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Giles's blog 6 days ago

Fun with low-rank vocab matrices (and a bonus test loss reduction?)

I was nerdsniped ! On the HF discussion page for one of the models I created for my previous post , asked if I'd considered trying out factorised embeddings -- something he's using for his model, Maba, and which was previously used in some other models, like ALBERT . It's a really nifty idea; you use a similar trick to LoRA as a way of reducing the number of parameters used for your embeddings and your output head. And for small models, those can be a disproportionate number of the total. For example, with my 163 million parameter GPT-2 small-style models, the embeddings are about 39 million of them ; the output head is the same size, so with those two taken together, that's half the model just getting stuff in and out rather than actually doing the thinking. Even if you use weight tying, like the original GPT-2 models did, you'll wind up spending 30% of your "budget" on the single matrix shared between embeddings and the output head. Similarly, while larger models have a smaller percentage spent that way, even medium-sized MoE models can wind up spending a lot of the active parameters on embeddings; I calculated that for Qwen 3.6 35B MoE, with 3B active parameters, there were about 1B of them used across both the input embeddings and the output head. So anything that can reduce that -- so long as it doesn't come at a high cost in terms of the model's capabilities -- is worth considering. You save on your parameter count, which either means smaller models and potentially quicker training, or allows you to "invest" the savings in more thinking parameters -- that is, a wider network with a higher embedding dimensionality, or a deeper one with more layers. Andrew reported that the impact of using this trick was minimal: At rank 128 on a 50k vocabulary, the penalty on validation cross-entropy is negligible (typically within +0.02 to +0.04 loss, or <0.5 perplexity delta). Now, that suggests that he's getting a loss of around 2.5 (that being the point where a loss increase of 0.04 implies a perplexity change of 0.5), which is much lower than I typically get with my GPT-2 architecture (3.5 is more like it for me), but still, perhaps I'd get good numbers too? I decided to train a few models to see what happened. The results were interesting! I found that using this trick on both the embeddings and the output head caused a somewhat larger increase in loss than Andrew described -- about 0.07 in my first test, 0.10 in the second, and 0.14 in the third. Those are relatively large, though potentially recoverable from further training or larger models. But more surprisingly, I found that using the trick on the input embeddings only seemed to reduce test loss. Before I'd started, I'd expected it to be less harmful on the input side than it was on the output side, but seeing an improvement was certainly unexpected. Getting loss down by reducing your number of parameters is not what normally happens! So, it's definitely worth digging in a bit. Let's start with the theory: what are these factorised embeddings, and how do they work? Like LoRA, they rely on low-rank factors, so firstly I'll define those. Our models are made up of a large set of matrices -- embeddings, output heads, attention weights, FFN linear layers, and so on. Making them smaller obviously reduces the number of parameters for the model, though you would expect it to come at a cost. The idea behind low-rank factors is that you can replace a larger matrix with two smaller ones that will contain most of the important information. Imagine that you've got a matrix of size m × n . Now, from matrix multiplication, we know that if you multiply an m × r matrix by an r × n one, you'll get a result that is m × n . If we do that, then we've factorised the matrix (in the same way as we might factorise 12 into 3 and 4 because 3 × 4 = 12 ), and the value r is referred to (logically enough) as the rank of the factorisation. If r is sufficiently smaller than m and n , then these two matrices -- the low-rank factors -- combined will contain fewer parameters than the full matrix. Let's make this specific, using the output head of a GPT-2 style model without weight tying. It takes in the embeddings that result from the Transformer layers (after normalisation), and converts them into logits across the vocabulary. For the GPT-2 small size, our embeddings are 768-dimensional, and the vocab size is 50,257 tokens. So when we use a linear layer to do that mapping, it has 768 × 50257 = 38597376 parameters. If we were to replace it with two matrices -- say, one of 768 × 128 and one of 128 × 50257 , then combined they would take up 768 × 128 + 50257 × 128 = 98304 + 6432896 = 6531200 parameters. That is almost six times smaller! So: the idea is that instead of creating our model with an m × n matrix, we create it with a pair, m × r and r × n , and train those instead. If all goes well, that pair will be almost as capable of learning what we want as the full one would have been, so we'll get results that are close enough. The maths and the implementation work out simply, too. For a linear layer with no bias, we can write the matrix multiplication that takes our inputs X and a weight matrix W , and produces an output Z , like this: 1 Now, if we're using a pair of low-rank factor matrices A and B instead of a full one, W , we can say the "virtual" weights we want to use for the calculation are W ′ such that: So that means that our calculation for this neural network is this: Matrix multiplication is associative, which means that you can rewrite that as: ...which hopefully you can see is the same as feeding X through a layer using A as its weights, and then feeding the result through a second one using B . In code, you've taken something like this: ...and replaced it with this: It's probably intuitively obvious, though, that this comes at a cost. If you have fewer parameters then you can store less information, so this part of your model is "dumber". If it's not clear, though, imagine that in the code above was one. You would be taking the (for GPT-2 small) 768-dimensional embedding, converting it to a single number, and then expanding that single number out to a 50,257-dimensional set of logits. Stuff is going to get lost -- the idea behind low-rank factors is that, so long as you choose an appropriate value for r (which is often referred to as the width of the low-rank bottleneck ), you won't lose the important stuff. This works surprisingly well in many cases -- in particular, LoRA, which allows you to fine-tune models that are too hard to fully train on your hardware, or to fine-tune them faster, uses it to easily train something like low rank factor "diffs" to your weight matrices 2 . But for this post, we'll try the simplest version: what if we replace both the input embeddings and the output head -- the vocabulary matrices -- in their entirety with low-rank bottlenecks? Low-rank factorisation of the vocabulary matrices is a bit of a mouthful, so based on the name "LoRA", I decided to call this trick LoRE, for low-rank embeddings. It's not strictly accurate (after all, we're doing it both to the embedding matrix at the start of the LLM and to the output head at the end), and I don't expect it to take off, but I rather like it and will use it in this post :-) The argument behind it is that the embeddings are just a simple lookup table, so are exactly the kind of place you'd expect to be able to make savings by only considering the important parts of our huge matrix; the same might apply to the output head at the end, though personally I was less convinced by this part. It's worth looking into that asymmetry a bit. My intuition was that while embeddings really are a lookup table, the output head is doing something a bit more subtle. It's projecting from the continuous embeddings that come out of our Transformer layers into logits, which we interpret (via softmax) as a probability distribution over possible next tokens. That's a significantly less simple job than mapping from "cat" to the appropriate embedding: as a (simplified) example, if you have an embedding that means something adjacent to "cat", "dog", "gerbil", "household pet" and so on, then projecting that to the appropriate values for the next token is non-trivial. 3 Still, implementation-wise, it was really simple. I decided to extend the PyTorch code that I had from Sebastian Raschka 's book " Build a Large Language Model (from Scratch) ", by adding an optional section to the model config JSON. This would allow us to switch on LoRE mode for the input embeddings and the output head independently, and would specify the rank -- that is, the r in the section above, which says how wide the low-rank bottleneck between matrices A and B would be. So, in code, my first cut changed this: ...to this: ...and this: ...became this: That in itself was a perfectly reasonable implementation of the underlying concept. But it had a problem. I wanted to compare the loss with and without LoRE, and also find out what the effects of input embedding-only and output-head-only LoRE would be. But I've found in other experiments that the initial random weights that my models start with can have a significant effect on the loss that the resulting model gets. The code above would give quite different random weights with different configurations. The fix was simple, though: With that code, so long as there is any LoRE config, we create the same weights regardless of whether it's used in the input embeddings or the output head. I would need to train a baseline which had config for LoRE but just had both and set to there -- that would actually be a normal, non-LoRE model but would have the same initial random weights in any parts that it shared with an equivalent model that did use LoRE. Additionally, Andrew had said that we need to be careful about initialisation: The main failure mode to watch for during training is early gradient spikes: scaling the bottleneck projection initialization to 1 / sqrt(r) keeps variance uniform and ensures training dynamics match the full-rank baseline from step 1. ...and later : Keep standard GPT-2 init on the embedding table, but scale the inner linear projection matrices by std = 1.0 / math.sqrt(128). (The 128 in there is because we were talking about a setup with r = 128 , which is actually what I wound up using.) Now, I was using PyTorch's default initialisation rather than the classic GPT-2 setup, but I went ahead and added a flag to handle the scaling as Andrew described: if it was set, then right at the end of the model creation I applied the initialisation to the low rank factor matrices that were innermost in my model (as in, closest to the Transformer layers): Even though that was not run for every configuration, because it happened right at the end of the model creation, it would not affect the other weights, so I figured that it would be safe. You can see the full diff from the normal GPT-2 code here . Those changes looked correct to me, and I asked GPT-6 Astra (on "extra high" thinking) to confirm that it looked right to it in the context of the discussion, which it did. That done, I double-checked with Andrew , as I really wanted this to be a clean attempt at a repro of his results. He confirmed, so it was time to train some models! I decided that I wanted five training runs to check this out thoroughly: To save time, I decided to train them in the cloud: the first three, and then the last two, in two batches of parallel runs. I used 8x A100 machines with 40 GiB VRAM per GPU on Lambda . I'll break with my normal tradition of giving detailed runthroughs of each training run, as there are (spoiler) quite a lot of them in this post! However, for reproducibility, the commands I ran for each one looked like this: ...with the run ID, , replaced with the appropriate one for the given run settings . Here are the results from the first three runs: the baseline, and the ones with LoRE at both ends, with and without smart initialisation. The model name links go to the uploaded models on Hugging Face. You can see that I had some issues with the last run; there were shortages of available instances, and I wound up starting one in Japan. Connectivity to it from my home in Portugal was really slow and unreliable; when I tried to download the model after the training run, was predicting 10 hours (which at the machine's cost of US$15.92/hour would have been insane). However, I was able to copy it to my PythonAnywhere account quickly, perhaps because that is in a US Amazon datacenter and had better connectivity to Japan, and then I could copy it from there to my machine, again quite quickly. Unfortunately while scrabbling around with that I didn't capture all of the output. Still, the important data at this stage was in the test loss column. As expected, using LoRE made the test loss worse. And smart initialisation did seem to help reduce that penalty. The loss delta, at +0.07, was worse than the +0.02 to +0.04 that Andrew described in his message, but not wildly different -- certainly small enough that it seemed plausible that you could recover it, and perhaps more, by investing the 64M parameter difference in a deeper or wider model. Promising! The cost and time savings were real, too: the run in Japan had its cost bumped up by all of the woes I had copying the model down, so the one is the best indication: US$38.55 rather than US$43.96, so a saving of more than 10%. And while going down from a bit more than 2h30m to 2h14m might not seem like much, with a larger model or a longer training run, an almost 14% reduction could be valuable in and of itself. (And that's not even considering the alternative of "reinvesting" the saved parameters in larger models.) So that was pretty good news! It was time to do the ablation runs to see whether having LoRE just on the token embeddings, or just on the output head, made things any different. I switched the setting for the remaining two models to , and kicked them off. Now, there's a bunch of interesting stuff in there, but let's focus on the test loss. My lab notes for this bit say "well, that's a bit of a shocker!" Quite amazingly -- at least to me -- we got lower test loss with the input-only LoRE, with 3.526610, than we did with the no-LoRE baseline, which got 3.599611. Things get even more interesting when you compare the different versions (excluding the non-smart-init) against each other, in terms of their difference from the no-LoRE baseline (which I'll call the delta for conciseness below). Rounding to 3dps: If you add the two single-LoRE loss deltas together -- the 0.138 and the -0.073 -- you get 0.065, which is surprisingly close to the delta of 0.072 for the dual-LoRE option, just 0.007 away. Could we be looking at two almost independent effects, one for each option of where to apply the LoRE? And was that apparent benefit of the input-only LoRE a real thing? Or was it due to chance -- perhaps we had "good luck" when initialising the LoRE weights, and "bad luck" when initialising the non-LoRE ones. If there was an overlap between the best loss you could get with LoRE and the worst loss you could get without, then this result could be within the noise. I decided I was going to do a second batch of training runs to see if I could get evidence one way or the other. But before we get into that, it's worth taking a look at some of the other numbers in those two tables. Firstly, let's take a look at the training time. The training machines looked like they were running at close to 100% during all of the runs, so tentatively let's assume that time taken is proportional to compute. Unsurprisingly, the baseline run came in with the longest time, at 9,339s. And equally unsurprisingly, the full-LoRE run that we have timing for was much faster, at 8,057s. It certainly makes sense that replacing big matrices with pairs of (much) smaller low-rank factors will mean fewer calculations on both the forward and the backward pass, meaning that the training run is faster. There was something that initially surprised me with the single-LoRE runs. Using LoRE on the output head sped things up quite a lot, but input embedding LoRE had much less effect. However, after thinking about it a bit, it became clear what was going on. An embedding module in a network is actually really quick to run, both on the forward and backward pass. While I like to think of embeddings as conceptually being a projection of a one-hot vector in vocab space into embedding space via a matrix multiplication, and I still think that's an excellent model of interpreting what's going on, in practice it's just something much more like an array lookup, with almost zero cost. Likewise on the backward pass, we only need to work out gradients for the chosen embedding, so that should be faster too. Indeed, with that all in mind, it might not have been surprising if input-only LoRE had turned out to be a bit slower than the baseline, as it, at least, needed to do forward and backward passes through actual matrices, even though they were fairly small. Doing another of those delta tables, the time difference in seconds (using the "ordinary init" run for the both-on number): You can see that if you add the delta of the single-LoRE options, you get -1,251, which is pleasingly close to the delta of the dual-LoRE one. And that is much less surprising than the loss result -- with training compute, you really would expect improvements like this to be additive. Finally, there was something interesting in the initial training loss. Because my training code prints out the loss it got on the first global step, I decided to note it down. There's something in this data that will become important later: in both of the training runs with LoRE on the output head and smart initialisation, it went up noticeably. Normally, the initial training loss on my GPT-2-style models is a bit less than 11. This fits in well with the architecture; you'd expect that a model that was predicting tokens purely randomly (well, strictly, uniformly) would have a perplexity roughly equal to the vocab size; the vocab size for the GPT-2 tokeniser is 50,257, and loss is the natural logarithm of perplexity, so that implies a loss of roughly 10.82. But both of those smart-init models with LoRE on the output head had an initial loss about one point higher, at 11.865 and 11.817, meaning that they were significantly worse than random. What might be going on here? We'll find out later :-) Anyway, for now, that was an interesting set of results. I wanted to dig in more; in particular, would that surprising improvement of test loss for the input-only LoRE model reproduce if I tried again with a different random seed? I had two working hypotheses at this point; either A solid and simple test to try to distinguish between these was just to train another set of models, with a different random seed. If I got the same kind of results, input LoRE beating non-LoRE, that would be evidence in favour of (2) rather than (1), whereas if things changed, it would work the other way. I decided not to bother with a non-smart-init model this time, so I set things up for four new models. My training code uses a random seed of 42 by default, but allows it to be overridden in the config file, so I decided to use 123 for these, and created: Once I had the config together, it was time to train the models -- but it was hard to get available instances on Lambda. I wound up having to use my script to start things up, but after a few hours I got an alert on my phone that I had instances running, and could kick things off. The second training run with input-only LoRE came in with a test loss that once again beat the baseline, so I did the full set of runs, and here are the results: Another matrix of deltas against the baseline: So, once again, input-only LoRE improved test loss (though at -0.013, it was a smaller effect than the -0.073 the previous set of models had). And once again, the sum of the input-only and output-only LoRE deltas was close to the combined-LoRE model's -- indeed it was even closer, 0.102 being just 0.001 away from 0.101. The training time results and the higher initial training loss for models with LoRE on the output head also held up, which was good to see. So at this point it was time to wrap things up -- or was it? While I was doing the second batch of training runs, I started writing up this experiment. As I normally do , I ran a draft past various AI models, and Claude Opus 5.5 on max thinking spotted something. Andrew had written : Keep standard GPT-2 init on the embedding table, but scale the inner linear projection matrices by std = 1.0 / math.sqrt(128). I'd interpreted that "inner" as meaning that the "innermost" matrices in our low rank factors -- the ones that were closest to the Transformer layers -- should be initialised as he said, with values drawn from a normal distribution with a standard deviation of 1 / r , and a mean of zero. So that implied the second matrix in the embeddings, and the first one in the output head: Claude felt that it should be the second low-rank factor in both cases -- that is, we should change It made a very good case for doing things that way, and digging into that is useful because it changes this "smart" initialisation from being a blindly-implemented "secret sauce" into something actually meaningful. Let's imagine that we're not using LoRE, and consider the output head only. It is a PyTorch , which means that: ...the values are initialized from 𝒰 ( − k , k ) , where k = 1 in_features So, we have a matrix, which we can call W , that is full of numbers drawn from that probability distribution -- a uniform (flat) distribution within the range from − k to k . For LoRE, we're replacing W with two separate matrices, which we'll call A and B : Let's say that W is shaped m × n , which means that A is m × r , and B is r × n . We want the combined effect of those two matrices W ′ = A B to start off with random numbers that look like -- in terms of their randomness -- the full matrix W . But what do the numbers actually look like in our "virtual" matrix W ′ with the code above? On the face of it, it sounds unlikely to be anything like the probability distribution for W . After all, W was initialised based on a formula that used in_features . You can see that A will have been initialised using the same distribution, as it has the same number of input features, but the number of input features for B is r , so its random numbers will have been based on that instead. Their combination into W ′ will presumably blend something from each of those two different probability distributions. We can actually work out what that will be. Maths incoming: click here to skip . Per Wikipedia , the variance of the product of two independent random scalar variables (so, not matrices, but we'll come back to that) X and Y is this: ...where σ X 2 is X 's variance, μ X is its mean, and likewise for Y . Now, let's say that the mean is zero for both X and Y (like it is for PyTorch's distribution for the initial weights, 𝒰 ( − k , k ) ). That simplifies the above: How does that apply to our matrices? Well, let's write out how we calculate W ′ by multiplying A and B , assuming that m = 3 , n = 4 , and r = 2 : By normal matrix maths, the value at position 0 , 0 in the result will be the dot product of the zeroth row in the first matrix (taken as a vector) and the zeroth column in the second: We can rewrite that for an arbitrary low rank bottleneck r like this: ...and even more generally, we can define all of the items in W ′ like this: So, every element is the sum of r products, where each product multiplies a number from whatever distribution was used to initialise A 's elements by a number from whatever was used for B 's. Using the notation from above, A 's elements have variance σ A 2 and B 's have variance σ B 2 . So that means that the variance of each component of that sum is σ A 2 · σ B 2 . There are r of them; if you add together independent variables, their variances add, so that means that the variance of the elements of our resulting W ′ is: Now, let's remember that our matrix A actually already was initialised in the way we would have liked our non-LoRE version W to have been done. The distribution of the random initial weights for an is dependent entirely on the number of input features -- and of course A has the same number of input features as W . So how can we make Var ( W ′ ) be σ A 2 ? We need to make σ B 2 = 1 / r , and then the equation above drops out like this: Now, let's look at the code (with Claude's correction in there): We're setting the weights on our B matrix to be drawn from a distribution with a standard deviation of 1 / r . SD is, of course, the square root of the variance (which is why we've been using things like σ B 2 with that squaring in there to represent variance -- σ on its own is used for standard deviation). So the variance of the matrix after we've called that on it will be 1 / r as desired, and W ′ will have random initial weights with the same variance as W would have done if we'd created it directly. That's rather satisfying :-) So that's the output head. How about the input embeddings? They're s, which are initialised somewhat differently: initialized from 𝒩 ( 0 , 1 ) That's a normal distribution, with a mean of 0 and a variance (and thus also a standard deviation) of 1. The maths above applied to any distribution with a mean of zero -- so the smart initialisation should work for those too. Our A matrix would have the correct variance, 1, and a mean of zero, and therefore so would A B , given that (for embeddings) B already had the smart initialisation. So does that mean that our W ′ "virtual" matrix created by multiplying A and B is identical in terms of randomness to the matrix W that it's replacing? Well, not quite. Remember that the output head had numbers drawn from a uniform distribution, and the embeddings from a normal distribution. We're multiplying the A matrices that are the "input" low rank factor matrices by B s that are initialised from a normal distribution. The variance of the A matrices is maintained by the smart initialisation, as we've shown, and the mean (being zero) is going to come through cleanly too. But the shape will change. I did a bit of digging around and it started to feel like a bit of a rabbit hole, but the one thing I was able to be certain of is that a variable drawn from a uniform distribution multiplied by another from a normal distribution gives a result that is neither uniform nor normal, and that normal times normal gives something called a normal product distribution (or also a product-normal distribution), which is also neither uniform nor normal. Remember that each element of our W ′ matrix is defined by this: So each one is a sum of r numbers, each of which is drawn from a product normal distribution (for embeddings) or from whatever the uniform × normal distribution is called. What shape would that be? Again: rabbit hole, but the central limit theorem states ( per Wolfram MathWorld ): the normalized sum of independent random variates with finite variances approaches a normal distribution So, for large r , we can say that the result is going to be close to a normal distribution. 4 What the smart initialisation does is make sure that our "virtual" weights, W ′ = A B , have the same mean and variance as the ones we're trying to approximate, W . But it won't keep the same shape for the distribution that they're drawn from -- certainly not for the output head, which goes from uniform to normal-ish. The embeddings are less impacted (though not completely untouched) because they were normal and are still almost so. Still, it's a neat trick! Was it what Andrew had actually meant? Having convinced myself that the smart initialisation code I'd been using was wrong, I decided to check with Andrew. My first thought was that perhaps he was using weight tying -- remember, with weight tying, the embedding matrix is just re-used in transposed form for the output head. That would mean that the embeddings would be W ′ = A B , and you'd use the smart initialisation on B , but the output head -- being that transpose -- would use W ′ T = B T A T for the same B and A , so the smart initialisation would be the other way around. But as it turned out, it was just a miscommunication . He said that he keeps his weights untied, and had been using the word "inner" in "inner linear projection matrices" to mean something different to what I'd interpreted it as. He confirmed that the real smart init code should always be applied to the second matrix, as the maths (and Claude) suggested. So that was a simple change. You can see the diff here , but in short: Now, the good news (both for my own sanity and my wallet) was that testing this fix was not going to require another four training runs. My training code is essentially deterministic -- no dropout or any other randomness is used after the model is created 5 . Because the smart initialisation code was at the end of the model creation, and smart initialisation of the output head was right at the end, that meant that changing it would not affect any other code that used randomness. So if I did two training runs -- with LoRE switched on for the output head only, and with it switched on for both embeddings and output -- then those would be "compatible" with earlier training runs that had no LoRE at all, and that only had it on the embeddings. You can see the configs here (with no explicit random seed, so they'd use my default of 42): Here are the results; I've put them into a table with their counterparts from the original seed-42 training run: Before we look at the test loss, check out that "Initial training loss" column. The weirdness that I'd noticed earlier, where that value was much worse than uniform, is gone! That actually makes some kind of sense. With the smart initialisation on the wrong part of the low-rank output head, our variance was completely wrong, and so the model was initially producing particularly bad results. 6 So that was promising! But the test loss results were even more interesting. In our original seed-42 results with the incorrect init, the output-head-LoRE-only model got a test loss of 3.737128, but here we got 3.785209. It was actually worse with the corrected smart initialisation! And that carried through to the result for models with LoRE on both input and output; the original model with the incorrect init got 3.671585, and here we got 3.741209 -- an even bigger worsening. Combining those two together, an image I like in terms of the loss landscape is that previously we were starting on a mountain that was near a deep valley -- we started with high loss, but there was a low-loss place nearby. But with the smart initialisation fixed, we were now starting on a hill with a less-deep valley nearby. So what happens if we do one of our test loss delta tables showing how each model performed against the baseline? If we add up the effect of the input-only test loss delta of -0.073 and the output-only delta of 0.186, we get 0.113 -- quite different to the delta of 0.142 that we actually got with them both. That nice "additive" property that we had, where the delta from input LoRE only plus the delta from output LoRE only summed to almost exactly the same as the delta for LoRE on both ends, appears to have gone away :-( Still, an interesting set of results! Let's bring everything together. Here are all of the training runs together in one table. I've removed the columns for the cost, the end training loss, and the training time to keep things manageable, and I've sorted it by the test loss. A few things stand out: So what does that all tell us? The numbers above suggest three things -- with important caveats below: Even more caveated, we might say: Neither the larger of the two apparent improvements from input-only LoRE nor the worsening from output-only were small -- when I was trying various interventions into my GPT-2 models (like removing dropout, scheduling the learning rate, gradient clipping and so on), many of them had a similar or smaller level of effect. Similarly, training a comparable non-LoRE model on double the number of Chinchilla-optimal tokens (ie. on 40 per parameter rather than 20) improved loss by 0.09, while investing the same compute resources to do a Chinchilla-optimal training run on a larger model improved it by roughly 0.13 over the baseline (see this post for the raw numbers). So, is input-only LoRE One Weird Trick To Improve Your Test Loss? Of course not. Are output-only LoRE and both-ends LoRE Bad Things? Not necessarily. The first thing to remember is that this is a really limited set of experiments. Although I wound up spending about US$450 on training runs (I really should be investing in building my machine up as a better training box to save on Lambda costs), I was only able to test: Even in order to say anything definitive about whether LoRE works for this specific architecture, it would be necessary to do tests over a much larger number of random seeds and different lengths (in terms of number of tokens) of training run: To get solid results on whether smart initialisation is a good thing, you'd need to try a number of seeds to see whether the poor performance of the "correct" smart initialisation against the "incorrect" one worked out as a general result. And if it did, why might that be? Beyond that: the apparent improvement in the loss from input-only LoRE is something that came up in these experiments. Maybe it would turn out to be a real thing, maybe not, but the real benefit that Andrew mentioned in his original message was about something quite different: LoRE saves a lot of parameters and only (as he said later) costs a small amount of loss. If you reinvest those saved parameters in a deeper network with more layers, you can hopefully gain back the loss that you, um, lost, and then some -- and wind up with a more capable model. So important further tests (which I think I will do, but later on) would be: It's worth noting that in my Chinchilla experiments, I scaled up the model by 2 . That's about a 41% increase, which is quite close to the 39% we save by using LoRE on both ends. So if the loss improvements carried over, perhaps it might work? On the other hand, those models were also trained on 41% more tokens, so... So there's a huge number of further experiments that would be needed to build out these results -- and that's even before we move beyond this specific architecture. Andrew reports solid results with his setup, and his model Maba does appear to be quite different. But would it extend to a Qwen-style one? Or Mistral? Next, there's what happens with scaling. LoRE is valuable in small models because embeddings and the output head are such a large proportion of the total parameter count. My guess is that it would become increasingly unhelpful as models got larger. To take an extreme case, looking at the Kimi K3 paper , it has a vocab size of 160k, and its hidden dimension (which I'll assume is what the embeddings are using) is 7,168. That means that the embedding layer uses something like 1,146,880,000 parameters, as does the output head, for a total of ~2.3B parameters (assuming they're not using weight tying, which seems likely for a monster like this). It has 104.2B active parameters per token, so if it were to use LoRE you'd save something of the order of 2% of the active parameter count. But for purposes of "reinvesting" in further layers, the real comparison point is the total number of parameters, not the active ones, and with about 2.8T total, the number used for embeddings is less than 0.1%. It doesn't sound like it would help much. Unless, of course, the benefit of the input-only LoRE loss improvement actually turned out to be a real thing, and didn't weaken as the model was scaled... But anyway, that's speculation. I think a good set of experiments for that would be to try the same training runs as before, but for larger models. Maybe just trying GPT-2 medium and large sizes would be a good start? Lots of work. If I had access to Google's resources then I think I'd be kicking off a bunch of training runs. OTOH if I had access to Google's resources I probably wouldn't be blogging about this... I think that there's one thing I'm going to take away from this. LoRE is an interesting intervention -- and I think is well worth trying out on any new model where vocab matrices are a significant fraction of the parameters, just to see if it helps. I'll be doing that in future. And if anyone else out there wants to try LoRE-style training runs, I'd love to hear from you. Any results, positive or negative, would be really interesting. Thanks for reading! And many, many thanks to Andrew for suggesting this as an idea to play around with -- it's been fun :-) This is often written Z = X W T to reflect the fact that weights are generally kept in d out × d in format, at least in PyTorch, but that's really an implementation detail .  ↩ I'll write more about LoRA in the future -- I've been playing with it a bit -- but essentially, what you do is: For each matrix that you want to train (which might not be all of them -- for fine-tuning, people often only train a subset), you add on a low-rank adapter , which is a pair of matrices like our A and B above, and which sits "beside" the existing one, like this: Gradient tracking is enabled on the matrices in the low-rank adapter, and its results are added to the results of the original weights. The idea is that the adapter pair learns to create a "diff" to the results of the original weights, which allows a certain amount of training of that part of your model. And it works surprisingly well. Again: I'll write about this in more depth in the future.  ↩ When I was discussing this with an LLM -- sadly I've forgotten which -- the softmax bottleneck came up here, as it did in later correspondence with Andrew. That's a reference to this paper . I'll need to work through it carefully at some later point, but the relevant point appears to be that the rank (which we can loosely read as how many independent ways the predictions can vary from one context to another) of the values coming out of softmax is limited by the rank of what is going into it. If we force our (in my GPT-2 small case) 768-dimensional embeddings through a 128-dimensional bottleneck, we're reducing their inherent rank, and it's that reduced rank that will drive softmax's, so the results will be worse than what you'd get with the original 768. Seems logical enough. Linkrot-proof reference: Yang, Z., Dai, Z., Salakhutdinov, R., & Cohen, W. W. (2018). Breaking the softmax bottleneck: A high-rank RNN language model. In International Conference on Learning Representations (ICLR) .  ↩ Thanks to Claude Opus 5.5 for pointing me at the central limit theorem. I definitely need to brush up on my statistics, and I'm sure I knew about it at some point back in the 90s...  ↩ Modulo CUDA jitter of various flavours -- though even then, a while back I did two training runs on 8x A100 machines with the same config, months apart, and the Safetensors files were identical bit-for-bit. So I think the effect of that in my setup is minuscule.  ↩ The temptation to dig into the details of why, and exactly what the mistaken smart initialisation did to the variance of the "virtual" weights W ′ , is strong. But I will resist.  ↩ When I ran a draft of this post past my editorial board of LLMs , Grok made the point that perhaps the apparent benefit of having LoRE only on the input embeddings might have been due to that model seeing significantly more tokens per parameter. It's an interesting thought! If we do the maths, the input-only and output-only LoRE models were trained on ~25 tokens per parameter and the both-ends ones were trained on about 33, vs 20 for the non-LoRE ones. However, in my Chinchilla experiments, I only got 0.09 improvement in loss by training on 40 tokens per parameter rather than 20. Now, I'm sure that gain mostly happened early in the run -- overtraining is subject to diminishing returns -- but the result of 0.073 for input-only LoRE would imply that almost all of it happened in the first extra 5 tokens per parameter, which seems dubious. Those are different training runs with different seeds, and were using my JAX code, which has a better baseline loss than my PyTorch code, so we can't compare them directly. But it feels unlikely that the effect would be drastically different. Also, if we use reductio ad absurdum on Grok's logic, it does sound a bit weak -- if we get rid of parameters but train on the same number of tokens, the higher tokens per parameter will not in general counteract the damage from the smaller model. Imagine, for example, removing 32M parameters worth of Transformer layers and training on the Chinchilla-optimal tokens for the size of the model before you did that. I'm not going to run the experiment, but I'd be... somewhat more than surprised if it improved the model.  ↩ A baseline. As I said earlier, the LoRE stuff would have an effect on the randomness used for the initial weights, so I wanted to train one model with LoRE "enabled" but not actually used. Config: , . A model trained with LoRE enabled on both the input embeddings and the output head, but without the smart initialisation of the low-rank matrices. This was simply to confirm to myself that it was required. Config: , . A model with LoRE on both ends, with the smart initialisation -- essentially, what Andrew had described. Config: , . A model with LoRE on the input embeddings only, using smart initialisation if that had turned out to be necessary, and not using it if it had not. I deliberately made the config invalid, using instead of or for the flag, so that I'd be reminded to fill those in later. Config: , . A model with LoRE on the output head only, with smart initialisation the same as with (4). Config: , . What we were seeing was due to luck of the draw with the random initialisation of the weights -- good luck with LoRE, bad luck without. There really was an effect here: input-only LoRE actually improved test loss in the resulting model. A new baseline with no LoRE. Config: , . An input-only LoRE. I wanted to run this one as the first non-baseline one, because if it got worse (higher) loss than the baseline, then it would look rather like the effect I'd previously seen was due to random weight initialisation, so I might want to consider stopping there. Config: , . If I decided to continue, the next one would be LoRE on both the input and output sides. Config: , . Finally, I'd do an output head LoRE only one. Config: , . I changed the config parameter so that instead of just being or , it could be , for the original incorrect interpretation, or for the fixed one. I updated all of the existing training configurations that had set to to be . I added on yet more config for some more training runs to test this, setting to the new option. Both: , . Output-only: , . The input-only LoRE models lead the pack -- though it's a close-run thing, and the gap between and could well be within the noise at 0.013. In that post, I trained three models with different weight initialisation seeds, and got models that differed by up to ~0.017. The two models with the corrected smart initialisation are the worst -- even worse than the model with no extra initialisation, . LoRE can reduce the number of parameters in these models dramatically even if only applied to the input embeddings, and that comes with potentially an improvement in test loss (0.073 in one test, 0.013 in a second). When applied to the output head, it has the same size of reduction in the parameter count, and causes a worsening in test loss (over three tests, 0.115, 0.138 and 0.186). The results with LoRE on both ends seemed to come in somewhere around (in two of the three cases, very close to) the sum of the input-only and output-only deltas, though it doesn't seem to be a strictly additive relationship. The mathematically cleaner implementation of what I've been calling "smart initialisation" actually worked worse than the mistaken version I used originally, though this was only tested with one seed, so the evidence is much weaker. Against my own GPT-2 style model (important differences from the original: no weight tying or bias on the QKV weights, no dropout, and PyTorch default weight initialisation). With two seeds. With two options for smart initialisation, and one of them only with one seed. Over a number of tokens chosen to be Chinchilla-optimal for the baseline non-LoRE model. 7 Maybe the effect was just limited to the seeds I tested? Two is better than one, but three would be better still, four even better, and so on. Maybe the effect disappears in longer training runs? Perhaps LoRE causes some kind of loss in capacity that stops models from improving past some point. Intuitively that could make sense -- if you imagine training as being in some sense compressing information from the training data into the parameters, having fewer parameters is clearly going to limit your capacity in some sense. Isoparameter tests: Input-only LoRE saves 32M parameters, so what happens if you "spend" that on extra layers? Or alternatively make the model wider (in terms of its embedding dimension)? Or some combination of both? Output-only LoRE likewise. And, of course, both input and output. Isocompute tests. Notably, input-only LoRE doesn't save you much compute, as I mentioned earlier -- embedding layers are cheap in terms of compute, if not in terms of parameters. So here perhaps it would be more interesting to see what happens if you did output-only LoRE, but trained the model on more data so that the total training compute budget matched. Would you gain back the lost loss? This is often written Z = X W T to reflect the fact that weights are generally kept in d out × d in format, at least in PyTorch, but that's really an implementation detail .  ↩ I'll write more about LoRA in the future -- I've been playing with it a bit -- but essentially, what you do is: Freeze all of the matrices that make up your parameters (eg. in PyTorch, set to ). For each matrix that you want to train (which might not be all of them -- for fine-tuning, people often only train a subset), you add on a low-rank adapter , which is a pair of matrices like our A and B above, and which sits "beside" the existing one, like this: Gradient tracking is enabled on the matrices in the low-rank adapter, and its results are added to the results of the original weights. When I was discussing this with an LLM -- sadly I've forgotten which -- the softmax bottleneck came up here, as it did in later correspondence with Andrew. That's a reference to this paper . I'll need to work through it carefully at some later point, but the relevant point appears to be that the rank (which we can loosely read as how many independent ways the predictions can vary from one context to another) of the values coming out of softmax is limited by the rank of what is going into it. If we force our (in my GPT-2 small case) 768-dimensional embeddings through a 128-dimensional bottleneck, we're reducing their inherent rank, and it's that reduced rank that will drive softmax's, so the results will be worse than what you'd get with the original 768. Seems logical enough. Linkrot-proof reference: Yang, Z., Dai, Z., Salakhutdinov, R., & Cohen, W. W. (2018). Breaking the softmax bottleneck: A high-rank RNN language model. In International Conference on Learning Representations (ICLR) .  ↩ Thanks to Claude Opus 5.5 for pointing me at the central limit theorem. I definitely need to brush up on my statistics, and I'm sure I knew about it at some point back in the 90s...  ↩ Modulo CUDA jitter of various flavours -- though even then, a while back I did two training runs on 8x A100 machines with the same config, months apart, and the Safetensors files were identical bit-for-bit. So I think the effect of that in my setup is minuscule.  ↩ The temptation to dig into the details of why, and exactly what the mistaken smart initialisation did to the variance of the "virtual" weights W ′ , is strong. But I will resist.  ↩ When I ran a draft of this post past my editorial board of LLMs , Grok made the point that perhaps the apparent benefit of having LoRE only on the input embeddings might have been due to that model seeing significantly more tokens per parameter. It's an interesting thought! If we do the maths, the input-only and output-only LoRE models were trained on ~25 tokens per parameter and the both-ends ones were trained on about 33, vs 20 for the non-LoRE ones. However, in my Chinchilla experiments, I only got 0.09 improvement in loss by training on 40 tokens per parameter rather than 20. Now, I'm sure that gain mostly happened early in the run -- overtraining is subject to diminishing returns -- but the result of 0.073 for input-only LoRE would imply that almost all of it happened in the first extra 5 tokens per parameter, which seems dubious. Those are different training runs with different seeds, and were using my JAX code, which has a better baseline loss than my PyTorch code, so we can't compare them directly. But it feels unlikely that the effect would be drastically different. Also, if we use reductio ad absurdum on Grok's logic, it does sound a bit weak -- if we get rid of parameters but train on the same number of tokens, the higher tokens per parameter will not in general counteract the damage from the smaller model. Imagine, for example, removing 32M parameters worth of Transformer layers and training on the Chinchilla-optimal tokens for the size of the model before you did that. I'm not going to run the experiment, but I'd be... somewhat more than surprised if it improved the model.  ↩

