Brain dump: JavaScript sandboxing

Another thing I’ve been researching is safe, sandboxed embedding of user-created JavaScript widgets… my last attempt in this direction was the EmbedScript extension (examples currently down, but code is still around).

User-level problems to solve:

  • “Content”
    • Diagrams, graphs, and maps would be more fun and educational if you could manipulate them more
    • What if you could graph those equations on all those math & physics articles?
  • Interactive programming sandboxes
  • Customizations to editor & reading UI features
    • Gadgets, site JS, shared user JS are potentially dangerous right now, requiring either admin review or review-it-yourself
    • Narrower interfaces and APIs could allow for easier sharing of tools that don’t require full script access to the root UI
  • Make scriptable extensions safer
    • Use same techniques to isolate scripts used for existing video, graphs/maps, etc?
    • Frame-based tool embedding + data injection could make export of rich interactive stuff as easy as InstantCommons…

Low-level problems to solve

  • Isolating user-provided script from main web context
  • Isolating user-provided script from outside world
    • loading off-site resources is a security issue
    • want to ensure that wiki resources are self-contained and won’t break if off-site dependencies change or are unavailable
  • Providing a consistent execution environment
    • browsers shift and change over time…
  • Communicating between safe and sandboxed environments
    • injecting parameters in safely?
    • two-way comms for allowing privileged operations like navigating page?
    • two-way comms for gadget/extension-like behavior?
    • how to arrange things like fullscreen zoom?
  • Potential offline issues
    • offline cacheability in browser?
    • how to use in Wikipedia mobile apps?
  • Third-party site issues
    • making our scripts usable on third-party wikis like InstantCommons
    • making it easy for third-party wikis to use these techniques internally

Meta-level problems to solve

  • How & how much to review code before letting it loose?
  • What new problems do we create in misuse/abuse vectors?

Isolating user-provided scripts

One way to isolate user-provided scripts is to run them in an interpreter! This is potentially very slow, but allows for all kinds of extra tricks.


I stumbled on JS-Interpreter, used sometimes with the Blockly project to step through code generated from visual blocks. JS-Interpreter implements a rough ES5 interpreter in native JS; it’s quite a bit slower than native (though some speedups are possible; the author and I have made some recent tweaks improving the interpreter loop) but is interesting because it allows single-stepping the interpreter, which opens up to a potential for an in-browser debugger. The project is under active development and could use a good regression test suite, if anyone wants to send some PRs. 🙂

The interpreter is also fairly small, weighing in around 24kb minified and gzipped.

The single-stepping interpreter design protects against infinite loops, as you can implement your own time limit around the step loop.

For pure-computation exercises and interactive prompts this might be really awesome, but the limited performance and lack of any built-in graphical display means it’s probably not great for hooking it up to an SVG to make it interactive. (Any APIs you add are your own responsibility, and security might be a concern for API design that does anything sensitive.)


An old project that’s still around is Google Caja, a heavyweight solution for embedding foreign HTML+JS using a server-side Java-based transpiler for the JS and JavaScript-side proxy objects that let you manipulate a subset of the DOM safely.

There are a number of security advisories in Caja’s history; some of them are transpiler failures which allow sandboxed code to directly access the raw DOM, others are failures in injected APIs that allow sandboxed code to directly access the raw DOM. Either way, it’s not something I’d want to inject directly into my main environment.

There’s no protection against loops or simple resource usage like exhausting memory.

Iframe isolation and CSP

I’ve looked at using cross-origin <iframe>s to isolate user code for some time, but was never quite happy with the results. Yes, the “same-origin policy” of HTML/JS means your code running in a cross-origin frame can’t touch your main site’s code or data, but that code is still able to load images, scripts, and other resources from other sites. That creates problems ranging from easy spamming to user information disclosure to simply breaking if required offsite resources change or disappear.

Content-Security-Policy to the rescue! Modern browsers can lock down things like network access using CSP directives on the iframe page.

CSP’s restrictions on loading resources still leaves an information disclosure in navigation — links or document.location can be used to navigate the frame to a URL on a third domain. This can be locked down with CSP’s childsrc param on the parent document — or an intermediate “double” iframe — to only allow the desired target domain (say, “*” or even “”). Then attempts to navigate the frame to a different domain from the inside are blocked.

So in principle we can have a rectangular region of the page with its own isolated HTML or SVG user interface, with its own isolated JavaScript running its own private DOM, with only the ability to access data and resources granted to it by being hosted on its private domain.

Further interactivity with the host page can be created by building on the postMessage API, including injecting additional resources or data sets. Note that postMessage is asynchronous, so you’re limited in simulating function calls to the host environment.

There is one big remaining security issue, which is that JS in an iframe can still block the UI for the whole page (or consume memory and other resources), either accidentally with an infinite loop or on purpose. The browser will eventually time out from a long loop and give you the chance to kill it, but it’s not pleasant (and might just be followed by another super-long loop!)

This means denial of service attacks against readers and editors are possible. “Autoplay” of unreviewed embedded widgets is still a bad idea for this reason.

Additionally, older browser versions don’t always support CSP — IE is a mess for instance. So defenses against cross-origin loads either need to somehow prevent loading in older browsers (poorer compatibility) or risk the information exposure (poorer security). However the most popular browsers do enforce it, so applications aren’t likely to be built that rely on off-site materials just to function, preventing which is one of our goals.

Worker isolation

There’s one more trick, just for fun, which is to run the isolated code in a Web Worker background thread. This would still allow resource consumption but would prevent infinite loops from blocking the parent page.

