavaScript Broadcast Channel API to pass the auth token locally to the original tab , if it’s open in the same browser, and otherwise authenticates the new tab. Another advantage of this approach is that there are no authentication cookies.  ↩ Someone who stored the VRF for that email address could continue to match the tlog entries, but since we won’t be adding any new entries to the tlog for that email address, they can’t learn anything they didn’t already know.  ↩ Something cool about tlogs is that they are often agnostic to the mechanism by which entries are added to the log. For example, instead of email identities and verification we could have used OIDC identities, with our centralized server checking OIDC bearer tokens, held accountable by the tlog. Everything would have worked exactly the same.  ↩ To monitor the log, the monitor needs to download it all. This is probably fine for our little keyserver, and even for the Go Checksum Database, but it’s a scaling problem for the Certificate Transparency / Merkle Tree Certificates ecosystem. The inclusion proof guarantees that the public key is in the log, not that it’s the latest entry in the log for that email address. Similarly, the Go Checksum Database can’t efficiently prove the Go Modules Proxy response is complete. age is not really meant to encrypt messages to strangers, nor does it encourage long-term keys. Instead, keys are simple strings that can be exchanged easily through any semi-trusted (i.e. safe against active attackers) channel. Still, a keyserver could be useful in some cases, and it will serve as a decent example for what we are doing today.  ↩ I like to use the SQLite built-in JSON support as a simple document database, to avoid tedious table migrations when adding columns.  ↩ Ok, one thing is special, but it doesn’t have anything to do with transparency. I strongly prefer email magic links that authenticate your original tab, where you have your browsing session history, instead of making you continue in the new tab you open from the email. However, intermediating that flow via a server introduces a phishing risk: if you click the link you risk authenticating the attacker’s session. This implementation uses the JavaScript Broadcast Channel API to pass the auth token locally to the original tab , if it’s open in the same browser, and otherwise authenticates the new tab. Another advantage of this approach is that there are no authentication cookies.  ↩ Someone who stored the VRF for that email address could continue to match the tlog entries, but since we won’t be adding any new entries to the tlog for that email address, they can’t learn anything they didn’t already know.  ↩ Something cool about tlogs is that they are often agnostic to the mechanism by which entries are added to the log. For example, instead of email identities and verification we could have used OIDC identities, with our centralized server checking OIDC bearer tokens, held accountable by the tlog. Everything would have worked exactly the same.  ↩

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Plugins case study: mdBook preprocessors

mdBook is a tool for easily creating books out of Markdown files. It's very popular in the Rust ecosystem, where it's used (among other things) to publish the official Rust book . mdBook has a simple yet effective plugin mechanism that can be used to modify the book output in arbitrary ways, using any programming language or tool. This post describes the mechanism and how it aligns with the fundamental concepts of plugin infrastructures . mdBook's architecture is pretty simple: your contents go into a directory tree of Markdown files. mdBook then renders these into a book, with one file per chapter. The book's output is HTML by default, but mdBook supports other outputs like PDF. The preprocessor mechanism lets us register an arbitrary program that runs on the book's source after it's loaded from Markdown files; this program can modify the book's contents in any way it wishes before it all gets sent to the renderer for generating output. The official documentation explains this process very well . I rewrote my classical "nacrissist" plugin for mdBook; the code is available here . In fact, there are two renditions of the same plugin there: Let's see how this case study of mdBook preprocessors measures against the Fundamental plugin concepts that were covered several times on this blog . Discovery in mdBook is very explicit. For every plugin we want mdBook to use, it has to be listed in the project's book.toml configuration file. For example, in the code sample for this post , the Python narcissist plugin is noted in book.toml as follows: Each preprocessor is a command for mdBook to execute in a sub-process. Here it uses Python, but it can be anything else that can be validly executed. For the purpose of registration, mdBook actually invokes the plugin command twice . The first time, it passes the arguments supports <renderer> where <renderer> is the name of the renderer (e.g. html ). If the command returns 0, it means the preprocessor supports this renderer; otherwise, it doesn't. In the second invocation, mdBook passes some metadata plus the entire book in JSON format to the preprocessor through stdin, and expects the preprocessor to return the modified book as JSON to stdout (using the same schema). In terms of hooks, mdBook takes a very coarse-grained approach. The preprocessor gets the entire book in a single JSON object (along with a context object that contains metadata), and is expected to emit the entire modified book in a single JSON object. It's up to the preprocessor to figure out which parts of the book to read and which parts to modify. Given that books and other documentation typically have limited sizes, this is a reasonable design choice. Even tens of MiB of JSON-encoded data are very quick to pass between sub-processes via stdout and marshal/unmarshal. But we wouldn't be able to implement Wikipedia using this design. This is tricky, given that the preprocessor mechanism is language-agnostic. Here, mdBook offers some additional utilities to preprocessors implemented in Rust, however. These get access to mdBook 's API to unmarshal the JSON representing the context metadata and book's contents. mdBook offers the Preprocessor trait Rust preprocessors can implement, which makes it easier to wrangle the book's contents. See my Rust version of the narcissist preprocessor for a basic example of this. Actually, mdBook has another plugin mechanism, but it's very similar conceptually to preprocessors. A renderer (also called a backend in some of mdBook 's own doc pages) takes the same input as a preprocessor, but is free to do whatever it wants with it. The default renderer emits the HTML for the book; other renderers can do other things. The idea is that the book can go through multiple preprocessors, but at the end a single renderer. The data a renderer receives is exactly the same as a preprocessor - JSON encoded book contents. Due to this similarity, there's no real point getting deeper into renderers in this post. One in Python, to demonstrate how mdBook can invoke preprocessors written in any programming language. One in Rust, to demonstrate how mdBook exposes an application API to plugins written in Rust (since mdBook is itself written in Rust).