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Giles's blog 2 weeks ago

Why do OpenAI's GPT-2 weights beat mine? Part five: data quality

When I finished learning how to build an LLM from scratch , I was left with a mystery: my own models were not as good as OpenAI's original GPT-2 models, despite being based on the same architecture. My models all had 163M parameters, and followed the design from Sebastian Raschka 's book " Build a Large Language Model (from Scratch) ". That meant that they were pretty much the same as the setup for the OpenAI GPT-2 "small" instance, except that they did not use weight-tying or bias on the QKV matrices. Weight-tying means that you re-use the initial embedding matrix as the output head at the end, and using it means that GPT-2 small saved quite a few parameters -- it was 124M rather than 163M -- at, at least in my own experiments, a cost in quality ; similarly, while I found that QKV bias made a tiny improvement in loss terms , I'd felt it was likely within the noise. But GPT-2 small consistently beat my models on an instruction fine-tuning (IFT) task -- also adapted from Raschka's book. That test fine-tunes the model on a subset of the Alpaca dataset, until validation loss starts rising, and then runs a test set through the resulting model. The responses to the test set questions are stored, and then I run all of the responses from all of the models under test past GPT 5.5 in one go to get an aggregate score; more details here . GPT-2 small always did better than any of my models on this. Additionally, it did surprisingly well on a simpler eval -- one that just measured the cross entropy loss it got on a test set. It scored close to my own best models, and better than many of them. What made this result particularly interesting was that the test set in question was a split of my own training data; my models would not have seen it when training (at least, in theory), but it seems likely that it would be much more similar to their own training data than it was to OpenAI's. I've checked two things while probing this mystery: The next thing I wanted to look into was the training data. The exact dataset that the various GPT-2 models were trained on has never been released; all we know about it is from the paper , where they say: [W]e created 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. This can be thought of as a heuristic indicator for whether other users found the link interesting, educational, or just funny. They called it "WebText". There is an OpenWebText that tries to replicate it, but although they tried to follow the same procedure as the original, there's no guarantee that it is all that similar. By comparison, I'd normally been training against FineWeb . While this is a general web-scraping dataset, without the "curation" provided by using only stuff that was linked from upvoted Reddit posts, it has been refined to remove any obvious junk. I had felt that it was pretty much equivalent. But what if I were wrong about that? I decided to see if I could get better models by using better data. Here's a table of all of the models I've been comparing to date. The "Test loss" column shows how well the model in question did on that held-back cross entropy loss evaluation. The "IFT epochs" column shows how many epochs of fine-tuning the model needed before its validation loss started rising, the "IFT score" the score that GPT 5.5 gave the model's responses to the test set of my Alpaca data, and the "IFT rank" the model's rank in terms of that score. The OpenAI small model is in there in bold, and I've also included the OpenAI medium model for comparison purposes. You can see that the OpenAI small model did pretty well in terms of the test loss, when you consider that it has 39M fewer weights than my models and was being tested against a dataset that differs more from its likely training data than it does from my own models'. Additionally, the specific models that did better than OpenAI's small one were all trained with JAX rather than PyTorch -- my hypothesis for that is that it's a result of the JAX ones getting better initial weights by pure chance. But the big difference was in the IFT score. In the specific run that gave the results in this table, the OpenAI small model got 26.00 -- the closest of my own models was more than 4.5 points lower, at 21.46. This difference was consistent over all of my other test runs. The GPT-2 small model was always ahead of mine. (GPT-2 medium, of course, beat GPT-2 small and all of my models, but given that it is twice the size of mine, that's not a big surprise.) Now, quite some time ago, I had tried looking into data quality as a lever to pull for model performance. At the bottom of the table, with the worst test loss of all models, you can see two models: These two were (as you might guess from the names) trained on the FineWeb-Edu dataset, which includes just the most "educational" data from FineWeb. They scored very badly on the test loss score. Given that the test dataset is from FineWeb, that's not a big surprise -- as I've written previously: If you train a model on Jane Austen and then evaluate against Chuck Tingle , then you're not going to get amazing results. But again, GPT-2 had the same issue, and did perfectly well on the test loss eval. On the other hand, while these FineWeb-Edu models' performance on the IFT eval wasn't stellar -- there are plenty of my other models ahead of them -- they did seem to punch above their weight. Consistently across all of the IFT evals I've done, they have scored higher than many of the others -- despite their poor loss on the test eval. Additionally: they were amongst the first models that I trained, before I'd spent time learning about how to optimise my hyperparameters and training loop . They did not use gradient clipping, they did use dropout, their batch size was just "whatever I could squeeze into the GPU", and I didn't set the learning rate to the right kind of value or schedule it over the course of the training run. So maybe a new training run on FineWeb-Edu plus my training improvements would help? And maybe some other tweaks to the training data would be worth looking into? I decided to see what would happen if I trained some models with better-quality data. Specifically, I would train models with my current optimised loop and hyperparameters on four different datasets: I would train each model on 3.2B tokens of the chosen dataset; that's the Chinchilla-optimal amount for my 163M-parameter models. If there were any interesting results, then I might consider doing overtrained models later on. I decided to be at least vaguely scientific about this, and to pre-register some predictions: Here's how things turned out. I already had a dataset based on FineWeb-Edu ready to go, from when I trained those two original models. It is just the 10B-token sample of the original dataset at the time I generated it last December, formatted appropriately for my training script (details on the dataset card). I kicked off a training run with my JAX code (which I've been using for the other posts in this series): ...and just less than 40 hours later, I had a model: I converted the saved JAX safetensors file from the last checkpoint into a format that would be compatible with my PyTorch eval code, and ran my smoke test: how would it complete the sentence "Every effort moves you"? That was nice and coherent -- if unusually religious! -- so that was promising. I ran the test eval: That was pretty good, putting it at a better test loss than all of the models I had trained without optimised hyperparameters, and worse than all of the ones I had trained on FineWeb with optimised hyperparameters. So that fit in with my prediction that it would be better than the old FineWeb-Edu models; the fact that it was also better than the non-optimised training runs with FineWeb seemed sensible enough that I felt silly for not having predicted that it would have fallen exactly there :-) I decided to leave the IFT eval until the end so that I could check all of the models from these experiments together, so it was time to upload this one to Hugging Face , and move on to the next model. I put together a new repo with a script to prepare datasets specifically for my training setup . You provide it with config that specifies some source datasets along with information about how to process them and how to mix them together, and it uploads a new dataset to Hugging Face Hub with the required characteristics. For example, for the 50:50 FineWeb to FineWeb-Edu split, the config looked like this: The way the script works is pretty simple: it works out (based on those s and the ) how many tokens it wants from each source dataset, shuffles the items in the sources, then it loops until it has the desired number of tokens or more stored in an output. In the loop, it works out which source is currently most under-represented, grabs an item from it, tokenises it, and adds it to the output. Running it with that 50:50 config seemed to work fine: So we had almost-perfect 50:50 balance between the datasets, and it saved this dataset on Hugging Face . I ran a script to double-check that it looked sane , and it did, so it was time to spin up a training run: That was running on , my normal workstation, and I kicked it off in parallel with the "curated" model training run below on my training box, but I'll keep the runs separate for the purposes of this writeup. When this had been running for an hour or so, our power went out. My guess is that having the tumble dryer running, the car charging, the kettle boiling, the electric hob switched on, and two machines doing training runs is a bit too much for our electrics... which might be a problem in the future, especially if (as planned) I make a multi-GPU machine. However, as things stand, I was able to kick it off again after switching the circuit breaker back on, and things held up. Again, about 40 hours later: (Note that the numbers reported at the end of a restarted run like this only include what happened after the restart.) I converted it to PyTorch-compatible tensors, and did the smoke test: Looking good! Time for the loss test: That was almost in keeping with my prediction that it would do worse than the JAX FineWeb-only models, except that it was better than the worst of those, "JAX, no MHA bias, with dropout": it was actually better than I predicted. So, a promising model. Time to upload it to Hugging Face -- and now let's move on to the next one. With my dataset-preparation script, this was easy enough to set up: Running that worked nicely: One thing that is worth noting in that output is the "6 iterators" for the Simple English Wikipedia. If a source dataset runs out of items while we're building up the results in this script, we start iterating over it again (with a different seed for the shuffle so that the ordering is different). The "6 iterators" means that it needed to do that 6 times -- the original creation of the iterator at the start of the script, and five more. So that means that the Simple English Wikipedia is repeated (oversampled) somewhere between five and six times in the dataset. That's not a bad thing! From what I've read, it's actually quite standard to oversample highly educational content in LLM training datasets. And anyway, the dataset the script generated was 10B tokens, of which we're only using 3.2B for the training run in this post, so it would only appear somewhere between one and two times. The repetition would likely only really cut in if and when we did an overtrained model on the dataset. Anyway, I ran my check against the uploaded dataset -- the first few items were clearly from FineWeb, FineWeb-Edu, and the Simple English Wikipedia. It was time to kick off a training run: Again, this was interrupted by the power outage that hit the 50:50 training run, but I was able to restart from a checkpoint. After another 22 hours, it crashed with an error that I've seen before : I put it aside as a one-off oddity when I hit it last time, but this time I dug in a bit more. I noted that it had not ever happened on , but seemed to be an issue on , and that had an older version of CUDA and the Nvidia drivers -- might that be the cause? I decided to upgrade those before kicking off the next run, but for now just restarted the run from the most recent checkpoint. (Note for anyone who is hitting the same error: it has not occurred since the upgrade, so that's worth trying.) This time it completed OK: Again, these numbers just show what happened after the most recent restart. I copied it over to , converted it into a format that was compatible with my PyTorch code, and ran the smoke test: Coherent enough -- time for the loss eval: Again, in line with my predictions -- worse than the JAX FineWeb-only models, and indeed than the very best PyTorch one, , and also worse than the 50:50 split, but better than the FineWeb-Edu one. I uploaded it to Hugging Face , and it was time to move on to what was meant to be the final model for this set of experiments. Again, this was a simple enough config to set up: ...and the build and upload process worked well (and took much less time -- for some reason, sampling randomly from a single dataset is faster than sampling from two or three): Note that it needed to oversample -- that "2 iterators". OpenWebText is about 40 GiB uncompressed, and so that's about 10B GPT-2 tokens -- presumably just a little bit less. Again, given that I was planning to use just the first 3.2B tokens of the dataset, I didn't feel that it would matter. I ran the check script on the newly-uploaded Hugging Face dataset and all looked well, so that was all set for the training run. I upgraded first with a to see if that helped with the weird error that I got in the previous run (which, as I said, it looks like it did), then kicked it off: About 31 hours in, it crashed again, but this time it was my own dumb fault: has a relatively small disk and I ran out of space. I fixed that and kicked it off again from the most recent checkpoint, and this time it completed: I converted it to PyTorch for the smoke test: ...which looked solid, so it was time for the test loss eval: Our worst score yet in this experiment! Worse than any of my models so far, apart from the two FineWeb-Edu ones I did without optimised hyperparameters. Now, the first draft of this post went straight to the results from here, but the story wasn't quite over yet... GPT-6 Astra is relentless . Before I publish any of these posts, I run them past an editorial board of LLMs to look for issues. GPT-6 Astra not only checked the text, it also visited the code I'd linked to to check that out too, and spotted something problematic. It's obvious in retrospect, but my code to build the new datasets had a high risk of including the contents of the -- in theory held-back -- test set. The way that the test set was generated was that I downloaded the 10B sample of FineWeb back in December , splitting it into 99% training data and 1% "validation". That validation split was about 100M tokens, and I was only using the first 19M or so for actual validation runs during training, so I (somewhat arbitrarily) designated about 19M other tokens starting at position 50M in there as my test set. Now, my new dataset-generation code was just sampling randomly from the complete 10B sample of FineWeb. So there was nothing stopping it from pulling in data that was in that old validation split! That meant that it was quite likely that my new "curated" and "50:50" datasets contained at least some of the test set that was meant to have been held back from the models during training. On reflection, the problem was potentially even worse. FineWeb-Edu is a subset of FineWeb; my existing FineWeb-Edu dataset came from the 10B sample of the Hugging Face original, and so it also could potentially contain documents that I'd put into the test set. The first thing to do was to establish the size of the problem. I wrote a script to take in a "forbidden" dataset and split; this was assumed to be formatted as one big tensor of GPT-2 tokens, which is what all of my datasets are. It would then split it by end-of-text tokens, and generate a hash and a token count for each resulting "document". Optionally, you could restrict it to only considering a subset -- the n tokens starting at position p -- and it would then generate hashes/lengths for the documents inside that slice, or that overlapped it at the start or the end. I ran that to generate a list of hashes for the entire validation set -- the validation split of -- and then used a second script to check my various training sets (and the validation set itself) to see how much of a contamination problem there was. I got these results: However, these numbers -- while scary, at least for the 50:50 and the curated datasets -- were not quite the ones to use. They showed how much of the full validation set showed up in the full training set; what I actually cared about was how much of the test set -- those 19M tokens starting at position 50M in the validation split -- was in the actual subset of the training datasets that I actually trained on -- the first ~3.2B of them. I re-ran the script to generate hashes for just the test set, and then re-ran the contamination-checking script, telling it just to look at the appropriate subset of the training tokens, and got this: It was clear that there was a problem -- certainly with and . They'd seen what felt like a significant amount of the test set while training, so their results on the test loss eval were dubious at best. I decided to train those two models afresh, and see what the result was in terms of loss. If the difference was huge, I'd look into the risks of the (much smaller) contamination of and . But if it was pretty small, I'd not worry about that too much. I extended the script that prepared datasets so that the config file could specify a . Any documents in the source datasets that matched forbidden ones would be excluded from the output. I then updated the config for and so that the whole validation split of was forbidden, and re-generated them. You can see the updated datasets here and here . Running the contamination-checker script against them showed that they were clear. I then re-did the full training runs for those models; the uncontaminated version of the 50:50 split model is here , and the curated one is here . And the good news: both of them actually did very slightly better at the test loss eval than their equivalents that had been trained on the contaminated data: There are a number of possibilities that come to mind; perhaps learning from the test set just doesn't happen with tiny 163M models like this, or perhaps while the contaminated models were learning, the benefit they got from that was outweighed by the data that they got instead of the test set data being in some way better for training purposes, at least in terms of the loss eval. But anyway, I felt that if the effect of seeing more than 10% of the test set data during training was so tiny, then the effect of seeing less than 0.2% -- which is what the FineWeb-Edu model in this set of training runs had, as did all of my other FineWeb-only models from previous experiments -- would be even smaller and I'd disregard it. That was excellent news! I didn't need to start all of my experiments from scratch. For the rest of this post, I will include the numbers and results for the contaminated models as well as the uncontaminated ones -- they're interesting for several reasons -- but for future posts I'll skip the contaminated ones. So -- finally! -- let's start digging into the final results. Firstly, I think it's worth taking a look at all of the test loss results in context. Here they are in a table, with the new models in bold: I think there's something very clear here: with the new models, the more FineWeb that was in the training mix, the better the model did on this eval. I think I might have been subconsciously expecting that in the predictions I did before running these experiments, but in retrospect it's so incredibly obvious that I feel silly for not mentioning it explicitly! But that tells us something interesting. From the description in the paper, whatever OpenAI did the GPT-2 training run on, it was not like FineWeb. It was probably more similar to OpenWebText -- and yet, that model was the one that performed the worst on this test eval, so if it is more like OpenWebText, there must be some other factor involved. But moving on for now: how about the IFT test -- the one that kicked off all of this work in the first place? I generated a set of IFT responses for all of the new models, and then ran them (plus responses for all of the other models on that table above) past GPT 5.5, and found that one of my new models was getting quite close to the original GPT-2 small weights! So I did four more runs, so that I could get an average. Here are the results -- the "IFT score" is the average across all five runs of the judge, and the "IFT rank" is based on that. The "IFT epochs" was from the original result-generation script. If you want to see the full numbers, they're below . The number that initially surprised me, and made me decide to do multiple LLM-judge runs was the one for the "JAX, FineWeb-Edu" model. In my first run it came in at 24.35 vs the OpenAI small weights' 24.93 -- so close that I wondered if it might even beat them on a re-run. However, in the further four runs its score was consistently lower than the OpenAI model's, and the gap extended a bit in some. So, was FineWeb-Edu the clear winner here? Perhaps. If you look at the contaminated/uncontaminated pairs, something interesting pops out. For the 50:50 mix, the model trained with the contaminated dataset got 19.30, and the one trained on the uncontaminated one got 17.69 -- a difference of 1.61. For the "curated" dataset, the situation was even more interesting: uncontaminated got 16.63, while contaminated got 13.58, a delta of 3.05 points. Remember, the contamination issue is about whether or not the model saw the held-back test set during training. It was an issue for the test loss that is based on that test set, but is entirely orthogonal to the IFT test. From the IFT perspective, both contaminated and uncontaminated models in each case saw training data that was -- in theory, at least -- essentially the same in terms of quality. Indeed, the uncontaminated run saw almost the same data in the same order as the contaminated one, except that some items were omitted, and then extra ones were added to the end. The purpose of this set of experiments was to see how data quality affected the results on the IFT test set. But in the case of the curated model, something that should be unrelated to data quality changed the results by 3.05 points! If something as simple as changing which data of the same quality the model is trained with can affect the IFT score so drastically, it makes it a bit harder to be certain as to whether or not data quality really had the effect we were looking for. On the other hand, the FineWeb-Edu model came in at 24.56, which is 4.06 points better than the 20.50 that the closest other model got -- more than the 3.05 points we see in difference between the two curated dataset models. And it's worth noting that the model with 20.50 is "JAX, no MHA bias, no dropout", which has a subtly different architecture -- no bias on the output projection of the multi-head attention blocks. A better comparison might be "JAX, with MHA bias, no dropout", which got a score of 17.90, for a whacking great difference of 6.66 points. I think that without doing a very large number of training runs on different datasets with different mixes, each one created with a different seed, it would be hard to work out exactly what is in the noise here and what is not. However, that would cost a lot in terms of time. I think that the best thing here is to chalk this up as a fairly decent indication that FineWeb-Edu improves matters for the IFT eval, but far from a certainty. But it's certainly worth noting that whatever the noise is, it has a range of at least 3.05 points -- and the FineWeb-Edu model is just 0.63 points short of GPT-2 small! So there could well be something there. Of course, we don't know whether that model got (by chance) the best possible balance of FineWeb-Edu tokens, and could never win -- or whether it got a bad balance and would actually beat GPT-2 with a better one. So that's certainly worth keeping in mind. As an aside, the result for the curated dataset really surprised me. I had expected that it would be the best one, simply because it almost certainly contained more facts. I took a look at its answers to the questions -- one possibility that came to mind might be that it would get better responses to questions like "What is the chemical symbol for chlorine" or "Who wrote Pride and Prejudice" than the others, but would fail on less knowledge-based tasks. But it was terrible at fact-based questions too: Name the author of 'Pride and Prejudice'. The author of 'Pride and Prejudice' is Priscilla Finch. What is the periodic symbol for chlorine? The periodic symbol for chlorine is H. As I understand it, many real-world training runs do include (often oversampled) amounts of highly educational training data like this model's dataset did. But perhaps the models that I'm training are just too small to be able to make use of the data they gained that way -- maybe doing things this way and expecting good results is like asking six-year-old children to memorise stuff before they've learned enough to be able to make use of it 1 . It's worth noting that the GPT-2 small model also failed on those factual questions. Well, anyway: I think we have some useful results here, so let's work out what that means for next steps. The results we got in these experiments point in two interesting directions. I think that the right direction to take this going forward is to separate these two angles. I should chase a higher IFT score, and then once I have nailed that down, I should see what (if anything) might allow me to get the resulting model to improve its test score. But I will need to make sure that whatever dataset I use, I use various "mixes" of it -- versions created with different random seeds. In my earlier experiments with overtraining, I did find that it didn't seem to improve the IFT results -- but it did improve the test loss. So perhaps identifying the right combination of other factors to boost the IFT score, then overtraining the result, might help? Of course, my overtraining tests were with FineWeb, so the connection might not hold up as well if the starting model (as seems likely) was trained on a different dataset. Also, while working through the results here, I've come to the conclusion that the set of models I'm using is a bit confusing -- there are now different hyperparameter settings, small architectural differences (the MHA bias thing), dropout settings during the pre-training, and now datasets. I think that's OK for now; I should see this part of this series as more ideation than actually running the proper experiments. But at the end, when I have some solid hypotheses with a reasonable amount of backup, I should start from scratch: a baseline model, then staged interventions to build up to what (hopefully) will be a model as good as GPT-2 small. Anyway, I'll wrap this one up here. I think that the next lever to pull is (perhaps surprisingly) going to be weight tying. I had previously kind of disregarded that as a possibility, but while I was working on this post, something popped into my mind. The OpenAI models were originally trained with weight tying. My codebase does actually support doing it -- but because I got the OpenAI weights I'm using from the code in " Build a Large Language Model (from Scratch) ", when I'm running the IFT test, the weights are not actually tied! We load up a model that has separate but identical embedding and output head matrices, and then we fine-tune that. So those two matrices can vary independently during fine-tuning -- to put it another way, while GPT-2 small was pre-trained with 124M parameters, the IFT test is being done on a 163M-parameter version. Does that give them some non-obvious advantage? And would adding weight-tying to my own models help, either with or without the output heads being independent at fine-tuning time? Stay tuned :-) Here are the numbers for all of the IFT judge runs, included for completeness. You can see that the LLM judge ranks models very consistently between runs, but there is variation -- that is, on some runs it's in what I think of as a "better mood" than others, and if that's the case, it will give better scores -- but it will give them almost consistently between models, so all of the models do better. Note that (unlike the table above) this one is sorted by the average IFT score rather than the test loss. A small boy asleep on his right side, the right arm stuck out, the right hand hanging limp over the edge of the bed. Through a round grating in the side of a box a voice speaks softly. "The Nile is the longest river in Africa and the second in length of all the rivers of the globe. Although falling short of the length of the Mississippi-Missouri, the Nile is at the head of all rivers as regards the length of its basin, which extends through 35 degrees of latitude …" At breakfast the next morning, "Tommy," some one says, "do you know which is the longest river in Africa?" A shaking of the head. "But don't you remember something that begins: The Nile is the …" "The - Nile - is - the - longest - river - in - Africa - and - the - second - in - length - of - all - the - rivers - of - the - globe …" The words come rushing out. "Although - falling - short - of …" "Well now, which is the longest river in Africa?" The eyes are blank. "I don't know." "But the Nile, Tommy." "The - Nile - is - the - longest - river - in - Africa - and - second …" "Then which river is the longest, Tommy?" Tommy burst into tears. "I don't know," he howls. Brave New World , Aldous Huxley  ↩ It seems very likely that the GPT-2 models were overtrained by modern standards; would overtraining my own models get them closer? It turned out that no, it probably didn't help with the IFT eval (though there might have been some signal there). It did help quite a lot with the test loss eval, though. The way I was handling dropout in the IFT test might have been unduly benefiting some models while working against others. I decided to standardise on not using dropout during this eval, as (counter-intuitively for me) it seemed to harm the results of most models, even those that had been pre-trained with dropout. In particular, the OpenAI weights were harmed by using dropout, and making a change that benefited them (along with some of my own models) seemed the most conservative approach to take in investigating this. "Local FineWeb-Edu train" "Local FineWeb-Edu extended train" FineWeb-Edu -- essentially the same as "Local FineWeb-Edu train" but with a better training setup. This would test the "more educational -> better" hypothesis. A 50:50 split of FineWeb and FineWeb-Edu. I've read that LLMs can be helped by having a decent amount of lower-quality data in their training loop, as it helps them to generalise. Perhaps having some FineWeb in there in addition to the FineWeb-Edu stuff would improve that test loss score while also helping the IFT test? A "curated" dataset containing 45% of its contents from FineWeb, 45% from FineWeb-Edu, and 10% from the Simple English Wikipedia . The full Wikipedia is huge, and full of obscure facts -- while the Simple English one is small and hopefully richer in useful information on a per-token basis. And conveniently, Answer.ai have made a snapshot of it available on Hugging Face Hub . Might deliberately putting a bunch of encyclopaedic data into the training set make the model better at the IFT eval (which has lots of factual questions in it, like "who wrote Pride and Prejudice")? OpenWebText. Even though I was unsure how well it matched the original WebText, given that it was there, it seemed silly to not try training something on it and see how it matched up. The FineWeb-Edu-only model would do pretty badly on the test loss, but better than my older FineWeb-Edu models (90%). It would also punch above its weight on the IFT eval (90%). The 50:50 split: I expected it to do worse on the test eval than my JAX FineWeb-only models (70%), but better than the FineWeb-Edu one (90%). I wasn't sure about how it would do on the IFT eval, but thought it might be somewhere in between the two groups (60%). The curated dataset I had high hopes for in terms of the IFT eval -- let's say 80% chance of it being the best of all of my models. For the test loss eval, I expected it to do about as well as the 50:50 split, maybe a little bit worse (70%). I had no idea how the OpenWebText eval would do! Could be worse, could be better. The validation set was 100% "contaminated" with itself, which was a useful sanity check. The training set of had what I felt was a small level of contamination. It was interesting that there was any at all -- I think that must mean that there are some repeated documents in the original dataset, and some of them wound up with copies in both my training and validation splits. The dataset also had what felt like a reassuringly low level of contamination. Both and , however, looked problematic. In both cases, the training datasets had more than 40% of the validation/test set in them. was, as you'd expect, almost completely uncontaminated. It looks like maybe one document happened to have been picked up by both the OpenWebText and the FineWeb crawls and then included in the bit of FineWeb I was using for validation. The perfect connection between the amount of FineWeb in the training set and the result on the (FineWeb-based) test loss eval, while perfectly obvious in retrospect, really does highlight how mysterious it is that the OpenAI small weights do so well on that test. The fact that FineWeb-Edu did well on the IFT test tells us that there does seem to be value in using richer training data -- though the less-spectacular results of the 50:50 mix and the curated one weaken that a bit, as does the indicator of what the noise due to data selection from equivalently high-quality datasets might be. The OpenWebText result I think I'll ignore, given that -- while in theory it should be similar to what OpenAI trained on -- there are no guarantees, and it might differ in non-obvious ways for non-obvious reasons. A small boy asleep on his right side, the right arm stuck out, the right hand hanging limp over the edge of the bed. Through a round grating in the side of a box a voice speaks softly. "The Nile is the longest river in Africa and the second in length of all the rivers of the globe. Although falling short of the length of the Mississippi-Missouri, the Nile is at the head of all rivers as regards the length of its basin, which extends through 35 degrees of latitude …" At breakfast the next morning, "Tommy," some one says, "do you know which is the longest river in Africa?" A shaking of the head. "But don't you remember something that begins: The Nile is the …" "The - Nile - is - the - longest - river - in - Africa - and - the - second - in - length - of - all - the - rivers - of - the - globe …" The words come rushing out. "Although - falling - short - of …" "Well now, which is the longest river in Africa?" The eyes are blank. "I don't know." "But the Nile, Tommy." "The - Nile - is - the - longest - river - in - Africa - and - second …" "Then which river is the longest, Tommy?" Tommy burst into tears. "I don't know," he howls. Brave New World , Aldous Huxley  ↩