However you’re back to the interpreter’s problem of having no DOM or user interface, and must build a UI proxy of some kind.

Additionally, there are complications with running Workers in iframes, which is that if you apply sandbox=allow-scripts you may not be able to load JS into a Worker at all.

Non-JavaScript languages

Note that if you can run JavaScript, you can run just about anything thanks to emscripten. 😉 A cross-compiled Lua interpreter weighs in around 150-180kb gzipped (depending on library inclusion).

Big chart

Here, have a big chart I made for reference:

Offline considerations

In principle the embedding sites can be offline-cached… bears consideration.

App considerations

The iframes could be loaded in a webview in apps, though consider the offline + app issues!

Data model

A widget (or whatever you call it) would have one or more sub resources, like a Gadget does today plus more:

  • HTML or SVG backing document
  • JS/CSS module(s), probably with a dependency-loading system
  • possibly registration for images and other resources?
    • depending on implementation it may be necessary to inject images as blobs or some weird thing
  • for non-content stuff, some kind of registry for menu/tab setup, trigger events, etc

Widgets likely should be instantiable with input parameters like templates and Lua modules are; this would be useful for things like reusing common code with different input data, like showing a physics demo with different constant values.

There should be a human-manageable UI for editing and testing these things. 🙂 See jsfiddle etc for prior art.

How to build the iframe target site


  • Subdomain per instance
    • actually serve out the target resources on a second domain, each ‘widget instance’ living in a separate random subdomain ideally for best isolation
    • base HTML or SVG can load even if no JS. Is that good or bad, if interactivity was the goal?
    • If browser has no CSP support, the base HTML/CSS/JS might violate constraints.
    • can right-click and open frame in new window
    • …but now you have another out of context view of data, with new URLs. Consider legal, copyright, fairuse, blah blah
    • have to maintain and run that second domain and hook it up to your main wiki
    • how to deal with per-instance data input? Pre-publish? postMessage just that in?
      • injecting data over postMessage maybe best for the InstantCommons-style scenario, since sites can use our scripts but inject data
    • probably easier debugging based on URLs
  • Subdomain per service provider, inject resources and instance data
    • Inject all HTML/SVG/JS/CSS at runtime via postMessage (trusting the parent site origin). Images/media could either be injected as blobs or whitelisted by URL.
    • The service provider could potentially be just a static HTML file served with certain strict CSP headers.
    • If injecting all resources, then could use a common provider for third-party wikis.
      • third-party wikis could host their own scripts using this technique using our frame broker. not sure if this is good idea or not!
    • No separate content files to host, nothing to take down in case of legal issues.
    • Downside: right-clicking a frame to open it in new window won’t give useful resources. Possible workarounds with providing a link-back in a location hash.
    • Script can check against a user-agent blacklist before offering to load stuff.
    • Downside: CSP header may need to be ‘loose’ to allow script injection, so could open you back up to XSS on parameters. But you’re not able to access outside the frame so pssssh!

Abuse and evil possibilities

Even with the security guarantees of origin restrictions and CSP, there are new and exciting threat models…

Simple denial of service is easy — looping scripts in an iframe can lock up the main UI thread for the tab (or whole browser, depending on the browser) until it eventually times out with an error. At which point it can potentially go right back into a loop. Or you can allocate tons of memory, slowing down and eventually perhaps crashing the browser. Even tiny programs can have huge performance impact, and it’s hard to predict what will be problematic. Thus script on a page could make it hard for other editors and admins to get back in to fix the page… For this reason I would  recommend against autoplay in Wikipedia articles of arbitrary unreviewed code.

There’s also possible trolling patterns: hide a shock image in a data set or inside a seemingly safe image file, then display it in a scriptable widget bypassing existing image review.

Advanced widgets could do all kinds of fun and educational things like run emulators for old computer and game systems. That brings with it the potential for copyright issues with the software being run, or for newer systems patent issues with the system being emulated.

For that matter you could run programs that are covered under software patents, such as decoding or encoding certain video file formats. I guess you could try that in Lua modules too, but JS would allow you to play or save result files to disk directly from the browser.

WP:BEANS may apply to further thoughts on this road, beware. 😉

Ideas from Jupyter: frontend/backend separation

Going back to Jupyter/IPython as an inspiration source; Jupyter has a separation between a frontend that takes interactive input and displays output, and a backend kernel which runs the actual computation server-side. To make for fancier interactive displays, the output can have a widget which runs some sort of JavaScript component in the frontend notebook page’s environment, and can interact with the user (via HTML controls), with other widgets (via declared linkages) and with the kernel code (via events).

We could use a model like this which distinguishes between trusted (or semi-trusted) frontend widget code which can do anything it can do in its iframe, but must be either pre-reviewed, or opted into. Frontend widgets that pass review should have well-understood behavior, good documentation, stable interfaces for injecting data, etc.

The frontend widget can and should still be origin-isolated & CSP-restricted for policy enforcement even if code is reviewed — defense in depth is important!

Such widgets could either be invoked from a template or lua module with a fixed data set, or could be connected to untrusted backend code running in an even more restricted sandbox.

The two main ‘more restricted sandbox’ possibilities are to run an interpreter that handles loops safely and applies resource limits, or to run in a worker thread that doesn’t block the main UI and can be terminated after a timeout…. but even that may be able to exhaust system resources via memory allocation.