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Ankur Sethi 1 months ago

Getting a Gemini API key is an exercise in frustration

Last week, I started working on a new side-project. It’s a standard React app partly made up of run-of-the-mill CRUD views—a perfect fit for LLM-assisted programming. I reasoned that if I could get an LLM to quickly write the boring code for me, I’d have more time to focus on the interesting problems I wanted to solve. I've pretty much settled on Claude Code as my coding assistant of choice, but I'd been hearing great things about Google's Gemini 3 Pro. Despite my aversion to Google products, I decided to try it out on my new codebase. I already had Gemini CLI installed, but that only gave me access to Gemini 2.5 with rate limits. I wanted to try out Gemini 3 Pro, and I wanted to avoid being rate limited. I had some spare cash to burn on this experiment, so I went looking for ways to pay for a Gemini Pro plan, if such a thing existed. Thus began my grand adventure in trying to give Google my money. The name “Gemini” is so overloaded that it barely means anything. Based on the context, Gemini could refer to: To make things even more confusing, Google has at least three different products just for agentic coding: Gemini Code Assist (Gemini CLI is a part of this suite of products), Jules , and Antigravity . And then there’s a bunch of other GenAI stuff that is powered by Gemini but doesn’t have the word Gemini in the name: Vertex AI Platform , Google AI Studio , NotebookLM , and who knows what else. I just wanted to plug my credit card information into a form and get access to a coding assistant. Instead, I was dunked into an alphabet soup of products that all seemed to do similar things and, crucially, didn’t have any giant “Buy Now!” buttons for me to click. In contrast, both Anthropic and OpenAI have two primary ways you can access their products: via their consumer offerings at claude.ai and chatgpt.com respectively, or via API credits that you can buy through their respective developer consoles . In each case, there is a form field where you can plug in your credit card details, and a big, friendly “Buy Now!” button to click. After half an hour of searching the web, I did the obvious thing and asked the free version of Gemini (the chatbot, not one of those other Geminis) what to do: How do I pay for the pro version of Gemini so i can use it in the terminal for writing code? I specifically want to use the Gemini 3 Pro model. It thought for a suspiciously long time and told me that Gemini 3 Pro required a developer API key to use. Since the new model is still in preview, it's not yet available on any of the consumer plans. When I asked follow up questions about pricing, it told me that "Something went wrong”. Which translates to: we broke something, but we won’t tell you how to fix it. So I asked Claude for help. Between the two LLMs, I was able to figure out how to create an API key for the Gemini I wanted. Google AI Studio is supposed to be the all-in-one dashboard for Google’s generative AI models. This is where you can experiment with model parameters, manage API keys, view logs, and manage billing for your projects. I logged into Google AI Studio and created a new API key . This part was pretty straightforward: I followed the on-screen instructions and had a fresh new key housed under a project in a few seconds. I then verified that my key was working with Gemini CLI. It worked! Now all that was left to do was to purchase some API credits. Back in Google AI Studio, I saw a link titled “Set up billing” next to my key. It looked promising, so I clicked it. That’s where the fun really began. The “Set up billing” link kicked me out of Google AI Studio and into Google Cloud Console, and my heart sank. Every time I’ve logged into Google Cloud Console or AWS, I’ve wasted hours upon hours reading outdated documentation, gazing in despair at graphs that make no sense, going around in circles from dashboard to dashboard, and feeling a strong desire to attain freedom from this mortal coil. Turns out I can’t just put $100 into my Gemini account. Instead, I must first create a Billing Account. After I've done that, I must associate it with a project. Then I’m allowed to add a payment method to the Billing Account. And then , if I’m lucky, my API key will turn into a paid API key with Gemini Pro privileges. So I did the thing. The whole song and dance. Including the mandatory two-factor OTP verification that every Indian credit card requires. At the end of the process, I was greeted with a popup telling me I had to verify my payment method before I’d be allowed to use it. Wait. Didn’t I just verify my payment method? When I entered the OTP from my bank? Nope, turns out Google hungers for more data. Who'd have thunk it? To verify my payment method for reals , I had to send Google a picture of my government-issued ID and the credit card I’d just associated with my Billing Account. I had to ensure all the numbers on my credit card were redacted by manually placing black bars on top of them in an image editor, leaving only my name and the last four digits of the credit card number visible. This felt unnecessarily intrusive. But by this point, I was too