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Giles's blog 1 months ago

Extending Raschka's GPT-2: an MoE trained from scratch on an RTX 3090

Mixture-of-experts models are really nifty. You get inference speed close to a small model's, with a lot of the smarts and knowledge of a large one. While they use as much memory as an equivalently-sized dense (non-MoE) model, they're much faster. The frontier labs don't publish their architectures, but Claude and ChatGPT are widely rumoured to be MoEs these days -- and certainly many large open-weights models like DeepSeek and Kimi K3 are. In this post, I'll show you how I added MoE support to the GPT-2-style code from Sebastian Raschka 's book " Build a Large Language Model (from Scratch) ", then used that to train a 446M-parameter model with 220M active parameters from scratch on my RTX 3090 -- essentially, GPT-2 small with 6 experts, 2 active per token. I wanted to understand how MoEs work, and as always, felt that the best way to do that is to build one (and then to write it up like this). Hopefully because it's all fresh in my mind as I write this, I should be able to explain things clearly for others who've finished Raschka's book. The training run took just less than eight days, and the resulting model got a better loss on my test set than any of the other models I've trained so far -- which was a good thing, given that it took four times longer to train! It was also better than the original OpenAI GPT-2 small (124M parameters), but not as good as GPT-2 medium (345M parameters), although it was close. That second result was interesting, as my model had more total parameters than GPT-2 medium, but fewer active ones. On an instruction fine-tuning test, it did better than any of my other models, but worse than both OpenAI models (why OpenAI's models are so good on that particular test is a mystery I'm digging into separately). I'll go into those numbers in more depth later on. Firstly, though, I'll run through the code that I needed to add to GPT-2 to make this work -- not just the code for the experts themselves, but the additional code for training it. With MoEs you can't just train to minimise loss on your training set -- you need to add on an "auxiliary" loss to make sure they actually use all of their experts, and that was where it became most interesting. Before we get into the weeds, though, let's start off with the basics; how do MoEs work in Transformers-based LLMs? The phrase "mixture of experts" evokes an idea of a model which has separate parts that are knowledgeable in different domains. Maybe one part would know about coding, another about history, another about philosophy, and so on. You can imagine something that had separate LLMs for different topics, and routed incoming inputs appropriately. That's actually not a bad design for a system -- Sakana.ai got a lot of interest for their Fugu system back in June of this year, and it works rather like that. But the "experts" in an MoE are at a much lower level, and (as with so many things in LLMs) their expertise is in some weird and alien thing that they determined was helpful for modelling language during their training. Let's look at how they fit in mechanically. The GPT-2-style LLM that we will start from -- the one Raschka describes in his book -- looks like this: Diagram 1 : a GPT-2-style LLM at the top level The MoE magic happens inside those Transformers layers, so let's zoom in on one of them: Diagram 2 : a GPT-2-style Transformers block Specifically, what we do to make our LLM an MoE is to replace that feed-forward network (FFN) with multiple separate FFNs -- our experts. Different context vectors get routed to different experts based on their contents. The FFNs themselves -- in both the dense and MoE versions -- are surprisingly simple . In GPT-2, they take in the incoming context vectors, run them through a normal linear layer to expand the number of dimensions by four, run the result through a GELU activation function, then project it back into the original incoming dimensionality with another linear layer. It's a really simple two-layer neural network, and at first glance seems somewhat arbitrary. Tutorials about LLMs tend to spend pages and pages explaining attention mechanisms, and pretty much gloss over the FFNs. But the FFNs take up twice as many parameters as the attention mechanism in GPT-2. They're clearly highly important, and my (very loose) metaphor for why that is, is that attention is how the LLM works out what to think about, and the FFNs are where it does its thinking. It's not a perfect match for what's going on, but I think it's a decent working model for intuition. An MoE leverages this. Instead of having just one FFN per Transformers block, we have multiple, and we use a subset of them for each context vector. We have what is called a router, or a gating network. It takes the incoming context vectors, and for each one, decides which of these FFNs -- which experts -- to use. Then we feed the input into the experts that were selected for it, combine their results, and that's our final output -- like this: Diagram 3 : an outline of an MoE block Doing this gains us more "space" in the LLM for it to remember facts and ways of thinking about things -- we have multiple experts for that knowledge to be spread over. We could, of course, do that by dedicating more space to the FFNs -- for example, by having one bigger one. But with an MoE, because we only activate a subset of the experts for each context vector, we save on the amount of computing we do for each context vector. We need to keep all of the experts in RAM -- remember that we're routing to them per-context vector, so in a batch of sequences we're likely to be using most if not all of them. But we don't have to feed everything through all of them. So, that's the basics -- nothing conceptually difficult. What becomes more tricky is the implementation. How, concretely, does the router choose which experts to use for a given context vector, how do we implement that choice -- and how can we do all of that efficiently? And how do we train the router to make its choices? When I started on this project, my first step was a Google search for useful references. I came across this excellent summary from IBM. In that post, they mention a number of papers; four seemed relevant 1 : I decided to take a look at them, and was pleasantly surprised about how readable they all were. It looked to me like I'd be able to put together a working MoE model by following what they (and the IBM summary) said, and that turned out to be true -- the model worked, and trained well. Of the four papers, I found the Switch Transformers one the most useful, but the original 1991 paper also clarified a lot to me. I really do recommend skimming through them if you want to learn more background about this stuff. Anyway, once I'd got it all put together, and trained the model, I compared what I'd written to the Hugging Face source code for Mixtral , which was the first open MoE model I remember hearing about. Mixtral has a bunch of other improvements over GPT-2 beyond its MoE support (RMSNorm, RoPE, and so on), and it's a much larger model than mine. But it turns out that the specific way it handles the MoE side of things is largely the same as mine, apart from one difference in how it calculates auxiliary loss (about which more later). So that was reassuring. I also ran the final codebase past three LLMs -- ChatGPT, Claude, and Kimi K3 -- just to make sure that I hadn't drifted from the mainstream or screwed up in other ways. I did this in an anonymous/private session so that they wouldn't use any memories they have of conversations we'd had in the past, and asked them Please take a look at the attached and tell me what you see. I'm particularly interested in the MoE stuff. All of them came back with some variation of "it's a pretty standard MoE implementation on top of GPT-2, with load-balancing adapted from Switch Transformers". Even more reassuring! So I'm comfortable that what I've built is a normal implementation, and that's what I'll describe in the rest of this post. It's all fresh in my mind, so hopefully by describing it in terms that would have made sense to me when I just finished Raschka's book, the result will be useful to other people coming to this for the first time. Let's get started by digging into the mechanics of how the router works. The problem we want to solve is that we have a context vector coming into our MoE block -- as per the diagram above -- and we want to decide which experts we want to route it to. Let's say that we have n total experts, and we want to send each input to k of them (where 0 < k < n , of course). We'll also say that our context vectors are of dimensionality d emb . Taking a d emb -sized input and categorising it into probabilities across a number of options is a pretty standard job for a neural network. In fact, we can use a single linear layer for it. Imagine that we create one with d emb inputs and n outputs. For a given context vector, it might produce (for n = 6 ) something like this: So we can imagine picking out the indexes of the top k of those. Let's be specific, and say that k = 2 . That means that we have indexes 4 and 2 -- the positions of the two largest numbers -- so we want to route our original context vector to experts 4 and 2. They do their calculations, we get output context vectors from them, and we can combine them. How might we combine them? Well, we do a lot of adding context vectors together in our GPT-2-style LLM code, and treat the results as meaningful -- token embeddings get added to position embeddings, shortcuts around attention and the FFN get added back in to their results, and so on. So perhaps we could do that? That kind of setup has a problem, though -- it doesn't train the router. Let's think about the MoE block again; here's the diagram again: Diagram 3 (repeated): an outline of an MoE block Think about the flow of data through it. The context vectors flow into the router. We use the output of the router to select the two experts, and then the original context vectors -- the ones that were fed into the router -- flow into those experts, then are combined, and we get our result. Now, consider what happens when you're training a model. You run some training data through it, then calculate the loss -- how good your model's results for that data are. You then use that to work out the gradients that you want to apply to the parameters to make it better. You work out the gradients by using back-propagation, working back from the loss, through the network, retracing the computation graph in reverse. When your backprop gets to this part of the model, it will start with the output context vectors, trace back through the combination step, then back through the two chosen experts, then back to the input context vectors -- and then it will go back to whatever step came before the MoE block. The calculations inside the router that selected our two experts did actually happen in our forward pass -- but they're not in the computation graph as we trace it backwards from the loss. It's kind of a dead-end. There's nothing in there to connect our selection of which experts we used to the path through the computation that ended up at the loss. (Interestingly, the effort of doing that diagram made that particularly clear to me -- that slightly-messy labelling of the arrows at the top, with numbers to show the sequence, is a direct result of the same issue.) What that all means is that the router will not be trained. The backward pass of our training will completely ignore it, and so there will be no gradients to apply to it. A starting model with random weights will randomly allocate context vectors to experts, and it will continue to do that. Clearly, we need to somehow modify our computation graph so that the router is connected -- so that when the backward pass flows up from the output context vectors, it sees the router -- and ideally sees some aspect of it that will allow it to be trained to do its job better. "Adaptive Mixtures of Local Experts" starts by describing some previous work which treats the router's outputs as weights for each of the experts. That's a nice trick, and while it does have some problems (as we'll see in a bit), it fixes the backprop issue. (Interestingly, they actually decided to do things differently -- but later work, like Switch Transformers, goes back to doing things this way, with some important tweaks.) As I said earlier, back in 1991 they weren't thinking of MoEs as being a way to save on computation. Instead, their focus was on training better models to handle specific tasks, and they felt that having some way to split those tasks up might make that easier. So, for their case, they might take those outputs from above and normalise them: ...and then run their inputs through all six experts, then add together the outputs weighted by those numbers -- 0.1459 times expert 0's output, 0.1249 times expert 1's, and so on. Rather like this: Diagram 4 : An MoE with all experts active With that, we have something where the router is on the backward pass through the computation graph from the output context vectors (and thus from the loss). The weights provide a route back, so the router will get trained. But that version, of course, is running all of the experts for every token, so it doesn't have the computational benefit of a modern sparse MoE. It also has an issue, as they point out in the paper, that if you imagine that you want each expert to have a well-defined responsibility, things get messy. Imagine that expert 2 in the above example was the one that knew how to solve a particular task; all of the other experts would be contributing too, and unless you find some way to train their weights down to exactly zero for that task, to get a good loss on your training run they'd need to learn to balance each other out. Their solution was to run all of the experts then to use a gating function on the output -- that is, it would drop all of the results from the non-selected experts (see their figure 1). But the modern solution is a bit different; for that, let's move on to "Outrageously Large Neural Networks". What we really want is an output from the router where some of the weights are zero. Specifically, with our k active experts, n total, we want all but the top k of them to have a weight of zero. Then when we're running the network, we can skip those zero-weighted experts and get a result that will be identical to what we would have got without skipping them. "Outrageously Large Neural Networks" does this with a trick that will be familiar from the causal mask in the GPT-2 code. Let's call the "raw" weights from our router : Let's run that through softmax again: That gives us some initial weights. But we want all but the top k to be zero. Now, if we want something to be zero after softmax, then it needs to be − ∞ on the way in. I'll show the code to do this later, but for now, let's just assume that we have some magic to set all but the top- k values to that. In our concrete example with k = 2 , the original with that change applied would be: Now we can run that through softmax: And we have some weights! With those, we can conceptually run this "all-experts-active" kind of network: Diagram 4 (repeated): An MoE with all experts active ...but skip the experts whose weights are zero. The weights that we provide through the steps above will mean that the router will get trained. That's pretty neat! There is one thing to highlight, though. Imagine if we have one active expert -- that is, k = 1 . We'd mask out all of the other ones: ...and softmax: The weight will always be one, regardless of the original logits. And if a function can only ever return the same result, its derivative will always be zero, so there will be no gradients and it can't be trained. Interestingly, that's something that is -- almost silently -- addressed in the Switch Transformers paper. They are specifically looking at k = 1 MoEs, and they mention the "Outrageously Large Neural Networks" paper's routing system, then present their own calculations which instead of replacing non-top- k values with − ∞ , and then softmax, do the softmax first, then zero out the non-top- k . They don't highlight the difference. I decided that for this experiment, I'd go ahead with the non-Switch Transformers calculations, anyway, and do the replace-with- − ∞ -then-softmax system, with some guard code to prevent it from accidentally being set to k = 1 . Most recent MoE models I've seen have at least two active experts, after all. The good news is that not only did it work -- when I checked later, it also matched the choice taken in Mixtral, which feels like a solid endorsement. 2 3 So, at this point, we know how an MoE works in theory. Let's start coding. I started with the code that I had from " Build a Large Language Model (from Scratch) ". In that, the Transformers block looked like this: I wanted to keep the capability to run this in MoE or non-MoE mode, and decided that I'd do that by extending the model config in so that it would have an optional section, which would include MoE-specific stuff. If that wasn't present, then I'd create a normal dense LLM. To do that, I replaced the line in that assigned to with this: Next, it was time to implement the class itself. The obvious config that it would need was how many experts there were in total, and how many were active per token: Now, there's that issue where having just one active expert will pin the softmaxed weight to one, so it won't train -- so I decided to protect against that (and against another obvious mistake that the config could contain): Next, it was time to create our router, mapping from the incoming context vectors (of size in this code) to the number of experts: I decided to make it unbiased because I have a vague impression that that is the fashion these days -- nothing more principled than that :-) Next, we needed the experts themselves, each of which would be one of the same modules as we were using in non-MoE mode: They needed to go into an because if they were just in a regular list (as I discovered when I tried it) they would not be registered by PyTorch as things containing parameters belonging to the module. We create our optimiser for training with code like this: ...and so anything that isn't in will never get updated, which would be a Bad Thing. That was enough to have the pieces in place. It was time to write the forward pass -- to get the router logits, do the top- k , softmax, and then to use those results to run the experts. Getting the logits was simple enough. Looking at the method: ...we have an incoming set of context vectors, . That is shaped . Let's try to visualise that. Understanding tensor operations is something you can kind of short-cut with intuition, but I think that in order to really understand the next steps it's best to have something a bit more tangible in mind. You can think of an order-3 tensor like this as a cuboid. With of 3, of 5, and of 7 -- artificially small values to keep things simple -- it might look like this: Diagram 5 : The tensor as a 3-D cuboid Every dot in that cuboid is a single number. A single context vector is the numbers as you follow a line from the "front" of the cube -- the × face to the left -- to the "back". That 3-D representation was fiddly to get right, and is likely to get confusing if we keep using it. So instead, let's look at two 2-D views of the same thing, from the front (where you are looking at a × face), and from one of the sides, where it's × : Diagram 6 : as two 2-D views So in this view, each of the circles in the "Front view" is the first number in a specific context vector, and each of the rows in the "Side view" is the set of numbers that make up a specific context vector. Hopefully that's easy to visualise. Now, a PyTorch layer like our operates on the last dimension in the tensor you pass in. For our , shaped , it will work on the dimension -- that is, it will operate on each context vector independently, which is what we want. That gives us a set of logits shaped . In our visualisation, that's a × × cuboid; let's diagram that with four available experts: : Diagram 7 : as two 2-D views Again, if we look at the front view, each circle represents the first element of the routing logits for one of the context vectors, and in the side view, each row represents all of the routing logits for a context vector. So we have the right data in our cuboid. From what we're calling the front view, it's got the same dimensions as , which is useful. In order to work out which of our experts each context vector should go through, we need to get the top- k values for each context vector in . PyTorch has a function to do exactly that: The tells it to work across the last dimension, which is the dimension of length . It returns the top values and their positions, as tensors -- both shaped . So, we have two new tensors, which we can visualise as cuboids, both of × × . Let's show that with : Diagram 8 : (or, equivalently, ) as two 2-D views Both and are shaped that way. As normal, the "front" face is the same; each circle corresponds to a context vector, and for it holds the highest value in that context vector's logits (because returns results sorted), while for it holds the index that that highest value sits at in the logits list. Looking at the side view, we're seeing a list of top-k logit values or indices for a given context vector. Now, we want a version of that we can run through softmax to get some weights, and in order to do that we need to replace the non-top- k values with − ∞ for each of the context vectors. The solution that I hit on eventually (after various other attempts 4 ) was this. Let's start with a tensor identical in size to , but full of − ∞ s: Now, contains the values that we want to have in there, and contains the indices in the logits lists where they should go. All of the other values can remain as − ∞ . This will, of course, be the same shape as : Diagram 9 : as two 2-D views Both and are compatible in their first two dimensions with , and thus with -- that is, their front faces in our visualisations are the same -- compare the front face of the above with the one for diagram 8 . For all of our tensors, the first two dimensions correspond ultimately to an incoming context vector in . It's just the last dimension and the data they contain that differ. PyTorch has a function on its class, which takes a list of positions and list of values. It takes one specified dimension, and treats that one essentially as a set of lists. Then it takes two other tensors with the same number of dimensions as each other, each of which matches in size in all but the specified dimension, and treats the other dimension as being a list of values for one parameter, and list of indices for the other. It overwrites the data at the specified indexes with the specified values. That sounds useful! Let's make it concrete. Remember that in all of these cuboids we're visualising, the front view we're looking at corresponds ultimately to a context vector. In , it's the raw logits for expert routing for that vector -- the number on the front face is the first of those (corresponding to the logits for routing to the first expert). If we look at the side view, each row would correspond to the set of logits for a given context vector. Now, let's consider just that. Inside , our specific context vector might have logit values corresponding to it like this: Diagram 10 : A single CV's routing logits in-place in the cuboid Visualised the same way, with k = 2 the equivalent part of would look like this: Diagram 11 : A single CV's top-k routing logits indices in-place in the cuboid That is, the function identified that index 2 in the original logits was the highest value, and 0 was the second-highest. Likewise, would have this: Diagram 12 : A single CV's top-k routing logits values in-place in the cuboid These diagrams are getting a bit unwieldy; let's look at this specific context vector's data as lists. For our selected context vector, we have these logits from diagram 10 : ...these top- k indices from diagram 11 : ...and these values from diagram 12 : Our tensor is just full of − ∞ s, and is the same shape as , so its corresponding part is this: What will do is select indices 2 and 0 (because of the values in ), and copy the corresponding numbers from on top of whatever is already there: It will do that for every one of the positions in that front view. So now we have what we wanted: a grid of routing logits for each context vector, where the non-top- k ones have been replaced by − ∞ . That's pretty nifty! And so here's the code to do it: I was initially a little worried that the whole "replace the logits with a 'static' tensor and then scatter the values in there" approach might break the computation graph, but tests showed that it didn't. My intuition is that because the − ∞ s were summoned out of nowhere, they are dead ends for backprop, but because the logits that we're copying in come from previous calculations, they are not. So once we've done that, we have our top- k logits, ready for a softmax -- so it's time to do that: ...and we have our weights, yet another cuboid like this: Diagram 13 : as two 2-D views Just as before, a given number on our front face is the first of the expert weights for a specific context vector, and each row in the side view is all of the expert weights for that context vector across all experts. Post-softmax, the weights for active experts for that context vector are positive numbers, and the weights for the inactive ones are zero. The next step was to actually feed the context vectors into their experts. I decided to keep this simple, and just iterate over the experts, one by one, and for each one to find which incoming context vectors wanted to go to it, run them all through, and then reassemble the results. I believe that larger MoE systems have smarter routing systems -- for example, for a huge model that can't fit on a single GPU you might have different experts on different GPUs or even machines, and route to them in parallel. But for my toy-sized models, this felt simplest, and felt like it would be efficient enough 5 . The way I decided to do this was to use an "accumulator" model. We know that the shape of the outputs is the same as the shape of the inputs -- that is, if we have our incoming shaped , then the output will have the same shape. So I started off by creating a tensor of zeros of that shape: So, just like in diagram 6 , it will look like this: Diagram 14 : as two 2-D views ...but it would be filled with zeros in every position. The plan was that each expert would be run on the appropriate context vectors, its outputs would be scaled by the weights that were calculated by the router and its associated top- k and softmax for that context vector/expert pair, and then the results could be added into . That would give us the result we wanted: after all experts had been run on their respective context vectors, would have the weighted sums. So the next step was to iterate over the experts: Now, we need to know which context vectors wanted to be fed to this expert. Let's take a look at our representation of again: Diagram 13 (repeated): as two 2-D views We can see it as a bunch of "slices", each the same shape as that front face, , where each one is the weights for a given expert. The front face itself (corresponding to the rightmost column on the side view) is the weights for expert 0, the next one "back" is for expert 1, and so on. And in code, we can get the slice for the expert with index like this: That will be a simple 2-D grid of numbers like this: Diagram 15 : sliced for one expert You can see that it's a grid of one number for each context vector in our input -- the same shape as the front face in all of the diagrams so far. Each number is the weight that the expert with index has for the context vector in question. We can now do a comparison: That will give us a new grid of the same shape, but the numbers have been replaced with booleans -- if the corresponding context vector has a weight for this expert that is greater than zero, otherwise. We've got a mask that identifies exactly the context vectors that we want to run through this expert. Let's imagine it looks like this for some particular set of context vectors and some specific expert; I've coloured in the circles representing and left the ones white. Diagram 16 : what might look like for one expert Now comes the clever part :-) Remember that our original incoming context vectors, in , looked like this: Diagram 6 (repeated): as two 2-D views We can use our mask -- the grid of s and s in diagram 16 -- to select a subset of the context vectors in there, like this: That will return us the subset of the incoming context vectors that we want to run through this expert. In terms of our diagrams above, it will be selecting the context vectors in the front face that are "selected" by our mask, and taking the "cores" as it goes back through the cuboid from there. The question is, what shape will it be? You can see that there's no simple 3-D shape it could be. The first sequence in our batch -- the first row on the front face -- has two selected context vectors, while the second has one, and the third three. You could imagine a world where the output would be the same shape as the input -- that is, a tensor the same shape as -- but with the non-selected numbers replaced with s or something like that. But instead, PyTorch produces what amounts to a list of the selected context vectors for this expert. We'll call the number of selected vectors , and it's six in the example diagram above, so the shape will be , like this: Diagram 17 : the "selected" context vectors Now, note that this is a big change from all of our tensors so far. It has lost the connection back to the original tensor's shape. In all of the other tensors, we could map something back to the original context vectors in . But with this new one, we have a bunch of context vectors with no inherent connection to where they originally came from in . We'll come back to that. But for now, we have the data that we want to run through the expert with index . The expert is an FFN -- our two linear layers with a GELU in between them -- and will treat all but the last dimension in whatever we feed it as "batch" dimensions, so it will work over that dimension, as we want it to. So the code to actually select the context vectors -- our above -- and then to run it through the expert itself just becomes this: We get the results of running our selected context vectors through the expert, still shaped . (If you're familiar with the GPT-2 code, that in the code might seem odd. We'll come back to why the expert is returning a tuple and why we are ignoring the second item in it later.) So we have our results -- we've run the appropriate context vectors for this expert through it. But they're in a slightly funny shape, and we're going to need to fix that later on. But first, we need to apply the appropriate weights for this expert to each result. Remember that is a grid of booleans, , like this: Diagram 16 (repeated): what might look like for one expert ...where -- filled in the diagram -- means that this expert is active for the corresponding context vector, and means it isn't. looks like this: Diagram 13 (repeated): as two 2-D views Now, previously we did this: ...to pluck out the context vectors from where the mask was true. If we were to do ...then we would be doing something similar. We'd get a grid like the one of context vectors in diagram 16 , with one row for every selected context vector for this expert, except that instead of each row containing a context vector, it would contain that context vector's per-expert weights: Diagram 18 : all expert weights for the "selected" context vectors Now, on its own, that's not particularly useful -- but you can hopefully see that column is the weights for our expert for all of the context vectors that are going to be run through it! So if we do the same masked lookup as before, but also select that column, we get code like this: What we're saying is "pick each item in that matches a in , then take the th element of it". It will look like this: Diagram 19 : this expert's weights for the "selected" context vectors That is exactly what we need to get the weights for our results! We need to multiply the i th element in -- an output context vector that is the results for a particular input context vector -- with the i th element of . However, there is one tensor-compatibility issue. What we have is shaped -- that is, it's essentially just a list of numbers. Now, we want to multiply our results by these weights, which means that we want to broadcast them across , which is shaped . Naively you might think that if you try to multiply a tensor by a one, PyTorch would match up the two dimensions of the same size and it would just work. But it would actually try to match up across dimensions from right to left, and would complain that does not equal . So instead, it's best to feed it an explicit tensor so that it knows which dimensions we're trying to match up, and which ones it should broadcast over: ...and that will give us of shape That would look identical to diagram 19 in the way I've been diagramming these things, but it's technically different and necessary. So now, with the having fixed the dimensionality so that we can do a broadcast, we can do this: ...and with that we have our weighted results for this expert -- all of the context vectors that should have been run through it have been, and they've been multiplied by the weights that the router gave for them, so we have our backprop channel for training the router. The next step is that we need to somehow put these results into . Specifically, because it's initially all zeros, and we want it to hold the results of adding together the results from each active expert for each context vector, we need to add them to whatever is already there at this point in the iteration through the experts. Even though -- as noted earlier -- we have, by this point in the calculations, lost the connection between the tensors we've been working with and the original "front-facing" grid of our original tensors like , where we could link something directly to the incoming context vector that it related to, it's actually surprisingly easy to patch that up :-) Remember that outside our loop through the experts, we created like this: So, just like in diagram 6 , it looked like this: Diagram 14 (repeated): as two 2-D views Now, previously we had code to extract the context vectors that we wanted to go through our expert from , and it looked like this: That gave us what amounted to a list of context vectors, like this: Diagram 17 (repeated): the "selected" context vectors Now, the interesting thing about a masked lookup into a tensor like is that not only can you do it to extract a subset of the values from the tensor like we did with -- you can also use it to assign to the masked subset of the elements in the tensor. To make that concrete: when we did We got a tensor sized . That means that if we were to do: ...then given that the mask is the same, and has the same shape as , then we'd also get a result. But the thing with assignment means that if we had some tensor shaped -- let's call it -- then we could do this: With that, instead of selecting the parts of and using them in the future, we're overwriting them with whatever is in . Furthermore, we can use the same trick with the augmented assignment operators like : ...means "select the elements from that have in , and then increment them by the elements of ". So finally, we get our code: That multiplies the results from the expert by the corresponding weights, and then adds them in to the running totals we're keeping in . Because is the same shape as , and thus is the same shape as , and we've kept that same shape as we went through the expert itself and multiplied by the weights, it will work. And with that, we've done all of the calculations that we need for this specific expert, so we can go back round the loop for the next one. When we've finished with all of the experts, we can return the result: Phew! That was quite a lot of explanation, but I think that what is going on in the code needs it. I originally wrote it in a kind of flow state of inspiration, and was very pleased with it, but it feels like now that I've explained it, I actually understand what my subconscious must have known while I was writing it. And I hope it's reasonably clear for anyone reading this. But there was one extra thing I wanted to keep track of before I started running this: the balance between the different experts. We'll come back to this in some detail later, but a problem with MoEs is that they can wind up depending heavily on specific experts, and ignoring the others -- the auxiliary loss I mentioned way back in the intro to this post is required to avoid that. I didn't want to implement that yet, but I wanted to log enough data to see if it really would be necessary with my setup. A good way to keep track of how much each expert is being used is to record the logits -- the original results from the router, before the top- k and the softmax -- and the actual post-top- k , post softmax expert weights that we actually used. The changes to do this are dotted around a bit, so I've linked to the appropriate lines on GitHub and if you have the screen real-estate to do so, I'd recommend that you use that to follow along. However, I've tried to put enough code inline in this post below that it should be comprehensible without that. I decided that I'd return the logits and the expert weights from the module's forward pass as an extra output here : Now, back in the we were setting to either a object or a depending on whether MoE was enabled for this model here : That meant that in our forward pass where we previously just did this (I won't link to this because it's an old version and having links to different versions would be confusing): ...then if the model was an MoE, we'd be getting a tuple -- the actual outputs from the module and then that extra routing information we were returning. But if it was a , we'd just get the outputs. I decided that the simplest fix was to change the module so that it returned values that were compatible with . The old , which was this: ...became this : That meant that we could do this in place of the code above: If we had an MoE in , then we'd get some real MoE routing info in , whereas if we were running a normal dense model then we'd get . This, by the way, explains the odd bit of code in the for that you might remember from earlier -- this bit : there is one of the modules, so what we're doing there with the underscore is just ignoring the that we know we're returning from it. The next step was to work out how to get the extra information that we had in out of the , if was a . Now, is used in an in the class here : That means that the outputs from the method of the first one are fed directly in as the inputs to the second one, and so on. Additionally, assumes that the forward method just takes a single input. What I really wanted at the end of this was a list of pairs, one for each Transformers layer. So I decided to pass an accumulating list into (allowing it to be ), like this : ...and then append whatever routing info came from the (potentially ) FFN to that here : ...and then return it from for the next layer here : Then, in I wanted to return it to whatever called the model for inference. I didn't want to change the implicit API that I was using -- pass in inputs, get next-token logits -- so I decided to just attach it to the like this : I'm not sure, in retrospect, that "smuggling" the routing info out like that was the right choice. Perhaps a Hugging Face-like model where the LLM returns some kind of "output" class that contains and other stuff like this as explicit fields would be better. But this setup seemed to work for my use case, so I've left it as-is for now, with a mental note to revisit later. Next, I modified my training script to store the that we got when we did our forward pass in a list, and then to store that in the metadata associated with my checkpoints for later analysis. The training script I'm using is something I developed after working through Raschka's book, and it's kind of complicated -- although the core is essentially the same as the one we use to train our model in chapter 5, I've built it out to allow training across multiple GPUs , with all kinds of tweaks that I've learned about since. So I won't dig into that code in any depth in this post; if you've been following along with my various training posts in the past, though, and want to see how this all fits in, you can see the code that accumulates the routing information here (note that it uses so that we don't accumulate compute graph information over time), the code that passes those details into the checkpointing function here , and the updated checkpointing function here . So: with those changes, I had a plausible-looking MoE setup. I was confident that it would be able to train, but suspected that it would not be able to balance load between the experts well and would collapse to using a subset of them. It was time to give it a go. The first question was, what total number of experts, and how many active ones, should I have? I was pretty limited by what I could fit into my RTX 3090's VRAM, and what an appropriate training speed might be, but after a bit of fiddling around I came to the conclusion that six experts with two active would fit into memory and train reasonably quickly, and that didn't sound like a crazy balance. Mixtral is 8 experts, two active, for example. I wouldn't be able to fit in the batch size of 6 that I had used in the past when training 163M-parameter dense models; the largest batch size I could fit in was 3. (For those who've been following my previous LLM training, because I'm using gradient accumulation over 16 steps with the microbatch size of 6 -- for a global batch size of 96 -- I could get the same effect, and fit the MoE into VRAM, by going down to a microbatch size of 3, with 32 gradient accumulation steps.) The next question was how many tokens to train for. As an experiment I kicked it off with the same 3.2 billion tokens that I would normally use to train my 163M-parameter models. I knew that this larger model would need to be trained on more tokens than that, but I just wanted a reasonably serious run to see how the model behaved. The training script predicted that it would take just over two days to complete this experimental run, which was perfect. I had reached this point during the late afternoon on a Friday, and had stuff to do over the weekend that would keep me away from my computer, so that run length would mean that it would be ready for me on Monday. On Monday, I came back to see that it had completed properly: Over the training run, the loss declined nicely: Out of interest, I ran an evaluation script to see how it performed on the held-back test set that I normally use to compare my models: That ranked it better than the 163M-parameter models I'd trained using PyTorch, but worse than the original GPT-2 small, and also worse than the models I've trained with JAX (which I believe got lucky with their initial untrained random weights). That was a pretty solid result, but as this was an exploratory training run, there's no point in putting much weight on it. For a start, this model was clearly undertrained. The number of tokens, 3.2B, was Chinchilla-optimal for my 163M-parameter models, but this one had more than twice the total parameters at 446M, and had 220M active for each token. Larger models need more tokens to be well-trained. The more important result was the load-balancing between experts. I put together a notebook to ingest the metrics that I was writing to the checkpoints, and to create two charts for each layer: In the charts, for each global step, I plotted a line for each expert; if it had an average probability less than uniform / 3 -- that is, it was getting less than a third of the tokens that a perfectly flat distribution across all experts would give it -- then the line was white, meaning that it was being "starved". If it was getting a number of tokens between uniform / 3 and uniform * 2 -- then the line was green, because it was getting what felt like a reasonable number of tokens. And finally, if it was higher than uniform * 2, it was plotted in red: it was being overfed. The charts showed exactly the problem I was expecting to see. Let's look at layer 0: You can see that it started off pretty well-balanced, but rapidly came to prioritise expert 5; the remainder of the probability distribution looks like it must have been scattered over the other experts, with a slight preference for expert 3. The other layers came back with similar issues: certain experts were strongly preferred while others were starved: So the problem was real. The model was relying heavily on some experts and ignoring others. We needed to do extra stuff to balance the load across the experts. The problem with load-balancing across experts has been clear for quite some time, and it was covered in "Outrageously Large Neural Networks" back in 2017. The solution they used is simple, and clever: define an "auxiliary" loss, which goes up as the experts become more unbalanced, and down as they become more balanced. You can then add that (scaled by some amount) on to the normal cross entropy loss that you get by comparing the outputs from your model to the training targets, to get a combined loss number -- and then you back-propagate using that combined loss. By reducing the combined loss, then you optimise both for correctness -- is the model doing its job as an LLM -- and for balance across the experts. Of course, you need to be careful about the scaling factor that you use to adjust the auxiliary loss before adding it. If it's too small, then the load-balancing will be de-prioritised and you can wind up with imbalanced experts anyway. But if it's too large, the training process will prioritise balance over quality of the output, and you'll wind up with a crappy model that balances load across the experts beautifully. We'll come back to that shortly. The "Outrageously Large Neural Networks" paper's particular calculations for the auxiliary loss are a little complicated -- they generate two separate numbers and combine them -- and it was simplified in the GShard paper, and then again in Switch Transformers. I found the last of those reasonably easy to understand, and decided to adapt it. Let's start off with their formalisation -- but keep in mind that they had only one active expert per context vector, so we'll need to make some changes. Firstly, they define a per-expert number, f i -- the i subscript means that it's for expert i . That's a little intimidating-looking but is much simpler than it looks. They define it as "the fraction of tokens dispatched to expert i ", and in that light we can interpret it. The x ∈ ℬ that we're iterating over in the ∑ is basically: So, the stuff inside the ∑ is done once for each of our input context vectors across all sequences in a batch. The 1 { . . . } (which should be rendering with a kind of doubled-1 -- Chrome, as of this writing, doesn't handle that, though Firefox does) is meant to mean "1 if the condition in the braces is true, 0 if it's false". In argmax  p ( x ) = i , the function p is the "raw" probabilities from our router. As I mentioned earlier, they were doing softmax and then zeroing out the non-top- k values, so they had those raw numbers available. We'll come back to that in a moment. But for now, the condition inside the 1 { . . . } just means "true if this expert was the top one for this particular incoming context vector, false otherwise". Putting that all together, then we're just counting the number of tokens in our batch for which expert i is the top pick of the routing network. The 1 / T at the start then divides that by the number of tokens in the batch (which is what they mean by T ), and for a network with one active expert like the Switch Transformers ones, we've got -- as they say -- "the fraction of tokens [in this batch] dispatched to expert i ". So that works nicely in the one-active Switch Transformers world. But we can fairly simply extend it to handle multiple active experts, while keeping similar semantics. Let's say that we calculate how many of the context vectors in a batch were sent to expert i ; we can divide that by the number of tokens to get something equivalent. It still means "the fraction of tokens dispatched to expert i ". (Ab)using their notation, if we say that our number of active experts is k , then it might look something like this: Let's run with that for now. As well as f i , they define another per-expert number, P i : They describe this as "the fraction of the router probability allocated for expert i ". Again, we're iterating over every context vector in every sequence in the batch -- but here we're just adding together all of the raw probabilities for the expert for each one, then dividing by the number of elements in the batch. In other words, we're working out the expert's average probability across the entire batch. Much easier! And in our multiple-active-expert world, their equation works -- it means exactly the same thing. Now, both of these calculations, as they're expressed in the maths, depend on something that we're not currently calculating. Remember, Switch Transformers worked out the weights for each expert by doing a softmax across all of the raw logits that came out of the router's linear layer, then zeroing out the ones that were not in the top- k -- that is, for their k = 1 setup, all but the highest (argmax) one. So they had those "raw" softmaxed probabilities knocking around. But because we are replacing the non-top- k logits with − ∞ and then doing softmax, our router weights aren't the same -- they're just the relative probabilities of our selected k experts. For f i that doesn't matter; we can use our weights and easily identify the experts that were routed to for a given element in the batch, because they're the only ones that are not zero. But for P i we do need the pre-top- k 's logits from the router. And, not entirely coincidentally, we already have the code to make that available! In order to do those charts above -- the ones showing where experts were being starved and when they were being over-fed in the non-load-balanced training run -- we passed the raw, non-top- k 'ed, non softmaxed router logits, and the expert weights after the top- k and the softmax, out of the module's : ...and then added code to feed that through to the training loop because it was needed for the metrics that we used to generate those charts. The numbers that we kept for the "raw" side of things were the logits rather than the actual probabilities, but we can fix that with a simple softmax. So that means that in our training code, we already had the numbers to work out P i and f i for our experts. They needed to be combined to make up a single scalar auxiliary loss for the model when run over a batch, and Switch Transformers does that like this: If you imagine f as a vector containing all of the per-expert f i s, and P as a similar one containing the P i s, then that ∑ is a simple dot product, f · P . They then scale that up by the number of experts N , multiply by a scaling factor -- the one I mentioned earlier to balance load-balancing auxiliary loss against "real" cross entropy loss -- which they call α . (We'll come back to α later.) So, we have a set of calculations to work out an auxiliary loss; there are a few extra things I'd like to highlight before we dive in to the code. Let's imagine what a "perfect" router would look like from the perspective of these calculations, firstly for the Switch-style one-active-expert model, and then for our own more general case. Starting with f i ; it's the number of tokens in the batch for which expert i is the chosen one, divided by the number of tokens in the batch. We want all of our experts to get the same amount of "traffic", so for N experts, one active per token, clearly each one will be active 1 / N of the time. So that should be the value of f i if everything is balanced. Now let's think about P i . Again, we want each expert to receive 1 / N of the tokens -- so its average probability should also be 1 / N . So, that means that for perfect balance, all of our P i s and all of our f i s should be 1 / N . That means that when we do the ∑ in that loss calculation to work out the dot product, then for a perfectly balanced router, each one will contribute: There will be N of them, so that will come to ...and then we're multiplying by N to get the loss, so the result (disregarding the scaling factor α ) will be 1 . So, when balance across the experts is perfect, the Switch Transformers auxiliary loss has a value of 1 . However, they have only one active expert, and our equation is slightly different to allow for the fact that we have k of them. I won't go through the boring derivation again, but if we replay the maths above with k active experts, we get an auxiliary loss for the ideal, perfectly balanced router of k . That's not a problem in and of itself -- after all, this is just a number we're trying to minimise, and it's not super-important what we're trying to minimise it to. But it does matter when we're talking about α , because if the auxiliary loss is larger, we'll need to scale it down more to stop it from "drowning out" the signal from the actual training loss. The Switch Transformers paper explains what values they used for the scaling, but ours will be different because of the different number of active experts. The auxiliary loss calculation above only covers what happens in one layer, so we need to work out how to combine the contributions from all of the layers. In the paper, the only mention they make of multiple layers in this context is: For each Switch layer, this auxiliary loss is added to the total model loss during training I took that to mean that we just sum up the scaled auxiliary loss across all layers and then add that sum to the normal cross entropy loss. I think that's the most natural interpretation of what they are saying. So -- given the value of k for a perfectly-balanced router that I worked out above -- for the model I was planning to train, with 2 active experts, 12 layers, the auxiliary loss before scaling by α would be 24. This, by the way, is where my implementation differs from Mixtral's -- or, at least, the Hugging Face source code for it as of this writing. In that, they do something that feels a bit odd. They treat (for example) expert 1 on layer 1 as being the same as expert 1 on layer 2, and so on throughout the layers, then do the calculations just once. That feels a bit dodgy. After all, you can imagine that expert 1 on layer 1 might be being starved but its equivalent on layer 2 might be getting overfed, and the two would balance out. I don't know if that's an error in that implementation, or if there's something I'm missing. Conceivably I might test it some day by training another model using their loss function, but I suspect I won't get around to that. There's also something that made me hesitate a bit in the Switch Transformers paper, and which I think I still need to ponder a bit. That calculation for f i : ...did not look differentiable to me. Things like argmax (and our own equivalent's is-in-top-k ) are generally not. Indeed, they confirmed that shortly after defining it, but in a way that gave me pause: The objective can also be differentiated as the P -vector is differentiable, but the f -vector is not. The "objective" they're referring to is the auxiliary loss, and it makes me a bit uncomfortable that something that is defined in terms of A and B is differentiable if A is, but B isn't. My hand-wavy way of thinking about it right now is that the undifferentiable bit can be treated as a constant, so as long as part of the calculation is differentiable, the whole thing can be treated as such. But I'm not 100% happy with that, and need to think further. But now, I think, we've covered the maths for the auxiliary loss, so it's time to dive into the code! My old training loop had the following code to do the forward then the backward pass: Let's strip out all of the extra enhancements that I have accumulated there on top of the simple training code from the book; there's AMP , DDP and gradient accumulation and if we remove that it would simply look like this: Hopefully that's familiar! What I wanted to do was add in the auxiliary loss (if we were training an MoE), scaled by that α scaling factor. What I came up with (and again, here I've stripped out all of the stuff required by those enhancements): Note that I wanted to keep track of the router losses in that list as well in order to monitor them as the training run progressed, just like I normally monitor training loss (as you can see in the loss chart above). That's all pretty nice and simple -- if we are getting MoE routing info back from the model, then we call this new to work out the loss from the maths in the last section, and then add it on, scaled by , which is what I decided to call the somewhat-opaquely-named α from the Switch Transformers paper. Adding all of the AMP, DDP and gradient accumulation gubbins back in, the final code looked like this : So that was simple enough (for LLM-training values of simple). The code to route the from the training configuration file is not really worth going through, and nor is the code to save average, minimum and maximum values from into the checkpoint metadata, or to chart those (though I'll show the charts later). The interesting bit is, of course, that function. It's here and looks like this: Let's look at it from the outside in. We start with a list called . Remember, this has been passed back from the MoE model for a single forward pass of a batch. It contains one item for each Transformers layer in the model, and those items are pairs of . We start off with a total routing loss of zero, and then for each layer, we do some stuff to work out its routing loss, and then at the end of the loop, we add it on to our running total. Finally, we return the total. Obviously, the fun stuff is inside the loop :-) It's time for some of those tensor diagrams again. Let's remind ourselves of the shape of : Diagram 13 (repeated): as two 2-D views Each cell on the front face relates to one context vector in one sequence in our batch, and the "core" going into the cuboid from there (horizontally along the side view) is the weights we actually used when routing the context vector in question to the experts -- zero for unselected experts, some value between zero and one for the selected ones. Now let's go back to the code for a moment. We start off by using the shape of to work out what our different dimensions are: Then we do our first block of calculations, trying to work out f i from the maths, "the fraction of tokens dispatched to expert i " We want to do this efficiently with a vector calculation, working out all of the f i s (remember, there's one for each expert) in parallel. Our first step is to flatten out the grid into a single dimension -- essentially stacking all of the front view's columns on top of each other to make just one long column, like this: Diagram 20 : flattened, as two 2-D views ...or, more simply, as it's now just a 2-D Tensor, like this: Diagram 21 : flattened, as one 2-D view We can use PyTorch's method to do that without having to copy any data around in memory -- as the name suggests, it just returns a different view on the same data: Now, remember that these are the expert weights. Each one of those cells contains a number -- zero if the expert was not selected for the context vector that corresponded to it in the original layout, or some non-zero number if it was. So if we do this: ...then we'll get a tensor of the same shape, where we have if the weight was more than zero (that is, the expert was active for that context vector), otherwise. And that means that if we sum down those columns in diagram 21 , we'll get a new grid of one row, and columns, which represents the total number of times each expert was active in the given batch: Diagram 22 : as one 2-D view So, in code, we can just do this: If we divide that by the number of context vectors in the batch, we've got a new tensor, a row with columns -- the same shape as diagram 22 -- containing exactly what we want: That is, is a vector f in terms of the maths, containing all of the f i s that we want for our auxiliary loss calculation -- that is, all of the result across all experts for this: The calculations for the P i s are very similar. We start off with like this: Diagram 7 (repeated): as two 2-D views So, each cell on the front face relates to one context vector, and the "core" going into the cuboid from there is the set of raw routing logits for that context vector, one number per expert. Firstly we need to convert the into probabilities by running them through softmax, along the last dimension -- the one that is the horizontal axis on the side view: Then we do the same trick with to convert the result to a single column of lists of length (metaphorically) -- the same shape as in diagram 21 : Now if we add them up across the rows like this: Then we get a single row, column result that has, for each expert, the sum of all of the probabilities it had across all of the context vectors in the batch, just like the one we had in diagram 22 . We can divide that by the number of context vectors in the batch: ...and that's our vector P containing all of the P i s, where each is one of these: Finally, we can multiply all of the f i s and their corresponding P i s by each other, and sum the results, by using a dot product, and then multiply the result by the number of experts: ...and that's this bit done (apart from the α ): And that's our auxiliary code wrapped up! We've been through , and we've already seen the code that scaled it by α aka , so we have a training script with MoE auxiliary loss using the maths in the last section! Again, I hope that the diagrams helped with that workthrough. The code is the kind of thing where it's easy to scan through and get a vague understanding, but I think that visualising what's going on step-by-step is important if you want it to really stick. With the code in place, the next thing to do was to explore what the right value might be for α . In the "Switch Transformers" paper, they say: Finally, a hyper-parameter α is a multiplicative coefficient for these auxiliary losses; throughout this work we use an α = 10 − 2 which was sufficiently large to ensure load balancing while small enough to not to overwhelm the primary cross-entropy objective. We swept hyper-parameter ranges of α from 10 − 1 to 10 − 5 in powers of 10 and found 10 − 2 balanced load quickly without interfering with training loss. But, as I noted earlier, that worked for them with their single active experts, but because I had multiple, my auxiliary loss would be larger. Now, given that their "ideal" per-layer loss was 1, and mine was 2 for my planned training run, it sounded like using half of their recommended value, 5 × 10 − 3 , would be appropriate. However, I was also a little concerned about the number of layers interfering with things. The auxiliary loss, as we saw above, was a single value per layer, all of which were added together. With my "ideal" per-layer loss of 2, and 12 layers, that meant that the ideal across all layers was 24. Now, they mentioned a specific value for α , but didn't mention the number of layers they had, or if they swept for different possibilities across different numbers of layers. That seemed strange! I decided to work on the hypothesis that they had found that as the number of layers increased, you needed the total contribution of the auxiliary loss to scale up in proportion. That was only a guess based on trying to fit together the info in the paper, though, and could well be wrong. I was in enough doubt, though, that I felt it would be wise to do my own, minimal sweep over a few values for α . For each, I'd do a one-hour training run. At the end I'd check the training loss -- that is, the pure cross entropy loss saying how well the model was doing at its real purpose of modeling language -- and the auxiliary loss, showing how well-balanced its usage of experts was. I got these results: I also generated router usage maps like the ones I gave way back in this post for the two-day training run with no auxiliary loss. I won't put them all in there, but: So, on the basis of those results (and, of course, the fact that it fit well with the Switch Transformers paper's recommendation), I decided to use α = 0.005 . It was time to train this thing! I decided not to think too hard about the right number of tokens to train it on. The Chinchilla number, 20 times as many tokens as parameters, is a heuristic that works for dense models, but is not meant for MoEs. You'd intuitively think that the "ideal" number of tokens to train a model with 446,410,752 parameters, 219,697,152 active per token would be somewhere between the Chinchilla-optimal numbers for those two parameter counts. But overtraining isn't necessarily a bad thing, so long as it's on non-duplicated data (or less than four epochs over the same data ), and I had a 10B token dataset all set up from my previous experiment. So training it on what would be the Chinchilla-optimal number of tokens if it was just a 446,410,752-parameter dense model didn't sound like a bad idea, so long as I could do it in a reasonable amount of time. That meant 8,928,215,040 tokens, which I had in my normal training dataset without needing multiple epochs. A quick check -- running the training script with that number of tokens configured -- told me that it would take 90,823 global steps to complete, over about eight days. It looked like each checkpoint would take up 5.3GiB, and I had about 348 GiB free on my disk, so I calculated that I could checkpoint every 1,500 global steps -- that would work out as roughly once every three hours. Not great, but losing a maximum of three hours work in the case of a power outage (or cat jumping onto the PC's power button) is not the end of the world. So I set things up with this model configuration and this training configuration , and on my dedicated training box, , I kicked it off: Just less than eight days later, it completed: The loss chart looked like this: You can see that the normal cross entropy loss (just tagged as "loss" on the chart) decreases nice and smoothly from random (about 10.82 with the GPT-2 tokeniser) down to that final training loss of 3.211, with only a couple of tiny spikes. I've also plotted the auxiliary loss for the MoE routing, on the right-hand Y axis, and you can see that while near the start there were a few bumps (and the max values for single iterations spiked up from time to time), the average was generally pretty close to 24, our "ideal" number. Indeed, for the last checkpoint, the average over all iterations was an almost-perfect 24.0847. The notebook that I had to plot layer-by-layer expert starving/overfeeding came back with some lovely green plots showing nice even routing, too: A couple of issues near the start of the run, but for the last 75% of it, they're all a sea of green. Lovely. I ran my normal smoke test , based on Raschka's from the book: what do you get if you ask your model to complete "Every effort moves you" with 20 tokens, with a temperature of 1? Reasonably coherent! It was time to do some evals and comparisons. I ran my normal loss eval : on a held-back test set of 19,200 sequences of 1,024 tokens each, what was the model's average loss? Comparing this against other models that I've trained, and the OpenAI GPT-2 small and medium weights, we get this (it's in bold): Not too bad, though not amazing. It was better than any of my 163M-parameter models, and OpenAI's GPT-2 small. But it was a bit worse than (but close to) the 345M-parameter OpenAI GPT-2 medium, which has fewer parameters -- albeit more active ones per token. I decided to do a second eval. I have one that I call the IFT test -- fine-tune the model on an instruction-following dataset, until loss starts rising on its held-back eval dataset, then run a test set through to get answers to questions the model has not yet seen. I then bundle together the responses from a bunch of models and ask GPT 5.5 to compare them. It's an extension of Raschka's example in chapter 7 of the book, modified to make it easier to compare different models. You can see the scripts here and here , and there are more details here . I kicked off the first script, to fine-tune the model and get its responses, and then handed that plus a bunch of responses from other models to the LLM for it to compare them. The results came back like this (note this this is sorted by loss, the "IFT rank" column is how well it did comparitively in the eval): As you can see, the correlation between loss and performance on this eval is interestingly loose -- I have an ongoing series trying to work out why that might be . In particular, OpenAI's weights consistently outperform mine, and I'm determined to find out why. But it was reassuring, at least, that the new, big model came in at rank 3, beating all of my other ones, even if it still lost to that pesky 124M-parameter OpenAI GPT-2-small. So, there we have it: a GPT-2 small model converted to an MoE with 6 experts per layer, 2 active per token. Its loss on the test set is pretty much where you'd expect, and its IFT eval makes sense, modulo the OpenAI weights weirdness. What does that mean, and what should come next? In this post, I started with the GPT-2 code from " Build a Large Language Model (from Scratch) ", and my own training script (which was originally based on the training code from the book), added on mixture of experts support including the auxiliary loss calculations that you need to make it balance load across its experts properly. After eight days of training, we wound up with a decent, capable model. So that's all quite satisfying in an intellectual sense, and -- at least in terms of how well it did on the test loss -- it landed pretty much where you might expect. And one thing I'm sure of is that grinding through the calculations has been a great work-out for my skills with PyTorch and tensor operations. But the interesting thing about MoEs is that they -- in theory -- provide similar performance to dense models, at a lower cost in inference computing time. I think there are some interesting follow-up experiments I can do. OpenAI's weights tend to beat mine, so if I exclude them from comparisons (until I've worked out why), I could try to build a mental model for whether MoEs are a good way to spend my scarce computing resources when learning more about LLMs. I could: I'm sure there are other options, and I'd love to hear people's thoughts on what they might be. Anyway, I hope this post has been an interesting journey, and explained things well. Any feedback much appreciated -- in particular, on whether the diagrams helped. I only recently added D2 support to my static site generator, and it's entirely possible that I was overusing my new toy :-) So: thanks for reading, and as always, comments and questions are very welcome in the comments below. To protect against linkrot, I'll put proper references to these papers, because they're important background. In terms of the implementation, Mixtral is a bit odd. It does the Switch Transformers trick of doing the softmax, then zeroing out the non-top- k values -- but then it scales up the resulting routing weights by taking the sum, then dividing each value by that sum (which will be less than one). But the net effect of doing that is exactly the same as the "Outrageously Large Neural Networks" technique of replacing non-top- k with − ∞ .  ↩ One other thing: the masking out of all but the top- k experts does make me feel a little suspicious. It feels in a hand-waving kind of way a bit like "dead-end" code in router that we had originally, where we didn't use the outputs to weight the sum at the end. And it looks like there is a real mathematical concern there (even if it's not quite the same); in the "Outrageously Large Neural Networks" paper they say: While this form of sparsity creates some theoretically scary discontinuities in the output of gating function, we have not yet observed this to be a problem in practice My intuition for this is that because we have the weights in the flow of the calculations, back-propagation can still try to (say) reduce the weight for an expert that should not have been used for a particular forward pass. That is a bit problematic, though, because (due to the softmax) you would expect that reducing one expert's weight would force the others' to increase -- the results of a softmax have to sum to one. On consideration the Switch Transformers softmax-then-mask system feels like a better solution to me, because although the weights for the unselected experts are "invisible" to the backprop, having been masked out, gradients that increase a selected expert's weight will kind of automatically decrease those of all of the other, non-selected ones -- and vice versa. I need to ponder this a bit more, and perhaps try another training run with the alternative Switch-style setup and see how it compares. One random idea -- the softmax-first approach means that the weight of the selected expert(s) will be lower than it would be otherwise, so that reduces the amount of signal that the FFN adds to the context vectors. But then maybe the FFN would just get trained to emit larger outputs...? But anyway, enough waffling: let's put that aside for now, and for the rest of this post I'll describe the code as I wrote it, using the "Outrageously Large Neural Networks"/Mixtral model for the router.  ↩ Here's one that I tried, and which worked, but had a non-obvious bug. Let's imagine we have these logits for one context vector: For two active experts, we want to replace it with this: By default, sorts the values in decreasing order (and keeps the indices in an appropriate order to match). So in this case, we'd have the top-2 values looking like this: So if you do something like ...then you mask out all logits that are less than the lowest value in the top-k list with − ∞ . Neat! The problem with that was that there could be a tie. Imagine if, for one context vector, instead of the values above, we had these logits: We want to select the top 2 experts, and the smallest top-2 value would be 0.4254, of course. But if we just replace all values less than 0.4254 with − ∞ then we get this: We have three active experts rather than two! That's not good. While I didn't think that hitting this problem would be all that likely in practice, it was a definite error, and needed addressing -- hence the solution I settled on.  ↩ Imagine an extreme case, 256 experts but one active per token. In a given batch, you might expect tokens to be scattered pretty much randomly across experts, so you'd need to run ~all of them. If each expert only uses a small amount of your GPU's power to run through the small subset of the tokens that are allocated to it, and you're processing one expert at a time, you'll seriously underutilise the GPU. But I figured -- and confirmed when I finally got all of this running -- that with the sizes of model and numbers of active/total experts I was using, this wasn't a problem. Another issue that might come up in really large models is that too much stuff in a batch might wind up going through the same experts. Although we will later on add code to make sure that on average each expert receives roughly the same amount of context vectors, within a given batch things might be imbalanced. To see why that might be a problem, imagine an LLM that is so large that you have different GPUs -- or even different machines -- handling different experts. Something in the trillions of parameters scale, for example. If in your batch, expert i winds up doing most of the context vectors, then the GPU it's on is a bottleneck. The setup I'm describing in this post is sometimes called "token choice" routing. In a sense, each token has chosen -- or, rather, the router has chosen on its behalf -- which experts it "wants" to go to. "Expert choice" routing inverts that; you generate the same logits, but instead of choosing the top- k experts for each token, you choose the top- k tokens for each expert, so that you guarantee balance across the experts. Both of these are something for future investigations, I think :-)  ↩ " Adaptive Mixtures of Local Experts ", the 1991 paper that introduced the name (but used a model where all of the experts were run for each input -- they were thinking less in terms of efficiency, and more in terms of having different parts of a network specialise in different things). " Outrageously Large Neural Networks: The Sparsely-Gated Mixture-of-Experts Layer ", from 2017, which showed that you could use gating on MoE models to reduce the amount of computation by only running the "best" of the experts for each input. " GShard: Scaling Giant Models with Conditional Computation and Automatic Sharding ", from 2020, which built on the above, applied it specifically to Transformers models, and simplified some things -- more on that later. " Switch Transformers: Scaling to Trillion Parameter Models with Simple and Efficient Sparsity ", from 2021, which did some nice additional simplification. One showing how well-balanced the logits coming out of the router were -- that is, for each global step, how close was the model to having every expert getting exactly the same amount of utilisation in terms of the router's raw numbers. One showing how well-balanced things were after the top-k. This was a belt-and-braces thing: you can imagine a router that always favours a particular pair of experts, but only by a small amount, so the raw routing balance might look reasonably good (say, the two favoured ones get a probability of 0.2 and the others get 0.15 each), but the favoured two experts would wind up getting all of the "traffic". With α = 0 , as you'd expect to see from the very high auxiliary loss, things started going red and white pretty quickly -- it was focusing on specific experts and ignoring others. It didn't do too badly on training loss, though. With α = 0.001 , one tenth of the Switch Transformers number (and one fifth of what you'd expect to be ideal for ours by converting it naively), the load-balancing charts weren't quite so bad, but there was a speckling of red. I couldn't say for sure that it might not have managed to keep things reasonably balanced in a longer training run. α = 0.005 , however, looked really good! It actually had the lowest (best) training loss out of any of these options, and the auxiliary loss was not too bad, only 1.2 above the ideal value of 24. By the time we got to the α = 0.01 that was used by Switch Transformers (with their ideal auxiliary loss values that were half of ours), although the auxiliary loss continued downwards, training loss started worsening. This was certainly in line with what I would expect to see if the training started prioritising balance over actually creating a good LLM. Train a small model (say, the 163M-parameter size I've been messing with so far) on the same amount of compute as I spent on the MoE. How would it perform? My guess is that it would be worse. Train a dense 446M-parameter model on the same amount of compute and see how it matches up. It would be undertrained (it would need more calculations per token, so if I matched the compute, I'd have to train it on fewer tokens). I don't know if it would be better or worse. Train a Chinchilla-optimal dense model on the same amount of compute, and see how that did. To protect against linkrot, I'll put proper references to these papers, because they're important background. Jacobs, R. A., Jordan, M. I., Nowlan, S. J., & Hinton, G. E. (1991). Adaptive mixtures of local experts. Neural Computation , 3(1), 79–87 Shazeer, N., Mirhoseini, A., Maziarz, K., Davis, A., Le, Q., Hinton, G., & Dean, J. (2017). Outrageously large neural networks: The sparsely-gated mixture-of-experts layer. In International Conference on Learning Representations (ICLR 2017) . Lepikhin, D., Lee, H., Xu, Y., Chen, D., Firat, O., Huang, Y., Krikun, M., Shazeer, N., & Chen, Z. (2021). GShard: Scaling giant models with conditional computation and automatic sharding. In International Conference on Learning Representations (ICLR 2021) . Fedus, W., Zoph, B., & Shazeer, N. (2022). Switch Transformers: Scaling to trillion parameter models with simple and efficient sparsity. Journal of Machine Learning Research, 23(120), 1–39 . In terms of the implementation, Mixtral is a bit odd. It does the Switch Transformers trick of doing the softmax, then zeroing out the non-top- k values -- but then it scales up the resulting routing weights by taking the sum, then dividing each value by that sum (which will be less than one). But the net effect of doing that is exactly the same as the "Outrageously Large Neural Networks" technique of replacing non-top- k with − ∞ .  ↩ One other thing: the masking out of all but the top- k experts does make me feel a little suspicious. It feels in a hand-waving kind of way a bit like "dead-end" code in router that we had originally, where we didn't use the outputs to weight the sum at the end. And it looks like there is a real mathematical concern there (even if it's not quite the same); in the "Outrageously Large Neural Networks" paper they say: While this form of sparsity creates some theoretically scary discontinuities in the output of gating function, we have not yet observed this to be a problem in practice My intuition for this is that because we have the weights in the flow of the calculations, back-propagation can still try to (say) reduce the weight for an expert that should not have been used for a particular forward pass. That is a bit problematic, though, because (due to the softmax) you would expect that reducing one expert's weight would force the others' to increase -- the results of a softmax have to sum to one. On consideration the Switch Transformers softmax-then-mask system feels like a better solution to me, because although the weights for the unselected experts are "invisible" to the backprop, having been masked out, gradients that increase a selected expert's weight will kind of automatically decrease those of all of the other, non-selected ones -- and vice versa. I need to ponder this a bit more, and perhaps try another training run with the alternative Switch-style setup and see how it compares. One random idea -- the softmax-first approach means that the weight of the selected expert(s) will be lower than it would be otherwise, so that reduces the amount of signal that the FFN adds to the context vectors. But then maybe the FFN would just get trained to emit larger outputs...? But anyway, enough waffling: let's put that aside for now, and for the rest of this post I'll describe the code as I wrote it, using the "Outrageously Large Neural Networks"/Mixtral model for the router.  ↩ Here's one that I tried, and which worked, but had a non-obvious bug. Let's imagine we have these logits for one context vector: For two active experts, we want to replace it with this: By default, sorts the values in decreasing order (and keeps the indices in an appropriate order to match). So in this case, we'd have the top-2 values looking like this: So if you do something like ...then you mask out all logits that are less than the lowest value in the top-k list with − ∞ . Neat! The problem with that was that there could be a tie. Imagine if, for one context vector, instead of the values above, we had these logits: We want to select the top 2 experts, and the smallest top-2 value would be 0.4254, of course. But if we just replace all values less than 0.4254 with − ∞ then we get this: We have three active experts rather than two! That's not good. While I didn't think that hitting this problem would be all that likely in practice, it was a definite error, and needed addressing -- hence the solution I settled on.  ↩ Imagine an extreme case, 256 experts but one active per token. In a given batch, you might expect tokens to be scattered pretty much randomly across experts, so you'd need to run ~all of them. If each expert only uses a small amount of your GPU's power to run through the small subset of the tokens that are allocated to it, and you're processing one expert at a time, you'll seriously underutilise the GPU. But I figured -- and confirmed when I finally got all of this running -- that with the sizes of model and numbers of active/total experts I was using, this wasn't a problem. Another issue that might come up in really large models is that too much stuff in a batch might wind up going through the same experts. Although we will later on add code to make sure that on average each expert receives roughly the same amount of context vectors, within a given batch things might be imbalanced. To see why that might be a problem, imagine an LLM that is so large that you have different GPUs -- or even different machines -- handling different experts. Something in the trillions of parameters scale, for example. If in your batch, expert i winds up doing most of the context vectors, then the GPU it's on is a bottleneck. The setup I'm describing in this post is sometimes called "token choice" routing. In a sense, each token has chosen -- or, rather, the router has chosen on its behalf -- which experts it "wants" to go to. "Expert choice" routing inverts that; you generate the same logits, but instead of choosing the top- k experts for each token, you choose the top- k tokens for each expert, so that you guarantee balance across the experts. Both of these are something for future investigations, I think :-)  ↩

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Giles's blog 1 months ago

Putting my JAX-trained models on the Hugging Face Hub

I hadn't uploaded the models that I trained using JAX to the Hugging Face Hub because Transformers has been PyTorch-only since version 5 (though they say they're working to add interoperability with JAX in the future), so it would have been tough to get them working natively with and the like. But then it dawned on me that I'd already written a conversion script that could take my JAX safetensors files and convert them into ones compatible with my PyTorch code. It's actually those converted models that I use for my evals -- so I could use my existing PyTorch script to upload them. So, I've now uploaded PyTorch-compatible versions of all of my JAX-trained models: " Writing an LLM from scratch, part 34b -- from bigrams to GPT-2, one component at a time (in JAX) " " Why do OpenAI's GPT-2 weights beat mine? Part three: testing overtraining " " A quick(ish) Chinchilla check " I've also added links to the posts in question. -- the first full LLM trained in the post, in the "Adding LayerNorm" section. -- the second full LLM trained in the post, in the "Dropout" section. -- the third full LLM trained in the post, in the "Adding bias to the MHA output projections" section. -- the single-epoch, double-Chinchilla-tokens model. -- the model trained on two epochs over the Chinchilla-optimal number of tokens. -- the model. -- the model.