I think it would be very interesting to extend Jupyter in two specific ways:

  • iframe-sandboxing the widget implementations to make loading foreign-defined widgets safer
  • implementing a client-side kernel that runs JS or Lua code in an interpreter, or JS in a sandboxed Worker, instead of maintaining a server connection to a Python/etc kernel

It might actually be interesting to adopt, or at least learn from, the communication & linkage model for the Jupyter widgets (which is backbone.js-based, I believe) and consider the possibilities for declarative linkage of widgets to create controllable diagrams/visualizations from common parts.

An interpreter-based Jupyter/IPython kernel that works with the notebooks model could be interesting for code examples on Wikipedia, Wikibooks etc. Math potential as well.

Short-term takeaways

  • Interpreters look useful in niche areas, but native JS in iframe+CSP probably main target for interactive things.
  • “Content widgets” imply new abuse vectors & thus review mechanisms. Consider short-term concentration on other areas of use:
    • sandboxing big JS libraries already used in things like Maps/Graphs/TimedMediaHandler that have to handle user-provided input
    • opt-in Gadget/user-script tools that can adapt to a “plugin”-like model
    • making those things invocable cross-wiki, including to third-party sites
  • Start a conversation about content widgets.
    • Consider starting with strict-review-required.
    • Get someone to make the next generation ‘Graphs’ or whatever cool tool as one of these instead of a raw MW extension…?
    • …slowly plan world domination.

Brain dump: x86 emulation in WebAssembly

This is a quick brain dump of my recent musings on feasibility of a WebAssembly-based in-browser emulator for x86 and x86-64 processors… partially in the hopes of freeing up my brain for main project work. 😉

My big side project for some time has been ogv.js, an in-browser video player framework which uses emscripten to cross-compile C libraries for the codecs into JavaScript or, experimentally, the new WebAssembly target. That got me interested in how WebAssembly works at the low level, and how C/C++ programs work, and how we can mishmash them together in ways never intended by gods or humans.

Specifically, I’m thinking it would be fun to make an x86-64 Linux process-level emulator built around a WebAssembly implementation. This would let you load a native Linux executable into a web browser and run it, say, on your iPad. Slowly. 🙂

System vs process emulation

System emulators provide the functioning of an entire computer system, with emulated software-hardware interfaces: you load up a full kernel-mode operating system image which talks to the emulated hardware. This is what you use for playing old video games, or running an old or experimental operating system. This can require emulating lots of detail behavior of a system, which might be tricky or slow, and programs may not integrate with a surrounding environment well because they live in a tiny computer within a computer.

Process emulators work at the level of a single user-mode process, which means you only have to emulate up to the system call layer. Older Mac users may remember their shiny new Intel Macs running old PowerPC applications through the “Rosetta” emulator for instance. QEMU on Linux can be set up to handle similar cross-arch emulated execution, for testing or to make some cross-compilation scenarios easier.

A process emulator has some attraction because the model is simpler inside the process… If you don’t have to handle interrupts and task switches, you can run more instructions together in a row; elide some state changes; all kinds of fun things. You might not have to implement indirect page tables for memory access. You might even be able to get away with modeling some function calls as function calls, and loops as loops!

WebAssembly instances and Linux processes

There are many similarities, which is no coincidence as WebAssembly is designed to run C/C++ programs similarly to how they work in Linux/Unix or Windows while being shoehornable into a JavaScript virtual machine. 🙂

An instantiated WebAssembly module has a “linear memory” (a contiguous block of memory addressable via byte indexing), analogous to the address space of a Linux process. You can read and write int and float values of various sizes anywhere you like, and interpretation of bytewise data is up to you.

Like a native process, the module can request more memory from the environment, which will be placed at the end. (“grow_memory” operator somewhat analogous to Linux “brk” syscall, or some usages of “mmap”.) Unlike a native process, usable memory always starts at 0 (so you can dereference a NULL pointer!) and there’s no way to have a “sparse” address space by mapping things to arbitrary locations.

The module can also have “global variables” which live outside this address space — but they cannot be dynamically indexed, so you cannot have arrays or any dynamic structures there. In WebAssembly built via emscripten, globals are used only for some special linking structures because they don’t quite map to any C/C++ construct, but hand-written code can use them freely.

The biggest difference from native processes is that WebAssembly code doesn’t live in the linear memory space. Function definitions have their own linear index space (which can’t be dynamically indexed: references are fixed at compile time), plus there’s a “table” of indirect function references (which can be dynamically indexed into). Function pointers in WebAssembly thus aren’t actually pointers to the instructions in linear memory like on native — they’re indexes into the table of dynamic function references.

Likewise, the call stack and local variables live outside linear memory. (Note that C/C++ code built with emscripten will maintain its own parallel stack in linear memory in order to provide arrays, variables that have pointers taken to them, etc.)

WebAssembly’s actual opcodes are oriented as a stack machine, which is meant to be easy to verify and compile into more efficient register-based code at runtime.

Branching and control flow

In WebAssembly control flow is limited, with one-way branches possible only to a containing block (i.e. breaking out of a loop). Subroutine calls are only to defined functions (either directly by compile-time reference, or indirectly via the function table)

Control flow is probably the hardest thing to make really match up from native code — which lets you jump to any instruction in memory from any other — to compiled WebAssembly.

It’s easy enough to handle craaaazy native branching in an interpreter loop. Pseudocode:

loop {
instruction = decode_instruction(ip)
instruction.execute() // update ip and any registers, etc

In that case, a JMP or CALL or whatever just updates the instruction pointer when you execute it, and you continue on your merry way from the new position.

But what if we wanted to eke more performance out of it by compiling multiple instructions into a single function? That lets us elide unnecessary state changes (updating instruction pointers, registers, flags, etc when they’re immediately overridden) and may even give opportunity to let the compiler re-optimize things further.