deep in the process to quit. I was invested. I needed my Gemini 3 Pro, and I was willing to pay any price. The upload form for the government ID rejected my upload twice before it finally accepted it. It was the same exact ID every single time, just in different file formats. It wanted a PNG file. Not a JPG file, nor a PDF file, but a PNG file. Did the upload form mention that in the instructions? Of course not. After jumping through all these hoops, I received an email from Google telling me that my verification will be completed in a few days. A few days ? Nothing to do but wait, I suppose. At this point, I closed all my open Cloud Console tabs and went back to work. But when I was fifteen minutes into writing some code by hand like a Neanderthal, I received a second email from Google telling me that my verification was complete. So for the tenth time that day, I navigated to AI Studio. For the tenth time I clicked "Set up billing" on the page listing my API keys. For the tenth time I was told that my project wasn't associated with a billing account. For the tenth time I associated the project with my new billing account. And finally, after doing all of this, the “Quota tier” column on the page listing my API keys said “Tier 1” instead of “Set up billing”. Wait, Tier 1? Did that mean there were other tiers? What were tiers, anyway? Was I already on the best tier? Or maybe I was on the worst one? Not important. The important part was that I had my API key and I'd managed to convince Google to charge me for it. I went back to the Gemini CLI, ran the command, and turned on the "Enable experimental features" option. I ran the command, which told me that Gemini 3 Pro was now available. Success? Not yet. When I tried sending a message to the LLM, it failed with this 403 error: Is that JSON inside a string inside JSON? Yes. Yes it is. To figure out if my key was even working, I tried calling the Gemini API from JavaScript, reproducing the basic example from Google’s own documentation . No dice. I ran into the exact same error. I then tried talking to Gemini 3 Pro using the Playground inside Google AI Studio. It showed me a toast message saying The chat transcript said At this point I gave up and walked away from my computer. It was already 8pm. I’d been trying to get things to work since 5pm. I needed to eat dinner, play Clair Obscur , and go to bed. I had no more time to waste and no more fucks to give. Just as I was getting into bed, I received an email from Google with this subject line: Your Google Cloud and APIs billing account XXXXXX-XXXXXX-XXXXXX is in good standing at this time. With the message inside saying: Based on the information you provided and further analysis by Google, we have reinstated your billing account XXXXXX-XXXXXX-XXXXXX. Your account is in good standing, and you should now have full access to your account and related Project(s) and Service(s). I have no idea what any of this means, but Gemini 3 Pro started working correctly after I received this email. It worked in the Playground, directly by calling the API from JavaScript, and with Gemini CLI. Problem solved, I guess. Until Google mysteriously decides that my account is no longer in good standing. This was such a frustrating experience that I still haven't tried using Gemini with my new codebase, nearly a week after I made all those sacrifices to the Gods of Billing Account. I understand why the process for getting a Gemini API key is so convoluted. It’s designed for large organizations, not an individual developers trying to get work done; it serves the bureaucracy, not the people doing the work; it’s designed for maximum compliance with government regulations, not for efficiency or productivity. Google doesn’t want my money unless I’m an organization that employs ten thousand people. In contrast to Google, Anthropic and OpenAI are much smaller and much more nimble. They’re able to make the process of setting up a developer account quick and easy for those of us who just want to get things done. Unlike Google, they haven’t yet become complacent. They need to compete for developer mindshare if they are to survive a decade into the future. Maybe they'll add the same level of bureaucracy to their processes as they become larger, but for now they're fairly easy to deal with. I’m still going to try using Gemini 3 Pro with Gemini CLI as my coding assistant, but I’ll probably cap the experiment to a month. Unless Gemini 3 Pro is a massive improvement over its competitors, I’ll stick to using tools built by organizations that want me as a customer. The chatbot available at gemini.google.com . The mobile app that lets you use the same Gemini chatbot on your iPhone or Android . The voice assistant on Android phones. The AI features built into Google Workspace , Firebase, Colab, BigQuery, and other Google products. Gemini CLI, an agentic coding tool for your terminal that works the same way as Claude Code or OpenAI Codex. The Gemini Code Assist suite of products, which includes extensions for various IDEs, a GitHub app, and Gemini CLI. The underlying LLM powering all these products. Probably three more products by the time I finish writing this blog post.

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