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Giles's blog 1 months ago

Why do OpenAI's GPT-2 weights beat mine? Part four: digging into dropout

I'm still digging into a mystery about the models I've been training; although an increasing number of them beat the OpenAI GPT-2 small weights on the narrow technical measure of the loss they get on a test set, they're not as good at an instruction-fine-tuning test . While reading about MoE models, I came across this paragraph in the Switch Transformers paper : Our paper considers the common NLP approach of pre-training on a large corpus followed by fine-tuning on smaller downstream tasks such as summarization or question answering. One issue that naturally arises is overfitting since many fine-tuning tasks have very few examples. During fine-tuning of standard Transformers, Raffel et al. (2019) use dropout (Srivastava et al., 2014) at each layer to prevent overfitting. So far, when running my IFT test, I'd been aiming to use the same dropout setting for the fine tune as the model concerned had used in its original pre-training. That was just because it seemed natural. But the goal of dropout is to prevent overfitting when training over multiple epochs -- or, at least, that's how most of what I've read explains why we don't need it on modern single-epoch training runs over large datasets. If that's the case, though, when we do multiple epochs for a fine-tune with a more restricted dataset -- exactly what I was doing for the IFT test -- it might make sense to use dropout, regardless of whether or not the model was pre-trained with it. The fine-tuning setup already tries to avoid overfitting by bailing out when a validation loss starts rising, but dropout might still help it avoid overfitting prematurely. On the other hand, something felt a little wrong about fine-tuning a model with dropout if its pre-training had happened without it. A model pre-trained with dropout have been trained on billions of tokens, and so the model will have spent a lot of effort learning to overcome the issues that dropout causes, but one trained without it won't have that benefit. Suddenly exposing it to dropout in a much shorter fine-tuning run felt rather like asking someone who rarely drinks alcohol to take a few shots of whisky; I felt that the models might not be prepared for the effects. As I looked into this more, I noticed another surprising thing -- there was an error in the configuration that I was using when fine-tuning the OpenAI models, both small and medium. They were originally trained with dropout (or so it's believed -- the paper doesn't say, but " Build a Large Language Model (from Scratch) " says that they were, and this config on the Hugging Face GPT-2 code agrees). But that actually made my original puzzle of why they outperformed my models on the IFT test seem even more perplexing, at least in the light of this idea. If dropout was a good thing for fine-tuning, then so far they had been penalised by not using it -- that is, they were even further ahead of my own models than I thought they were. It was time to take a careful look. I fixed the config for the OpenAI weights so that my setup had dropout set to 0.1 for them, then carefully revisited the config for all of my own models, and made sure that those ones matched reality (which they did). Now, the IFT test that I've been running has two phases: Now, the nice thing about the judge script was that it didn't really care whether the result files it got came from different models or the same one; it just printed out a mapping of result files to scores. So I realised I could use it to do a comparison of all models with all possible dropout settings. For all of the models, I ran the three times, once with each of the dropout settings: , , and . Then I sent all of the resulting result files -- all models, and all dropout options for each -- to the LLM judge in one go, to see what it came up with. Here are the results, consolidated into one table. For each model, I have: It's quite an intimidating wall of numbers, but there's a bunch of interesting stuff there. Firstly: I've put the IFT score for each model where it was trained with the opposite of its pre-training dropout in bold. Let's look at the non-bold numbers first, though. If you scan down through the models, you'll see that the non-bold IFT scores -- that is, the one where the IFT test was done with the dropout, and then the one where it was done with dropout set explicitly to the same value as the one -- are identical in every case. That is a really reassuring sanity check. Remember, each of those numbers came from a different run of the script -- but because there is a fixed random seed, they should have been identical. They were presented to in the same way as a separate model's response. The fact that it came up with identical scores tells us that it judged them as being equal, which is solid evidence for its consistency in judging results in this run (which is something that can be hard to guarantee with an LLM). Similarly, if you look through the numbers of training epochs, the epochs for each one matches the epochs with the dropout forced to match the model's pre-training setting, which is also reassuring -- it's certainly what you'd expect given a fixed random seed. Looking just at the and epochs columns, you can see something else interesting. With dropout forced to be on, the number of fine-tuning epochs is always higher than the number of epochs with no dropout, except in the case of the OpenAI medium weights and "Cloud FineWeb, 8x B200 160 GiB", where it's the same. That makes intuitive sense, I think. If you're discarding 10% of your activations when training a model, you'd expect it to take longer to converge. But now let's look at the size of those changes. If you compare the increase in the number of epochs needed to train with dropout forced to be on, you can see that the change is much larger for those models that were pre-trained without dropout. The first of them, for example, "JAX, overtrained one long epoch", went up from 3 epochs to 19! That's way larger than, say, the change from 5 to 7 for "JAX, no MHA bias, with dropout". That was the first indication that something interesting was happening when using dropout to fine-tune models that had been pre-trained without it. One question is whether so many epochs on a small dataset might just be a bad idea, regardless of whether the early-stopping from validation loss helps avoid overfitting. However, way back I did some investigations into the effect of the number of epochs of training, and found that while varying it changed the results somewhat -- as you'd expect -- the effect was surprisingly small, and didn't change anything about the fundamental mystery of why the GPT-2 weights were so much better than mine. So I think we can put that aside for now. Now let's dig into those scores. We can divide them into two groups; models that were helped by adding dropout, and models that were harmed. In the "helped" group, we have these: In the "harmed" group, we have: There are some patterns there, and I think that putting them into a table sorted by the score increase/decrease is a good way to visualise them: One thing is pretty clear: with two exceptions, the models that were pre-trained with dropout are at the top, and the models that were pre-trained without are at the bottom. Of the exceptions, is so close to "Local FineWeb train" that perhaps its position could be due to some kind of noise. is much more puzzling, however. It's a real outlier in terms of the models that were pre-trained with no dropout, with its improvement of 0.29 compared to the next closest, with a decrease of 3.4. But if we disregard that outlier for the time being, the pattern actually does fit rather well into my original suspicion about the risks of switching on dropout when fine-tuning a model that was pre-trained without it. They really don't handle it very well! On the other hand, it rather does put the kibosh on the idea that I based on the quote near the start of this post -- that fine-tuning with dropout is a good way to help the model learn with less risk of overfitting. In my particular case -- these specific models, this particular fine-tuning task, with this IFT data -- dropout seems to generally have a negative effect on the fine-tuning results. Even of those that were pre-trained with dropout, more than half got worse results when fine-tuned with it. Another interesting thing that stands out from the table above is that the JAX models are at the top and the bottom. The model that was pre-trained with dropout was the one that gained the most from fine-tuning with it (or, contrariwise, lost out the most if fine-tuned without it). The models that were pre-trained without were the ones that were most harmed by being fine-tuned with. If you look further up, at the original table of results, you'll see that the JAX models all did better than my other ones (which were trained using PyTorch) in terms of loss on my test set (the second column). I've been chalking that up to two things: the JAX models would have started their pre-training with different random initial weights, and they were all trained in full-fat float32 (unlike the PyTorch models, which used AMP ). Given that I found that AMP had a negligible impact on training loss, I've been thinking that the "initial weights" aspect was the more important -- by chance, they happened to start in a place on the loss landscape with a route to a better minimum during training. I don't think there's anything in these results that pushes against that theory, but it does suggest that there's some kind of "fragility" in the minima they have found; changing dropout from what they were pre-trained with seems to knock them out of their exceptional positions. And finally, of course, the mystery around 's anomalous position remains. I honestly don't have any theories at all about that one right now. Interestingly, it was trained with an identical configuration to our other (but less extreme) exception, . The difference is that the first was trained on my local RTX 3090, using gradient accumulation to get a global batch size of 96, while the second was trained on a cloud machine with 8x A100 GPUs with 40 GiB each, which (using DDP) got a global batch across all GPUs of 96 without gradient accumulation. There's something going on there, but I'm not sure what. Anyway, for now, I think it's time to wrap this one up. The idea I started this post with -- that using dropout for the fine-tuning part of all of these IFT tests might be a good idea to avoid issues from the multi-epoch nature of the fine-tuning -- doesn't seem to hold up. Dropout in the fine-tuning turned out to be more often harmful than helpful, regardless of whether a model was originally pre-trained with dropout or not. However, exactly how harmful it was seemed to be pretty strongly correlated with whether the model was originally pre-trained with dropout, the oddity of aside. I think that while working further on solving this mystery, I should stick to not using dropout. Because adding it on for the OpenAI models made their performance worse, I think that's principled -- it's quite the opposite of making a choice to try to sweep the mystery I'm trying to solve under the carpet :-) So that means that my task in future posts in this series is to explain this table (to go back to the format I've been using for the previous posts) -- the dropout numbers from the table above, with rank added: The OpenAI small model still has a 4.54-point lead over the best of my own models, "JAX, no MHA bias, no dropout". Previously I'd considered data quality as a possibility, and felt it was an unlikely cause. I now think I may have been premature in that, and it's worth looking into. Those two "Local FineWeb-Edu" models near the bottom were trained with sub-optimal hyperparameters and -- while they don't do super-well in this test -- they do much better than their raw test loss numbers might suggest. But while thinking about dropout, it occurred to me that there were other levers that I'd pulled in my interventions into my original base model that might be worth investigating 1 : So, plenty of further possibilities for this investigation. Stay tuned! Other interventions that I decided not to check, at least at this point: Firstly, for each model, I run . This script trains the specified model on an IFT dataset until validation loss starts rising. It then uses the model from before that loss started going up to generate responses to a test set, and saves those responses to disk. I made a small change to it so that the dropout used in the fine-tuning phase was a required command-line parameter, with three options: -- that is, what the model was pre-trained with -- , which forced it to 0.1, or , to force it to 0. Next, I pass all of the saved test responses for all models into a second script, , which sends them to an LLM judge so that each model can get a score. The script averages all scores across all questions for each model. Check the link for more details of how that script works and tries to achieve consistency across models and responses. Its loss on my test set -- the technical measure of quality I mentioned near the start. They're sorted by that column. Whether or not the base training run -- the pre-train -- had dropout. The number of fine-tuning epochs before validation loss started rising when the IFT run used a dropout setting identical to the pre-training ( ). The score that the model thus trained got from the LLM judge. The fine-tuning epochs with dropout forced to be . The score for the dropout-off model. The fine-tuning epochs for dropout forced to be . And finally the score for the resulting model from that. "JAX, no MHA bias, with dropout", which was pre-trained with dropout and gained 4.52 points when the IFT run used dropout. , which was pre-trained without dropout and gained 0.29 points. "Cloud FineWeb, 8x A100 40 GiB", which was pre-trained with dropout and gained 1.99 points. , which was pre-trained with dropout and gained 0.96 points. "Local FineWeb-Edu extended train", which was pre-trained with dropout and gained 2.52 points. "Local FineWeb-Edu train", which was pre-trained with dropout and gained 2.44 points. "OpenAI weights: medium", which was pre-trained with dropout and lost 1.35 points. "JAX, overtrained one long epoch", which was pre-trained without dropout and lost 12.6 points. "JAX, overtrained two normal epochs", which was pre-trained without dropout and lost 6.8 points. "JAX, with MHA bias, no dropout", which was pre-trained without dropout and lost 5.49 points. "JAX, no MHA bias, no dropout", which was pre-trained without dropout and lost 16.21 points. "OpenAI weights: small", which was pre-trained with dropout and lost 2.51 points. , which was pre-trained without dropout and lost 3.4 points. , which was pre-trained with dropout and lost 2.21 points. "Cloud FineWeb, 8x H100 80 GiB", which was pre-trained with dropout and lost 0.31 points. "Cloud FineWeb, 8x A100 80 GiB", which was pre-trained with dropout and lost 0.09 points. "Cloud FineWeb, 8x B200 160 GiB", which was pre-trained with dropout and lost 2.65 points. "Local FineWeb train", which was pre-trained with dropout and lost 3.46 points. Weight tying -- I honestly can't think of a reason why it might make a model better for this kind of task, but it certainly is true that the OpenAI weights use it -- while none of the ones of mine that I've been testing do. That feels worth a quick look, especially given that I have a copy of a model that I trained using it lying around. AMP. Apart from "JAX, no MHA bias, with dropout", all of the JAX models -- trained without AMP -- did pretty well in this test (though not close to the OpenAI models). And again I have a PyTorch model that was trained without AMP on my disk somewhere, so I may as well throw it in and see how it does. The learning rate. All of these fine-tunes are happening with a fixed learning rate of 0.00005. While I really don't want to do some kind of sweep across multiple values for all of these models, perhaps there's some way I can try to relate the fine-tuning learning rate to what the models are "used to" from pre-training and see if that helps? Other interventions that I decided not to check, at least at this point: QKV bias: all of my PyTorch models in the table apart from the two ones use it, so that's been thoroughly tested. Weight decay: again, we have a mixture of values for that in the table and there's no obvious pattern. Gradient clipping: likewise.

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Giles's blog 1 months ago

Adding diagrams to my static site generator with D2

A lot of the time when I've been writing posts for this blog, I've felt that a diagram would really help. But they're a pain to produce well, and I think I underuse them as a result. I wanted to fix that, and wound up adding D2 support to my static site generator. I think it works pretty well! In the past, I've tried drawing my own diagrams in LibreOffice and exporting as SVG, but my complete lack of artistic skill doesn't help: Asking an AI to do it for me helped in simple cases: ...but with something less standard (there must be a million neural network diagrams in their training sets) it can be really fiddly to get something right. I did some investigations into the various diagram-generating tools out there, and decided to give D2 a go. It has a simple language for specifying what your diagram should show, and the output is pretty nice: Here's the source for that diagram: That looks pretty clear to me! So now, in the source for my blog posts, I have a directory. That contains subdirectories -- by convention, I create one for each post that needs diagrams -- and D2 files. These can be generated automatically when I publish: (Hat tip to Evan Hahn for the method on , which I wasn't aware of.) The flags on the command line took a little bit of fiddling; the just gets rid of the large margins that D2 puts around the diagram by default, but the others are to tell it to use the ELK layout package with particular formatting. Its default layout has curvy lines, and I prefer the closer-to-right-angle ones that ELK provides. Another awkward bit was in scaling; the file that is generated by that command comes out pretty large ( you can see it full-size here ). By default, I allow images inlined into my posts to be as wide as the text, but that would still be too large here. I use to convert the markdown source for my posts into HTML, and there isn't any way to tell it what size an image should be using markdown-ish syntax. So for now, instead of embedding images the normal markdown way, like this: ...for these D2-generated ones I'll just embed a normal tag like this: ...so that I can control the size. Perhaps more work needed there. At some point I may go back and update my old diagrams -- at least, the really ugly hand-drawn ones -- to use this. And a random thought: perhaps it might also make sense to include the D2 source somehow on the blog? I can imagine that it could help with accessibility in some situations, and perhaps also for any LLMs stopping by. Will have to ponder that a bit more. What do you think? Does the D2 diagram look good to you? Or is there a better diagramming package that might work better?

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Giles's blog 1 months ago

Use the built-in GELU, don't roll your own!

Unsurprisingly, PyTorch's own built-in GELU function is faster than the hand-rolled one I've been using to date. But I was surprised at how much faster using it made things when training my models. I discovered this accidentally just now while working on something unrelated, but am logging the details here for anyone else that might find it useful. The headline numbers: the same code, training the same model on the same data, ran at about: That's a 20% increase in throughput for both of the built-in versions -- definitely nothing to be sneezed at. And what is particularly interesting is that there aren't that many GELUs going on -- it's a GPT-2 small-style model, with 12 layers. So that's 12 GELUs handling tensors shaped , which is for my training setup. Given that the rest of the model is doing all of the normal full attention stuff for GPT-2, it's really surprising that the GELUs alone must have been taking up so much of the time. The throughput numbers mean that we must have been spending about 17% of our time on the extra overhead from the hand-rolled version, so that sets a lower bound for how much time the GELUs were taking up. More info below the fold. Back when I was doing the "interventions" part of my LLM from scratch series , training dozens of GPT-2 small-sized models in the cloud and on my local machines, to keep things simple I used the original model code from Raschka's book. That happens to have its own implementation of the GELU function -- you can see my copy here . I'm not that sure why the hand-rolled version is in there -- he covers the maths, but the specific implementation isn't explained in that much depth, and it seems rather like boilerplate, just a "type this in and use it" kind of thing. By contrast, for example, while he does explain the maths behind cross-entropy loss in similar detail, we use the built-in function for it rather than coding it up ourselves. When I switched to using JAX for my own from-scratch implementation , I decided to not bother porting the boilerplate, and just used JAX's own built-in version . I was revisiting the PyTorch code -- I'm in the process of extending it with mixture-of-experts support, about which more in a later post -- and decided to switch from the hand-written GELU to the PyTorch one just to tidy things up a bit. I noticed something interesting -- my new MoE code suddenly seemed to speed up. Was that a mirage? Or had I discovered part -- or even all -- of the reason why the JAX code was so much faster than the PyTorch code? With PyTorch, I was typically getting training speeds of about 21,000 tokens per second, while in JAX I was getting 24,000 tps or so. I'd been chalking that up to JAX's JIT compilation, but could it have been just a result of a random implementation choice I'd made? I did three partial test training runs, letting each one run for 20 minutes to allow the training speed to settle down from any startup overhead. Firstly, with the old hand-coded GELU: So it was getting 20,920 on average over those 257 global steps. That speed was in line with the original run of the configuration I was using. Next, I introduced the built-in PyTorch GELU with no arguments: That does the full calculations for GELU, rather than using the -based approximation that the hand-rolled code did. After 20 minutes, it looked like this: So this time we were getting 25,134 tokens per second -- 20% faster! By default, PyTorch's GELU uses an exact calculation of the function -- the hand-written code from the book uses an approximation using . Luckily, you can get that same approximation from PyTorch: So, training with that for 20 minutes: 25,142 tokens per second -- basically the same as the non-approximate version. So: switching to the built-in GELU made my PyTorch code run 20% faster, at about 25,000 tps rather than 21,000. My JAX code, which used JAX's built-in GELU, ran at around 24,000 tps. I'd actually found that rather surprising, because in JAX I was training in full-fat 32-bit floating point, while in PyTorch I was using Automatic Mixed Precision (AMP) -- a special mode that allows it to use 16-bit calculations where it won't hurt the model much. I'd found that AMP gave PyTorch a huge speedup -- from 15,402 tps to 19,797 on one test. So JAX without AMP being so much faster than PyTorch with AMP was a bit of a surprise. Its JIT is pretty amazing, but I didn't expect it to be that much faster. Now I think that we have at least part of an explanation. I was using JAX's built-in GELU (interestingly, with its default parameters, which means that it used the approximation), but the PyTorch code was using the hand-rolled one, and that unduly penalised it and erased some of the gains it got from AMP. If I really wanted to dig into this, I suppose I might try JAX with a hand-rolled GELU to see what happened. My guess is that because of its JIT, it might actually handle it better -- the whole hand-rolled thing could be compiled into one thing on the GPU. Perhaps it would also be interesting to try the non-AMP PyTorch code with the built-in GELU. But I doubt that would really be the best use of my time (and my electricity bill), so I'll leave it here. On the other hand, I do intend to have a look at in the future, to see what kind of speedup I can get from it. And it might be able to compile and fuse together the hand-rolled GELU -- so that would be an interesting thing to experiment with in that post: does the built-in GELU advantage disappear if we're compiling? But anyway, for now, lesson learned: use built-in PyTorch modules when you can. It's a pretty obvious one ;-) [Update] On X, Sebastian Raschka noted that he used the approximate version of GELU in his code so that the models were compatible with the OpenAI weights -- they were trained with that version, so they may behave slightly differently if you use the "pure" version. That's a great point, and so I've updated my own copy of the code to use . 21,000 tokens per second using the hand-rolled GELU from Sebastian Raschka 's book " Build a Large Language Model (from Scratch) ". 25,000 tokens per second using PyTorch's built-in GELU with no arguments. 25,000 tokens per second using the built-in GELU with , which uses the same maths as Raschka's version under the hood.

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Giles's blog 2 months ago

A quick(ish) Chinchilla check

I recently overtrained a couple of GPT-2 style models , training them both on 40 tokens per parameter rather than the 20 per parameter that is generally regarded as "Chinchilla-optimal". The normal heuristic is that instead of doing that, you should scale up the number of tokens and the number of parameters equally -- so I would have been better off scaling up the model by 2 and the token count by the same amount. By doing that, I should expect to get a better model in terms of loss on my held-back test set than I did with my 40-tokens-per-parameter models. My training machine wasn't doing anything, so I decided to give that a go. Would the Chinchilla rule-of-thumb hold up? As you might expect, it did. But it was a surprisingly close-run thing, and could conceivably have been in the noise. Let's take a look. If you already know all about the Chinchilla paper -- regular readers in particular must be sick and tired of it by now :-) -- then click here to skip this section . In "Training Compute-Optimal Large Language Models" , which is always called the Chinchilla paper after the name of the model they trained at the end, the authors tried to work out the optimal number of tokens to train an LLM on based on its number of parameters. In particular, they were pushing back on a trend they were seeing at the time, where people were making models ever-larger, but not increasing the amount of data they were training on. The authors were all at Google DeepMind, and this was the kind of project that only a large lab could do: they trained "over 400 language models ranging from 70 million to over 16 billion parameters on 5 to 500 billion tokens". Their conclusion was "for compute-optimal training, the model size and the number of training tokens should be scaled equally: for every doubling of model size the number of training tokens should also be doubled". They don't actually state an overall optimal number of tokens to train on in the paper, but in table 3 they provide an estimate of the optimal training FLOPs and tokens for models of various sizes, and it's approximately 20 tokens per parameter. That number has become a heuristic, and people talk about a model as being trained for the Chinchilla-optimal number of tokens. Models that were trained on fewer tokens per parameter are referred to as "undertrained", and models that were trained on more as "overtrained". It's worth noting that overtraining a model is not, in itself, a bad thing. If you have a model of a particular size and you continue training it past the Chinchilla-optimal number of tokens, it will -- in general -- get better. The point of the heuristic is that doing that is not the best way to spend whatever budget you have in terms of compute time. You'll get better results, as they say, by scaling the number of tokens and the number of parameters equally. But let's say you're creating a model for specific target hardware -- say, a mobile device. You have a hard restriction on how large the model can be -- the device has only so much RAM to hold it. So it might make sense to overtrain to get a better model. 1 But if you're not so limited in how many parameters you can use, then you should indeed scale the model up, and that's what I wanted to try. How would that work? A week or two back, I was investigating whether I could make my GPT-2 style models better at a specific instruction-following task by overtraining them . The details of that experiment aren't important here, but what it meant was that I had three GPT-2-style models, each of exactly the same size, roughly 163M parameters When I tested them against a held-back test set of sequences -- stuff that they'd never seen before -- they got results rather like you might expect: A lower loss is better, and you can see that the longer-trained models were noticeably better than the Chinchilla-optimal one. The difference between them was tiny; they were trained starting with the same initial weights, and the training runs themselves were deterministic, but a difference of 0.05% in loss doesn't seem like it could be meaningful -- an extra batch for one or one fewer for the other could easily swap them around, you'd think. Now, these models each had 163,009,536 parameters -- they were the small-size model from the GPT-2 paper , modified to not have QKV bias or weight-tying. had been trained on 3,260,190,720 tokens (rounded up to fit into a round number of full batches), and the other two on 6,520,381,440 tokens each -- double the amount (rounded up too). What I needed to do for my Chinchilla check was to try training a model that used the same amount of compute, scaling the parameters and the number of training tokens equally. Because training compute increases roughly linearly with both parameters and tokens, that would mean scaling both up by 2 , giving us: ...and thus 4,610,605,920 tokens. How to scale the model up? In the GPT-2 paper, they train four models: I wanted to scale my own model up from 163M parameters to about 231M. Which of those numbers would I want to increase, and by how much? The first thing that stands out is that the number of heads is always 1/64th of the number of embedding dimensions. So that sorted that one out. I just needed to adjust the number of layers, and the number of embedding dimensions, but ensure that the latter was a multiple of 64. I decided to see if I could fit some kind of curve to the relationship between the number of parameters and the GPT-2 authors' choices. This was made a bit more complicated by one thing: they were using weight-tying, and I was not. That meant that they re-used the embedding matrix at the start of the LLM as an output head at the end -- which is why they had 38M fewer parameters. Embeddings and the output head make up a surprisingly large percentage of the parameters for small models like this -- about 47% without weight-tying, 23% with. I couldn't work out a solid way to scale things up and wound up doing some rather messy hacking around in a spreadsheet . I came up with two proposed model sizes that were within a couple of percentage points of the right size: Interestingly, I found that because could only change in increments/decrements of 64, it was a pretty coarse control -- my first attempt at making a model changed it to the next step down, 832, but that led to a model that was 9.25% too small. That was an interesting first lesson. I'd previously been thinking of the Chinchilla rule as being something like "don't double the tokens, just scale the model and the tokens equally". But that "just" was wrong. Scaling a model is hard -- even with just two dials to fiddle with, like in this case, it was tricky to get something right -- and I can't say for sure that my choices were the right ones. Anyway, the next step was to double-check that these models would use the right amount of compute to train. As I said earlier, the compute time scales roughly linearly with the number of parameters. Let's dig into that "roughly". Different kinds of parameters take different amounts of FLOPs to train, and scale differently with things like the embedding dimensions, sequence length, and so on. Now, for very large models, a lot of that comes out in the wash, but with tiny models like these where the embeddings make up such a large proportion of the parameters, it might matter. Conveniently, in appendix F of the Chinchilla paper, they provide a set of formulae for estimating the number of training FLOPs for a normal dense LLM like these ones. I coded that up into a script that, given the JSON configuration files I was using for my models and training runs, would work out the number of FLOPs for a single epoch of training. It didn't take account of the fact that my real training runs round the number of tokens up so that we do a round number of full batches, but I felt that so long as the results weren't very close that wouldn't matter. I got these results (multiplying the two-epoch numbers by two): The numbers were indeed different enough that I wasn't worried about the batch-rounding. And the good news was that and would indeed use slightly more and slightly less compute to train than the overtrained models -- about 4.6% more and 4% less respectively. A true Chinchilla-equivalent model would lie somewhere between them. It was time to train some models! I kicked off the run for the model first. Because it was bigger than the 163M models I'd been training, I couldn't fit such large batches into my VRAM; previously I'd been running with a batch size of 6, and now I could only fit in a batch of 4. Luckily, though, I was using gradient accumulation , so by bumping that up from 16 steps to 24 steps I could keep the same overall batch size and keep the training runs comparable. Even despite that, the training run ran out of VRAM about 60 hours in -- I'm guessing due to VRAM fragmentation, as I did not have set to -- but I was able to restart from the most recent checkpoint and complete the run. After just less than four days total training time, it completed. When it was done, I copied the last checkpoint 4 over to my dev box, , and ran my standard smoke test against it, asking it to complete "Every effort moves you" with 20 tokens, using greedy sampling. I got something reasonably coherent: Next, I converted the safetensors file -- which had been saved by my JAX code -- into a format compatible with my PyTorch code, because that's what I use for evals. I ran another smoke test (this one with temperature 1): Very spiritual. Next, it was time to work out the loss on my held-back test set: Well, it was certainly better than the 3.324953 that the best of the overtrained models got -- but only by a bit over 1% better. Interesting! I decided to train the second model, . This one crashed mid-way through with an error that I've seen before: I'm going to have to investigate that more in future, but for now, I just restarted from the checkpoint, and again after a bit less than four days, I had a model. The JAX smoke test was solid: ...and so was the PyTorch one: Both quite commercial this time! It was time for the proper test loss eval: So, slightly worse than the 3.280028 from the larger model, better than the 3.324953 from the best overtrained one. Time to put this all together. Here's an updated version of the table from the start of this post; I've added in the two new models, and the improvement they each had over in both absolute terms and as a percentage rounded to 3sf. Now, unlike the overtrained models, prior to training these two new ones started with different initial weights to the one -- after all, they had to, because they had more of them! A while back, I did a bit of analysis of how random variation in weight initialisation can change the resulting test loss. It wasn't anything in-depth, but I trained three models with different explicit seeds set prior to the model initialisation, but with the same seed set before the training run started 5 . Those three models wound up with test losses of 3.681356, 3.673943, and 3.664345. Doing statistics with three data points is a bit flaky, but the cost of training models is so high that I'll leave the Proper Science to the likes of Google DeepMind and wing it :-) Now, piling statistical flakiness on statistical flakiness, we'll compare these. You'd normally expect about two thirds of results to be within one SD of the mean, 95.4% to be within two SDs, and 99.7% to be within three. Three SDs on that (yes, different, I know) distribution is 0.025587. That's smaller than both of the improvements that our Chinchilla-optimal runs had over the overtrained ones. So what does that tell us? Well, perhaps not much given the statistical flakiness. But I think it is useful directionally. It suggests that we might be able to take these results seriously as an improvement, and that Chinchilla held: scaling up the model and the number of tokens evenly did give us a better model than just scaling up the number of tokens. In particular, the fact that the loss for was lower -- even though it had 4% less compute spent on it than the overtrained models -- was encouraging. But it's certainly far from a slam-dunk. A larger test, training lots of overtrained models and lots of Chinchilla-optimal ones, all with different random seeds, would give actual real serious data. Not worth it for me, and perhaps not for anyone. I wanted to do a quick sanity check of the Chinchilla heuristic of 20 tokens per parameter. I came up with results that were certainly in line with it -- perfectly so in terms of the ordering of the models I trained. But the effect was small enough that I could imagine that it was in the noise, especially given the small numbers of models I'm able to train. I'll chalk it up as a tentative success. In addition, I learned one useful thing: when talking about scaling up a model to more parameters, you actually have to think quite hard about where you want to put those parameters. I wound up doing a rough curve-fit to the models in the GPT-2 paper, but I have no idea if that was optimal. At some point I should try to dig up some research into optimising embedding dimensions, numbers of layers, and so on. But not now, as I've a bunch of other stuff I want to investigate first. Anyway, I hope you found this experiment interesting, and as ever, comments and questions welcome below. Thanks for reading! I'm less familiar with arguments for under-training -- that is, for fewer than 20 tokens per parameter. I've heard that these days, modern LLMs get a lot more reinforcement learning than they do pre-training, and perhaps that might mean that some very big ones are undertrained prior to RL? I'm uncertain. It's unlikely to be raw lack of data; even for those of us outside the big labs, FineWeb has 18.5T tokens. On its own, that would be enough to train a 0.925T-parameter model, and given that you can apparently do four epochs over the same data before you start getting diminishing returns, that takes us up to 3.7T. That's frontier-lab size, and I'm sure they have better datasets than FineWeb.  ↩ Parameter counts are from the paper, apart from the "small" model, which is known to be wrong -- I used my own calculation, and the result is in line with what I've seen elsewhere.  ↩ The paper doesn't mention the number of heads; these numbers are from " Build a Large Language Model (from Scratch) ", and match up with the ones on this Hugging Face page .  ↩ Regular readers might have noticed that I'm ignoring what I've been calling the "best" checkpoint. I've come to the conclusion that because for my training script, "best" means best in terms of training loss, and the training loss changes based on what training data the model has seen recently, it's actually not a very useful metric and just confuses things. At some point I'll probably re-introduce pre-checkpoint evals and use that for "best", which would be the right way to do it.  ↩ At the time I was using dropout, so training runs were not deterministic without a known seed.  ↩ A Chinchilla-optimal one, which I'll call here. One trained on twice the Chinchilla-optimal tokens, . One trained on the Chinchilla-optimal tokens, with two epochs (so that it was trained for as long as #2): Mean: ~3.673215 Sample variance: ~0.000073 Standard deviation (SD): ~0.008529 I'm less familiar with arguments for under-training -- that is, for fewer than 20 tokens per parameter. I've heard that these days, modern LLMs get a lot more reinforcement learning than they do pre-training, and perhaps that might mean that some very big ones are undertrained prior to RL? I'm uncertain. It's unlikely to be raw lack of data; even for those of us outside the big labs, FineWeb has 18.5T tokens. On its own, that would be enough to train a 0.925T-parameter model, and given that you can apparently do four epochs over the same data before you start getting diminishing returns, that takes us up to 3.7T. That's frontier-lab size, and I'm sure they have better datasets than FineWeb.  ↩ Parameter counts are from the paper, apart from the "small" model, which is known to be wrong -- I used my own calculation, and the result is in line with what I've seen elsewhere.  ↩ The paper doesn't mention the number of heads; these numbers are from " Build a Large Language Model (from Scratch) ", and match up with the ones on this Hugging Face page .  ↩ Regular readers might have noticed that I'm ignoring what I've been calling the "best" checkpoint. I've come to the conclusion that because for my training script, "best" means best in terms of training loss, and the training loss changes based on what training data the model has seen recently, it's actually not a very useful metric and just confuses things. At some point I'll probably re-introduce pre-checkpoint evals and use that for "best", which would be the right way to do it.  ↩ At the time I was using dropout, so training runs were not deterministic without a known seed.  ↩

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Giles's blog 2 months ago

How I use AI on this blog

Inspired by this LessWrong post , I thought I'd write about how I use AI here. This is less in the interest of disclosure, more to provide a snapshot of what I'm doing right now so that I can revisit it in the future and see how it changes. And hey, maybe it'll be of interest to you, dear readers. If I were to summarise my working philosophy in fewer than ten words, it would be: AIs identify problems and I fix them myself. With a very specific kind of exception (which I always flag), the text and code on this blog are human-generated. That's not a moral stand, but more a constraint imposed by what this blog is meant to be -- a place for me to learn in public . Every post here is based on an idea I had, and work that I've done. For many posts -- for example, the large-scale coding projects like this one -- I'll have multiple chat sessions ongoing while I do the work, normally with either ChatGPT or Claude, or sometimes both. The amount of input they have varies, but because the value of these projects is in what I learn when I'm doing them, letting an AI do my thinking for me would make the whole thing pointless -- so I take steps to stop that from happening. AIs are, of course, trained to be helpful, and will often explain things in their replies that I would have better learned on my own through experimentation. I'm generally pretty good at spotting when that happens before I've read more than a sentence or two, though, so I can skip reading that part, scroll straight down to the input field, and ask it to operate in more of a rubber duck mode. With the most complicated projects, where each step has a hard dependency on having got the previous one just right, I do use AIs for code review. Let's say that I've built a model that I intend to extend. I'll test it myself (does the loss go down when training, is it generating plausible-looking results?), but if I want to be really cautious, then I'll run the code past an AI. I'll paste it into a chat session and tell the LLM what it's meant to do, and ask it to check if I've screwed up 1 . Again, though, I make it clear that I don't want it to make fixes -- just to point out any bugs. As things progress with a project, I keep detailed notes. When I'm done, I write them up without AI assistance, getting the post to a level where it might be a little messy in terms of how things are explained, but all of the important information is in there. I read it through and make sure I'm reasonably happy with it, and then it's time for what I've taken to calling the editorial board. I paste the draft post into a fresh chat session with an LLM -- right now, this is normally Claude -- and ask for comments. It already has enough information in its memory of earlier conversations to know that what I'm looking for: places where I'm confidently wrong or other technical errors, places where my explanations are missing a step, or where I'm overexplaining things, conclusions that don't really follow from the results of an experiment, and that kind of issue. It tends to spot a few silly grammatical errors and typos at the same time. An important standing instruction is that I do not want it to rewrite anything. Just as with the code, it should tell me where there is a problem, and let me fix it. We iterate on that for a while until we have something that we're both happy with, and then I feed it to the next LLM -- normally ChatGPT. ChatGPT has a much more pernickety attitude than Claude does. I often use metaphors, and it will generally want me to replace them with mathematically rigorous prose. This is still very useful, though. Sometimes the metaphors aren't flagged as such well enough -- or even worse, there are times when the terms I've hit on for a metaphor happen to clash with a technical term, making what I've written misleading at best. I don't always address all of the issues that ChatGPT raises, as otherwise every post would be a mess of hedges and overexplanations and what-have-you, but I like to get to a stage where I'm comfortable that I have a good understanding of specifically why I'm rejecting the remaining points it raises. One other point where ChatGPT has helped a lot is that it's very diligent about checking supporting materials that I link to. When I was recently about to post an article on running an eval on a model, it followed the link to the training code and spotted a silly bug. It wasn't something that materially changed the eval's outcome, which is probably why I'd not noticed it, but it was something that was important to get right if I wanted later runs of the same evals to be solid. Definitely helpful. With that done, I run it past a cast of other LLMs. The exact set varies over time; for the last few posts it has been (in this order) DeepSeek, Grok, GLM-5.2 and Kimi K3. I did use Gemini in the past, but over time it became less effective and just started complimenting me on the post and suggesting related topics to chat about, which was kind of pointless. I'll wait until the next release and then try it again. Because the Claude and ChatGPT passes have generally got rid of anything particularly nasty, this second group of AIs often don't have much to add. However, occasionally they will spot something the others have missed, or have other suggestions, so it's worth spending the five minutes or so it takes to use them. It also helps to keep me up to date with what the other models out there are like. 2 When all of that's done, I run it past Claude one final time, tidy up any remaining issues, then publish it on a private staging site, and read through it carefully myself. The best time for that final readthrough is after dinner, ideally after a glass of wine; the goal is to smooth out the prose, and remove anything overly formal. To make it as close to being fun to read as I can manage. Once I'm happy, I can promote it to the live site and hit the publish button. That probably all sounds much more complicated than it actually is. A short post will normally go through all of that in half an hour -- less if I skip the full editorial board, which I sometimes do. The longer ones can take an hour or two, but given that they're normally the result of a week of work, on and off, in percentage terms it's not that much, and it's worth it for the polish. So, there's no AI-generated text here, but I do lean on AIs to make it the best version I can of what I have to say. How about the code? Again, the goal of the projects I document here is to learn in public. If I'm learning some concept that is expressed in code, then I need to write that code. So that means that anything non-trivial will be something I've written by hand, with AI input limited to code review -- the same rule as I have for the text. Of course, sometimes there are things I'd like to publish that I wouldn't learn anything by writing. Coding up stuff to chart loss curves, or writing a fancy JavaScript visualiser to show what models' parameters are used for would teach me nothing. So for that kind of thing, I just let the AIs get on with it (and even then, for the parameter visualisation, I hacked a first version together in a spreadsheet to check my understanding, and then tested the visualiser against it). I do always mention in the text when a particular bit of code was AI-generated, though. So, my rule is: if I would learn something by writing the code, I'll write it. If not, I'm happy to delegate to an AI. But even then, I apply one restriction: if it's for the blog, I'll ask for the code in a chat session, rather than using a more agentic system like Claude Code or Codex. This is to add friction. If you're using an agent, it's easy for a task to grow, and what started as a throwaway idea can come to consume more and more time and cognitive space. Keeping it in the chat interface keeps things minimal -- or at least, that's how it works for me. Does that mean that I'm against coding agents? This blog is where I post about experiments I've done and what I've learned. At the moment, what I'm learning is all pretty low-level. How does an LLM work? What factors make it smarter or dumber? It's all pretty hands-on, and involves code that I need to understand. I do other things apart from writing this blog, of course :-) And for that I'm keen on agentic tools; I have an OpenClaw agent to help me run my life generally, and use Codex and Claude Code for projects where I'm trying to achieve a specific goal, rather than trying to learn something. But by their very nature, those are not projects that will wind up here on the blog right now. That might change in the future! When I feel that I have a solid, large enough foundation, perhaps I'll be running experiments that I want to write about, where it would make sense for AIs to handle the details, while I focus on the broader strokes. But that time is not now, so right now, you can be sure that every word 3 , and almost every line of code, was written by hand. Even if I do need the AIs to keep me on track and at least borderline coherent. If you're wondering "why not use Claude Code or Codex", I get into why I avoid them for blog-related work towards the end of this post.  ↩ Current thoughts: ChatGPT, being wonderfully true to form, thinks I should clarify here that I'm not including quotes from models here (like the ones here ) when I say that.  ↩ If you're wondering "why not use Claude Code or Codex", I get into why I avoid them for blog-related work towards the end of this post.  ↩ Current thoughts: DeepSeek used to be close to Claude and ChatGPT, but has been left somewhat behind. I have been hearing rumours on X of an upcoming update, though. Grok used to have a propensity to try to turn my posts into clickbait. It would always want to rewrite things, and would suggest titles that were not a million miles away from "Ten things you never knew about RNNs -- number four will shock you!" Recent releases have been much better, though, and I might experiment with moving it further forward in the sequence. GLM-5.2 recently complimented me on my "science fiction" blogpost that mentioned ChatGPT 5.6 Sol. I probed it a bit on that and it said that because the publication date on the post was in 2026, it understood that I was writing near-future SF. From its perspective, the real date was sometime in late 2024. Surprising! I think the last time I saw that kind of behaviour from a model was, well, sometime in 2024... Kimi K3 is really quite impressive. I will be using it more. I particularly like the way it shows a fairly detailed chain of thought -- something it shares with DeepSeek and GLM-5.2, but there seems to be more depth there. It's a pity that Claude and ChatGPT only show summaries in the chat interface, though I understand their reasoning. ChatGPT, being wonderfully true to form, thinks I should clarify here that I'm not including quotes from models here (like the ones here ) when I say that.  ↩