A start is to combine runs of instructions that end in a branch or system call (QEMU calls them “translation units”) into a compiled function, then call those in the loop instead of individual instructions:

loop {
tu = cached_or_compiled_tu(ip)
tu.execute() // update registers, ip, etc as we go

So instead of decoding and executing an instruction at a time, we’re decoding several instructions, compiling a new function that runs them, and then running that. Nice, if we have to run it multiple times! But…. possibly not worth as much as we want, since a lot of those instruction runs will be really short, and there’ll be function call overhead on every run. And, it seems like it would kill CPU branch prediction and such, by essentially moving all branches to a single place (the tu.execute()).

QEMU goes further in its dynamic translation emulators, modifying the TUs to branch directly to each other in runtime discovery. It’s all very funky and scary looking…

But QEMU’s technique of modifying trampolines in the live code won’t work as we can’t modify running code to insert jump instructions… and even if we could, there are no one-way jumps, and using call instructions risks exploding the call stack on what’s actually a loop (there’s no proper tail call optimization in WebAssembly).


What can be done, though, is to compile bigger, better, badder functions.

When emscripten is generating JavaScript or WebAssembly from your C/C++ program’s LLVM intermediate language, it tries to reconstruct high-level control structures within each function from a more limited soup of local branches. These then get re-compiled back into branch soup by the JIT compiler, but efficiently. 😉

The binaryen WebAssembly code gen library provides this “relooper” algorithm too: you pass in blocks of instructions, possible branches, and the conditions around them, and it’ll spit out some nicer branch structure if possible, or an ugly one if not.

I’m pretty sure it should be possible to take a detected loop cycle of separate TUs and create a combined TU that’s been “relooped” in a way that it is more efficient.

BBBBuuuuutttttt all this sounds expensive in terms of setup. Might want to hold off on any compilation until a loop cycle is detected, for instance, and just let the interpreter roll on one-off code.

Modifying runtime code in WebAssembly

Code is not addressable or modifiable within a live module instance; unlike in native code you can’t just write instructions into memory and jump to the pointer.

In fact, you can’t actually add code to a WebAssembly module. So how are we going to add our functions at runtime? There are two tricks:

First, multiple module instances can use the same linear memory buffer.

Second, the tables for indirect function calls can list “foreign” functions, such as JavaScript functions or WebAssembly functions from a totally unrelated module. And those tables are modifiable at runtime (from the JavaScript side of the border).

These can be used to do full-on dynamic linking of libraries, but all we really need is to be able to add a new function that can be indirect-called, which will run the compiled version of some number of instructions (perhaps even looping natively!) and then return back to the main emulator runtime when it reaches a branch it doesn’t contain.

Function calls

Since x86 has a nice handy CALL instruction, and doesn’t just rely on convention, it could be possible to model calls to already-cached TUs as indirect function calls, which may perform better than exiting out to the loop and coming back in. But they’d probably need to be guarded for early exit, for several reasons… if we haven’t compiled the entirety of the relooped code path from start to exit of the function, then we have to exit back out. A guard check on IP and early-return should be able to do that in a fairly sane way.

function tu_1234() {
// loop
do {
// calc loop condition -> set zero_flag
ip = 1235
if !zero_flag {
ip = 1236
// CALL 4567
tu = cached_or_compiled_tu(4567)
if ip != 1236 {
// only partway through. back to emulator loop,
// possibly unwinding a long stack 🙂
// more code

I think this makes some kind of sense. But if we’re decoding instructions + creating output on the fly, it could take a few iterations through to produce a full compiled set, and exiting a loop early might be … ugly.

It’s possible that all this is a horrible pipe dream, or would perform too bad for JIT compilation anyway.

But it could still be fun for ahead-of-time compilation. 😉 Which is complicated… a lot … by the fact that you don’t have the positions of all functions known ahead of time. Plus, if there’s dynamic linking or JIT compilation inside the process, well, none of that’s even present ahead of time.

Prior art: v86

I’ve been looking at lot at v86, a JavaScript-based x86 system emulator. v86 is a straight-up interpreter, with instruction decoding and execution mixed together a bit, but it feels fairly straightforwardly written and easy to follow when I look at things in the code.

v86 uses a set of aliased typed arrays for the system memory, another set for the register file, and then some variables/properties for misc flags and things.

Some quick notes:

  • a register file in an array means accesses at difference sizes are easy (al vs ax vs eax), and you can easily index into it from the operand selector bits from the instruction (as opposed to using a variable per register)
  • is there overhead from all the object property accesses etc? would it be more efficient to do everything within a big linear memory?
  • as a system emulator there’s some extra overhead to things like protected mode memory accesses (page tables! who knows what!) that could be avoided on a per-process model
  • 64-bit emulation would be hard in JavaScript due to lack of 64-bit integers (argh!)
  • as an interpreter, instruction decode overhead is repeated during loops!
  • to avoid expensive calculations of the flags register bits, most arithmetic operations that would change the flags instead save the inputs for the flag calculations, which get done on demand. This still is often redundant because flags may get immediately rewritten by the next instruction, but is cheaper than actually calculating them.

WebAssembly possibilities

First, since WebAssembly supports only one linear memory buffer at a time, the register file and perhaps some other data would need to live there. Most likely want a layout with the register file and other data at the beginning of memory, with the rest of memory after a fixed point belonging to the emulated process.

Putting all the emulator’s non-RAM state in the beginning means a process emulator can request more memory on demand via Linux ‘brk’ syscall, which would be implemented via the ‘grow_memory’ operator.