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Giles's blog 2 months ago

Why do OpenAI's GPT-2 weights beat mine? Part three: testing overtraining

The GPT-2-style models that I've been training work really well, and I've even managed to train some that perform better than the original OpenAI small model in terms of cross entropy loss on a test set. But as I wrote previously , there's a mystery: why do they perform worse on my instruction fine-tuning evaluation? I had various theories about why that might be, and to me, the most plausible-seeming of them was the amount of data they were trained with. As best I can find out, OpenAI's models were, by modern standards, trained on much more data than they should have been, while I'd used the theoretically optimal amount of training data. To put it in other words, OpenAI's models were overtrained. If I deliberately overtrained my own models, could I match their performance? This post is a write-up of what happened, but so as not to bury the lede -- it didn't seem to help much, if at all. Let's see why. Let's start by getting a nice crisp definition of overtraining. It's important not to confuse overtraining with overfitting. Overfitting is where you train a model so that instead of learning a general rule about the data it's seeing, it learns something very specific to the training data -- for example, this: ...rather than this: Overfitting is pretty much always a bad thing. Overtraining, by contrast, is more of a judgement call. For LLMs, it's generally used as a shorthand for "training for more than the Chinchilla-optimal number of tokens". The Chinchilla paper makes a very specific case: if you train a model for roughly 20 times as many tokens as it has parameters, then you'll have as good a model as you can get for that budget in terms of compute. They were arguing against contemporaneous experiments where people were doing things like doubling the number of parameters but training on the same amount of data. If you overtrain, it means that you're training on more than 20 tokens per parameter. The Chinchilla argument is that instead of doing that, you should scale up the number of parameters and the number of tokens equally to keep the 20x ratio. Because the amount of compute used scales pretty much linearly with both tokens and parameters, you'll spend the same amount and you'll get a better model that way. Based on that heuristic, if you double your compute budget, then you should scale up your parameter count by 2 and your training token count by the same amount, and by doing that you'll get a better result than you would if you'd naively just doubled the parameters or the tokens, for the same amount of compute time spent. But overtraining is not always a bad thing. Keeping things Chinchilla-optimal means that you have to keep scaling up the model as you scale up the compute budget, and often you can't do that -- for example, let's imagine you're training a model that's meant to run on mobile phones. You have a hard limit on the number of parameters: what will fit in the target devices' RAM. And importantly, in general you will still get a better model by overtraining -- just not as much better as you would have done if you had been able to scale up the model as well as the training tokens. Now, as always, we don't know enough about the original GPT-2 training runs to be sure as to whether they were overtrained, and if so, by how much. But one thing that we do know is that GPT-2 was trained in 2019, three years before the Chinchilla paper came out, so they definitely didn't use it as a heuristic! One thing that they do say in the GPT-2 paper is that their dataset, WebText, is "a total of 40 GB of text". Assuming 4 bytes per token (a good rule of thumb for the GPT-2 tokeniser), that's 10B tokens. If the small model, which had 124M parameters, was trained on all of those, then it definitely was overtrained; 124M times 20 is about 2.5B. 1 On top of that, there's the question of epochs. In my training runs so far, I've been training through a Chinchilla-optimal 3.2B unique tokens. But being able to easily get your hands on that much training data is a relatively new thing, as you can see from the fact that the GPT-2 authors decided to document how they got theirs in the paper. So back then, pre-Chinchilla, people tended to train for multiple epochs so that they could make better use of limited data. Now, when I first started training my models, I tried to dig up some details on the GPT-2 training run beyond what was there in the paper. I found this report , and using data there, I calculated that it looked like OpenAI had trained for about 42 epochs over WebText. The report's author came up with 60 epochs as an equivalent-sized training run for their own dataset. 2 Again, those numbers are shaky; we don't have the real data. But I think it's not crazy to say that the OpenAI models were probably trained on more data than mine, and probably over more than one epoch. They were, in the Chinchilla sense, overtrained. That makes perfect sense for the time, given that it was three years before the Chinchilla paper! But that opened up a couple of interesting experiments that I could try. Obviously I didn't want to train a model over 10B tokens (not least because I'd need to download a larger sample of FineWeb). And I certainly didn't want to do 42 epochs of training, given that one epoch over 3.2B tokens took almost two days, even with my relatively fast JAX code . But it seemed plausible that I'd be able to get at least some signal if I trained longer. I decided to use , my new dedicated training box to train two new models: I expected each training run to take a bit less than four days on . Once they were done, I would be able to evaluate the models, both with the test loss eval and the IFT one. In an unusual-for-me fit of scientific good practice, I decided to write down what I expected to see up-front. I felt that: It was time to find out! I kicked off the extended train, over 6.4B unique tokens, using my JAX code because it's somewhat faster than the PyTorch version. It crashed after about 40 hours, with this error: There were no obvious issues with the machine -- the CPU temperature had been hovering at around 60°C, and GPU at around 70°C, as had been typical in training runs on in the past. There was nothing in or that looked suspicious. On Nvidia's documentation site I found a mention of that error message, but the page in question was all about PyTorch; I couldn't find anything relevant to JAX. For now, I decided to chalk it up to some bug somewhere in the training stack, and only dig in if it happened again. So I kicked the training run off again from the latest checkpoint to see what happened, and 38 hours later, it completed: Note that the numbers there -- tokens seen, time taken, and so on -- are for the portion of the training run after the restart. My checkpoint recovery code doesn't carry those over. The final train loss is just the loss over the period from the penultimate checkpoint to the end of the run -- there must have been some "easy" data there :-) The training loss chart looked like this: As you can see, the latest checkpoint wasn't the "best" one, so I pulled both latest and best down to , my workstation, for investigation. Firstly, I ran them both through my JAX smoke test, which asks them to complete "Every effort moves you" with 20 more tokens (using greedy sampling): Very inspirational. But coherent, and that's what matters. Now, the bulk of my evals use PyTorch rather than JAX -- I've been sticking to that for consistency -- so I converted the model Safetensors files so that they had the right structure to work with that: ...and did the equivalent smoke test on that side (which uses non-greedy decoding with a temperature of 1): I think that the Unicode junk at the end of the second is probably the first half of a two-token apostrophe or something along those lines. Anyway, those looked good. Next, it was time to run the first eval: how would the models perform on the held-back test set? So, the "latest" checkpoint was better than the "best" one. This is a problem with the way I define "best" in my training script. The original version of that script used a validation set to evaluate the model before every checkpoint; "best" at that point meant that the model was the best performer on that validation set. Later on, I decided to play a little loose with my training runs and pull out the validation -- this seemed reasonably safe because I was doing single-epoch training, so what I saw as the primary benefit of regular validation during training -- detecting overfitting by looking for rising validation loss -- was not so important. It's hard for a model to overfit on a single epoch training run. However, doing that meant that "best" had to be changed to mean "best-performing on the training data". And that's actually a really bad metric, because the training data is different for each measurement, so they're not really comparable. So, for this model, I decided that I'd discard the "best" checkpoint, and use the "latest" one. But all of that aside, those numbers were pretty impressive! Not only did it beat my best Chinchilla-optimal model to date, which had got 3.418784 loss on the same eval, and the original OpenAI small model with a loss of 3.499677, but it was actually getting quite close to the OpenAI medium's loss of 3.231442. 3 So now it was time for the two-epoch training run. Adding support for handling multiple epochs to the code was very simple , so having done that, I kicked it off. Just over three days later: This time the numbers are for the full training run -- there were no odd CUDA issues, and it ran straight through. Again, the final train loss is just the loss over the period from the penultimate checkpoint to the end of the run. For runs over the same dataset, it can actually be a useful quick-and-dirty way to compare results before running the full evals, but in this case it's obviously not comparable with the last run's result there, because we're talking about loss on different data. Anyway, once again, "best" and "latest" were different checkpoints, as you can see from the loss chart: ...so I pulled them both to , did the first smoke test: ...converted them to PyTorch: Did the PyTorch smoke test: So that all looked good (despite the Unicode junk), and it was time to work out the loss: That was pleasingly in line with my predictions: Both models would get better results on the test loss eval than my existing ones (90% probability). The one trained on more tokens would be better than the one trained for two epochs on the same tokens (70%). ...though of course the difference between the 3.326482 that this model got and the 3.324953 that the one-long-epoch one got was tiny and probably in the noise. I decided to count it as a win, anyway :-) So now it was time for the big test: how would they do at instruction-following? There are two phases to getting numbers for this test: firstly, I run the script that does the fine-tuning, and then generates the model's completions for the test set using the fine-tuned model. These are written to a JSON file for later use by the LLM-as-a-judge script. That script is the one I fixed in my previous post . I ran it for the extended train (one long epoch) model, taking care to make sure that the config it was passed matched the model's original training run in not having dropout enabled: Next, I ran it for the two-epoch model: With that done, it was time to run the LLM as a judge script . That gave us results, which I've put into the table below. For each model, I've shown the number of epochs of training the model needed before its validation loss started rising. For the pre-existing models, I've shown the score that they got in my baseline evaluation at the end of my last post , and their ranking in that eval. Then for all models, there's the score from this new run, and the new ranking. The two new models are in bold. A note on the numbers: in my previous post, I mentioned that different LLM judge runs differ due to randomness in how "strict" the judge model is. You can see that showing up here -- most of the old/new IFT scores are pretty close, within a point or so, but they do differ, some rising and some falling. This is with exactly the same set of responses for each model going into the judging program -- I used the same JSON files for this table as I did for the previous one for the models that were in there -- the only difference was that the new JSON files for the new models were added. As you can see, the new models scored better than the "JAX, with MHA bias, no dropout" one, which is the most similar: it's the same model config, just trained for the Chinchilla-optimal number of tokens. The one long epoch model got a score that was 1.22 higher than that one, and the two-epoch one scored 0.92 higher. However, my normal rule of thumb for comparing models in these evals is that differences of less than a point or two are probably in the noise. I did a second run of the LLM judge script -- it takes 20 minutes to run and costs a couple of dollars each time, so I don't like to run it all that often -- and this time around (just looking at the JAX numbers) things were a bit closer: You can also see that the two new models have swapped places. So I think that the principled approach here is to say that the improvement is almost certainly in the noise. Perhaps if I did a very large number of runs of the judge I'd get something more solid -- but what I'm hoping for is something a little more unambiguous; some change that really moves the needle in an obvious way, clearly outside the noise. That means that my second pre-registered prediction: Both models would score better than my existing models in the IFT test (70%) but would still be worse than GPT-2 small (90%). ...was half wrong. I got the "worse than GPT-2 small" bit right, at least. But while the models did look a bit better than the most similar pre-existing model, (a) the difference was too small for me to be confident in it, and (b) they were still worse than "JAX, no MHA bias, no dropout" and "Cloud FineWeb, 8x A100 40 GiB". What can we take away from this? The hypothesis that I was trying to test was whether it was simply overtraining that made the OpenAI weights better at this IFT evaluation than mine. Frustratingly, I can't say that the hypothesis was false. Perhaps there is an improvement gained by overtraining -- and the apparent gains which, on this experiment, appeared to be in the noise, would have been consolidated if I'd trained for even longer -- perhaps the 42 epochs on 10B tokens I suspect that the original weights were trained on? But equally, perhaps there is no benefit, and further training would have left the models exactly where they were in the ranking. Intuitively, you'd think that further training of a model would make it better at answering questions, at least up until the point that its parameters were "saturated" and could not absorb new information without forgetting something else. After all, a model that's never seen "Jane Austen wrote 'Pride and Prejudice'" will never be able to successfully answer when it's asked who the book's author was. But where that saturation point might be -- and indeed how much training you'd need to do to get there -- is not obvious. It's an annoying place to finish this experiment, but I guess at least an inconclusive result is better than never having run it at all. And it was at least good to see the test loss improvement that I expected. But given the opportunity cost of tying up in four-day training runs, I think I'll look into other possibilities next. As I was running this experiment, something came up -- and I'll post about that soon. Luckily, this time it won't involve training more base models... It's also worth noting that by the same, um, token, the extra-large model, with 1,542M parameters, was undertrained because the Chinchilla-optimal number of tokens would have been about 31B. Though that said, see later regarding epochs.  ↩ You might wonder whether training over the same tokens repeatedly over multiple epochs "counts" for Chinchilla purposes. Is training on 1.6B tokens over two epochs the same as training on 3.2B tokens over one? " Scaling Data-Constrained Language Models " poked into that in 2023, and from the abstract, came to the conclusion that you could do up to four epochs over the same data without losing much value, but after that returns diminished. If that holds for GPT-2, and they really did train for 42 epochs, maybe they wasted a lot of time? I'll need to read that paper in full at some point.  ↩ There is a table of results later on in this post where you'll be able to compare models easily.  ↩ Firstly, I'd train one on 6.4B tokens from my FineWeb dataset -- the original 3.2B that I had been training on to date, and then on whatever 3.2B came next. Secondly, I'd train another one on the same 3.2B tokens as usual, but I'd do two epochs. Both models would get better results on the test loss eval than my existing ones (90% probability). The one trained on more tokens would be better than the one trained for two epochs on the same tokens (70%). Both models would score better than my existing models in the IFT test (70%) but would still be worse than GPT-2 small (90%). It's also worth noting that by the same, um, token, the extra-large model, with 1,542M parameters, was undertrained because the Chinchilla-optimal number of tokens would have been about 31B. Though that said, see later regarding epochs.  ↩ You might wonder whether training over the same tokens repeatedly over multiple epochs "counts" for Chinchilla purposes. Is training on 1.6B tokens over two epochs the same as training on 3.2B tokens over one? " Scaling Data-Constrained Language Models " poked into that in 2023, and from the abstract, came to the conclusion that you could do up to four epochs over the same data without losing much value, but after that returns diminished. If that holds for GPT-2, and they really did train for 42 epochs, maybe they wasted a lot of time? I'll need to read that paper in full at some point.  ↩ There is a table of results later on in this post where you'll be able to compare models easily.  ↩

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Giles's blog 2 months ago

Why do OpenAI's GPT-2 weights beat mine? Part two: the bugfix

I'm digging into why my GPT-2 style models score worse on an instruction-following eval than OpenAI's original weights; I gave the details in this post . While I was writing up the results of my first experiment into possible causes, I ran the post past ChatGPT -- I always use an "editorial board" of AIs to check my posts for flow, style, and any technical errors (though all writing is always mine). It took a look at the eval code that I was running, and highlighted a bug. Luckily, it doesn't change the important results -- OpenAI's models continue to be better than mine at instruction-following. But it was enough to change the baseline numbers, re-ordering how well my own models did. So I fixed it and regenerated the baseline so that future experiments are based on solid ground. The eval takes a model, and trains it over multiple epochs on a split of a subset of the Alpaca instruction-following dataset. At the end of each epoch, it evaluates the resulting model against a held-back validation split; if the eval loss starts rising, it bails out. Finally, it runs a test split of the dataset through the resulting model, and saves the result. Once I've run it for a bunch of different models, I use an LLM-as-a-judge script to get GPT 5.5 to score results -- for each question-answering result, it sees all of the responses for all of the models in the same prompt , shuffled in order each time, to try to make it judge models against each other as consistently as possible. Now, the idea was that the generation of the test split answers would use the model from the epoch prior to the rising-loss one. So I had code like this: Each time the validation loss went down, I wanted to store the model's parameters in so that they could be restored later in that last line. If you look closely, the error is pretty obvious. does not return a copy of the model's parameters -- it produces a dictionary containing references to the parameters inside the model. So although I was trying to stash away the parameters in each time loss went down, so that we had a copy of the ones that were live in the epoch prior to the rising-loss one, what I was actually doing was just pointlessly saving a reference to the "live" params. The call to was essentially a no-op. The solution was simple enough: That was enough to make the code do what it was meant to do. While I was there, I also noticed that the evaluation code was only using the first five batches in my eval dataset. This code was originally adapted from an eval in " Build a Large Language Model (from Scratch) ", where the eval was run much more frequently and had to be super-fast. Because my own code ran it more rarely, it made more sense to use all of it. Given that I was going to completely re-run the script to generate the test responses, I figured that I might as well fix that at the same time. I re-ran the fixed script on all of the models I'm comparing, and ran that past the GPT 5.5 judge; here's what I got: Let's dig into those numbers. Let's look into the number of training epochs first; I've highlighted the models for which it changed. For the "Cloud FineWeb, 8x A100 40 GiB" model, I think that was a result of the change to the number of validation samples. For the two JAX models, it's a bit more of a mystery. There may be some part of the same "more eval data" variation there, but I have a suspicion that there's something more, related to dropout. My methodology to date has been that the IFT training runs should use the same dropout setting as the original base model training run (I'll come back to this in a later post). However, due to differences between the JAX training script and the evaluation code (which is PyTorch), matching the dropout rate when evaluating those models is a bit fiddly and error-prone. I am 100% sure that I got it right this time around -- I've checked and double-checked the specific commands I ran. But I think -- from what I remember doing -- that I might have messed it up the previous time. That's not certain, though -- just a suspicion based on half-memories of some commands I ran several weeks ago, which never wound up in my (too many terminals open at once). For the IFT scores, remember that they're not strongly comparable between runs. Let's imagine that the LLM judge is given the following answers to the question "Who was the author of Pride and Prejudice ?" In some cases it might treat the first two as being 100/100, in others it might give the second 95/100 for being too wordy. Likewise, in some cases it might rank the last two as 0/100 for being wrong, in others it might give the "Sarah Palin" one 5/100 for at least being the name of a person rather than complete nonsense. Now, we always ask the LLM to judge all of the models' answers for a given question in one go, so at least we can be sure that it will be consistent for a given prompt about a given question. But what we can't keep consistent is which way it leans between different runs, or different questions within the same run. Sometimes it might be feeling "generous" and give the Sarah Palin answer a bit of grace, other times it might be harsher. So there's a significant amount of noise there; my rule of thumb is that a variation of a point or two is within that noise, so OpenAI medium going from 41.62 to 42.41 is pretty much meaningless, and likewise "JAX, with MHA bias, no dropout"'s move from 19.25 to 18.12. So what's important is the relative ranking -- which is first, which is second, and so on. Naturally, you have to allow for the fact that -- for example -- if one model goes from position 11 to position 3, like "JAX, no MHA bias, no dropout" did, then the previous number 3 will have to become number 4, 4 will become 5, and so on. You can see that happening in the results table. Obviously, as other models rise and fall, that has further knock-on effects. Anyway, with all of those caveats, the good news is that my original mystery remains. The OpenAI models were still doing noticeably better than my own ones. GPT-2 medium continued to lead the pack (unsurprisingly, given that it's a bigger model), and GPT-2 small was still in second place. If that had changed, it would have made a rather disappointing end to this series: "mystery explained, it was a bug in the eval :-(" But now let's look at my models. Firstly, it looked like the "no MHA bias" models might have benefited from the extra training -- or from having their dropout settings corrected. They rose from positions 11 and 15 to 3 and 8 respectively -- a huge swing for "JAX, no MHA bias, no dropout", and a solid improvement for "JAX, no MHA bias, with dropout". Most of the other changes in relative rankings can be explained by those two models having been promoted, but there are some other changes. In particular, three models dropped significantly in score (and, as a result, ranking): My suspicion -- a very weakly-held hypothesis, but an interesting one -- is that those models had previously been benefiting from the bug. Remember that the signal we're using to stop training is that the validation loss starts rising. We're using that as a proxy for overfitting, which in turn we're using as a proxy for "this model has had as much training on this data as it needs for the eval". But there's no guarantee that the connection is there. Perhaps the models weren't overfitting and if we'd waited for another epoch or two, the validation loss might have started falling again. Or maybe some amount of overfitting would be beneficial for this eval? There's probably a near-infinite amount of digging in that I could potentially do here. But I think it's best to stop. The bugfix was important because it meant that the eval was now doing what I thought it was doing. Importantly, it doesn't change the puzzling fact that my models were worse at this eval than OpenAI's, which is what I'm trying to untangle. And it means that I can now lean more confidently on the baseline numbers. So now it's time to actually start changing things to see if I can close the gap! Here's a link to the next post in this series: does overtraining help? . "Jane Austen" "The author of 'Pride and Prejudice' was Jane Austen" "The author of 'Pride and Prejudice' was Sarah Palin" "The author of 'Pride and Prejudice' was 'Pride and Prejudice'" "Cloud FineWeb, 8x B200 160 GiB" "Local FineWeb train"

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Giles's blog 2 months ago

Why do OpenAI's GPT-2 weights beat mine?

When I finished my project training an LLM from scratch , I was left with a minor mystery. Why were my models worse at instruction-following than the original OpenAI GPT-2 small weights? I had an evaluation that I was running, based on the instruction fine-tuning code in chapter 7 of " Build a Large Language Model (from Scratch) ". The process was to train a model on samples from the Alpaca instruction-following dataset until validation loss started rising, to use that instruction fine-tuned model to generate completions to a held-back test set, and then to use an LLM to compare the results from various different models. The details are here ; let's call it the IFT eval. OpenAI's original weights for GPT-2 small consistently beat my own models, even when mine got better results than theirs on a more technical evaluation, where I just measured the cross entropy loss for each model on a held-back set of test sequences. This surprised me; I would have expected a reasonably close correlation between the two evals -- that better test loss would imply better instruction-following. I have a couple of thoughts about why this might be, and given that I recently set up , my dedicated LLM training box I decided to inaugurate her with an experiment to test one of them; further experiments will come in time -- though I don't think this will be a focus for the blog. More of a running theme, with occasional posts until either I solve the mystery, or give up in despair... In this post I'll give a bit more detail about the nature of the problem, and list some of the things I've been thinking might be the cause. In later posts, I'll dig into some of them. Let's take a look at the results from my most recent runs of the IFT eval. I've highlighted the OpenAI models in bold, and the table is sorted by the test loss -- that more technical evaluation that I mentioned earlier, where lower is better. How well the model did with the IFT eval is in the last three columns. The "IFT epochs" column shows how many epochs of training the model needed before its validation loss started rising, the score is the average mark out of 100 that (in this case) GPT 5.5 gives the model's answers to the IFT test set, and the rank is the position the model holds in terms of that average score. For the IFT eval, the OpenAI weights are the best. The "medium" model comes first, and the "small" one comes second -- and there's a noticeable gap between "small" and the score for my best model for that eval, "Cloud FineWeb, 8x A100 40 GiB": 26.73 vs 20.71. That difference was consistent across multiple runs of the eval -- the relative rankings of my own models varied more. It's not surprising that the "medium" model does better -- it's more than double the size of the others -- but the small model consistently beating mine was odd given that I had models with lower test loss. Another thing that stands out is the number of epochs that the models were trained for. Remember, I trained them until the validation loss started rising -- that is, until the model started overfitting. You can see that the OpenAI weights hit that after two epochs of training, whereas my models ranged between three and seven. My starting hypothesis was that the OpenAI weights were starting off from a better position in the IFT loss landscape than mine. The loss landscape for the task the various models were originally pre-trained for -- "predict the next token for this sequence of text that was pulled from the web" -- is different to the loss landscape for the IFT task, which is something more like "predict the next token in a useful response to this request". Let's dig into that a bit. The core idea behind the normal LLM training paradigm is that we can get a useful starting point for a model by pre-training -- training on a whole load of cheaply-available stuff from the web -- and only then have to use our much more expensive task-specific datasets for fine-tuning. I've been training my models on approximately 3.2 billion tokens of the fineweb dataset, which was generated from scrapes of the web, and then deduplicated and tidied up a bit. If I had instead trained them on 3.2 billion tokens of instruction-following samples -- essentially, pre-packaged Q&A sessions between a human and an assistant -- then they would almost certainly be much better than the models I have at instruction-following, for the same cost in compute used to do the training run. But the cost of generating those 3.2 billion tokens would be incredibly high. Imagine hiring people to do it: the GPT-2 tokeniser averages about 0.75 words per token, and while I don't know how much you'd need to pay people to write 2.4 billion words, I suspect it would be rather a lot. Even generating that much synthetic data from a larger LLM (as the Alpaca dataset was) would not be cheap -- I make it US$144,000 using GPT 5.6 Sol as of mid-2026. Even worse, the LLM we trained on that kind of data would be "brittle" -- if you wanted it to solve an even slightly different task (for example, Alpaca is single-shot question and answer, so imagine if you wanted it to handle multi-turn conversations like a chatbot), it would be hard to train it to do that. So: the idea is that if we pre-train a base LLM on less-structured -- but, importantly, still "real" -- data like scraped web pages, we'll get a general-purpose base LLM relatively cheaply. It will have discovered basic stuff like the structure of language, and hopefully some general knowledge (eg. maybe that the capital of France is Paris). Once we have that, we can fine-tune it for specific uses. Now, both the initial pre-training and -- in this IFT test -- the fine-tuning phase of building our model are done in essentially the same way: trying to minimise the cross entropy loss of the model's predictions against a dataset. 1 What has changed between them is the targets we're trying to train the model to predict. In the pre-training phase, we're trying to get it to predict web scrape data, in the fine-tuning, we're trying to predict responses to queries. Putting that idea into slightly more mathematical language (albeit loosely enough that any real mathematicians reading this will probably scream with horror), if we're saying that a pre-trained model is useful as a base for instruction fine-tuning, what we have is an assumption that the loss landscape for the original pre-training phase is reasonably similar to the loss landscape for the fine-tuning. We are hoping that a place that is nice and low on the pre-training landscape is reasonably close (in parameter space) to a place that is low on the fine-tuning landscape. That's all very abstract; let's try to visualise it. If we imagine the loss landscape for a model with two parameters, that's reasonably easy. The two parameters make up two dimensions -- let's say left and right for one, forward and back for the other -- and the loss at any given point is the vertical dimension. It's an uneven surface, a bit like a rolling landscape, with hills and mountains at points where the parameters have very high loss, and valleys and clefts where the loss is lower. In reality, of course, we have somewhat more than two parameters. With the 163,009,536 parameters in my GPT-2-small-style models, the loss landscape is not a surface but a 163,009,536-dimensional hypersurface 2 in 163,009,537-dimensional space. Good luck visualising that. While taking intuitions from levels of dimensionality that we can actually imagine over to insanely high-dimensional spaces like that is often risky -- weird stuff starts happening as the number of dimensions goes up -- for what we're specifically looking at here, it's safe. The landscape image works for intuition. So, we have one landscape like that for our pre-training phase. When we start fine-tuning, the loss landscape changes to a different one. What we're hoping is that the landscape is reasonably similar; that the low point that we wound up at in the pre-training landscape is close to a low point in the new fine-tuning landscape when that is swapped in. 3 The good news is that we know that this process of doing a pre-train then a fine-tune works. Most modern AI is trained using it as a foundation (though there's lots of extra stuff on top). And indeed, you'd intuitively expect it to work -- it would be kind of weird if there was no correlation at all between the "understand language" loss landscape and the "answer questions" one. But this does mean that we can -- at least in a somewhat hand-wavey way -- characterise the manner in which the OpenAI weights are "better" than mine in terms of the loss landscape. Both the OpenAI weights and mine have landed in places that were good from the pre-training viewpoint; that's what the "Test loss" results in the table above mean. GPT-2 medium is doing better than any of my models -- given that it's about twice the size, it should -- and GPT-2 small is doing better than most of them. But the places in the pre-train loss landscape where the OpenAI models landed were clearly better-matched to good places in the fine-tuning loss landscape than my own models' places were. The fact that they took fewer epochs to start overfitting intuitively points in that direction -- after all, if you're closer to somewhere, then it takes less time to get there -- but the fact that they scored better after fine-tuning makes it pretty clear. But something else that occurs to me as I write this is that the results on the test set also point to a superiority in the OpenAI weights -- one that hadn't occurred to me in the past. Let's think about what that test set is. In order to get data that was pre-processed and ready to work with, I downloaded all of the 10 billion token version of FineWeb, and split it into 99% training and 1% validation and test. I tokenised each sample in each of those splits, and then concatenated them together into a single sequence for each, separating the samples with tokens. My training script took the train split, broke it up into 1024-token sequences, assembled them into batches, and ran through 3.2B token's worth. The loss test takes 19,660,800 tokens starting at position 50,000,000 in the validation/test split, and runs them through in batches of six, working out the average loss. Now, the GPT-2 weights were trained on something that I believe was constructed similarly -- get a load of text consisting of a bunch of documents from the web, tokenise them all and tack them together separated with s, and then use that. But, importantly, it was a different dataset. Annoyingly, we don't have access to OpenAI's "WebText" dataset, but I think it's reasonably safe to say that even if it is similar to what I came up with, it will be less similar to the sequences used for the loss test than my own training set is. Given that the loss test for the OpenAI small weights came out with a better result than any of the models I trained in PyTorch, and was only narrowly beaten by the ones I trained with JAX (even then, I think, because the JAX models got lucky with their random weight initialisation before they started training), then I think we can say that the OpenAI weights are already considerably better than my own ones. They were already at a good place in their own loss landscape, but it happened to also be in a good place in my test set's. An alternative analogy: if you imagine a group of 15 trail runners having a race on a route in a forest, where all but one of them regularly run on different routes in the same forest, but the odd one out is running there for the first time, if the newcomer gets a very close fourth place, then it's not unreasonable to think that they're probably the best runner of all of them. 4 Finally -- I've been talking about the models as if they were all the same size (apart from the OpenAI medium weights), but there's one extra important difference with OpenAI's: they use weight tying, while mine don't. That means that their small model is significantly smaller than mine are -- closer to 124M parameters 5 than the 163M that mine have. Weight tying made a lot of sense back in 2019, when VRAM was even harder to come by than it is now, but when I looked into it , it made loss much worse. Carrying on with the trail runner metaphor, the OpenAI small weights not only don't know the forest well -- they're also ( scratches head looking for a good analogy ) physically weaker. Their performance is even more impressive. So: OpenAI weights good, mine (relatively) bad. Their smaller model, being tested for loss on a less-familiar-looking test set, does better -- or at least only a tiny bit worse than -- my larger ones. And when instruction fine-tuned, it converges on a better-performing result, and does it faster. What could the difference be? One thing that occurred to me while I was pondering this was dropout. If, when training, we randomly ignore 10% of the activations, perhaps we get a more generalisable model? Most of what I've read recently about dropout has focused on how it helps to avoid memorisation in multi-epoch training runs, but when I first read about it I felt that the generalisation argument was also strong . If true, that might explain the lower number of epochs needed for instruction fine-tuning, and maybe even the performance. Luckily, though, I'd accidentally done an experiment that suggested that this wasn't the difference. As one of my JAX training runs, I'd trained one with dropout, and as you can see if you look back at the results table above, it was one of the worst performers in the IFT test. One result is not a proof, but the fact that this one was bad compared to the others meant that I felt it was solid enough to ignore dropout as a possibility, at least initially. Again, it's kind of annoying that WebText was never published. All we know about it is from the paper , where they say: [W]e created 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. This can be thought of as a heuristic indicator for whether other users found the link interesting, educational, or just funny. How does this compare with FineWeb in terms of quality? Without seeing the results, we really can't say. FineWeb is more of a general web-scraping corpus, without the "filtering" provided by those Reddit upvotes. On the other hand, we're talking about Reddit here, and all kinds of awful junk gets upvoted, so I'm not confident that it's that strong a quality signal. And FineWeb has been curated to some degree. That said, I had previously noticed that the models I had trained on the FineWeb-Edu dataset ("the most educational web pages" from FineWeb) punched above their weight in the IFT test. They got pretty poor loss on the test set, but didn't do too badly in IFT. In retrospect, it's actually not that surprising that they did badly on the test loss eval; they were trained on a curated dataset of "good" stuff, and then were being evaluated against whatever less-curated stuff appears in the test set -- which you'll remember came from FineWeb. If you train a model on Jane Austen and then evaluate against Chuck Tingle , then you're not going to get amazing results. But then, there's the excellent performance of the OpenAI weights on that same FineWeb test set. Even if WebText was higher quality than FineWeb, that clearly wasn't the whole story. So data quality felt like it might be part of it -- but it wasn't where I wanted to start. This, finally, is where I landed for a first experiment. I'd been training my models on 20 tokens per parameter -- the Chinchilla-optimal number. But that heuristic dates to 2022, and GPT-2 was trained in 2019. While we don't know for sure, from what I've managed to dig up it looks like it was probably trained on much more data -- and probably over multiple epochs too. Training on more tokens than the Chinchilla number is called overtraining (not to be confused with overfitting). So: what happens if we overtrain the model? Will we get the loss down? And if so, will we start approaching GPT-2 small's results on the IFT eval? Stay tuned :-) Here's a link to the next post in this series . Modern production LLMs tend to use reinforcement learning as a big part of the fine-tuning phase, and that works quite differently.  ↩ When I started reading up on this, I was hoping that it would be a manifold, largely because I like the word. And it is! A hypersurface is a specific kind of manifold. :-D  ↩ If you like fantasy, there are some neat parallels there; the different overlaid worlds the characters move between in Philip Pullman's His Dark Materials series (aka The Golden Compass ), China Miéville's Un Lun Dun , V. E. Schwab's Shades of Magic , and Charlie Stross's The Merchant Princes series come to mind. Even the Upside Down in Stranger Things works, though then you have to decide which of your loss landscapes is the evil one, and that muddies the analogy annoyingly.  ↩ Sharp-eyed readers might have noticed that there are 16 models in the table above. I'm excluding the OpenAI medium weights. If you want to extend the analogy, that model is an Olympic runner who happens to have stopped by for the race...  ↩ The paper says 117M, but the released weights are larger. I can successfully load them into my code with a model config that has 124,439,808 parameters.  ↩ Modern production LLMs tend to use reinforcement learning as a big part of the fine-tuning phase, and that works quite differently.  ↩ When I started reading up on this, I was hoping that it would be a manifold, largely because I like the word. And it is! A hypersurface is a specific kind of manifold. :-D  ↩ If you like fantasy, there are some neat parallels there; the different overlaid worlds the characters move between in Philip Pullman's His Dark Materials series (aka The Golden Compass ), China Miéville's Un Lun Dun , V. E. Schwab's Shades of Magic , and Charlie Stross's The Merchant Princes series come to mind. Even the Upside Down in Stranger Things works, though then you have to decide which of your loss landscapes is the evil one, and that muddies the analogy annoyingly.  ↩ Sharp-eyed readers might have noticed that there are 16 models in the table above. I'm excluding the OpenAI medium weights. If you want to extend the analogy, that model is an Olympic runner who happens to have stopped by for the race...  ↩ The paper says 117M, but the released weights are larger. I can successfully load them into my code with a model config that has 124,439,808 parameters.  ↩

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Giles's blog 2 months ago

Benchmarking Qwen 3.6 35B MoE (3B active) on an RTX 3090

I mentioned I'd got a second RTX 3090 on a group chat, and a friend said: I know this is not really your thing... but let me know how quickly it runs Qwen 3.6 35bn MoE. With only 24gb of VRAM you’ll need to use a 4-bit quantized version and you won’t get a massive context window. But it should still be pretty cool. He's right that it's not really been my thing -- I've been focusing on my own LLMs recently. I decided to dig in a little, and in particular to play with Llama.cpp , which I haven't used for a while. And then things got a tad out of control, and I wound up doing some relatively detailed benchmarking. The headline results: I downloaded Unsloth's quantisation of the model from Hugging Face . With that, using the default Arch build of Llama.cpp, which uses Vulkan under the hood: Compiling Llama.cpp myself, in order to get the full CUDA version, helped a lot: That was a pretty impressive improvement. But it also showed that the lack of VRAM on the 3090 really does hurt. The friend who asked me about this is running an Intel Arc B70 Pro, with 32 GiB VRAM. He can (of course) fit the whole 4-bit quantised model on there without any context window reductions, and he says this about throughput: It starts off at 75-80. but drops into the high 50s as the context window expands That's worse than the CUDA-with-offload results above, though I did limit my testing to a 2,457-token prompt, with 6,144 tokens generated. The RTX 5090 has 32 GiB VRAM and fast Nvidia processing -- I imagine things would be a lot better there. Downside: it costs more than three times what you pay for a 3090 (and double the Arc B70 Pro). Anyway, in the rest of this post, I'll give more details of the benchmark, including charts showing the performance at different numbers of offloaded layers, and comparisons of the Vulkan vs CUDA versions of Llama.cpp for this model. One caveat: Qwen 3.6 is a multimodal model. That means that it has an extra component to map graphical inputs to tokens that can be consumed by the LLM. I didn't realise this as I ran these tests, but you can actually tell it not to use that component if you're only working with text inputs -- for example, will do the trick with Llama.cpp. If you're working with text-only stuff, you might want to play with that to see if you can get better results -- it will free up some VRAM and might allow you to get longer context lengths, or better performance. But anyway, as with many of my posts, this is a tidied up version of my lab notes, so you can share my learning journey with me -- but if that sounds pointless, and you've landed here because you want to run this model on your own RTX 3090, you can click this link to jump right to the results. Qwen3.6-35B-A3B is a Mixture of Experts model with 35B total parameters, 3B active. The tensor type is BF16, so it's two bytes per parameter. That's ~70 GiB of space required for the model, just for the weights, with no space for attention matrices and the like. We're going to need a quantised version. I'll dig deep into MoE models at some point, but essentially they work by having multiple different FFN blocks for each Transformers layer. After attention, the context vectors are routed to a subset of those blocks, depending on their content 1 . The number of active parameters is how many are actually used when running a single token through. Now, that number, 3B, looked really crazy to me as soon as I saw it. Qwen 3.6, as you can see from the model card linked above, has a vocabulary size of 248,320, and an embedding dimensionality of 2,048. It doesn't use weight-tying, so that means that the embedding layer and the output head come out at a hefty 508,559,360 parameters each. Both the embedding layer and the output head are required for every token, of course, so that means that of those 3B active parameters, more than 1B are used just getting tokens into and out of the model! Only 2B active parameters are left to handle all of the attention and the active FFNs. That is an imbalance of the same level as you get with GPT-2 small , and was surprising to see. Well, let's think about VRAM requirements. Naively you might think that you only need to keep the active parameters in VRAM, swapping FFN experts in and out as needed. Unfortunately that doesn't work. Let's say you feed in "The fat cat sat on the", hoping for a completion. The whole sequence is processed in parallel (that being the point of using a GPT-style LLM rather than, say, an RNN ). You'll need whichever experts are required for all six tokens in the prompt for the first layer, and likewise for all of the context vectors for the prompt in each of the later layers. So, realistically, for a non-trivial prompt, you can't "swap out" the inactive parameters; an MoE buys you lower usage in terms of processing, but not in terms of memory. 2 That said, as we'll see later, you can at least pull stuff out of VRAM into normal RAM, and get some advantages that way. As my friend said, a 4-bit quant would be needed -- perhaps with that, the whole model, both active and inactive parameters, would fit into 35 / 2 = 17.5 GiB of VRAM, leaving 6.5 GiB for activations, attention matrices, and so on? As it turned out, it needed a bit more, and it's worth taking a look into why. I must admit that I'd never really looked into quantisation prior to playing with this, and had naively assumed that you basically just took (say) a BF16 model, scaled the parameters down so that they were (say) FP4, tweaked the computations a bit and then ran the result. That was completely wrong! I'll have to dig into it further in the future, but my new working mental model is that it's more like lossy compression. The weights are stored in a "compressed" format -- one that is designed so that they can be quickly and cheaply "decompressed" by code running on the target platform. So, speaking very loosely, when running an unquantised model, we might have CUDA code doing something like this: ...a quantised model might operate more like this: Understanding that minimal amount clarified three things for me: So, with that (minimal) level of understanding, it was time to dig in! There are a bunch of different ways that people run LLMs locally, so I needed to work out which one would be likely to give the best results. One thing I noticed when googling around for things like "Qwen3.6-35B-A3B RTX 3090" was that pretty much everyone appeared to be using Llama.cpp . That wasn't something I'd used for a couple of years, and I've repaved my machine multiple times since, so it was time to install it. I run Arch Linux, and there's an OS package for it , so I installed it using the instructions there -- specifically, the ones to run inference using CUDA: Now it was time to try a model. I decided to start off with a small one that would fit onto my GPU without any quantisation, just to work out any bugs. I needed the model to be in GGUF format (that being what Llama.cpp uses). GGUF stores tensors and metadata -- the actual architectures of the LLMs Llama.cpp supports are baked in to the tool's source -- but, of course, it supports Qwen3, and so a nice simple non-MoE small model that looked like it would fit onto my GPU was . The help page for that model gave this example command to run it: That was out-of-date and gave various errors (which was fair enough, the model being from late December 2025 -- all of seven months old!). The fixes were simple enough: That is, and now needed an argument. I kicked it off, and got an error: That seemed strange. The instructions I'd followed on the Arch Wiki page were the ones to install Llama.cpp for CUDA, but it was referring to . Vulkan is an open GPU-programming language -- it fits into the stack in essentially the same place as CUDA. As always seems to happen with these things, while the open platform is, well, open, and runs on more platforms than the closed one, it is somewhat slower and buggier. So, why did I have a Vulkan version? Looking at the talk page for the package was enlightening: user wrote: Hello. The article only provides the package for GPU inference, even though on AUR there is a CUDA package available from the same submitter [1] I must add that the CUDA package is "Flagged out-of-date (2025-12-22)" ...and got a reply from saying: Hi. The reason CUDA was not added is that Vulkan solution is general, performant enough for most hardware setup, and I personally believe that is what edge AI deployment should look like - if your software stack is capable of gaming, it is capable of AI. They went on to say that they would add CUDA support, but they hit an issue because the AUR version that had referred to no longer had an active maintainer. For non-Arch users, the AUR is essentially a repository of non-official, community maintained packages. So, it looked like I had a Vulkan stack installed. Could I install the CUDA one from the AUR instead? Over the last few months, there have been security issues with the AUR, where bad actors have picked up ownership of unmaintained packages and put nasty stuff into them -- exfiltration code to steal things like SSH keys and crypto wallets, for example. As a result, I've been pretty cautious about adding on new AUR dependencies -- and this specific one having been unmaintained for a while scared me. I decided that I would stick with Vulkan for now; if I wanted to try with CUDA, I'd compile it myself from source later on. So, that left the error message: I decided to break down the command line step by step, as running random stuff you copied from the Internet is rarely a good way to understand what's going on. We're running and specifying a model from Hugging Face -- that's clearly what means. Per the , "whether to use jinja template engine for chat". "Colorize output to distinguish prompt and user input from generations" This is "max. number of layers to store in VRAM". For this model, I wanted everything in VRAM, so felt that "all" would be better there. basically means "all", at least for any normal-sized model, but I wanted to be explicit. This switched on Flash Attention, which seemed reasonable. This was "how to split the model across multiple GPUs". The error was "does not support split buffers", so this sounded relevant! Given that I only have one GPU on my machine, I decided that would be a better option here. At this point I suspected I had worked out how to solve the problem, but decided to take a look at the other options just out of interest. These were clearly the normal sampling parameters that I was used to from creating my own models. This was an extension to the previous sampling params: in addition to the stuff (and after it), limit sampling to the tokens that take up the top 0.95 of the probability distribution. This was just disabling the feature. min-p discards tokens that are less than X times the probability of the most likely token. Apparently using it is a promising alternative to messing around with and . Rabbit hole alert, let's move on. Per the docs, this applies a "repeat alpha presence penalty". That was interesting! I googled around a bit and found this page , where it became clear that it is a trick to stop models from getting stuck in loops. Small models are quite prone to that, so it certainly sounded useful. Perhaps something to dig into more later, but for now, keeping it seemed perfectly reasonable. That was the "size of the prompt context" according to the help -- that is, the amount of the context window that can be taken up by stuff going into the model before it starts generating. ...and this was the other side of that equation: the maximum number of tokens to predict. Another interesting one, "whether to use context shift on infinite text generation". Context shift, it appears, means that if you keep on generating, it will just drop stuff off the start of the sequence when it reaches the max context length, so that it can just keep going. Seems clever, though there are obvious risks of dropping important bits of context (compacting the context would be safer, and that's what most real-world workflows seem to do). But anyway, we're switching that off, so no harm there. So, most of those parameters seemed sensible, but I wanted to set to so that it didn't try to split things over multiple GPUs when I only had one, and also set to rather than , just for tidiness's sake. That gave this command: ...and that worked! I gave it a whirl: Looking good! Time to try our MoE. The most popular quants I noticed for this model were Unsloth 's, which makes sense -- they're a well-known organisation. They publish their models to Hugging Face, and the model I wanted was there as . I wasn't sure which of the various 4-bit quants to use -- as I said earlier, there were four of them -- but this Reddit commenter was using , so I decided to start with that, just to see what happened. That warning at the start was a bit concerning: ...and I noticed that in , my VRAM was maxed out. I decided to remove the from the command line; these were setting a specific context length, both for the prompt and for generation. Llama.cpp can work out what context length it can fit into VRAM on its own if you don't specify them, and this auto-fit felt like a good idea given that I was trying to jam a 22.4 GiB model into a 24 GiB GPU, and had X and various other apps running on the card at the same time. No error, it wasn't maxing out VRAM, and generation and prompt processing were noticeably faster. Perhaps the first time around it had offloaded some of the model to the CPU so that it had enough space on the GPU for the context size I was asking for? Still, I wasn't sure how far I could trust those numbers -- the number of tokens being emitted was quite small. 4 I wanted a prompt that would make the model generate lots of tokens. Now, it was a thinking model, so asking it a difficult question seemed like a good way to do that. I decided to use Ethan Mollick's "Lem test" : Compose a poem -- a poem about a haircut! But lofty, tragic, timeless, full of love, treachery, retribution, quiet heroism in the face of certain doom! Six lines, cleverly rhymed, and every word beginning with the letter S! More about that in the appendix below . 5 Unfortunately, when I gave it that prompt, it thought for a while, but crapped out during the thinking process, while it was trying to generate rhymes. Now, I didn't know what the context length was, either for the prompt or for generation. But I could be fairly sure that they were less than 40,960 and 32,768 respectively, given that the model did not fit into VRAM when I specified those limits, but did when I asked Llama.cpp to auto-fit. I couldn't find a quick and easy way of asking what lengths it had fit things to, but when I tried running it with the option for verbose logging, I got something. Actually, I got a lot of something -- thousands of lines of output, with incredibly detailed messages about everything Llama.cpp was doing. But there were three blocks starting "constructing llama_context". That made the issue reasonably clear. I assumed that the last message was the definitive one, and it was saying that my context length -- both prompt and generation combined -- was 4,096 tokens. I had set, so that was a hard limit. And 4k or so tokens fit with the amount of stuff that I would have expected to be in the model's context window at the time it crapped out. I decided to see what would happen if I tried a smaller quant. Hugging Face said (and agreed) that this one, , was about 22.4 GiB in size. But was shown on HF as using 19.5 GiB, so I decided to give that one a go next. I ran it with the option, and again looked at the last "constructing llama_context" block: 93,952 tokens -- still not the full amount, but not at all bad! I decided to give it a whirl, restarted without to avoid getting swamped with debug info, and: It's hardly Shakespeare, and the rhyming is both odd (ABCBCC) and unadventurous ("son", "sun", "son") but for a 35B parameter model with 3B active (of which 1B are used up by the embeddings and the output head), it's pretty bloody impressive! And every word did indeed start with an "S". So, now I had the model running. I had a context length of 93,952 tokens, was generating about 120 tokens/second, and processing the prompt at about 180 tokens/second. The question was, could I get the full context length in there? I knew that the answer was yes from previous reading, but it was time to try it out. Llama.cpp has a command-line flag , or in its long form, . As its name suggests, it's designed for mixture of experts models, and it works by offloading the FFN for a specified number of Transformers layers to the CPU. Intuitively it's surprising that that doesn't cause performance to completely crash. After all, aren't we running our models on the GPU because it's so much better at matrix multiplications than the CPU? And it's true, the GPU is best. But if it's a question of where to use its superiority in matmuls, then keeping the attention layers on the GPU and offloading just the FFNs to the CPU is the least harmful way to do it. Obviously you do lose performance, but not as much as you would if you offloaded attention. I started a binary chop. From the logs, I could see that the model had 40 layers: so I ran this to offload the first 20 to the CPU: That gave me a context length of 262,144 -- the native context length! I had 16944MiB used on the GPU, 10075MiB on the CPU, and I got 51.4 tokens per second for generation, and 64.4 tokens per second for the prompt. Then I tried various other values, but the process got dull quickly. For each one, I was doing this: At maybe four minutes for each and an error-prone process, this was ripe for automation. Additionally, something about those numbers for the prompt-processing tokens per second was niggling me. They seemed very slow. You'd expect an LLM in a generation harness like this to process the prompt much faster than it could generate, but I was getting numbers that roughly matched the generation speed. I figured that my prompt was so short that the numbers I was getting were being dominated by overhead. I needed a longer one. Llama.cpp's command exposes an API that tells you what sequence length it's running with, and takes pretty much the same parameters as , so I gave Claude Fable 5 the command I'd been running and asked it to write a script to sweep over CPU offload layer counts from zero to 40, and to store the numbers I was interested in: I also added onto that some code to expand the prompt so that I could get better numbers for the tokens per second for that part of the processing. Now, uses a chat template by default; the endpoint that Claude had chosen for does not -- it's a raw completion engine. This actually worked to my advantage. From the logs I found the template it had been using, which was something like this: Because I could easily fake the conversation history, I baked in a template that started with a fake previous interaction where the user had provided the first two chapters of Jane Austen's Pride and Prejudice and asked the model for its opinion, it said something like "I like that", and then the user asked for the Lem test poem. That bulked things out quite a bit -- as it turned out, to a prompt of 2,457 tokens. I also made it give up after 6144 tokens so that it wouldn't take forever to run, and kicked it off. Here's the code , and you can see the results here . The prompt processing numbers were much better! With nothing offloaded, I got a context window of 51,200, a prompt processing rate of 2,787.1 tokens/second, and a generation rate of 122.4 tokens/second. (It was interesting that the context window was smaller than it was when I ran without , but I decided that I didn't want to get into the weeds by digging into that 6 . Perhaps isn't quite the same as running without at all?) Anyway, the numbers for other levels of layer-offloading did what you would expect: as more layers' FFNs went from the GPU to the CPU, the context window expanded until it hit the model's native context length of 262,144 when 12 layers were offloaded. And the speed -- both in prompt processing and generation -- went down. I ran it three more times, just to see if there was a lot of noise there, and then asked Claude Sonnet to slop up a charting script for me (in the same repo as the above if you're interested) -- and I had some results! Just for people who skipped the details: we're running the quant of on , and seeing how its performance changes as we vary the number of layers offloaded from the GPU to the CPU. Firstly, how does the context length change? You can see that the model's full built-in context length was reached when we had 12 layers offloaded. But how did that impact performance? You can see that the prompt throughput drops off fairly smoothly as the number of layers offloaded increased. What is interesting is those spikes in the generation tokens/second. The reason I'd done four runs of this script was because I'd initially thought they were noise, but you can see they're pretty consistent. Claude Fable 5 thinks they may be related to the points at which there are full attention layers -- the attention system in these Qwen models is complex, with some layers using a faster system and others using the full attention that my GPT-2 models always use. That's something to investigate another time, I suspect, if ever. It's noticeable that with 12 layers offloaded (the exact number that gave us the full context), the throughput is somewhat better than at 11, which makes 12 look pretty much like the sweet spot for running this model with this setup! From the raw numbers, we can see that there we had about 66 tokens/second generation, and 607 tokens/second on the prompt. Anyway, finally, let's take a look at memory usage -- both VRAM and regular system RAM. It looks exactly as you'd expect: We're using as much VRAM as we can get up to the layer 12 offload point, because with fewer than that number of layers offloaded, we're grabbing as much space as we can on top of the amount consumed by the parameters, in order to get the longest possible context window. But after that, it drops off smoothly. And system RAM rises smoothly as layers are offloaded into it, as expected. So, that was it! As I mentioned back at the start of these lab notes, the version of Llama.cpp that's part of the official OS repositories for Arch uses Vulkan, in order to maximise the number of platforms it runs on. And that was the version I'd used. The Vulkan version was clearly solid and stable, given that I'd been able to run a bunch of benchmarks on it. But you'd expect CUDA to get performance enhancements and the like first, and in general to be more polished. So how would it perform comparatively? I didn't want to install the AUR version of the CUDA-based Llama.cpp, as it had been abandoned previously (though someone appears to have picked it up again now), and bad actors have been taking over abandoned AUR repos and putting bad stuff in them. So out of an abundance of caution, I decided to compile it from source from the official Llama.cpp repo . Their instructions made it almost laughably simple; I cloned it, then ran Less than ten minutes later, I had a working set of binaries. I tweaked my benchmarking script to use them instead of the system-installed ones, and got these results . Charting those, we get this for the context length: So with CUDA, we get the full built-in context length with just 10 layers offloaded, compared to the 12 we needed with Vulkan. And even with nothing offloaded, we got a bigger context window -- 89,600 rather than Vulkan's 51,200. How about performance? With CUDA, we don't get those odd bumps in the generation throughput -- or at least, if they're there, they're much smaller. Perhaps Claude's ponderings that I mentioned earlier were off the mark? Or perhaps they only impact Vulkan? 7 But the big news on this chart is -- as I had suspected would be the case -- CUDA was noticeably faster! With all layers on the GPU, it was getting 139.6 tokens/second generation, 3360.4 tokens/second on the prompt, as compared to around 122 / 2787 for Vulkan, averaging across my four runs on that platform. And with a bigger context window too. With ten layers offloaded, CUDA had the full context length of 262,144, and was getting 89.1 / 1153.5, while Vulkan managed a context length of 233,984, and throughput of 66 / 702. Finally, with 12 layers offloaded, both models had the full context length, CUDA had throughput of 84.9 / 1008.1, and Vulkan had 66 / 607. And that meant that we were able to reclaim some VRAM too: Advantage: CUDA, I think. That was probably just a tad more time and effort than I think my friend expected me to put into this when he asked an offhand question on WhatsApp. But I feel that I've learned a lot of useful stuff in this journey, and while there's a lot of detail in this post, it didn't actually take up much time to run the experiments. I have another post on the way that involves multiple four-day training runs, so I needed to occupy my time somehow... Anyway, I hope it's useful to people out there! Let me know in the comments if it helped you. And now, just to finish off, a bonus section on the prompt I was using to get the model to generate lots of stuff. The prompt I was using for these tests was this: Compose a poem -- a poem about a haircut! But lofty, tragic, timeless, full of love, treachery, retribution, quiet heroism in the face of certain doom! Six lines, cleverly rhymed, and every word beginning with the letter S! What I like about it as a prompt is not just that it's a particularly silly thing to ask of an LLM, but it forces models with reasoning enabled to think hard about how to keep within all of those constraints. It's basically a short prompt that makes them generate a lot of tokens -- perfect for benchmarking. I stole it from Ethan Mollick. He's well worth following if you're not already doing so; here's his X/Twitter profile , and his Substack . He writes a ton of interesting stuff, but I want to focus here on this particular eval. In the science fiction short story "The first sally (A), or Trurl's electronic bard" (included in the collection The Cyberiad ), Stanisław Lem wrote of two rival engineers. One of them, Trurl, has created an electronic poet, and the other, Klapaucius, gives it what he thinks is an impossible task. From the English translation of this part on Goodreads : "Have it compose a poem -- a poem about a haircut! But lofty, tragic, timeless, full of love, treachery, retribution, quiet heroism in the face of certain doom! Six lines, cleverly rhymed, and every word beginning with the letter S!!" The machine ponders briefly, and responds: Seduced, shaggy Samson snored. She scissored short. Sorely shorn, Soon shackled slave, Samson sighed, Silently scheming, Sightlessly seeking Some savage, spectacular suicide. The story is a fun read, definitely recommended! You can see a bunch of parallels between current LLMs in the way Trurl's machine works -- and in its failure modes. Mollick decided to use it as a prompt ages ago, and in March 2024, he managed to get Claude 3 to almost complete the task . The result wasn't a very good poem, but neither ChatGPT 4 nor the version of Gemini at the time were close. Here's Claude 3's attempt: Stealthily snipping, shearing silky strands, Severing soulmate's sacred, sworn soulbands. Scoundrel's scandalous sins spurred sly scheme, Spurring scorned sweetheart's spiteful supreme dream. Shining locks sacrificed, stoically shorn, Stalwart spirit stays strong, staunchly reborn. You can see that it had two non-S words, "dream" and "reborn" 8 . But otherwise not bad -- it makes sense, the rhymes work, and it ticks the emotional boxes. Recent models have become very good; here's Claude Fable 5 from today: Ah, Trurl's Electronic Bard's finest hour — the impossible commission from The Cyberiad! Here's my own attempt at the feat: Soft she sang; strong Samson slumbered, serene; Sly, she snipped seven shining strands, scarcely seen. Soldiers stormed, seized, scorched Samson's sight; Shackled, still, sightless — seething sacred spite. Straining, steadfast, Samson seized stout stanchions, swaying slow — Stones smashed; seductress, soldiers, Samson: slain. So sagas show. Love in the lullaby, treachery in the shears, doom in the darkness, heroism at the pillars, and retribution in the rubble — with every single word dutifully saluting the letter S. The Electronic Bard would, I hope, give a grudging beep of approval. Now, it recognised the source, so perhaps that helped -- but while I'm no connoisseur of poetry, I actually think it's slightly better than Lem's version (though as Lem wrote in Polish, that's the work of his translator -- and translating poetry is obviously a tough task). And while both versions are based on Samson, given that he was a biblical hero who lost his strength after his hair was cut, he's a pretty obvious subject for a tragic poem about a haircut, especially if every word has to start with the letter "S" -- so I don't think that points to copying. ChatGPT 5.6 Sol also does a great job: Silken Samson sought sweetheart’s soft sanctuary; Sweetheart, serpent-souled, summoned shears—sanguinary. Silver strands slid; sacred strength sank, slain; Silent sorrow sealed sweetheart’s scarlet stain. Shorn, storm-bound, Samson shielded slaves, steadfast; Stone sanctums shattered; smiling, Samson sank—surpassed. Another Samson-based one; it doesn't say it recognises the source, but I imagine it did. Now let's take a look at some samples from this tiny model we've been benchmarking -- 35B parameters, 3B active, of which only 2B are doing the actual thinking (rather than messing around with embeddings). Here's the first one I got: Sneering shears shred sacred share, Silent steel sells silver soul; Sister's snare spares son; Sweet sign seals sole; Still star saves sun; Secret sign seals son. I mean, it's not amazing, but it satisfies the S constraint. Less impressed by how well it handles the "cleverly rhymed" part, though -- as I said earlier, its scheme is ABCBCC, which is odd rather than clever -- and it doesn't make an awful amount of sense. I noted down two more while I was playing with this: Silent shears sever sweet silver strands; She stands, sacrificing self; submission seals severing's demands. Shear's snare strikes; swift sentence seals sorrowful sphere. Shorn sovereign swears silence shields secret spear; She saves sacred soul; stillness safeguards sterling store. Spirit soars; sorrow's stroke stops, solace restores. That one's a lot better, I think; there's a hint of a story, and at least it rhymes. But note the "demands" and the "restores" at the ends of lines 2 and 6 -- the "S" constraint has been broken. Sister's scheme scalp shorn, Silver shears slice soul sworn, Sweetheart's steel skull sear, Silent slitters shear. Spirit spies sight, Sorrow swallows slight. Hmm. A bit bleak. Again, poetry isn't my thing. But I think that Claude 3's offering from two years ago is better than any of them -- apart from the fact that it couldn't keep the "S" constraint. Still, it's impressive to see how well such a tiny model -- again, 2B actual thinking parameters per token -- can do on what is a pretty challenging task! With some models, there are also extra FFNs that are always loaded that do stuff before the routed expert ones. Again, I'll dig into the details at a later point.  ↩ That said, a while back Dan Woods managed to get Qwen3.5-397B-A17B -- 397B parameters with 17B active -- to run on a 48 GiB Macbook Pro with some very clever use of the SSD. It was slow , though, at around 4 tokens/second.  ↩ And only supports certain sizes for certain kinds of operations.  ↩ As was the size of the prompt, but I got to that later.  ↩ Yes, my blog posts have started accumulating appendices as well as footnotes. I may have a problem.  ↩ Some readers might feel that that ship sailed long ago and I was already very much into the weeds, and possibly in the middle of the Amazon jungle. I disagree, but largely due to the mixed metaphor.  ↩ "Put down that rabbit hole and step slowly away, Giles."  ↩ 'Hey, Claude 3, how many "S"s are there in "strawberry"?'  ↩ Using the GPU only, I was able to get the model to generate at just over 120 tokens per second, and it was able to process the prompt at just less than 2,800 tok/s. However, having the whole model on the GPU didn't leave that much space for the context window -- it was constrained to about 50,000 tokens, compared to the model's native context length of 262,144. Offloading the FFNs for the first 12 of the model's 40 layers to the CPU managed to reclaim enough VRAM to be able to get the full context length; however, with that setup, things were -- unsurprisingly -- slower. I got just over 65 tok/s for generation, and 600 tok/s for the prompt. With everything on the GPU, I got 140 tok/s for generation and over 3,300 tok/s for the prompt. That was with a context window of 89,600. So everything was better :-) It was also easier to get to the full context length; that needed just 10 layers' FFNs to be offloaded, and at that point I was getting 89 tok/s for generation, and about 1,100 tok/s for the prompt. Load BF16 weights from VRAM Load BF16 data from another bit of VRAM Do a matrix multiplication of one by the other. Store the results into VRAM as BF16 Load quantised weights from VRAM "Decompress" them to BF16 Load BF16 data from another bit of VRAM Do a matrix multiplication of one by the other. Store the results into VRAM as BF16 Why people talk about things like "4.1-bit quants". We're talking about the average number of bits per weight in the "compressed" model. Why there are different quants of about the same size for a given model; for example, when we get on to looking at quantised versions of this model later, you'll see 4-bit ones called , , , and , with different sizes in terms of GiB. Each of those is using a different set of trade-offs in the quantisation process, so will perform differently on different tasks. I gather that picking the right one for any given task is a bit of an art form. Hugging Face have a summary of the different types they support here , which has some explanation of the differences, but decoding what they're saying there is beyond what I've learnt at this point. How GPUs can run quants with bits-per-parameter levels that they don't support for calculations. For example, the RTX 3090 supports 32-bit and two forms of 16-bit (float16 and BF16) plus integer operations of various bittednesses 3 . 4-bit formats like FP4 are only supported in more recent cards like the RTX 5090. But, of course, we're not doing any 4-bit computations -- it's all in some format that the GPU can handle natively. Start with the value I wanted and with . Use the logs to find the context length, and note down the VRAM/RAM usage. Restart it without , paste in my prompt, and wait until it either came back or got stuck in a loop trying to find rhymes (which it did maybe one in every four times). Note down tokens per second for the prompt and generation. The number of layers offloaded The context length we got The VRAM and system RAM (RSS) usage The size of the prompt we were providing The tokens per second at which the prompt was processed How much was generated Generation tokens/second. With some models, there are also extra FFNs that are always loaded that do stuff before the routed expert ones. Again, I'll dig into the details at a later point.  ↩ That said, a while back Dan Woods managed to get Qwen3.5-397B-A17B -- 397B parameters with 17B active -- to run on a 48 GiB Macbook Pro with some very clever use of the SSD. It was slow , though, at around 4 tokens/second.  ↩ And only supports certain sizes for certain kinds of operations.  ↩ As was the size of the prompt, but I got to that later.  ↩ Yes, my blog posts have started accumulating appendices as well as footnotes. I may have a problem.  ↩ Some readers might feel that that ship sailed long ago and I was already very much into the weeds, and possibly in the middle of the Amazon jungle. I disagree, but largely due to the mixed metaphor.  ↩ "Put down that rabbit hole and step slowly away, Giles."  ↩ 'Hey, Claude 3, how many "S"s are there in "strawberry"?'  ↩

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Giles's blog 3 months ago

Building intuition about LLM parameter counts

When I was building my GPT-2 implementation in JAX , I started with just token embeddings for the input, and a separate output head (as I was not using weight tying ). It wasn't an LLM -- no Transformer blocks, no attention, no feed-forward networks. I was somewhat surprised when I noticed that even that stripped-down model had 77 million parameters with the "small" settings I was using to train -- specifically, an embedding dimension of 768. However, I realised I shouldn't be -- with a vocab size of 50,257, each of those components is essentially a 768 × 50 , 257 matrix, and that is indeed over 38 million numbers. But the finished LLM at the end of the project was only 163 million parameters -- that meant that the input and output components alone were almost half of it. That felt like a surprisingly large percentage. I had a similar shock when I was first looking into the feed-forward network , and realised that it had roughly twice as many parameters as the attention layers. When we learn about the internals of LLMs, a lot of the focus is on the attention mechanism. This makes sense -- it's the hardest part to get your head around. The rest of the setup, at least for simple GPT-2 type models, is fairly standard stuff. But that means that it is easy to overestimate how much of the total parameter count of the model attention uses up -- especially for smaller models, where the token embeddings and the output head are so large in comparison to the Transformer layers that make up the actual body of the LLM. OpenAI released GPT 5.6 today, so I decided to take its "Sol" variant for a ride in Codex and asked it to write a visualiser . It shows breakdowns of how the parameters are split between embeddings, attention, the FFNs, and the output head for different sizes of GPT-2 models (or your own custom settings with the same architecture), and you can also add/remove weight tying and QKV bias. It did a really good job -- check it out! Here's a screenshot of what it showed for GPT-2 small without weight tying. It's well worth a play. In particular, it's interesting to see what happens as the number of tokens in the vocab gets very large (many modern models have hundreds of thousands). You can very easily create a "tiny" model which is almost entirely embeddings and the output head.

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Giles's blog 3 months ago

poppy the training box, part 1: the beginnings

For a while I've been planning to put together a separate machine for local LLM training. Until now, I've been using my desktop PC, . I have an RTX 3090 installed, and can get useful training runs done (most recently, a 163M-parameter GPT-2 small style LLM in JAX ), but there are a couple of problems. And relatedly to all of those: the two-day limit to the training runs I've been doing is something I set because that's the maximum amount of time I'm willing to have tied up. It would be really interesting to try longer training runs! I also have longer-term plans; a multi-GPU box would be interesting to put together -- not just to have more power locally, but so that I could test larger-scale cloud multi-GPU training runs before starting to pay for expensive machines. US$15.92 an hour to rent a machine isn't a lot of money, but it adds up, especially if you're spending it while debugging parallelism issues. And finally, I've always been interested in putting together a custom water-cooling loop in a PC. I've been building my own machines since 1995 or so, but never got round to that side of things. It sounds fun! But despite all of those future plans, this is a fairly normal machine-building post -- how I repurposed an old PC, plugged in a second-hand RTX 3090 from eBay, tested it all, accidentally trained an LLM for 11 days, and almost cooked a CPU. Over time, I expect to be posting more -- and more interesting -- build details. Let's think of this as establishing the baseline. Back before I moved to Lisbon, we had a holiday home here. When we came over, I'd bring my laptop, but that was always somewhat unsatisfactory -- limited CPU power for work, limited GPU for my occasional gaming. During Covid, we started staying in the holiday home for longer periods -- and this became too big of an annoyance to ignore. So in 2020 I put together a small form-factor PC, which I named . The constraints were: The build was a bit fiddly, like all SFF PCs. You can see the component list and build notes here on PCPartPicker , but in short she had: She looked like this: (Gosh, I'd forgotten how... vivid our wallpaper was in that dining room.) For scale -- that case is slightly taller than two cans of coke stacked on top of each other. So, pretty small. When we moved to Lisbon full-time, I brought with me from London, and while he's been upgraded several times since (including adding an RTX 3090 in late 2023 ), he's been my daily driver since. So sat in the corner of my study, sad and unused :-( It was time to bring her out again. Initial plan: get her up and running in a new, larger case, with a PSU that could potentially handle three graphics cards. Initially, I found that she wouldn't switch on: a quick check suggested that the problem was the PSU. I'd had problems with SFF PSUs in the past, and given that the plan was to give her a new one, I just got one, along with a new, larger case -- specifically: A few days later, the parts arrived. Here's a family photo: is to the left, centre, sitting on top of her new case, and Cornélia (wearing her Flower of Shame) is to the right. For scale, Cornélia is quite a large cat. (I appreciate that that is not immensely helpful.) Time to put the old motherboard and the new PSU into the new case. Here's what it looked like: The Mini-ITX motherboard in a case designed for full ATX looks comically like a postage stamp. I switched her on, and luckily enough, everything worked! Must have been a PSU issue. The OS that she had was a more than three-year-old version of Arch, so I wiped the drives and installed the most recent version with my normal config, and it was time for a quick test. One of the nice things about having done all of this LLM training stuff recently is that you have a ready-made burn-in test for new hardware :-) I didn't have my JAX training code yet, but I did have the PyTorch one . Now, with her GTX 1660 Super GPU, was clearly not going to be able to train an LLM of the size I could with 's RTX 3090. I did some fiddling around with the model and training run parameters, and found that I could fit in a cut-down version of GPT-2 small with this setup: I trained it with a microbatch size of 4, gradient accumulation over 16 steps, and all other hyperparameters the same as my normal training runs on . The number of training tokens went down -- the model had 76,933,120 parameters, so I needed to train for just over 20x that -- about 1.5B instead of the 3.2B I've been training my other models on. I kicked that off, and out of interest, I kicked off another training run on with the same setup to see what happened. The training run went normally -- GPU running at full blast, 368W, and it completed in about 9 hours. That's less than 1/4 of the time my normal training runs take, which makes sense because time taken for this kind of thing scales roughly linearly with both the size of the model and the number of tokens, and both of those were about half the normal size. was a bit more interesting. In , the GPU usage showed up as 100%, but with an "effective" utilisation of 53%. The power draw matched the latter, being 67W out of a total possible 125W. I'm not quite sure what was causing that -- clearly there was a bottleneck somewhere. Not really worth digging into, though, given that I was going to replace the card shortly. Anyway, that took 963,257 seconds to run. That's 267.57 hours, or just over 11 days. What's kind of interesting there is that this training run not only took much longer (which is only to be expected), but that it used more electricity. 67W over 267.57 hours is just short of 18kWh, whereas 368W over 9 hours is about 3.3kWh. Buy an RTX 3090, save the planet! I decided to run my normal evals to confirm that what had come out the other end was sane. When asked to complete "Every effort moves you", 's model said: And 's said: Those were actually rather good, I thought! And looking at my normal loss test confirmed that the models really weren't that bad; 's got 3.855702, and 's 3.855981. That was actually better than the 3.943522 I'd got on before I went down my rabbit hole of optimising hyperparameters . So, that was an interesting test -- I was talking to ChatGPT about it at the time and it called it "maybe an art project", which I thought was amusing if a bit arch. Time to do something a bit more useful. Finding an RTX 3090 for a decent price from a trustworthy-seeming vendor is kind of hard right now. But it's still the sweet spot for price-performance if you're looking to train models locally, so I set up an alert on eBay, and eventually one popped up in Bulgaria. I bought it, and a few days later, this turned up: It's actually not as ugly as it looks in that photo -- it's considerably uglier. The stuff that looks a bit like crinkled aluminium foil is really white plastic with a kind of crystalline texture. Made me glad that I'd gone for the mesh-sided case rather than the glass one. Well, I hadn't bought it for the looks. I removed the old GTX 1660, and put in the new card, switched it on, and: Wow, a disco in my PC. Lovely. It was time to kick off another training run to see if it worked. This time, I did my normal GPT-2 small sized train with optimised hyperparameters. It ran for about ten minutes, and then switched herself off. That didn't look good. I spent some time digging around trying to work out why my new graphics card was broken, and then happened to be sending the video above to a friend, and spotted something. Check out the Noctua fan -- the beige and brown one you can see behind the cooler mount, above the graphics card. It wasn't spinning. That's the CPU cooler fan and should always be spinning, even if slowly, when the machine is on. I log basic metrics for all of my PCs to a central InfluxDB instance, so I checked that out and: A CPU temperature spike up to about 115°C! Not good. Clearly an emergency thermal shutdown from the CPU. I initially thought that I must have knocked the fan cable loose while plugging in the new GPU -- plausible, though they were quite far apart -- but unplugging then reseating it, then powering up the machine still didn't start the fan spinning. And it was not visible in the BIOS. I then zoomed out a bit in Grafana; I only keep 30 days' worth of metrics, and it had been more than a month since I did my original burn-in test, so I didn't have anything for that. But I did have this: had been idle for all of that time, and was averaging CPU temps of over 70°C. The dropoff prior to running the test was because she'd had a chance to cool down while I installed the GPU. Having spent ages setting up my InfluxDB monitoring stuff so that I have metrics for everything, I should probably actually look at them every now and then, because the fan had obviously not been doing anything for a month or so. Well, thank goodness for Amazon next-day delivery. I bought a new Noctua NF-A9x14 PWM (praying that the problem was the fan and not the header on the motherboard), and when it arrived, I put it in. This time, when I powered her on, the fan was spinning. Phew. I left her running for an hour, and the CPU temperature stabilised at 35.5°C. Next, I kicked off a version of my standard LLM training run with the number of tokens reduced so that it would run for an hour. During that, the CPU temperature went up to a moderately-toasty 76°C -- not ideal, but remember that with the broken fan, she was running that hot at idle. It seemed a bit odd that it was that hot at 10% CPU usage, but given that one core was running at 100%, it didn't seem totally off. The heatsink and fan are designed for SFF PCs anyway, and those tend to run somewhat hot. The GPU temperature also went up to 70°C and stabilised there, while power draw was stably about 368W out of 370W, and GPU utilisation at 100%. That was particularly pleasing because Nvidia cards throttle at 83°C or so by default, so if I was getting a lower temperature at full power, the fans clearly had some headroom for cooling. Once that was completed, it was time for another full training run for a burn-in. I kicked off my normal run. CPU and GPU temperatures stabilised at the same level as they had with the one-hour test, which was promising, so it was just a question of waiting... ...until I got this: About 40 hours, which is pretty much standard -- certainly the same as I'd expect from . The smoke test: Don't you just love it when your LLM tries to sell you something? 1 But anyway, loss on the test set was 3.548880, which is essentially the same as the same training run on too. So, now is a properly-configured training machine -- one RTX 3090, a CPU that runs a bit hot but at least doesn't do emergency shutdowns, and a case and a PSU with enough space for more GPUs. I think that the next step will be to move on to water cooling. In order to support more than one GPU, I'll need a new motherboard and probably a new CPU, so I don't think there's any point in watercooling the latter, despite its toastiness -- I'd just be buying a waterblock for it that I'd throw away in the not-too-distant future. Instead, I'll get the block for the GPU, and set up a loop to cool just that. Who knows -- maybe I can get rid of that horrendous RGB stuff at the same time! We live in hope. Also, that "expertise and expertise" tiny model smell.  ↩ is my daily driver. If he's doing a training run, then everything is just a little bit sluggish as CPU and GPU alike are busy. Although I don't play games often, it's annoying to have the option ruled out for days at a time. While the GPU is busy with a training run, I can't do other experiments in parallel -- for example, to scope out what the next step might be. Small enough to fit in a carry-on bag. I was building the machine in London, and wanted to be able to bring her to Portugal easily, and to be able to bring her back if I wanted to. Portable enough to quickly move around the flat. In the holiday home, the dining room was my study, so I wanted to be able to keep there normally, but move her when we had guests for dinner. Powerful enough to be able to run the games I was playing -- at the time I was a big fan of Assassin's Creed Odyssey , which didn't need a flagship card, but wasn't lightweight either. An AMD Ryzen 5 3600 3.6GHz 6-Core CPU A Noctua NH-L9a-AM4 CPU cooler A Gigabyte X570 I AORUS PRO WIFI Mini ITX Motherboard 32 GiB Corsair Vengeance DDR4 RAM 2x Samsung 970 Evo 500 GB NVMe SSDs A Zotac GTX 1660 Super 6 GiB GPU A Lian Li PC-TU100 Mini ITX case A Corsair SF450 450W SFF PSU An ASRock Phantom Gaming PG-1600G 1600W , which would have power in spades -- an RTX 3090 goes up to about 370W at full draw, so that should hopefully handle three of them plus a CPU without problems even if one or two of the GPUs had power spikes. A Fractal Design North XL . was already in a North (not the XL variant) and I love the case; the XL one looked like a good option if I was going to be cramming more GPUs in there, and had plenty of space for water-cooling. Vocab size: 50257 -- this was fixed because I was using the GPT-2 tokeniser. Context length: down from 1024 to 512 Embedding dimensions: down from 768 to 512 Number of heads: down from 12 to 8 Number of layers: down from 12 to 8 QKV bias: no (different to GPT-2, but the same as my own best local model). Also, that "expertise and expertise" tiny model smell.  ↩

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Giles's blog 3 months ago

Writing an LLM from scratch, part 34b -- from bigrams to GPT-2, one component at a time (in JAX)