64-bit math

WebAssembly supports 64-bit integer memory accesses and arithmetic, unlike JavaScript! The only limitation is that you can’t (yet) export a function that returns or accepts an i64 to or from JavaScript-land. That means if we keep our opcode implementations in WebAssembly functions, they can efficiently handle 64-bit ops.

However WebAssembly’s initial version allows only 32-bit memory addressing. This may not be a huge problem for emulating 64-bit processes that don’t grow that large, though, as long as the executable doesn’t need to be loaded at a specific address (which would mean a sparse address space).

Sparse address spaces could be emulated with indirection into a “real” memory that’s in a sub-4GB space, which would be needed for a system emulator anyway.

Linux details

Statically linked ELF binaries would be easiest to model. More complex to do dynamic linking, need to pass a bundle of files in and do fix-ups etc.

Questions: are executables normally PIC as well as libraries, or do they want a default load address? (Which would break the direct-memory-access model and require some indirection for sparse address space.)

Answer: normally Linux x86_64 executables are not PIE, and want to be loaded at 0x400000 or maybe some other random place. D’oh! But… in the common case, you could simplify that as a single offset.

Syscall on 32-bit is ‘int $80’, or ‘syscall’ instruction on 64-bit. Syscalls would probably mostly need to be implemented on the JS side, poking at the memory and registers of the emulated process state and then returning.

To do network i/o would probably need to be able to block and return to the emulator… so like a function call bowing out early due to an uncompiled branch being taken, would potentially need an “early exit” from the middle of a combined TU if it does a syscall that ends up being async. On the other hand, if a syscall can be done sync, might be nice not to pay that penalty.

Could also need async syscalls for multi-process stuff via web workers… anything that must call back to main thread would need to do async.

For 64-bit, JS code would have to …. painfully … deal with 32-bit half-words. Awesome. 😉


WebAssembly initial version has no facility for multiple threads accessing the same memory, which means no threads. However this is planned to come in future…

Processes with separate address spaces could be implemented by putting each process emulator in a Web Worker, and having them communicate via messages sent to the main thread through syscalls. This forces any syscall that might need global state to be async.

Prior art: Browsix

Browsix provides a POSIX-like environment based around web techs, with processes modeled in Web Workers and syscalls done via async messages. (C/C++ programs can be compiled to work in Browsix with a modified emscripten.) Pretty sweet ideas. 🙂

I know they’re working on WebAssembly processes as well, and were looking into synchronous syscalls vi SharedArrayBuffer/Atomics as well, so this might be an interesting area to watch.

Could it be possible to make a Linux binary loader for the Browsix kernel? Maybe!

Would it be possible to make graphical Linux binaries work, with some kind of JS X11 or Wayland server? …mmmmmmaaaaybe? 😀

Closing thoughts

This all sounds like tons of fun, but may have no use other than learning a lot about some low-level tech that’s interesting.

ogv.js 1.4.0 released

ogv.js 1.4.0 is now released, with a .zip build or via npm. Will try to push it to Wikimedia next week or so.

Live demo available as always.

New A/V sync

Main improvement is much smoother performance on slower machines, mainly from changing the A/V sync method to prioritize audio smoothness, based on recommendations I’d received from video engineers at conferences that choppy audio is noticed by users much more strongly than choppy or out of sync video.

Previously, when ogv.js playback detected that video was getting behind audio, it would halt audio until the video caught up. This played all audio, and showed all frames, but could be very choppy if performance wasn’t good (such as in Internet Explorer 11 on an old PC!)

The new sync method instead keeps audio rock-solid, and allows video to get behind a little… if the video catches back up within a few frames, chances are the user won’t even notice. If it stays behind, we look ahead for the next keyframe… when the audio reaches that point, any remaining late frames are dropped. Suddenly we find ourselves back in sync, usually with not a single discontinuity in the audio track.


The HTMLMediaElement API supports a fastSeek() method which is supposed to seek to the nearest keyframe before the request time, thus getting back to playback faster than a precise seek via setting the currentTime property.

Previously this was stubbed out with a slow precise seek; now it is actually fast. This enables a much better “scrubbing” experience given a suitable control widget, as can be seen in the demo by grabbing the progress thumb and moving it around the bar.

VP9 playback

WebM videos using the newer, more advanced VP9 codec can use a lot less bandwidth than VP8 or Theora videos, making it attractive for streaming uses. A VP9 decoder is now included for WebM, initially supporting profile 0 only (other profiles may or may not explode) — that means 8-bit, 4:2:0 subsampling.

Other subsampling formats will be supported in future, can probably eventually figure out something to do with 10-bit, but don’t expect those to perform well. 🙂

The VP9 decoder is moderately slower than the VP8 decoder for equivalent files.

Note that WebM is still slightly experimental; the next version of ogv.js will make further improvements and enable it by default.


Firefox and Chrome have recently shipped support for code modules in the WebAssembly format, which provides a more efficient binary encoding for cross-compiled code than JavaScript. Experimental wasm versions are now included, but not yet used by default.

Multithreaded video decoding

Safari 10.1 has shipped support for the SharedArrayBuffer and Atomics APIs which allows for fully multithreaded code to be produced from the emscripten cross-compiler.

Experimental multithreaded versions of the VP8 and VP9 decoders are included, which can use up to 4 CPU cores to significantly increase speed on suitably encoded files (using the -slices option in ffmpeg for VP8, or -tile_columns for VP9). This works reasonably well in Safari and Chrome on Mac or Windows desktops; there are performance problems in Firefox due to deoptimization of the multithreaded code.