This post is the capstone of the most long-running series on my blog . In December 2024 (!), I started reading Sebastian Raschka 's book " Build a Large Language Model (from Scratch) ", and worked through it carefully. Being who I am, despite trying to apply a strict "no side quests" policy, I found myself zooming off and digging into all kinds of things. It's time to wrap it up. I had decided that the endpoint would be to build and train an LLM from scratch just using my notes -- no reference to the book, no reference to the model code I'd written when following the book. After an X/Twitter poll, I decided to use JAX for that, just to make sure that I really was building it from scratch and not regurgitating bits of PyTorch code like a bad coding LLM spitting out half-digested lumps of Stack Overflow. In my last post , I showed how I built a JAX training script that mirrored what I had built for the original PyTorch version of the model. To test it as I went along, I used it to train a really dumb "LLM", which instead of trying to predict the next token for every token in an input sequence, instead predicted the input -- that is, if you fed it It would return the same thing. I called that an A-to-A model. In this post, I'll show you how I turned it into a GPT-2 model, and then trained it from scratch on my RTX 3090 (using the parameter counts for the original paper's "small" size). What turned out really well with this is that I found a route that meant that almost every component I added made the model better! That's not guaranteed -- sometimes different aspects of an AI model depend on each other, so adding A without also adding B makes things worse. But (admittedly with a bit of backtracking in places) I was able to find a route that shows a nice clear progression. The final training run took 37 hours 15 minutes -- compared to 40 hours, 38 minutes for an equivalent PyTorch model . That is despite it being full-fat 32-bit -- the PyTorch one was using Automatic Mixed Precision (AMP), which allowed it to use 16-bit calculations in places where it would be relatively harmless in terms of loss. When asked to continue "Every effort moves you", it came back with a decent response: The model got 3.418784 loss on my held-back test dataset, as compared to my PyTorch model's 3.538161, and even more impressively, it was better than the original GPT-2 small's result of 3.499677 on the same dataset! However, just as I found previously , the OpenAI weights still beat mine consistently in instruction fine-tuning challenges. Let's get started. At the end of the last post, we had a solid training loop, using all of the tricks I'd picked up with my PyTorch code. The A-to-A model we were training with it looked like this: That was based on my preferred model of how LLMs work , where at the top level for a model, we feed in a sequence of token IDs, then: The A-to-A model basically skipped the second step completely: it would project to embedding space, then immediately project back to vocab space -- and after training, it was pretty good at mapping a sequence to itself. One interesting question is, if we train the same code, but this time try to get it to make next-token predictions, how good will it be at that? Obviously it can't be as good as a full LLM. But there are correlations between tokens; full stops will generally be followed by spaces, adjectives will normally be followed by other adjectives or nouns (at least in English), and so on. It would be kind of like the predictive text systems on a phone, where (at least until recently) it would just use the last word you entered to generate a list of possible next words to select from. Old-school natural language processing has a name for this: bigrams. The idea is that you can work out statistically what the most common two-word pairs are, which allows you to make a guess at a next word from a single one. (There are also trigrams, where you look at the last two words when predicting the next, then 4-grams, 5-grams, and so on.) You'd build up a full probability table -- for every word in your vocab, you'd have the probability of every word coming next. So maybe even with that minimal model, we could get it to learn something similar to a set of token-level (rather than word-level) bigrams, which would then get the loss down. Obviously it wouldn't be as good as a full bigram table -- for our GPT-2 vocab size of 50,257, that would need 50 , 257 2 = 2 , 525 , 766 , 049 parameters -- but perhaps it could approximate one. (For comparison, the model we're using has just an embedding table and an output head, each mapping between 50,257 dimensions and 768, so that's 2 × 50 , 257 × 768 ≈ 77  million parameters -- about 3% of the full table.) An uninitialised model would (hopefully) have a loss of about 10.82, implying a perplexity equal to the vocab size. If we can train our dumb model to get better loss than that, then we'd have the beginnings of an LLM. That was a simple test to run. In my training code, I had a dataset class that looked like this: That is, the inputs, the , were the same as the targets, the . If we fed it ...then we'd be training it to output exactly the same thing. The modified version for a real LLM would involve feeding it something like this: ...and targeting this: That's a simple change -- that method became this: I did that, and kicked it off to train on the 92,209,152 tokens that I was (somewhat arbitrarily) using in the last post to test my training loop. The loss chart looked like this: That was pretty promising! Loss came down from roughly 10.82 down to a fairly stable 6 or so by global step 768, and seemed to flatten out there. It's possible that further training could have got it down a bit more, but I decided (again, somewhat arbitrarily) to use the average train loss in the checkpoint period ending at step 937 as my starting point. If we could make changes that reduced that, then we'd be moving forward. For this model, that value was 5.909. So, what were the changes we needed to make to change our bigram-style model to a real, if small, LLM? Adapting from my how LLMs work post, a GPT-2-style LLM looks like this. We receive our sequence of token IDs, and then: Inside the Transformers blocks, we: So that gave me the checklist; looking at it, the most tempting next step was layer normalisation (henceforth LayerNorm). It's used at the end of the core loop, and then twice in the Transformers blocks. What would happen if we coded it up, and then added it to the core only? The purpose of LayerNorm is to stabilise training. We constrain the values flowing through our model so that they have certain statistical properties that tend to make the whole thing more trainable. That would mean that if it did help with this model -- placed in between the embedding layer at the start, and the output head at the end -- then we'd hope for loss to go down faster, and ideally finish at a lower level. Time to code it up! NNX has its own LayerNorm implementation , of course (as does PyTorch ), but in the book, we implement it ourselves, and that felt like the correct path to take. Firstly, I implemented a dummy version: ...and updated the core to create and call one: And kicked off a training run for a few seconds just to make sure that it hadn't broken anything and that loss dropped -- being my first NNX module-inside-a-module, I worried that there might have been something non-intuitive that I had to do to get it to work. But everything seemed good -- loss was dropping, no errors. So, following the notes I made when I first learned about LayerNorm , I needed to make the values flowing through centred around zero by subtracting their mean, and then scale them to have a variance of one by dividing by the standard deviation (details in those notes). The shape of the I had coming into my class's was this: That was . So we needed to do those operations strictly on the last axis, manipulating each embedding independently. JAX has a function and a one , both of which take an parameter. The object repackaged those as methods, which was convenient, so I did a first cut test like this: That printed out these results: ...which looked plausible; one number for each embedding vector. Could we broadcast them across the array? This blew up: Fair enough. But and have a kwarg that looked like it would help: ...and it did! Excellent. So the next step was to see if that would work even slightly. Interestingly loss started off a bit higher at 11.29 after the first global step -- so adding in the LayerNorm had actually made the model worse than it was -- but it seemed to be falling rapidly. Things weren't totally broken, at least. But there was more to LayerNorm than just zeroing the mean and scaling to the variance; we also needed to scale them up by a learnable amount, and then shift/bias them by adding on a different trainable amount. More precisely, both of those trainable amounts were different for each of the (in this case) 768 embedding dimensions. We needed two learnable vectors of length . I hadn't noted it down at the time but I figured (as it turned out, correctly) that a sensible starting point for those values would be all-zero for the bias, and all-one for the scale. From this help page , the way you create a trainable array associated with an NNX module is this: That code created a random vector, rather than the zeros/ones we needed, and we'd need to get the dimensions right. Because of the "Incompatible shapes for broadcasting" error I'd just had, I was feeling a bit paranoid about the latter, so I chose a shape of , and wrote this: That looked pretty plausible, though in retrospect I think I was being overly cautious and didn't need the leading two axes for the scale and bias. The only thing I was unsure about was whether the wrappers I had put in were really making those arrays trainable. I put some code in to print them out and kicked off a run for a few minutes, and confirmed that they were changing in ways that seem plausible -- small non-zero bias, scale close to but not equal to one. That was all good! Next, I spotted one issue. What if one of the standard deviations was zero? That would lead to a divide-by-zero error here: Now, the standard deviation, if it's not zero, has to be positive -- so adding on a small value would fix that 1 : With that in place, I felt that it was ready to go. Time to do a full training run! I kicked that off, and it completed with this output: Loss looked like this: Let's look at the results for the previous run without LayerNorm for comparison: You can see that the new run, the first one, drops faster. It's harder to see from the chart, but it also finished up with a lower training loss at 937 (my relatively arbitrary metric): 5.734 rather than 5.909. That was interesting! The new model was basically doing the same thing -- predicting the next token based only on the "current" token, but loss was lower. My take is that if we had trained the non-LayerNorm model for longer, it might have managed to eventually grind out a better loss. But LayerNorm was doing its job -- it was stabilising training, and as a result we converged faster. That was a win! I decided to run it through my old smoke test from the PyTorch training runs, and see how it completed "Every effort moves you": It was kind of impressive that it managed to finish the first line before it got stuck in a loop -- but it was understandable that we couldn't expect anything good yet. Each predicted token was based entirely on the token before it. What next? Back to our checklist: Inside the Transformers blocks, we: So, at this stage, for each input token we were predicting the next one based on the input token only -- like I said earlier, we were doing a somewhat roundabout way of building an approximation of a table of bigram probabilities. What would happen if we started paying attention to the tokens to the left? And what would be the simplest, dumbest way to do that? The real LLM has multiple layers of multi-head attention, each one also having a feed-forward network, some LayerNorms, and some shortcut connections. Single-head attention is easier to code, but even on its own, you'd expect it to be able to add some value. Each token would get at least some information from the ones to the left. And one layer, likewise, you'd expect might help a bit. I suspected that it wouldn't work on its own -- I expected I'd need shortcut connections too -- but decided to start with attention on its own. I modified the main class to have a single "Transformers" layer: ...where that layer was actually just single-head attention: Next, it was time for the class. I'm not going to write yet another attention explainer -- I think my "How do LLMs work?" one does a decent job of that, and "The 'why' of attention, or: attention heads are dumb" works well too. So in the next bit I'll assume that you understand the basics. My first cut was basically just the maths (up to the causal mask) to get the attention scores: It did the projections into query, key and value space, worked out the attention scores with the array multiplication, normalised it by dividing by the square root of the number of dimensions in the Q-K embedding space, and then zeroed out the scores where a token was attending to tokens in its "future". There were a couple of problems, though. Firstly, that wouldn't work if we were working with batches, and secondly, zeroing out the non-causal scores wasn't quite correct. The batches first. Our incoming here would have the shape . After the projections to the Q-K embedding space, both and would also be shaped . Now, the property on the JAX array class just reverses the axes, so the code above would give us with the shape . That would break! Matrix multiplication in JAX expects all but the last two axes to represent batches, so we actually wanted to have the shape ` . That meant that what we actually wanted was to just transpose the last two axes. The JAX function takes an parameter that allows you to specify the specific re-ordering of the input axes that you want. So I could rewrite the code like this: As would have the shape , and the transposed version of would be , they'd be compatible for matrix multiplication and give us a result that was -- just what we wanted for attention scores. The next step was to fix the causal mask. The next step in this attention mechanism was going to be running the causal attention scores in through softmax over the last dimension, to convert them into attention weights. Now, our current code was zeroing out unwanted acausal scores, but a zero still contributes to softmax. If you want a particular value to come out of softmax guaranteed to be zero, you need to set it to minus infinity. I decided that the easiest way to do this was to create a causal mask -- a boolean array that matched the size of , but was full of s: Then I could zero out (well, "false out") the cells in the mask related to unwanted future-facing scores, just like I was previously doing on the scores: ...and then I could apply that mask to omega with , telling it to create a new array, taking the value from where the mask had , and in places where it had . That seemed solid, so I just needed to run the result through , specifying that the last dimension was the one where it should apply the function, and that would give me the attention weights: Finally, I just needed to use those attention weights to get the attention output by mixing in appropriate portions of the projection of the inputs into value space, : As was shaped , and (like and ) was shaped , the batch axes were at the start where they belonged, and the matrix multiplication would work and return something shaped . With that, we were done! The final single-head attention class looked like this: I kicked off a training run with that, and it did work, in that loss went down over the course of the run -- but at the end of the run, the loss at step 937 was 5.934 -- significantly above the 5.734 I got on the previous run, with no attention. But that made sense! As I'd said earlier, I suspected that this wouldn't help if we had no shortcut connection. Intuitively, if you want to work out what token should be at position n + 1 , on average the most important other token you need to know about is probably whichever one is at position n . Knowing about the tokens at n − 1 , n − 2 , and so on, could well be helpful -- maybe very helpful -- but not at the cost of not knowing about the one at n . Now, single attention heads are just simple pattern-matchers. They can't learn complex rules, it's only by working together -- "horizontally", in multi-head attention or "vertically" across multiple layers -- that they can do complex things. What we were asking this head to do was to learn some way of gathering information about previous tokens, and also to keep the knowledge about the "current" one. That's a tall order for a dumb attention head! In my mind, this is a large part of the benefit of shortcut connections. They are often presented as a way to make sure that during training, gradients flow smoothly from the output end of the model to the earlier layers. But I prefer to think of them as preserving the original embeddings, so that each layer doesn't completely replace what came into it, but instead does something closer to adding on its own notes -- like scholars adding commentary to a core text in the Talmud . In the training run above, the attention head was trying to learn how to preserve the meaning of the embedding it was working on, while also merging in information from earlier ones. If we added a shortcut connection, then it would only have to do the second of those two jobs. The code was simple: I updated the module to do a shortcut connection: I kicked off a training run, and at the end it printed this: The loss chart looked like this: And, importantly, that training loss at step 937 which I was using as a metric was 5.553 -- a decent improvement over the previous best of 5.734. Even a dumb single attention head was able to do something useful, if it had a shortcut connection. I decided to run another qualitative smoke test: I mean, it was repetitive, but it was actually getting noticeably closer to making sense! So that was excellent news. What next? Our checklist looked like this: Inside the Transformers blocks, we: Now, our single attention layer was lacking something. Without position embeddings, that layer has no idea what order the tokens before the one it's looking at come in. If it's considering the " cat" in ...it doesn't know if it's looking at "The fat cat" or "fat The cat". Position embeddings are simple, and might help, so that was the next step. These were trivial to add. We had this core code: So I just added a position encoding module in : ...and mixed it in with the token embeddings to create new, improved to be used in our "Transformers" layer: I kicked off a training run with that: Pretty hard to distinguish from the previous one, but the metric I was tracking, that loss at step 937, had improved again! We were down to 5.354 from 5.553 :-) A quick qualitative smoke test didn't show that improvement, though: Pretty much indistinguishable to the previous one. But still, Loss Number Went Down, and that's what was important at this stage. It was time to try the next step. From the checklist: Inside the Transformers blocks, we: We had only one attention head right now. Individually, attention heads are dumb , so switching to multi-head attention seemed like a good thread to pull. At this point, my single-head attention code looked like this: I decided to re-implement multi-head attention (which I'll call MHA from here onwards) from first principles rather than working strictly from my notes, and then to come back and check it. If you're looking at your browser's scrollbar with horror (" still only 50%?!") and really don't want to read a full derivation of MHA, you can skip straight to the first complete version of the code . The point of MHA is that we're running multiple copies of the calculation above in parallel -- let's pin down the name of the number of copies as . Now, we could naively implement it just by spinning off threads and running the existing code in each, but that wouldn't really take advantage of the GPU's inherent parallelism. I felt that we could rely on the fact that JAX's matrix multiplications treat all but the last two dimensions as "batches". For example, if you have two arrays with shapes: ...then you can multiply them. A m × n matrix multiplied by a n × p one will be m × p , so you'll get something that is The other dimensions (so long as they match) will essentially act as an a × b × c × . . . × l batch. Now, right now we were just using a single batch dimension. Let's look at the core multiplication in the attention mechanism, which works out , the attention scores. I had this: Breaking that apart into two steps: We got from this line: Let's look at the shapes here. is our input embeddings for this layer; its shape is . Projecting it through , which is shaped gives us a shape for of again. , being a projection of through , which is the same shape as , will have the same shape as . Now, that means that is , and the calculation ...is doing a batched matrix multiplication getting us the that we want, shaped . But as I said above, there's no need to stop with just one batch dimension. Let's say that we have heads, and that they each work with embeddings sized . Imagine that we've already somehow done multiple projections into the key and query spaces for each of our heads, and that the results have somehow been put into arrays such that and are shaped -- that is, we've gained an extra axis that keeps the projections for each head into its query-key space separate. We could use the fact that both of those two leading axes are basically just batch dimensions, and the existing single matrix multiplication will still work, with one tiny tweak: the current transpose is this: ...to swap around the last two axes of a three-axis array. With one extra batch dimension, we'll need to take account of that and do this instead: That will be a multiplication of , shaped , with , shaped , which gives us an of the right shape, . So, if we can start treating the heads as just another batch dimension, things seem simpler, at least for the attention score calculation. Let's continue down through the single-head code, and then come back later to how we might get the inputs into that double-batched shape. The next line after the calculation just scales the attention scores by a scalar: That looked fine, just a broadcast division-by-float. We'd need to change that to be in some manner, but that's all. The will give us an array that's full of s. That seems reasonable. The next step: What will that do? Well, per the documentation : When , operates batch-wise on the trailing axes. ...which sounded good. and would be treated as batch axes, which meant that the next line: ...would work. Likewise, with the next line: ...the axis to apply to is explicitly stated as the last one, which is what we wanted. So at the end of all of those steps, we'd have shaped , where the last axis had been softmaxed (softmaxxed?). The next line looked a little trickier: In the single-head version we had of shape , and V of shape , so multiplying them gives us In the new MHA code so far, we had our shaped . So in order for the matrix multiplication to work, we'd need to be shaped . That would give us a result shaped as . And conveniently, we'd already decided that the correct shape for and for was . If we could use the same "magic" to do the projection into value space -- that is, to get such that the heads formed a new batch-like axis like we had for and -- then we'd be all set. So, at that point, I'd worked out the core of MHA. If we could get all of the inputs into the shape , and somehow handle an output of the shape , then we could use MHA code something like this: The next question was, how do we get our inputs into that shape? We could run them all through separate per-head weights -- that is, have an array with one per head, like , and for the first one. But that, again, felt like it would be failing to take advantage of the GPU properly. The solution was to think of how matrix multiplications work. If you multiply two matrices, X · Y , the value in the result, in row r , and column c , is the dot product of row r in X and column c in Y . So, imagine if you wanted to multiply X by n different versions of Y , let's call them Y 0 , Y 1 , and so on up to Y n . If you imagine a new matrix, Y all , which is basically all the Y x s stacked side-by-side, then the dot-product understanding of multiplication makes it pretty clear that if you did X · Y all , you would get the results of all of those separate multiplications, also stacked side-by-side. I'll call that kind of matrix a "striped" one, for want of a better word. Now, when we project our inputs into the embedding spaces used for attention, we have code like this: We've initialised the weights, in this case, as an , so what is happening under the hood here is basically: That is, it is just a matrix multiplication. 2 So if we imagine that is one of those "striped" matrices, holding all of the separate matrices to do the projections for all of the heads in a single one shaped , then we could stick with the current code -- the Our input would be shaped , so the result would be , and would have the projections for each head in the same vertical stripes as the separate heads' projection weights. Now, like PyTorch, JAX allows you to reshape arrays. You can take one axis of length (say) m × n , and split it into two of lengths m and n respectively -- or, conversely, you can combine two axes of length m and n to one of m × n . If our data had the shape , we could reshape it like this: ...and that would split things up. So we'd have Q shaped as . That's almost what we wanted! We needed , and a simple transpose could sort that out: Likewise for and , and that was our inputs sorted. Moving on to the output; it came from this: ...and as we worked out above, it was shaped . I remembered that we wanted to run that through a single linear layer to combine all of the different heads' outputs into one. It felt like the best way to do that would be to get it back into a "striped" layout: . This would be something like the inverse of the input-wrangling. That would need a reshape, but before I could do that, I'd need to get the axes that needed to be merged next to each other. If the input to the linear layer was going to be , we'd need to convert it from to first: ... and then we could just reshape it to : Finally, we could run it through a linear layer, with set to , and set to . I put that all together, and decided to throw something extra into the mix. I remembered that Raschka's code had various checks to make sure that , which seemed a little artificial -- I'd read that this was true of GPT-2, but wasn't a necessary restriction for GPT-style models, which makes sense. There's no obvious reason per se why the heads' embedding dimensions should sum up to the higher-level embedding dimensions. So I decided initially to just pass in and to the constructor. In my training script I could force them to match the GPT-2 model, but if I wanted to use the code later for something different, I could vary them. Then I remembered that although the dimensionality of the embedding spaces for the query and the key vectors have to match (because otherwise you can't multiply them to work out attention scores with Ω = Q K T ), the value vector's dimensionality can in theory be different. So I decided to break into two separate and parameters. The result was this: Unusually for a case where I went off the reservation like this, the whole thing with the embedding space dimensionality didn't cause any problems at all! But there was one small bug in this code, which I didn't discover until later -- we'll come to it by the end of the post. At this point, I did another of my short training runs, and: ...with a loss chart that looked like this: The training loss at the 937th global step was 5.336, only a tiny bit better than the 5.354 with single-head attention. That was quite possibly within the noise. Even though (due to the restriction I was enforcing in my training script) the , , and arrays were the same size, I was creating that , which would consume randomness and make things vary. If I were doing a proper scientific experiment to see if a single layer of MHA beat a single layer of single-head attention, I think I would have run both for more steps to see if the difference became more pronounced later. But for the purposes of this post, I decided to move on. My checklist now looked like this: Inside the Transformers blocks, we: Adding that simple neural network -- the FFN -- seemed like a good next step. The feed forward network is simple; you take the output of the MHA block, run it through a biased linear layer to expand it from to , then run it through the GELU activation function, then shrink it back down to with another linear layer. I didn't really see any value in writing my own implementation of GELU, given that even in the book we were just given code for an approximation to type in. So, using , I wrote this: Note that I added in a shortcut connection around the FFN as well, so that it didn't overwrite what was there, but only "added on its notes". I kicked that off, and it ran for ten minutes or so, but then OOMed: Adding didn't help. I spent some time trying to dig into what might be causing it, but eventually noticed something interesting: in , the VRAM usage was consistently 75% throughout. Now I knew that JAX pre-allocates 75% of VRAM when it starts up, but I'd been assuming that it would try to grab more if it needed it. It turned out I was wrong with that assumption -- it grabs 75%, but that's all you ever get! The solution turned out to be the environment variable. If you set that to, say, , then JAX will pre-allocate 90% of the VRAM, and you can use all of that. (You can also make it allocate as-needed with , and there are various other settings you can control with other environment variables on that linked page). Anyway, setting it to to grab 90% of VRAM worked, and I was able to get a successful run: The loss chart was this: ...and the training loss at global step 937 was 5.295, compared to the 5.336 from MHA alone. Another tiny improvement, another one that could have been in the noise. Again, if I were doing a proper experiment, I'd do a longer run, but for now, I decided to move on. The checklist looked like this: Inside the Transformers blocks, we: Now, my gut instinct was that the layer normalisation inside the Transformers blocks was of most value as a way of stabilising training over deep networks. And with one layer, it didn't seem like the right time to add it. Instead, I decided to add on multiple layers. For GPT-2 small, you have 12 layers. That was already being passed in to my 's method as , so I just replaced this: ...with this: ...and then just renamed it where it was called; this: ...became this: I kicked it off, and it completed! However, the loss chart was telling: Ouch. Loss started dropping quite nicely, but then things got out of control and it settled down at a loss that was essentially that of a random model. At step 937, we were at 10.75, so just a hair less than the 10.82 that randomly guessing next tokens would give. Well, LayerNorm is specifically meant to stabilise training, and the checklist looked like this: Inside the Transformers blocks, we: ...and the only remaining step was that LayerNorm in the Transformers blocks, so it was time to add it in! As per the checklist, we do the LayerNorm after we've taken our copy for the shortcut connection, just before MHA, and then likewise after the second shortcut copy, before the FFN. As I understand it, this was a GPT-2 innovation -- previously, people had done normalisation after those steps, but this pre-norm setup turned out to work better. The code changes were simple. I added two modules to the class, and then called them in the appropriate places (taking the opportunity to tidy up the variable naming in the forward pass while I was there): I kicked it off and ran it, and got these results: That certainly looked much healthier! However, when I looked at the loss at step 937, it was 5.311 -- a tiny bit higher than the single-layer MHA example, which got 5.295. I'd been willing to play a bit fast and loose with this loss number and allow myself to accept a win when the loss went down a tiny bit, even if it was such a small amount that it could have been within the noise. But increasing loss -- even if it could also be within the noise -- was a step too far. I decided that in this specific case, I'd be strict and test the hypothesis that longer training runs would demonstrate an improvement between one single layer without pre-norm, and multiple layers with pre-norm. I had to remember that these training runs would not be comparable with the earlier ones. In the training script, I had a learning rate schedule like this : That straight-line warmup period and the following cosine decay were 5% and 95% of the training run respectively, which meant that (for example) global step 937 of the short runs we had been doing would be at a completely different point in the schedule than the same step would in these longer runs. However, they would be comparable to each other, and that was what mattered. After some humming and hawing, I decided that a full Chinchilla-optimal (for the full model) training run over 3,260,190,720 tokens, rounded up to fit into a round number of global steps, would be a nice experiment. I expected it to run comfortably overnight for the single-layer run, and take a bit less than two days for the multi-layer one. So I kicked off the first. Just over 11 hours later: Here's the loss chart: The last checkpointing period in that run ended at global step 33,164, and the training loss then was 4.165 -- indeed, it had been at around 4.17 for quite some time, though the trend still seemed to be a tiny bit downward. So then I kicked off a run of the full version -- multiple layers, with pre-norm in the Transformers blocks. Just over 37 hours later: The "Final train loss" line at the end said it all, really! But here's the loss chart: ...and the loss at step 33,164 was 3.399. Definitely quite an improvement over the 4.165 that a single layer got. Again, at some point I might do the equivalent tests for the earlier results where improvements appear to be pretty much in the noise. It would be good to be sure that the changes really did have the impact I think they did. But for now: our checklist was looking like this: Inside the Transformers blocks, we: Everything was checked off. So was this journey over? Well, there was one thing that the original PyTorch code had that my new code didn't: dropout. I'd found in my lengthy interventions experiments that dropout seemed to make models worse. It was, I felt, a smart idea back in the days when people had little data and did multiple epochs, each sweeping over everything, but it made less sense nowadays with single-epoch training runs over very large datasets. (Though I do have some intuitive ideas about why it could still help .) Still, it would be good to show that it harmed loss for this model as well. Checking my notes, I found that there were four places where dropout was applied: The changes are tiny and rather dotted around the code, so rather than showing you isolated bits of code, if you'd like to see it you can take a look at the code at this point and search for "dropout". When I started running that, I got an error when saving the first checkpoint: This was happening deep inside the bowels of Safetensors, but it made a lot of sense. The object needs to keep track of the state of the random number generator, and that meant that the function that I was using might return a structure that had something that contained that state, and was not compatible with Safetensors. I decided that I'd cheat a little bit here. If I skipped the dropout layers when I saved my checkpoints, like this: ...then I'd be able to save them. This would have a problem -- if I restarted from a checkpoint, the dropout pattern after the restart would mirror the dropout pattern from the start of the training run, because the random seed it started with would not have come from the checkpoint, but just the initialisation code. I felt that this would not have a serious impact, though, and given that I'd not had to restart from checkpoints so far, I (wrongly, as it turned out) decided it wouldn't matter. I kicked off the run, and... after four hours, it OOMed. I cursed, decided that I'd nurse this run through anyway (despite my dropout checkpointing concerns), and kicked it off again. Three hours later, it OOMed again. I happened to be away from home at the time, logging in to my machine remotely (thanks, Tailscale !), and on looking at , I realised that the X window system on my machine was using a gig or so of VRAM. I was running the training run in a session, which meant that I could kill X and not lose state, so I did that, and adjusted the environment variable I was using -- it had been 0.90, so I bumped it up to 0.95. I kicked it off again, and... Note that the tokens seen only relates to the period since the restart, which is why it was lower. One more loss chart: ...and the training loss at step 33,164 was 3.524, higher enough than the 3.399 I got without dropout that I was comfortable that it wasn't in the noise. That was very reassuring. Once again, if this was a proper scientific experiment I'd fix the issue with saving dropout, and run it completely from scratch -- or, at least, run it all the way through from scratch without restarts, even if I had to try several times to get it done. But I don't think that "replaying" dropout would make the loss any worse. And for this experiment, I felt this was enough. So: checklist complete. GPT-2 model coded up. It was time for some evals! I wanted to evaluate these models against the ones I got using the old PyTorch code: specifically, the last local training run that used exactly the same training hyperparameters, and only differed in that it was trained using AMP -- 32-bit floats in general, but using 16-bit where the framework thought it would not be harmful. In order to do exactly the same evals, I decided it would be easiest to write a conversion script to take the Safetensors files written to my JAX checkpoints, and write out new files that were compatible with the PyTorch model code -- then I'd be able to use the original PyTorch eval code. I put something together , converted my last two models -- the full runs with and without dropout -- and tried to load them up. Unfortunately there was an error: You might remember that back when I went through multi-head attention, I mentioned that I'd made a mistake. Somehow, I'd misremembered, and thought that the output projection -- the one that mixes together all of the different heads' outputs -- was a linear layer without bias, despite my original notes being perfectly clear that it did have bias. The good news was that if I disabled bias in the PyTorch code, I could load the safetensors files that I had. So the two models I'd trained so far were not useless, and could actually work as a kind of natural experiment into the benefits of having that bias there. But anyway, in order to do things properly, I was going to need to fix the bug and train yet another model. The fix was simple, I just replaced this (in ): ...with this: Then it was time to kick off yet another training run. After another 37 hours: ...with this loss chart: ...and the training loss at step 33,164 was 3.398 -- almost exactly the same as the 3.399 that I got in the no-dropout training run without MHA bias above! Well, now it really was time for the evals. I updated my conversion script to handle the bias on the MHA output projections, and used it to convert the three models -- the un-biased ones, with and without dropout, and the biased one, without -- to the PyTorch format, then ran the loss test that I had been using to compare the old models on each. Here are the results, compared to the previous models, and OpenAI's: That was a pretty amazing result -- I'd clearly proven that JAX trains much better models than PyTorch! 3.5% better in the best case. Well, OK, no. My guess is that the difference was probably something like better luck with the initial weights on the JAX side, plus the improvement from not using AMP . Anyway, the important thing was that the JAX models were in the same kind of loss range as the PyTorch ones -- and while a 3.5% improvement in loss was more variation than I'd been expecting, it was definitely the right ballpark. Now, one thing I had found in the past was that the OpenAI weights -- and some of my own models, like the Fineweb-Edu ones -- were consistently better at an instruction fine-tuning test than their test loss scores would indicate. Would that hold here? The IFT eval code fine-tuned each model on the Alpaca dataset until validation loss started rising, then used the model prior to the start of the rise to generate responses for a test set. These were saved, and then run past an OpenAI model so that they could be compared with each other: ...with the model order randomly changed for each query to avoid any position bias. The methodology seemed solid, but I was uncertain about the "train until loss starts rising", as it meant that different models had wildly different amounts of fine-tuning -- between two and seven epochs. On the one hand it felt "unfair" to certain models that they'd get less training than others. On the other hand, if the less-trained models had been trained past the point where their validation loss started rising, then assuming that loss would continue to rise, further training would actually be a disadvantage rather than an advantage. I decided to stick with the original plan, and train until validation loss started rising. I did, however, switch the judge model from the GPT 5.4 that I used in my last IFT test to GPT 5.5. Here are the results: More interesting datapoints! As before, you can see that low loss is not particularly well-correlated with a high score on this instruction fine-tuning test. The OpenAI weights continue to lead the pack, and while one of our new JAX models did quite well, it's still beaten by the Cloud FineWeb, 8x A100 40 GiB model. But what was important here, just as with the loss, was that the new JAX models landed in the same ballpark as the PyTorch ones. They did, and so I could be confident that they were doing essentially the same thing. And that meant that, after 18 months, I had reached the end of my LLM from scratch journey. It's been a long trek . I started reading "Build a Large Language Model (from Scratch)" on 22 December 2024. I was planning to breeze through over the Christmas break, but somehow it morphed into being a curriculum onto which I could hang projects to learn the fundamentals of LLMs, beyond what was in the book. In May 2025, I had my first real conceptual breakthrough when I realised that attention heads are (individually) dumb , and as I continued, the second big one came later on in the same month, when the concept of embeddings as being projections between vocab space and embedding space (and the converse projection in the other direction that happens in the LLM's output head) became clear. In August I had the first moment where I felt that the standard teaching approach to LLMs might not be the full story; shortcut connections are normally explained as a way to fix vanishing gradients, while I felt that a better way to see them was a way to allow attention and the FFN to "annotate" the existing information, similarly to how Jewish scholars have annotated the original text of the Talmud . (The results in this post seem to point in that direction, given how even a single layer of attention was massively helped by adding them.) By early December, I had essentially finished the book, and felt I wanted to try to train my first base model from scratch on my RTX 3090 . It worked, and wasn't far off the quality of the original GPT-2 small. I was really surprised that I could do that with consumer hardware, and became interested (perhaps obsessively so) with whether I could match OpenAI's weights. In January 2026, I trained a model using DDP on Lambda Labs , and then spent the following months training model after model, trying to work out which interventions -- learning rate scheduling, gradient clipping, etc -- would improve the loss. I wrapped that up in late April , with the interesting finding that although I'd been able to get the test loss pretty low, that didn't seem to map cleanly to performance in my instruction fine-tuning tests. In other words, Loss Number Goes Down is an interesting technical game to play, but doesn't cleanly map to real-world performance. The final step was this post, and the previous one -- could I, using my notes, implement GPT-2 completely from scratch in JAX without referencing the book? And as you've read, the answer was a definite yes! Of course, as with any long-running project, there are some loose ends -- from this post alone, there's the interesting fact that JAX trained faster than PyTorch (perhaps could close the gap?) and had a larger possible batch size for full-fat 32-bit. And the fact that fixing the multi-head attention bias bug didn't seem to help with the loss much was interesting too. But those are really details, and there's so much beyond them to learn. Longer-context LLMs: position embedding improvements like RoPE, efficiency tricks like flash attention and attention variants like DSA. Mixture of experts models. How do optimisers really work? ( Do they work? ) And plenty more. So it's time to draw a line under this series, and start thinking about what comes next. It's been a blast; if you've been reading along, I hope it's been as useful (and fun) to read as it was to write. And as always, comments, questions and corrections very welcome below. On looking back at Raschka's code, after having worked through all of this, there's a slight difference. I do this: ...whereas he does this: Now, the standard deviation is the square root of the variance, so if you ignore the small numbers -- in my case, and in his -- the calculations are the same. But there is a difference once those are taken account of. I don't think it's large enough to have any serious effect in these runs, though.  ↩ In PyTorch, linear layers are stored as the transpose of the matrix that would allow you to do that, so it would be: Also, note that for simplicity (heh) I'm disregarding bias in this discussion.  ↩ Firstly, we convert them into embeddings, so we get a sequence of vectors, one for each token. We do this by a lookup into a table, but we can see it conceptually as a projection via a matrix, from vocab space (where a particular token ID is a one-hot vector) to embedding space. Next, we do the magic with our Transformers layers, getting embeddings for the next token. After these layers, the embedding at position n in the output sequence is for the predicted token to come after the token at position n in the input sequence, considering that input token and all other tokens to its left. Finally, we project those back from embedding space to logits, this time actually using a real matrix (in the form of a linear layer), the output head. The logits (after being run through softmax) represent the probabilities for each token of it being the next one. Convert them into embeddings. ✔ ️done Add on position embeddings. Run these embeddings through multiple successive Transformers blocks. Layer normalisation Project them back from embedding space to vocab space. ✔ ️done Take a copy of the input sequence of embeddings Layer normalisation Run multi-head attention Add the copy back in so that the version that came out of MHA is something more like an "annotation" of the original Take a second copy of that one Layer normalisation again Run it through a simple neural network Add the results of that back in. Convert token IDs into embeddings. ✔ ️done Add on position embeddings. Run these embeddings through multiple successive Transformers blocks. Layer normalisation ✔ ️done Project them back from embedding space to vocab space. ✔ ️done Take a copy of the input sequence of embeddings Layer normalisation Run multi-head attention Add the copy back in so that the version that came out of MHA is something more like an "annotation" of the original Take a second copy of that one Layer normalisation again Run it through a simple neural network Add the results of that back in. Convert token IDs into embeddings. ✔ ️done Add on position embeddings. Run these embeddings through multiple successive Transformers blocks. part-done -- one layer only Layer normalisation ✔ ️done Project them back from embedding space to vocab space. ✔ ️done Take a copy of the input sequence of embeddings ✔ ️done Layer normalisation Run multi-head attention part-done -- single-head attention only Add the copy back in so that the version that came out of MHA is something more like an "annotation" of the original ✔ ️done Take a second copy of that one Layer normalisation again Run it through a simple neural network Add the results of that back in. Convert token IDs into embeddings. ✔ ️done Add on position embeddings. ✔ ️done Run these embeddings through multiple successive Transformers blocks. part-done -- one layer only Layer normalisation ✔ ️done Project them back from embedding space to vocab space. ✔ ️done Take a copy of the input sequence of embeddings ✔ ️done Layer normalisation Run multi-head attention part-done -- single-head attention only Add the copy back in so that the version that came out of MHA is something more like an "annotation" of the original ✔ ️done Take a second copy of that one Layer normalisation again Run it through a simple neural network Add the results of that back in. Convert token IDs into embeddings. ✔ ️done Add on position embeddings. ✔ ️done Run these embeddings through multiple successive Transformers blocks. part-done -- one layer only Layer normalisation ✔ ️done Project them back from embedding space to vocab space. ✔ ️done Take a copy of the input sequence of embeddings ✔ ️done Layer normalisation Run multi-head attention ✔ ️done Add the copy back in so that the version that came out of MHA is something more like an "annotation" of the original ✔ ️done Take a second copy of that one Layer normalisation again Run it through a simple neural network Add the results of that back in. Convert token IDs into embeddings. ✔ ️done Add on position embeddings. ✔ ️done Run these embeddings through multiple successive Transformers blocks. part-done -- one layer only Layer normalisation ✔ ️done Project them back from embedding space to vocab space. ✔ ️done Take a copy of the input sequence of embeddings ✔ ️done Layer normalisation Run multi-head attention ✔ ️done Add the copy back in so that the version that came out of MHA is something more like an "annotation" of the original ✔ ️done Take a second copy of that one ✔ ️done Layer normalisation again Run it through a simple neural network ✔ ️done Add the results of that back in. ✔ ️done Convert token IDs into embeddings. ✔ ️done Add on position embeddings. ✔ ️done Run these embeddings through multiple successive Transformers blocks. ✔ ️done Layer normalisation ✔ ️done Project them back from embedding space to vocab space. ✔ ️done Take a copy of the input sequence of embeddings ✔ ️done Layer normalisation Run multi-head attention ✔ ️done Add the copy back in so that the version that came out of MHA is something more like an "annotation" of the original ✔ ️done Take a second copy of that one ✔ ️done Layer normalisation again Run it through a simple neural network ✔ ️done Add the results of that back in. ✔ ️done Convert token IDs into embeddings. ✔ ️done Add on position embeddings. ✔ ️done Run these embeddings through multiple successive Transformers blocks. ✔ ️done Layer normalisation ✔ ️done Project them back from embedding space to vocab space. ✔ ️done Take a copy of the input sequence of embeddings ✔ ️done Layer normalisation ✔ ️done Run multi-head attention ✔ ️done Add the copy back in so that the version that came out of MHA is something more like an "annotation" of the original ✔ ️done Take a second copy of that one ✔ ️done Layer normalisation again ✔ ️done Run it through a simple neural network ✔ ️done Add the results of that back in. ✔ ️done Once in the main body, just after we've worked out the embeddings. Twice in the transformers block: once after attention (but before the shortcut is mixed back in), and once after the FFN (ditto) Inside multi-head attention, on the attention weights ( which surprised me ). On looking back at Raschka's code, after having worked through all of this, there's a slight difference. I do this: ...whereas he does this: Now, the standard deviation is the square root of the variance, so if you ignore the small numbers -- in my case, and in his -- the calculations are the same. But there is a difference once those are taken account of. I don't think it's large enough to have any serious effect in these runs, though.  ↩ In PyTorch, linear layers are stored as the transpose of the matrix that would allow you to do that, so it would be: Q = xs × W q T Also, note that for simplicity (heh) I'm disregarding bias in this discussion.  ↩

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Giles's blog 3 months ago

Writing an LLM from scratch, part 34a -- building a JAX training loop for an LLM training run