This actually works in iOS 10.3 as well — however Safari on iOS seems to aggressively limit how much code can be compiled in a single web page, and the multithreading means there’s more code and it’s copied across multiple threads, leading to often much worse behavior as the code can end up running without optimization.

Future versions of WebAssembly should bring multithreading there as well, and likely with better performance characteristics regarding code compilation.

Note that existing WebM transcodes on Wikimedia Commons do not include the suitable options for multithreading, but this will be enabled on future builds.

Misc fixes

Various bits. Check out the readme and stuff. 🙂

What’s next for ogv.js?

Plans for future include:

  • replace the emscripten’d nestegg demuxer with Brian Parra’s jswebm
  • fix the scaling of non-exact display dimensions on Windows w/ WebGL
  • enable WebM by default
  • use wasm by default when available
  • clean up internal interfaces to…
  • …create official plugin API for demuxers & decoders
  • split the demo harness & control bar to separate packages
  • split the decoder modules out to separate packages
  • Media Source Extensions-alike API for DASH support…

Those’ll take some time to get all done and I’ve got plenty else on my plate, so it’ll probably come in several smaller versions over the next months. 🙂

I really want to get a plugin interface so people who want/need them and worry less about the licensing than me can make plugins for other codecs! And to make it easier to test Brian Parra’s jsvpx hand-ported VP8 decoder.

An MSE API will be the final ‘holy grail’ piece of the puzzle toward moving Wikimedia Commons’ video playback to adaptive streaming using WebM VP8  and/or VP9, with full native support in most browsers but still working with ogv.js in Safari, IE, and Edge.

Testing in-browser video transcoding with MediaRecorder

A few months ago I made a quick test transcoding video from MP4 (or whatever else the browser can play) into WebM using the in-browser MediaRecorder API.

I’ve updated it to work in Chrome, using a <canvas> element as an intermediary recording surface as captureStream() isn’t available on <video> elements yet there.

Live demo:

There are a couple advantages of re-encoding a file this way versus trying to do all the encoding in JavaScript, but also some disadvantages…


  • actual encoding should use much less CPU than JavaScript cross-compile
  • less code to maintain!
  • don’t have to jump through hoops to get at raw video or audio data


  • MediaRecorder is realtime-oriented:
    • will never decode or encode faster than realtime
    • if encoding is slower than realtime, lots of frames are dropped
    • on my MacBook Pro, realtime encoding tops out around 720p30, but eg phone camera videos will often be 1080p30 these days.
  • browser must actually support WebM encoding or it won’t work (eg, won’t work in Edge unless they add it in future, and no support at all in Safari)
  • Firefox and Chrome both seem to be missing Vorbis audio recording needed for base-level WebM (but do let you mix Opus with VP8, which works…)

So to get frame-rate-accurate transcoding, and to support higher resolutions, it may be necessary to jump through further hoops and try JS encoding.

I know this can be done — there are some projects compiling the entire ffmpeg  package in emscripten and wrapping it in a converter tool — but we’d have to avoid shipping an H.264 or AAC decoder for patent reasons.

So we’d have to draw the source <video> to a <canvas>, pull the RGB bits out, convert to YUV, and run through lower-level encoding and muxing… oh did I forget to mention audio? Audio data can be pulled via Web Audio, but only in realtime.

So it may be necessary to do separate audio (realtime) and video (non-realtime) capture/encode passes, then combine into a muxed stream.

Canvas, Web Audio, MediaStream oh my!

I’ve often wished that for ogv.js I could send my raw video and audio output directly to a “real” <video> element for rendering instead of drawing on a <canvas> and playing sound separately to a Web Audio context.

In particular, things I want:

  • Not having to convert YUV to RGB myself
  • Not having to replicate the behavior of a <video> element’s sizing!
  • The warm fuzzy feeling of semantic correctness
  • Making use of browser extensions like control buttons for an active video element
  • Being able to use browser extensions like sending output to ChromeCast or AirPlay
  • Disabling screen dimming/lock during playback

This last is especially important for videos of non-trivial length, especially on phones which often have very aggressive screen dimming timeouts.

Well, in some browsers (Chrome and Firefox) now you can do at least some of this. 🙂

I’ve done a quick experiment using the <canvas> element’s captureStream() method to capture the video output — plus a capture node on the Web Audio graph — combining the two separate streams into a single MediaStream, and then piping that into a <video> for playback. Still have to do YUV to RGB conversion myself, but final output goes into an honest-to-gosh <video> element.

To my great pleasure it works! Though in Firefox I have some flickering that may be a bug, I’ll have to track it down.

Some issues:

  • Flickering on Firefox. Might just be my GPU, might be something else.
  • The <video> doesn’t have insight to things like duration, seeking, etc, so can’t rely on native controls or API of the <video> alone acting like a native <video> with a file source.
  • Pretty sure there are inefficiencies. Have not tested performance or checked if there’s double YUV->RGB->YUV->RGB going on.

Of course, Chrome and Firefox are the browsers I don’t need ogv.js for for Wikipedia’s current usage, since they play WebM and Ogg natively already. But if Safari and Edge adopt the necessary interfaces and WebRTC-related infrastructure for MediaStreams, it might become possible to use Safari’s full screen view, AirPlay mirroring, and picture-in-picture with ogv.js-driven playback of Ogg, WebM, and potentially other custom or legacy or niche formats.

Unfortunately I can’t test whether casting to a ChromeCast works in Chrome as I’m traveling and don’t have one handy just now. Hoping to find out soon! 😀

JavaScript async/await fiddling

I’ve been fiddling with using ECMAScript 2015 (“ES6”) in rewriting some internals for ogv.js, both in order to make use of the Promise pattern for asynchronous code (to reduce “callback hell”) and to get cleaner-looking code with the newer class definitions, arrow functions, etc.

To do that, I’ll need to use babel to convert the code to the older ES5 version to run in older browsers like Internet Explorer and old Safari releases… so why not go one step farther and use new language features like asynchronous functions that are pretty solidly specced but still being implemented natively?

Not yet 100% sure; I like the slightly cleaner code I can get, but we’ll see how it functions once translated…

Here’s an example of an in-progress function from my buffering HTTP streaming abstraction, currently being rewritten to use Promises and support a more flexible internal API that’ll be friendlier to the demuxers and seek operations.

I have three versions of the function: one using provisional ES2017 async/await, one using ES2015 Promises directly, and one written in ES5 assuming a polyfill of ES2015’s Promise class. See the full files or the highlights of ES2017 vs ES2015:

The first big difference is that we don’t have to start with the “new Promise((resolve,reject) => {…})” wrapper. Declaring the function as async is enough.

Then we do some synchronous setup which is the same:

Now things get different, as we perform one or two asynchronous sub-operations:

In my opinion the async/await code is cleaner:

First it doesn’t have as much extra “line noise” from parentheses and arrows.

Second, I can use a try/finally block to do the final state change only once instead of on both .then() and .catch(). Many promise libraries will provide an .always() or something but it’s not standard.

Third, I don’t have to mentally think about what the “intermittent return” means in the .then() handler after the triggerDownload call:

Here, returning a promise means that that function gets executed before moving on to the next .then() handler and resolving the outer promise, whereas not returning anything means immediate resolution of the outer promise. It ain’t clear to me without thinking about it every time I see it…

Whereas the async/await version:

makes it clear with the “await” keyword what’s going on.

Updated: I managed to get babel up and running; here’s a github gist with expanded versions after translation to ES5. The ES5 original is unchanged; the ES2015 promise version is very slightly more verbose, and the ES2017 version becomes a state machine monstrosity. 😉 Not sure if this is ideal, but it should preserve the semantics desired.

Scaling video playback on slow and fast CPUs in ogv.js

Video playback has different performance challenges at different scales, and mobile devices are a great place to see that in action. Nowhere is this more evident than in the iPhone/iPad lineup, where the same iOS 9.3 runs across several years worth of models with a huge variance in CPU speeds…

In ogv.js 1.1.2 I’ve got the threading using up to 3 threads at maximum utilization (iOS devices so far have only 2 cores): main thread, video decode thread, and audio decode thread. Handling of the decoded frames or audio packets is serialized through the main thread, where the player logic drives the demuxer, audio output, and frame blitting.

On the latest iPad Pro 9.7″, advertising “desktop-class performance”, I can play back the Blender sci-fi short Tears of Steel comfortably at 1080p24 in Ogg Theora:

The performance graph shows frames consistently on time (blue line is near the red target line) and a fair amount of headroom on the video decode thread (cyan) with a tiny amount of time spent on the audio thread (green) and main thread (black).

At this and higher resolutions, everything is dominated by video decode time — if we can keep up with it we’re golden, but if we get behind everything would ssllooww ddoownn badly.

On an iPad Air, two models behind, we get similar performance on the 720p24 version, at about half the pixels:

We can see the blue bars jumping up once a second, indicating sensitivity to the timing report and graph being updated once a second on the main thread, but overall still good. Audio in green is slightly higher but still ignorable.

On a much older iPad 3, another two models behind, we see a very different graph as we play back a mere 240p24 quarter-SD resolution file:

The iPad 3 has an older generation, 32-bit processor, and is in general pretty sluggish. Even at a low resolution, we have less headroom for the cyan bars of the video decode thread. Blue bars dipping below the red target line show we’re slipping on A/V sync sometimes. The green bars are much higher, indicating the audio decode thread is churning a lot harder to keep our buffers filled. Last but not least the gray bars at the bottom indicate more time spent in demuxing, drawing, etc on the main thread.

On this much slower processor, pushing audio decoding to another core makes a significant impact, saving an average of several milliseconds per frame by letting it overlap with video decoding.

The gray spikes from the main thread are from the demuxer, and after investigation turn out to be inflated by per-packet overhead on the tiny Vorbis audio packets… Such as adding timestamps to many of the packets. Ogg packs multiple small packets together into a single “page”, with only the final packet at the end of the page actually carrying a timestamp. Currently I’m using liboggz to encapsulate the demuxing, using its option to automatically calculate the missing timestamp deltas from header data in the packets… But this means every few frames the demuxer suddenly releases a burst of tiny packets with a 15-50ms delay on the main thread as it walks through them. On the slow end this can push a nearly late frame into late territory.

I may have further optimizations to make in keeping the main thread clear on slower CPUs, such as more efficient handling of download progress events, but overlapping the video and audio decode threads helps a lot.

On other machines like slow Windows boxes with blacklisted graphics drivers, we also benefit from firing off the next video decode before drawing the current frame — if WebGL is unexpectedly slow, or we fall back to CPU drawing, it may take a significant portion of our frame budget just to paint. Sending data down to the decode thread first means it’s more likely that the drawing won’t actually slow us down as much. This works wonders on a slow ARM-based Windows RT 8.1 Surface tablet. 🙂


Thoughts on Ogg adaptive streaming

So I’d like to use adaptive streaming for video playback on Wikipedia and Wikimedia Commons, automatically selecting the appropriate source format and resolution at runtime based on bandwidth and CPU availability.