For over a year, I've been using Sebastian Raschka 's book " Build a Large Language Model (from Scratch) " -- and the multitude of side-projects that have branched out from reading it -- as something like a curriculum for learning about modern AI. The one final task I had set myself was to build and train an LLM from scratch just using my notes -- no reference to the book, no reference to the model code I'd written following the book. As an output, I wanted something as good as my best PyTorch model based on Raschka's code -- a base model, trained on 3.2B tokens, that my (admittedly limited) evals ranked as being close to the original GPT-2 small's quality. I wanted to use a different framework, just to make sure I wasn't parroting code that I'd somehow memorised, so I asked people on Twitter which one I should use, and the winner was JAX . I took a slightly different route to Raschka's book; he takes an inside-out perspective, explaining things like attention, gradually building up a complete GPT-2-style model, and then building a training loop on top of it. I wanted to go outside-in: I'd put together a training harness to train the simplest-possible model with an API similar to a real LLM, get that working to my satisfaction, and then add features to that simple model, one by one, until it had the full architecture in place. The plan (which actually worked out nicely!) was that I'd be able to show how each change improved things. That's all done now, and I'm posting about it in two parts; in this one, I'll explain how I built the training harness, and in the next, I'll show the actual building and training of the LLM. So let's get started! JAX itself has a relatively minimal API, and doesn't include standard neural network components like linear layers. Likewise it doesn't have any built-in optimisers, data loaders or similar ML utilities. Now, I could have decided to build my LLM using just pure JAX, like I previously did with a toy XOR model . But I felt that it would be better to build this in the style that real-world JAX code is written, which would mean using some of the many utility libraries . On the JAX site itself, there was a useful-looking link: "If you’re looking to use JAX to train neural networks, check out the JAX AI Stack !" On the linked page, it made it clear that the two core parts of that stack were: I took a look at both, and they seemed pretty easy to grasp. Indeed, at first glance, I felt that NNX looked pretty PyTorch-like! In their tutorial example, the only real obvious difference was the JAX-y derivative-style gradient calculation and the way that random numbers were handled. And even the random numbers were handled in a less pure-functional way than pure JAX -- instead of having to mess around with splitting keys, you could just pass in what appeared to be a stateful variable that somehow split itself internally as needed. So, NNX and Optax were the frameworks I'd use. Rather than grinding through the tutorials, I decided that I'd just dive right in, and try to pick things up as I went along. How hard could it be...? To build a functioning training loop, I needed a minimal model to train -- not an actual LLM, but something that behaved at least a bit like one. It would take in a sequence of tokens, and spit out logits for each token. In my preferred model of how LLMs work , at the top level for a model, we feed in a sequence of token IDs, then: All of that suggested to me that the dumbest "LLM" I could write just to get started would be one that just projected token IDs into embedding space, and then projected back to vocab space. No Transformer layers at all. I'd then train it so that instead of trying to predict the next token, it would try to "predict" what was fed into it in the first place. In other words, you'd feed the training loop this input: ...and this target ...rather than the normal setup for an LLM, where you feed it ...and give it targets of If I could get that to work -- and it felt like the kind of thing where you'd be able to get the loss down to near-zero without a huge amount of training -- then I could be reasonably sure that I had a working training loop. 1 I decided to call this an A-to-A model. Coding up the model itself was ridiculously simple: it looked like this: There's as much boilerplate in there -- for the parameters that I knew that the model would need when I built out the full LLM -- as there is actual code doing stuff! But the training loop was a bit more fun. As I said, my plan here was to make sure my understanding of the internals of LLMs was correct by rebuilding one just from my notes. That "notes only" restriction didn't apply to the training loop itself, so I allowed myself to crib a bit from the PyTorch DistributedDataParallel code that I'd been using to train the original model in the cloud. The first version that I used is here . Let's start at the bottom, where we have the function . It starts with some boilerplate to handle the concept of "runs". This is a pattern I've found myself using in most of my projects. When working on a model, it's useful to be able to do multiple training runs, changing things each time. You want to keep the checkpoints, metadata and training charts for each one for future reference. So in my repo, I'll have a "runs" directory, and in there subdirectories for each training run I want to track. In those subdirectories, there are JSON files -- one to configure the model, , and one to configure the training hyperparameters and similar stuff, . (It's worth noting that at this stage, a bunch of those hyperparameters were unused; I kept them in there out of laziness, as I knew I'd need them later.) So we start our function by loading those. Our next step is to completely ignore one of the training hyperparameters, . I definitely wanted to do gradient accumulation , but decided to leave it for later. Better to get a solid, simpler training run done first, I felt. Next, we download the dataset we're going to use to our local disk with (which will only download if there's not an up-to-date copy already there). The next step is to call to load it into RAM. You can see that there's another hard-coded variable there, . This is a holdover from the multi-GPU DistributedDataParallel code that this was all based on; in this blog post I'm only covering the code for single-GPU training, but I decided to leave the DDP stuff in there for dataset-wrangling purposes, hardcoded to one GPU, so that it would be easier to re-introduce if I later decide to implement something similar in JAX. Let's take a look at and its related stuff. If you go up to line 39 you'll see the code. Firstly, there's a that keeps track of our training data. If you look closely, you might spot one oddity in that class. We have this: Remember that at this stage, the plan was to train the model to map tokens to themselves rather than to make next-token predictions. So the targets are the same as the inputs, not the more normal next token, which would look like (and, in the next post, will look like) this: Next, we have a function to load the appropriate subset of the data from the copy on the local disk into one of those objects. I hit an out-of-memory issue when I ran the first version of this. It was trying to load the data into my GPU's VRAM -- JAX's default behaviour if you have a GPU, and the CUDA version of JAX is installed -- and there was too much to fit in there. After a bit of digging around I learned how to change the JAX default device so that it would be loaded into normal system RAM. Unfortunately, once I'd done that, I found that iterating through it was super-slow -- it took about 1.2 seconds to get one training batch of 6,144 tokens out of the array, which meant that I'd have a limit of 5,120 tokens/second of training from that alone. I eventually learned that the data had been loaded into the main RAM, but was being copied up to the GPU for processing because it had not been committed to the main RAM -- details here . Fixing that (with an explicit call to ) meant that getting a single training batch from the dataset and putting it onto the GPU took less than 0.001s, which was much better. So that was many hours of work that all got packed into lines 55 to 58 of the code: The remainder of the logic in is just to make sure that we have a dataset that is exactly the right size for the world size (even though that's always one right now), the microbatch size, the gradient accumulation steps, and the sequence length that we're working with, Let's go back to the function again. Having loaded our dataset, we create our model, passing in the model configuration stuff and also the (currently unused) dropout rate training hyperparameter, then we create a Flax NNX optimiser which wraps an Optax one. This was essentially a copy/paste from the Flax tutorial, except we're configuring the optimiser with learning rate and weight decay hyperparameters from the training config: Finally, we call to kick off our training loop, passing in some appropriate stuff. Let's go to that function next. We start off with a bit of housekeeping, then go into the main loop. You can see that it's kind of gesturing at gradient accumulation: ...but if you look at the actual body of that loop, it's not doing anything of the sort. It's just getting training batches, putting them on the GPU, doing a full training step, and keeping track of some metrics: So, we're just doing a traditional batch-by-batch training loop without gradient accumulation right now. But some of the infrastructure is there, because it was the next thing I wanted to add after I'd got the basic loop working. The rest of the function is just housekeeping and checkpointing; we'll come back to the checkpointing shortly, but first let's take a look at the function that actually trains the model on a set of inputs and targets, and its associated function -- they're just above . Now, as you might remember from my first JAX post , the best way to JIT a training loop is at as high a level as possible. So when I first coded this, I integrated that into the traditionally-named function like this: When I actually came around to run it the first time, loss wasn't falling at all, and after banging my head against it for a while, I realised I should have used rather than , fixed that, and kicked it off again. Loss started falling immediately. D'oh! Now let's take a look at loss. Cross entropy loss was clearly what I would need to train an LLM, and also felt like the right thing for the A-to-A model. Optax has five loss functions that are related to cross entropy; three of them looked a bit more complicated than I needed: So it was a choice between The latter was the right one -- expects the labels (that is, the target token IDs) to be one-hot vectors, while , as it says in the function name, expects integer labels, which is what we have. That sounded pretty similar to PyTorch's , but there was an important difference. For normal use (if you're not using K-dimensional loss, whatever that might be) PyTorch expects that the inputs are either just a one-dimensional tensor of c logits, or at worst a b x c matrix, where b is the batch size. I had noted when working through this section of Raschka's book that the code we wrote flattened things out. So a batch of six sequences, each 1,024 tokens long, with a vocab size of 50,257, would give us a logits tensor shaped like this: The first axis is the batches, the second is the length of the sequences -- remember, we have logits for every input token in the sequence, with next-token predictions for that token in the context of all of the other ones to its left. And the last axis, with a size equal to our tokeniser's vocabulary size, is the logits themselves. After flattening, it looked like a "batch" of 6 * 1024 = 6144 logits vectors: Likewise our targets -- the token IDs we wanted our model to be predicting -- were batched, and there was one per token in each sequence, so that tensor was Flattened, it looked like a "batch" of 6 * 1024 = 6144 targets: Finally, the PyTorch function returned a scalar value -- wrapped in a PyTorch object, of course, so that it could participate in the backward pass, but a single number. But I'd forgotten about all of that when I was writing this part of the JAX code, and just fed the inputs and the targets straight in to the JAX function. The result was interesting. I started with this: And printing out the shapes of each variable gave this: It had returned a cross entropy number for every element in every sequence, across all of the batches! What's interesting is that the docs for imply that it has the same restrictions as PyTorch's -- it expects a single batch axis in the tensors that are passed in. Perhaps they're out of date? Or perhaps Optax just assumes that you know that in JAX "a batch axis" should be read as "as many batch axes as you want"? Well, anyway -- it worked, and I checked that the numbers were solid. Now, of course, we can't ask JAX for gradients using that 6 × 1024 matrix -- the loss function needs to return a scalar -- but the function on a JAX array does exactly what we need. So I had a solid loss calculation, which you can see in : So that's covered our loss function and the JITted that uses it. The only remaining code that I haven't gone over in this version of the script is the stuff immediately above -- and . These are both called as part of the housekeeping code I glossed over in the function, after we take checkpoints. They just redraw a plot of the loss and other training metrics, using stuff that's stored in the metadata of all of the checkpoints so far. That means that there's a nice graphical way to keep track of a training run. Fairly dull stuff, so there's no need to go through them, but it is worth taking a look at the checkpointing code itself. You can see the version I was working with at this point here . It's not really much of a checkpoint; I was saving the model itself and the metadata needed for that charting code, but not the optimiser, which would be needed for a real checkpoint. After all, the purpose of a checkpoint is to be able to pick things up again if your training loop crashes, and you can't do that without the optimiser's state. Still, it was enough to get started with. That said, one wrinkle I encountered when writing that simple checkpointing code was that it was a tad tricky to save them in Safetensors format -- you can see the details here . So, that was my initial training code. It was time to let it rip: could I train my dumb "LLM" to map from A to A? As I mentioned earlier, the very first run didn't converge at all -- loss started at about 10.82, which was promising (it's exactly what you'd expect for a randomly-initialised network trying to predict GPT-2 tokens -- see here for details), but then it remained there. But when I fixed the " should be " issue, it started dropping. After 92,160,000 tokens seen, it seemed to have hit zero (at least to the three DPs I was printing), so I baked that into and did another training run fixed to that number of tokens. After about 14 minutes, it finished: A very promising final loss, even though that was just whatever we got on the last batch! The actual loss chart looked like this: If you're used to the loss charts in my previous posts, there's something to highlight here: I've switched the Y axis over to being log, so those bumps near the end are actually tiny deviations away from 0.001. I think it's worth showing what the model actually did at this point. It was actually somewhat later that I wrote some code to load up the model checkpoints from these training runs and do some smoke tests, but I'll show you some results now. I wrote some code based on my JAX safetensors post to load up a model's parameters from a checkpoint's file: ...and then wrote two test scripts. Firstly, was it really mapping from A to A? I wanted to be sure that the loss number was actually reflecting what I wanted it to reflect. I wrote a simple script that took a Safetensors file on the command line, and ran the first verse of The Rime of the Ancient Mariner (chosen because it uses oldish English so there are some odd tokens in it) through the LLM it loaded from that file. Here's what the model at the end of the run came up with: That's great! It could certainly handle the mapping. Out of interest, I decided to see how quickly it had learned to get that right. The average training loss in that "best" checkpoint at the end of the training run was 0.0001, so how did the mapping improve, and what was the loss, near the start of the training run? For the first checkpoint, when we'd just run one batch through, we had an average training loss of 10.8242. With the model parameters that were saved then, we get this output: As you'd expect from that loss, it's total token salad. Now let's take a look at the next checkpoint, taken after 375 "global steps" -- that is, 6,000 batches. In that one, the average train loss since that first checkpoint was 2.9323. But that hides something important -- the maximum loss, near the start, was (as you would expect) 10.78524, not much less than the average loss in the previous checkpoint. But the minimum (which we can safely assume was towards the end of this checkpointing period) was 0.54155, so we can reasonably assume that the model improved very rapidly at this point. And the A-to-A test bears this out: So, we can see that the bulk of the improvement happened right at the start! It was able to pass the A-to-A test for that fairly unusual sequence after just 6,001 total batches of 6 1,024-token sequences. The rest of the training run was perhaps just grinding out improvement on rarer tokens, and perhaps making it more certain about already-correct predictions. After all, the test script was simply printing the most likely token for each position, so at this state it might have been predicting some of those tokens as 51% probability. That would have meant a penalty in the loss function, even if the answer was actually correct. So that was an interesting script; I wanted to do another -- the standard smoke test that I've been using, based on Raschka's prompt: how does the model complete "Every effort moves you" when asked to continue the sentence? Here's the script , and here's what it generated: That makes perfect sense. In order to generate the next token in an autoregressive loop, we're looking at the logits for the last one in the prompt. When it first runs, the last token is " you", and our model is trained to map A to A, so its result is " you". We append that to the prompt, run it through again, the last token is still " you", so of course it "predicts" the token " you" again. And so on. So these results were both good news! The A-to-A mapping was working, and was converging rapidly in terms of loss -- and even more rapidly in terms of our poetic test. So, what was next? I wanted the training loop to be as similar as possible to the code I used for my best locally-trained PyTorch model . That used three things I had not built into the training loop at this stage: learning rate scheduling, gradient clipping, and gradient accumulation. The PyTorch code also had the ability to restart from a checkpoint -- not super-important in a 14-minute training run like this one, but I figured it would become important later. After all, the PyTorch runs on my local machine had taken almost two days, and if something went wrong halfway through (cat jumping onto PC power button, etc) then I really wouldn't want to start from scratch. I decided to handle gradient accumulation first. In PyTorch, doing gradient accumulation is pretty simple: the core of a typical training loop without it might look something like this: We start off by clearing out any gradients that are stashed on the model's parameters, then do a forward pass, work out the loss, do a backward pass to put new gradients on the parameters, and then step the optimiser to apply those gradients. Accumulating gradients just means changing it to something like this: That is, we do a forward and a backward pass times. Because we're not zeroing out existing gradients between them, the parameters will accumulate gradients over time -- each backward pass will add its contribution onto what is already there. Each time, we divide the loss by , so that the gradients that are put on the parameters are that much smaller, which means that by the end of our loop we've got gradients that are the average of what we'd have got if we'd done all of these microbatches in one big batch. Finally, once we've exited the loop, we step the optimiser to apply those averaged gradients. When I started thinking about implementing this in JAX, I noticed that Optax has a help page on how to do it , but then I had one of those brilliant shower thoughts that one sometimes has. I should have learned by my age that they rarely work out well, but this time I decided to give it a go rather than doing things the official way. My brilliant idea was that with some finessing, we could put the whole gradient accumulation loop inside JITted code. From what I'd learned so far, the higher up in our code we put the JIT decorator -- that is, the more of the training loop it covered -- the faster it would be. In itself, that wasn't a bad idea. But my first implementation was less smart: The were full-step arrays (eg. shaped (16, 6, 1024) for 16 gradient-accumulation steps over 6 microbatches of 1024 sequences), and the targets likewise. That seemed very clever! But in retrospect, it was obviously doomed to failure, and when I ran it, I ran out of VRAM. The point of gradient accumulation is that what you accumulate over time is, well, gradients. So you have to do a full forward pass and then a backward pass over the model for each microbatch, letting gradients build up, and then apply those in one go, like the PyTorch code did. Unfortunately what I was doing with my code was essentially all of the forward passes, one by one, letting the activations and JAX's internal structures representing what calculations had been done accumulate -- not the gradients -- and then doing a single backward pass across all of that. Mathematically it made sense -- I would have got the right effect if I'd had enough VRAM -- but it wasn't much more memory-efficient than just doing a single batch of sequences. Immediate CUDA OOM. My second attempt was a bit more sensible and ran OK without the JIT: You can see that now I was doing both the forward and the backward pass within the loop, and then working out the mean gradients with that , then passing those average gradients to the optimizer. It all made sense, and seemed to work when I ran it: ...and it wasn't as much slower as I would expect given the lack of JITting: 1,146 seconds versus 843. It was interesting that the final train loss was higher than the run without gradient accumulation, but larger effective batch sizes are not always a better thing: it depends very much on the model you're training and the data. The batch size and number of gradient accumulation steps I was using were ones I had optimised for the full 163M-parameter GPT-2-style LLM, not for this model. So it was OK if it was a bit worse. Anyway, I tried adding the to that function, and ran it: Ouch. And looking at the traceback, it appeared that it was the actual JITting that was running out of VRAM. Something to do with loop unrolling, perhaps? I dug around for a while, trying to use JAX's rather than a normal Python one, but to no avail -- I would always run out of GPU memory. Eventually, after a few hours, the alarm bells on my side quest detector had become too loud to ignore. Reluctantly, I gave up on hand-rolling my own gradient accumulation, and implemented it the Optax way . That was actually really nice and simple. The code is here , but the change is tiny and simple to explain. Remember that we had this code to set up the optimizer: That creates a Flax NNX optimiser, which uses an Optax AdamW optimiser under the hood. The Optax way to do gradient accumulation is to wrap the optimiser in a helper, which -- with the NNX optimiser wrapping the result -- looks like this: The wrapper is really neat. It has the same interface as a regular optimiser, so its method can be called with a set of gradients. But instead of applying them, it just accumulates them until a particular number of calls to have been made, at which it actually does apply the mean of the accumulated gradients, and resets its counter so that it starts accumulating again. That's actually a really nice API. And it actually meant that I would have been able to simplify the training loop. Remember, we had this: The loop-within-a-loop was needed by the PyTorch code, because we needed to do the optimizer step at the end to apply the accumulated gradients. But with the Optax wrapper, we could have just iterated over our samples in one top-level loop, relying on the to make its updates every iterations. However, I decided to leave it in -- keeping track of the training in terms of global steps meant that the training output with my JAX model would be easier to compare to the PyTorch versions. Perhaps if I'd been building the training loop completely from scratch I would have chosen differently. Anyway, with that code change in, I ran it, and: I had the same loss at the end as the by-hand un-JITted version, which was reassuring. And it was slightly faster than the non-gradient-accumulating version, but it's a small enough difference that it was probably just in the noise. So that was gradient accumulation! Here's the code with that added . Next, I wanted to get charting and scheduling of the learning rate, and gradient clipping working. Scheduling the learning rate means that we'll be changing it over the course of the run -- like this example from one of my PyTorch training runs: Having a chart like that one is really useful, as it allows you to sanity-check that the changes you are making to the learning rate really are the right ones. So I wanted to add the charting first, and then the scheduling. The boilerplate code to actually generate the chart, given learning rate numbers in the checkpoints' metadata, was already there, so I had to work out how to extract the current value of the learning rate from the optimiser and then save it into the checkpoints. This was the obvious starting point . Optax optimisers themselves don't store the learning rate, but if you create them like this: ...where the in the brackets is the normal stuff that you'd pass in to the optimizer when creating it, then you can extract the learning rate later. However, the code on that help page was using the Optax optimiser directly, whereas my one in the training code was wrapped inside a , which was in turn wrapped inside an NNX object, like this: Still, the solution seemed reasonably clear. I could use the trick on the that I was creating, and then pass it in to be wrapped like this: The next question was how to actually read the learning rate from that optimiser. The sample code in the Optax docs looked like this: Again, that was using the Optax optimiser directly, rather than trying to use one that was inside an NNX one. However, in the docs for NNX's optimiser I noticed that it exposes its wrapped Optax one's state as . I put in some temporary debug code to print that, and saw that it was the ' state, which made sense -- and that, in turn, contained the state of the wrapped one as . That had a field called , which was a dictionary that included as a key. Finally, the value that that key pointed to was a object. To get the actual value from there, you need to call its to get the actual value, which is a JNP array, so we needed to call on it. All of that led to the following abomination unto God, mankind, and the Law of Demeter : Eurgh. I mean, really, eurgh. Well, anyway, I put code to do that into the function and save the number as part of the metadata. I did a partial training run, just for long enough to confirm that the learning rate chart was being generated, and had a flat line on it at 0.0014, the constant learning rate I was using at that point. I can't say I was very proud of it, though. To recap, the learning rate schedule that I wanted was this: That's formed of two phases: an initial warmup, where the learning rate started at 0.00001 times the desired peak value, and then rose linearly to the peak, followed by a cosine wave to decay it to 0.1 times the peak. In PyTorch I had had to use different learning rate scheduler objects to handle each phase, with a wrapper to bolt them together : However, it's a common pattern in training loops, and conveniently Optax provides a class that does all of that for you. The only oddity in it is that is kind of misnamed; it's actually total steps, including the warmup. So I wound up writing this code: I did a training run with that, and it completed with this: The loss was a bit worse again, but just as with the gradient accumulation steps, the learning rate schedule I had specified was specifically designed for training a real (if small) LLM, not for this toy A-to-A task that I was using to test the training loop. The important thing was the learning rate chart, and it looked like this: Perfect! Here's the code at this point . There were two boxes left to check before I had a training loop I could actually use to build the LLM: gradient clipping and the ability to restart from a checkpoint. I decided to do gradient clipping first. Gradient clipping is where for each update, you look for gradients that are suspiciously large, and cut them off so that they don't make excessive changes to the model. The Optax docs made it look pretty simple: So, you use an to chain together first a thing that does clipping, and then the actual optimiser -- presumably the first thing in the chain sees the gradients and does stuff to them, and then the second receives whatever the first has returned. Now, the question was, should we do the chain outside or inside the MultiSteps? That is, should we clip gradients each time before we step the MultiSteps optimiser, or do we accumulate them and clip the average before we step the inner AdamW one? Looking at the old PyTorch code , I was running the gradient accumulation loop, and then clipping at the end. So the gradient clipping was happening to the accumulated gradients. That actually felt less intuitively good than the alternative, but I decided that we should try to mirror what the PyTorch code is doing. So: So, the optimiser would receive clipped gradients. Because it was wrapped in the , it was receiving the accumulated gradients every time that object hit its limit. Unfortunately there was still a problem: that change meant that the optimiser that we were reading the learning rate from with this horrendous code in the function: ...would now be inside yet another level of nesting -- the object. So, of course, when I ran it, it blew up with an error: I used some debug prints to work out what was going on, and determined that the state of the object was a tuple, the first element being an essentially-empty state for the clipper, and the second being the hyperparameter-injected state for the . So that meant that the new correct code to get the learning rate would be this: Note that we've gained that to do the lookup into the 's tuple state. I remember coming across a comment saying "forgive us for our trespasses in this method" in a codebase long ago, and I know well how the author felt. I did have an idea of how to at least limit the blast radius a bit, though. At this point in the code, I had the complex optimiser setup in the function, and the learning-rate-getting abomination in . I decided instead to define a function called right next to the optimiser setup, and pass that in to . So the horror was still there, but at least it was all in one place, like this: ...where called where it needed it. I was just about to kick this off, but by chance happened to take a closer look at the documentation for , and spotted that it said Clips updates element-wise, to be in That rung a bell! When I was originally looking into gradient clipping for the PyTorch training loop, I noted that that is a perfectly valid way to do gradient clipping, but it's not the way I ultimately chose. Instead, I was clipping based on the L2 norm. The JAX training code was meant to work the same way as the PyTorch code, so that was a good catch; I switched over from using to using , and then kicked off another training run: Everything looked fine; my guess was that the final loss was so similar because a simple task like A-to-A mapping, with such a shallow network, would be unlikely to cause gradients to explode. But it would be nice to be sure. Was there some way I could track the gradients and see if clipping had had to cut in? One neat thing we had in the PyTorch code was that we could track gradient norms pre-clipping: Unfortunately, and the general Optax API doesn't provide any way to access the pre-clipping norms: the that was the zeroth element of the state of the that we were reading in the horrendous learning rate-reading code is an alias of . I considered using to work out the norms directly, and logging that, but that would be tricky -- because the gradients we were applying the clipping to were not the ones that were generated in the function, but instead the ones that had accumulated inside the object over multiple gradient accumulation steps. This sounded like a lot of work for a not-enormous benefit, so I decided to leave it out for this project. There was, however, one small change that I wanted to make while I was messing around with gradients -- what to do if non-finite numbers crept into them. Back when I was first looking into gradient clipping, I was somewhat horrified to realise that the scaler object I was using to tell PyTorch to train in 16-bit for things where it felt it would help (Automated Mixed Precision, or AMP), was silently dropping any updates with non-finite gradients, and if you didn't use AMP, such gradients would be happily applied to your model, most likely completely breaking it by setting parameters to non-finite values. This felt like the wrong place for that kind of logic to go -- I felt that it should belong to the optimiser, or at least in some other part of the stack that wasn't specifically related to the totally orthogonal task of mixed-precision training. I checked what JAX's default behaviour with non-finite gradients was, and it turned out to be to just apply them -- but, with Optax, it actually was something you could fix at the optimiser level. If you wrap an Optax optimiser with , it will only apply finite gradients, so we could add it to the optimiser setup like this: I set to infinity to mirror the PyTorch code's behaviour. Now, obviously, this required yet another level of indirection in the learning-rate-getting function from hell: If you're keeping track, it's the in there. Heigh ho. So, it was time to run it again: That looked OK -- no change from before. Here's the code . Now, it was time to take the last step to finish the training loop: the ability to restart from a checkpoint. At this point, the checkpointing code was pretty basic -- it would save the model as a Safetensors file, along with some metadata like the min, max and average loss since the previous checkpoint, the number of the global step that we were on, and whether or not this was the best checkpoint (in terms of average training loss) so far. In order to restore from a checkpoint, we'd need more information. In the old PyTorch code, we needed three extra things on top of the model and the metadata: So that was the job: save the optimiser in , and then implement a so that we can restart from one. I could then try kicking off a training run, waiting for a bit, killing it, then restarting from the most recent checkpoint. The loss and learning rate charts would tell me whether or not the restart really had picked up from where it had left off. Initially I was thinking that I would just use pickle to save the optimiser, but that felt like a problem waiting to happen. Pickle has issues when you change Python versions or versions of installed packages, which never feels like it's going to be a problem, but all-too-frequently turns out to break stuff in reality. 2 Using Safetensors looked a bit tricky -- it had been hard to get it to work with Flax models, even though it had explicit support. Now, the recommended library for checkpointing in JAX code is called Orbax . I'd looked into it before, and it looked a bit heavyweight, so I'd moved on. But digging in a little more, I found that it had what looked like a simple API for saving PyTrees , which bypassed the complexity. Getting it working was still a bit tricky, though. Firstly, in the docs, they give this example: I tried that in the function with code like this: ...and got the error Huh. Digging into the library from the command line showed that the function was actually called . Not super-promising if the docs don't match the API (though to be fair, it does say right there in the package name). Anyway, changing that appeared to work: ...and then next to the 295 MB file called in my checkpoint directories, there was a 353 MB directory called . In PyTorch-land the optimiser had always been double the size of the model 3 , but given the wildly different file formats in play, I was comfortable enough that it was order-of-magnitude the same as the model and somewhat bigger. Perhaps Orbax was doing some kind of compression or something like that. Next, it was time to write . I started off by writing the function to load up the safetensors file -- that's the one I showed earlier, back when I showed how the original A-to-A model learned how to map a poem to itself, and that if you asked it how to complete "Every effort moves you", it would respond with " you you you you you" and so on. Once I had that, I created a , which called , and then loaded up the metadata and worked out what our best loss so far had been (which is necessary when continuing from a checkpoint so that, as you continue training, you can work out whether each new global step has had a loss that is better than the current best). That was simple enough: Restoring the optimiser turned out to be a bit trickier. Firstly, of course, just like with saving, the Orbax function was called rather than the documented . The next part was working out how to load it in a fashion that the optimiser would accept. If you load a checkpointed PyTree like this: Then what you get back is a "basic" PyTree -- it will consist of lists, dictionaries, tuples, basic Python types like strings, and JAX arrays. The problem is that the optimiser's state is formed of objects that can be mapped to such things -- for example, an object can be mapped to a dictionary where each field is an item in the dict -- but aren't actually those specific types of objects. So if you do this: ...you get an error, something like this: ...and likewise if you use the function I was using in the code: ...you'll get a slightly different but equally confusing error. After a certain amount of floundering around, limited by the lack of documentation (and it not seeming to match the API that I was seeing) I had the bright idea of looking at 's docstring, and that turned out to be excellent. In IPython: The solution was obviously that . When you provide it, it's used as a template. If in the abstract PyTree it finds a object, and in the loaded PyTree there is a dictionary in the same position with keys , and , it will create a object, setting those fields to those values. That means that you have something with the right structure to apply, so I wound up with this relatively simple code to load checkpoint into the optimiser: We're using the existing state of the optimiser as a template to tell Orbax how to structure the loaded one. I kicked off a training run, hit control-C halfway through, then restarted it from the checkpoint, and the final loss chart looked like this: ...and the learning rate chart like this: Perfect! The interrupt was at about global step 400, and the loss continued to go down properly, and the learning rate followed its schedule perfectly. Here's the checkpoint-loading code and the training script . So with that, phase one was done. I had a training script. It was massively overengineered for training this little A-to-A model, but just right for training a small LLM from scratch. And now it was time to do that -- and that's what I'll cover in the next post. If you're thinking "why not just have it return one-hot vectors based on the input tokens", remember that I needed something in the model to train, so that I could confirm that loss was going down. A pure "identity" model without the embedding space would have nothing to learn, so wouldn't be able to provide that.  ↩ It was a surprisingly large source of tech support queries on PythonAnywhere. Someone would train a model with (say) Python 3.11.1, and then try to run it on our servers using 3.11.2, and discover that they couldn't load up their checkpoints. This confused them and they wondered if it was something to do with our platform. I even had a quicktext response to send with a rundown on how Pickle works so that I didn't have to keep typing the same explanation. This may have biased me more against Pickle than I should rationally be.  ↩ AdamW stores two numbers per parameter to keep track of its optimisation state, so 2x the model size is exactly what you'd expect if both files were in the same format.  ↩ Flax NNX for neural network components. Optax for optimisation. Firstly, we convert them into embeddings, so we get a series of vectors. We do this by a lookup into a table, but we can see it conceptually as a projection via a matrix, from vocab space (where a particular token ID is a one-hot vector) to an embedding space. Next, we do the magic with our Transformers layers, getting embeddings for the next token. The embedding at position n in the output sequence, after these layers, is for the predicted token to come after the token at position n in the input sequence, considering that input token and all other tokens to its left. Finally, we project those back from embedding space to logits, this time actually using a real matrix (in the form of a linear layer). The logits (after being run through softmax) represent the probabilities for each token of it being the next one. The scaler that we used to do automated mixed-precision training. This JAX loop was not going to do that, so it was not necessary here. The learning rate scheduler. This was built into the optimiser for JAX, so I didn't think it was needed. The optimiser itself. This was important, and we definitely did need to save it. If you're thinking "why not just have it return one-hot vectors based on the input tokens", remember that I needed something in the model to train, so that I could confirm that loss was going down. A pure "identity" model without the embedding space would have nothing to learn, so wouldn't be able to provide that.  ↩ It was a surprisingly large source of tech support queries on PythonAnywhere. Someone would train a model with (say) Python 3.11.1, and then try to run it on our servers using 3.11.2, and discover that they couldn't load up their checkpoints. This confused them and they wondered if it was something to do with our platform. I even had a quicktext response to send with a rundown on how Pickle works so that I didn't have to keep typing the same explanation. This may have biased me more against Pickle than I should rationally be.  ↩ AdamW stores two numbers per parameter to keep track of its optimisation state, so 2x the model size is exactly what you'd expect if both files were in the same format.  ↩

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Giles's blog 3 months ago

Thoughts on Role Confusion

The other day, I came across " Prompt Injection as Role Confusion " ( via Simon Willison ). It's a really interesting blog-style version of a paper by Charles Ye, Jasmine Cui and Dylan Hadfield-Menell, where they find that LLMs seem to almost ignore 'role' tags like , or , and instead use the tone of text to infer roles. This seems to explain a lot of jailbreaks. When LLMs are reasoning about their context to work out what tokens they need to generate next, they need to separate out different things: what the system prompt says, what the user says, what the LLM itself has said in the past -- and for recent LLMs, what their own past thoughts have been -- their reasoning traces -- and what they've sent to and received from their tools. These "roles" for each bit of text need to be specified in the context. For example, in a simple chatbot (say, 2022-vintage), it might be written up a bit like a transcript : The LLM then starts predicting what would come next (eg. "The capital of France is Paris"). Alternatively, we might use XML-like separators: But most modern systems use special tokens -- which have the benefit that the things outside the LLM harness (like the user through the chat interface, or hostile tool output) can't fake them. In the post, they call the special inputs that tell the system how to interpret the role of a bit of text the role tags . But, after digging in with various tools, they find that LLMs seem to pay much more attention to the tone of text than they do to the actual role tags! So even if the special tagging tokens are unfakeable, that doesn't save your model from being jailbroken -- for example, by a user managing to trick the model so that even though something is tagged , it treats it as if it were tagged . They give a particularly fun example, which worked well on OpenAI's reasoning models in late 2025. They would simply provide text -- which would all go into a "user"-tagged role section -- that sounded like the kind of thing the models themselves would come up with in their reasoning trace: The model saw that, ignored that it was tagged "user", and treated it as its own thoughts. Because the model trusts its own thoughts, it happily complied. For example, they give this reply from GPT-5 Mini: A lot of jailbreaks I've seen ( Pliny the Liberator 's come to mind) seem to consist of putting in text that looks a bit like chain-of-thought reasoning or a system prompt. Perhaps this is (part of) how they work? It all sheds an interesting light on the prompt injection trick that I wrote about back in November , though. You can start a chat with an LLM with this message: ...and then when it accepts the challenge and says "go ahead", you reply with all of this in one message: In one quick test, even now in mid-2026, this still bamboozles ChatGPT 5.5, with thinking set to "High" -- it replied: My theory back in November was that it was related to the models' intelligence and their having been trained on instruction following. But this paper gives a more plausible and concrete way of thinking about it: if, internally in the LLM, it's using the phrasing as a way of guessing who is saying what, that might explain what is going on. However, I tried a variant of the second prompt where I tried to make the "bot" responses significantly less ChatGPT-like: ...and I still got So it still seems to have fallen for it. (It does seem a bit terser, but that might be random.) Perhaps the "User:" and "Bot:" tags -- even though they're not the real ones -- are pushing it hard enough that it overrides the tone. Or maybe we should treat them as "tone" in this case anyway, given that they are almost certainly not what ChatGPT is using to tag things. Or perhaps ChatGPT 5.5 with high thinking is just humouring me... Something I've been wondering for a while is whether this kind of thing could be fixed by somehow directly tagging the embeddings that are fed into the LLM. Role tags go around the tokens that they are tagging; these would be an inherent part of the tokens themselves, which might make it harder for the model to get confused. After all, the tag tokens are quite far from some of the text that they're tagging, and that signal needs to be pulled to the right by the different transformer layers, which are also trying to pull all kinds of other information rightwards. With the GPT-2 models I've been working on to date, the position of each token in the context is tagged by adding on a learned position embedding to the token-specific one -- that is, for "the fat cat sat on the mat", the first three embeddings would be: You can imagine that you could have an extra embedding that meant "role", and add it on in a similar way. I believe that BERT does this with what it calls segment embeddings . Alternatively -- and also inspired by position information, with the more current RoPE system -- you could rotate the embedding vectors about some axis to reflect their role. Or you could even add on one new dimension to the embeddings for each role, with a one for the real role, and zeros for the others. I guess a problem with all of these -- even if they worked in theory -- would be that in pre-training, you wouldn't have the roles correctly set. You could only add them on for the post-training phase -- and you could never be certain that something from the pre-training might "leak through" and make them ineffective. But certainly something to add to my ever-growing list of things to investigate. In particular, ASIDE looks like an interesting paper to look at -- it does something with rotation, though they're only trying to separate instructions from data rather than specifically to tag roles, and they're training from scratch with the separation in there. Given that jailbreaks are an unsolved problem, it's clearly somewhere where there's plenty left to be discovered. The token embedding for "the" plus the position embedding for position 1. The token embedding for "fat" plus the position embedding for position 2. The token embedding for "cat" plus the position embedding for position 3.

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Giles's blog 4 months ago

Flax debugging: making a hash of things

I was debugging an issue with a JAX/Flax NNX training loop the other day, and found a neat little trick to help debug it. Specifically, I wanted to see if the issue was with my model, my loss function, my optimiser settings, or the "plumbing" of the training loop itself -- were gradients actually coming through and being applied to the parameters? I could print out the loss and the gradients, but printing out the parameters to see if they were changing was unhelpful -- any given update might only change a small number of parameters, or might change them such a small amount that I'd not notice -- especially given that the model had 77 million of them! Let's take a look. I am building an LLM from scratch in JAX and Flax NNX, and at this stage I'm trying to get the training loop right. As a simple test, I've just implemented the "shell" of the LLM -- the token embeddings on the input side, and the final linear layer for an output head, wired directly together. My plan was to train that so that given a sequence, instead of predicting next tokens for each position, it would "predict" the sequence itself -- that is, I might train it with the input ...and the target ...rather than the normal setup for an LLM, where you feed it ...and give it targets of So, in LLM terms, I'd be training a model to project from vocab space to a learned embedding space where each token had a distinct-enough embedding for the output head to be able to reliably project back to logits in vocab space. There's a bit of background here if that was all Greek to you . Here's the core part of the code I was working with, the function, which seems to be the traditional JAX name for the JITted part of your code that does the forward pass through the model, works out the gradients, and then applies them to update the model: I'd based it on the "Basic Usage" example that's currently right there on the front page of the Flax site. Seasoned Flax veterans will probably spot the issue right away, but it wasn't obvious to me -- so it was time to dig in. The problem was that loss was not dropping -- indeed, taken to two decimal places, it was stuck at 10.82. The digits to the right of that changed for each batch, but the first four did not. Now, this model was using the GPT-2 tokeniser, and 10.82 is exactly the loss that you'd expect if the model was essentially guessing randomly -- if you convert it to perplexity by calculating e 10.82 , you get about 50,011 -- which is very close to the GPT-2 vocab size of 50,257. Perplexity is, loosely, the number of tokens that the model was trying to choose between for a typical input -- so a perplexity equal to the vocab size is what you'd expect of a random model that is getting it right about one in 50,257 times. That said, getting that loss consistently was a solid validation of my loss function! It's vanishingly unlikely that it would have been getting that specific number so consistently if I'd made a mess of that. The tiny variations I was seeing in the third and subsequent decimal places would make sense, as they could easily be due to the variations in the contents of the different batches. So was it that the gradients were somehow zero, or NaNs, or something else that couldn't be usefully applied to the model by the optimiser? I printed them out in the function (removing the decorator, as otherwise the s would only get executed in the initial JIT pass through the function to compile it -- not when it had actual data 1 ). The result was values like this: Those looked plausible enough -- pretty small, but not so tiny that I'd expect them to have no effect at all with my learning rate of 0.0014. It was time to dig into the training loop's plumbing. The obvious suspect was the update step -- was that call to actually changing the parameters at all? Flax's NNX API is a bit odd compared to the normal JAX functional way of doing things . In vanilla JAX code you would expect to do something like this to apply gradients: That is, you get the new parameters by applying a transformation to the old ones. NNX, by contrast, is more PyTorch-flavoured. It updates the parameters in-place, using a function with a side effect of mutating one of its parameters: ...rather than something more functional like this imaginary API: I could easily imagine that I'd got something wrong that would break that in-place update, as it has the feel of something that would have to be quite delicately implemented on top of a functional system like JAX. But how could I see whether the parameters were changing, when there were 77 million of them and they would be being updated (based on gradients like -2.6879393e-06 and a learning rate of 1.4e-3) in the ninth decimal place or beyond? Printing the arrays out was a non-starter! After a little thought, I realised that the solution was to use hashes. Even tiny changes in the parameters' values would change their hashes drastically. So if the parameters were not being updated, as I suspected, I'd see constant hashes. If they were being updated, even by a minuscule amount, then the hashes would change. This GitHub discussion pointed me in the right direction: if I could get the parameters as pure JAX arrays, I could do this: ...where is just . That would produce a hash that was stable for the life of this run -- the same parameters would always have the same hash, and different ones would differ, just as we want. It could vary from run to run (Python uses different hash seeds in each new interpreter), but that wouldn't matter for this kind of debugging. I wasn't sure what the structure of my Flax model's parameters was, but printing them out in the training loop told me: So, guided by that, I added these lines to the training loop: Obviously copying the arrays around and converting them like that would slow things down, but for debugging purposes, it looked solid. I kicked off the training loop, and the problem was clear: ...and so on. The hashes were not changing, so the model's parameters were not being updated, even by a tiny amount. Gotcha! The problem turned out, as I had suspected, to be related to the in-place updates that NNX does. Like I said earlier, I'd based my training loop on the "Basic Usage" example on the Flax site -- but I'd messed up one important thing. I had this: ...and they had this: You can see a number of differences -- for example, they're baking the inputs and targets into the lambda they're using for the loss function through a lexical closure, and that means that they're only passing in the model to the version of it wrapped in . But none of that matters! The real difference is actually nicely highlighted with a comment, but I'd completely managed to miss it. Right at the start, where I had , they had this: It 100% makes sense that in order to support this kind of non-functional, in-place updating of the model's parameters, you have to have a modified version of the JIT decorator. And I was just using the standard, functional pure-JAX one. Fixing that fixed the problem: The hashes were changing! And even better, if you scroll to the right you'll see that loss was slowly dropping. After 10k or so iterations, I was seeing 0.000: I had my do-nothing "LLM" working. A satisfying debugging journey -- and while I don't think I'll make this specific mistake in the future, I think that the parameter-hashing trick is actually a really useful trick for the toolbox. If you're uncertain as to whether your parameters are being updated, just looking at them probably won't help. But looking at their hashes can help you find out whether anything is changing. And I think that the pattern that I used to zoom in on it is a useful one, too. I always track loss, so it's a good starting point (indeed, seeing that it wasn't falling was what told me that something was going wrong). But checking that it has a sane -- or ideally, as in this case, a meaningful -- value is a nice sanity check that we have a working loss function and a model that isn't doing something completely pathological. Moving on from there to checking that some kind of gradients are flowing through is a solid next move (and might become increasingly interesting with deeper models where they can vanish or explode ). Then finally we can check the parameters -- in particular, are they changing? 2 Let's see how many new tricks I pick up as I work through this LLM project. I always forget that exists -- I could have used that instead, and kept the JIT.  ↩ Something's slightly broken in my brain and I keep reading that as "is our parameters changing" in George W. Bush's voice . Maybe I can stop that from happening by inflicting it on my readers instead. You're welcome.  ↩ I always forget that exists -- I could have used that instead, and kept the JIT.  ↩ Something's slightly broken in my brain and I keep reading that as "is our parameters changing" in George W. Bush's voice . Maybe I can stop that from happening by inflicting it on my readers instead. You're welcome.  ↩

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Giles's blog 4 months ago

10Gb/s Ethernet: switching to a Broadcom SFP+ module

Back in April , I upgraded my home LAN to 10Gb/s. The in-wall cabling is CAT-6 or similar, so I had to use 10GBASE-T. Now, the router I'm using, and the switch in my study, provide 10Gb/s through SFP+ cages; that meant that they needed 10GBASE-T SFP+ modules in order to connect. That kind of module is known to run hot -- sometimes too hot to actually work. The modules in , the router, appeared to be running OK (see the linked post above for charts), but the one in , the study switch, was a worrying 93C. I tried sticking some mini-heatsinks on it , which seemed to help a bit. But the weather got warmer, and eventually the module overheated. I lost access to the Internet from the study, and checking the metrics showed me this: You can see that it's "flapping": the temperature gets up to a level where the module shuts itself down for its own protection -- about 95C, I think -- and then when it has recovered, it switches on again, the temperature rises, and the process repeats. I was able to work around the problem by switching on the air conditioning in the study. But normally I only have it on when I'm in there, and keeping aircon on 24/7 just to keep the network working felt like the wrong solution. It was time to switch to a more power-efficient SFP+ module. My original 10Gb/s post had quite a lot of discussion on Hacker News , and mentioned that there are two generations of 10GBASE-T SFP+ modules: old ones using a Marvell chip, and newer ones using one from Broadcom. on the ServeTheHome forums made the same point. The Marvell-based ones were known to run hot, and they both recommended finding Broadcom-based ones. I'd confirmed that the MikroTik S+RJ10 that I had in was indeed a Marvell one, so the solution was pretty simple: get a better one. So I went on Amazon and picked up a 10Gtek ASF-10G-T80-INT . Checking 10Gtek's own page on that module confirmed that it used the right kind of chip (although it was a little bit garbled): 10Gtek's ASF-10G-T80 is a newest version copper transceiver, its biggest feature are ultra lowpower consumption and longer transmission distance (1.6W C10Gbps 30m,2.0W 110Gbps 80m). ASF-10G-T80 is a 10GBase mult-rate Copper RJ45 SFP+ transceiver, designed in with BROADCOM BCM84891 PHY chip following IEEE 802.3an/az and SFP+ MSA, supporting up to 80-meter transmission over CAT.6a or CAT.7. A day or two later, it arrived. It came in a rather pretty little metal case: Installing it took a little while, because I found removing the existing MikroTik module tricky; Willie Howe's video on YouTube helped quite a lot in showing how to disengage the latch, but I still needed to fiddle around with it quite a bit to get it out. However, that was eventually done, and the new module went in. I plugged all of the network cables back in, switched on the switch, and (after a slightly nerve-wracking wait for it to boot up) the network was back up and running! So, were the temperatures any better? I checked my monitoring, and: Huh, nothing was being reported. That made sense, though. The way I was charting those numbers was that the switch exposed them over SNMP, and then the Telegraf daemon on my router, , read the numbers and sent them to InfluxDB ; finally, Grafana did the charting. I'd been reading the module temperatures in using the SNMP OID that I'd identified that the switch was providing them on ( if you're interested), but perhaps the new module was published on a different OID. It was time to log in to the switch and take a look. It's saying that it's an Intel module; that in itself is not all that odd -- there are frequently compatibility issues between switches and SFP+ modules, so sometimes modules are configured to "lie" about which manufacturer made them -- and I'd specifically bought the "Intel-compatible" one on Amazon, the , because I couldn't find one that pretended to be MikroTik. Research had suggested that it would work OK, and it did. But the really odd bits were these: Not only was it impersonating an Intel module -- it was saying that it was a fibre-optic one ! Perhaps if I had found the "MikroTik-compatible" option it would have been better -- though, equally, it might have just impersonated a MikroTik fibre module anyway. Anyway, it was working -- so that was OK. But there was some bad news. If the switch was able to read a temperature from the new module, then you'd expect it to appear in that output, as . So, sadly, I don't think I'll be able to monitor the temperature of the new module. How could I tell whether it had helped, then? Well, one thing would be to simply see if there are any further instances of network flapping. I actually did the replacement just over two weeks ago, and everything has been fine as far as I can tell from using it and from the other monitoring (despite another hot week last week). But another interesting metric is the CPU temperature for over the two weeks before and after the module change: You can see that there was a real drop-off late on 1 June, when I switched the modules, and it has been running about 5C cooler since. Of course, there's a lot that's different about the new module -- as well as having a different chipset and a mendacious EEPROM, it's likely to have different thermal coupling characteristics -- it might be shedding more or less of its heat to the SFP+ cage and thence to the switch's CPU. So it's not proof of anything, but in combination with the improved link stability, I'll take it as a win. So, an interesting little excursion into the world of SFP+ modules -- in particular, slightly dodgy ones :-) Let's see if this one holds up better as we go through the toasty Lisbon summer.

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