For Safari, Edge, and IE users, that means figuring out how to rig a Media Source Extensions-like interface into ogv.js to let the streaming handler inject its buffered data into the demuxer and codecs instead of letting the player handle its own buffering.

It also means I have to figure out how to do adaptive stream switching for Ogg streams and Theora video, since WebM VP8 still decodes too slowly in ogv.js to rely on for deployment…

Theory vs Theora

At its base, adaptive streaming relies on the ability to feed the decoders with data from another stream without them freaking out and demanding a pause or reset. We can either read packets from a subset of a monolithic file for each source, or from a bunch of tiny segmented files.

In order to do this, generally you need to switch on video keyframe boundaries: each keyframe represents a point in the data stream where the video decoder can reset its state.

For WebM with VP8 and VP9 codecs, the decoders are pretty good at this. As long as you came in on a keyframe boundary you can just start feeding it packets at a new resolution and it’ll happily output frames at the new resolution.

For Ogg Theora, there are a few major impediments.

Ogg stream serial numbers

At the Ogg stream level: each Ogg logical bitstream gets a random serial number; those serial numbers will not match across separate encodings at different resolutions.

Ogg explicitly allows for “chaining” of complete bitstreams, where one ends and you just tack another on, but we’re not quite doing that here… We want to be able to switch partway through with minimal interruption.

For Vorbis audio, this might require some work if pulling audio+video together from combined .ogv files, but it gets simpler if there’s one .oga audio stream and separate video-only .ogv streams — we’d essentially have separate demuxer contexts for audio and video, and would not need to meddle with the audio.

For the Theora video stream this is probably ok too, since when we reach a switch boundary we also need to feed the decoder with…

Header packets

Every Theora video stream sets up start codes at the beginning of the stream in its three header packets. This means that encodings of the same video at different resolutions will have different header setup.

So, when we switch sources we’ll need to reinitialize the Theora decoder with the header packets from the target stream; then it should be safe to feed new packets into it from our arbitrary start position.

This isn’t a super exotic requirement; I’ve seen some provision for ‘start codes’ for MP4 adaptive streaming too.

Keyframe timing

More worrisome is that keyframe timing is not predictable in a Theora stream. This is actually due to the libtheora encoder internals — it allows you to specify a maximum keyframe interval, but it may decide at any time to insert a keyframe on its own if it thinks it’s more efficient to store a frame that way, at which point the interval starts counting from there instead of the last scheduled keyframe.

Since this heuristic is determined based on actual frame data, the early keyframes will appear in different times and places for renderings at different resolutions… And so will every keyframe following them.

This means you don’t have switch points that are consistent between sources, breaking the whole model!

It looks like a keyframe can be forced by changing the keyframe interval to 1 right before a desire keyframe, then changing it back to the desired value after. This would result in still getting some early keyframes at unpredictable times, but then also getting predictable ones. As long as the switchover points aren’t too often, that’s probably fine — just keep decoding over the extra keyframes, but only switch/segment on the predictable ones.

Streams vs split files

Another note: it’s possible to either store data as one long file per source, or to split it up into small chunk files at each keyframe boundary.

Chunk files are nice because they can be streamed easily without use of the HTTP ‘Range’ header and they’re friendly to cache layers. Long files can be easier to manage on the server, but Wikimedia ops folks have told me that the way large files are stored doesn’t always interact ideally with the caching layer and they’d be much happier with split chunk files!

A downside of chunks is that it’s harder to download a complete copy of a file at a given resolution for offline playback. But, if we split audio and video tracks we’re in a world where that’s hard anyway… Can either just say “download the full resolution source then!” Or provide a remuxer to produce combined files for download on the fly from the chunks… 🙂
The keyframe timing seems the ugliest issue to deal with; may need to patch ffmpeg2theora or ffmpeg to work around it, but at least shouldn’t have to mess with libtheora itself…

WebM seeking coming in ogv.js 1.1.2

Seeking in WebM playback will finally be supported in ogv.js 1.1.2. Yay! Try it out!

I seeked in this WebM file! Yeah really!

This took some fancy footwork, as the demuxer library I’m using (nestegg) only provides a synchronous i/o-using interface for seeking: on the first seek, it needed to be able to first do a seek to the location of the cues in the file (they’re usually at the end in WebM, not at the beginning!), then read in the cue data, and then run a second seek to the start of a cluster.

On examining the innards of the library I found that it’s fairly safe to ‘restart’ the operation directly after the seek attempts, which saved me the trouble of patching the library code; though I may come up with a patch to more cleanly & reliably do this.

For the initial hack, I have my i/o callbacks detect that an attempt was made to seek outside the buffer range, and then when the nestegg library function fails out, the demuxer sees the seek attempt and passes it back to the caller (OGVPlayer) which is able to trigger a seek on the actual, asynchronous i/o layer. Once the new data starts arriving, we call back into the demuxer to read the cues or confirm that we’ve seeked to the right place and can continue decoding. As an additional protection against the library freaking out, I make sure that the entire cues element has been buffered before attempting to read it.

I’ve also fixed a bug that caused WebM playback to occasionally die with an out of memory error. This one was caused by not freeing packet data in the demuxer. *headdesk* At least that bug was easier. 😉

This gets WebM VP8 playback working about as well as Ogg Theora playback, except for the decoding being about 5x slower! Well, I’ve got plans for that. 😀