wasp-hls v0.4.2

Wasp-hls is an HLS media player (the library, streaming engine part, not the UI/application part which has to be built on top of it) for the web which:

1. Relies the most possible on WebAssembly (Written in the Rust language before being compiled).

2. Runs mostly in a Web Worker (even for media buffering when APIs are available), to reduce the influence an heavy UI can have on playback (and in some situations vice-versa).

Note that this is only a personal project as well as a proof of concept and it is still heavily in development.

What's this exactly? HLS?

Streaming protocols and HLS

To provide media contents at large scale to their customers, most streaming actors (Netflix, Amazon Prime Video, YouTube, Twitch, Canal+, Disney+ etc. you got it), rely on the few same HTTP-based adaptive bitrate streaming protocols: majoritarily MPEG-DASH and Apple's HLS (some rely on only one like Twitch with HLS, others may rely on both depending on the case).

Those protocols all have similar concepts: all expose a central file listing the characteristics of the content (for example, the video qualities, audio tracks, the available subtitles etc.) and allow a client to request the media through small chunks, each containing only the wanted quality and track for a specific time period.

Schema of how HLS basically works, found on Apple's Website which is the one behind HLS

This architecture allows to:

• only load the wanted media data (thus not e.g. also loading all unwanted audio tracks with it)
• allows efficient seeking on the content, by allowing to load the content non-sequentially (e.g. you can directly load the data corresponding to the end of the content if you want to).
• facilitate live streaming by continuously encoding those chunk and adding it to the central file progressively
• profit from all the goodies of relying on HTTP(S) for content distribution (compatibility with the web, firewall traversal, lots of tools available etc.)

For HLS specifically, this so-called central file is called the "Multivariant Playlist" (a.k.a. "Master Playlist") and is in the M3U8 file format. HLS also has the concept of secondary "Media Playlists", also as M3U8 files, which describe specific tracks and qualities.

The Media Source Extensions™

To allow the implementation of such adaptive streaming media players on the web (and thus with JavaScript), a W3C recommendation was written: the Media Source Extensions™ recommendation, generally just abbreviated to "MSE".

Basically, it builds on top of the HTML5 <video> element, adding a new set of browser API accessible from JavaScript to create media buffers, called SourceBuffers, and allowing to push aforementioned small chunks of media data to it for later decoding.

The media player library

The role of an HLS media player library like this one is thus to load the Multivariant Playlist, detect which characteristics (bandwidth, quality, codecs, preferred language, accessibility etc.) are wanted and to load the right media data at the right time, then communicating it to the browser through the MSE APIs so it can be decoded.

The Wasp-hls player reading a Multivariant playlist from Twitch. You can see on the top right the requests performed - mostly of media segments, and some logs on the bottom right. You can also see a cog logo before the requests' url indicating that they are all performed in a WebWorker.

This may look relatively simple at first, but there's a lot of potential optimizations to make and special cases to handle which make quasi-mandatory the need to develop separately the media streaming library (the part "understanding" the streaming protocol and pushing chunks) and the application (the part relying on the library and implementing an UI and the business logic on top).

Most people generally mean the latter when they talk about a "player", here, I'm "only" implementing the former and the application has to be developped separately.

The demo page do also implement a UI part, but this is just to showcase the library and is not actually what's exposed by this package, only the library is (though you can copy the demo's code if you want to).

What's WebWorker and WebAssembly doing with all that?

Amongst the main characteristics of this player is that it relies on a WebWorker, to run concurrently with the application, and WebAssembly, to do so optimally. It thus allows to have a theoretically efficient media player, allowing to avoid stalling in the content if the UI do some heavy lifting and vice-versa.

This is even more important when playing what is called "low-latency contents" (contents with a small-ish latency between the recording and the watch-ing, in the few seconds range) which have amongst its characteristics the fact that only very small data buffers are constructed - allowing to play closer to live but more exposed to rebuffering risk if the segment loading pipeline takes more time than expected).

Playing low-latency contents is one of the main goal of this project. On that matter as an amusing note, playing low-latency contents through a media player with a WebWorker + WebAssembly combination is exactly what Twitch is doing, though their players isn't open-source. This one is (it's also able to play Twitch contents if you succeed to work-around their CORS policy, only not with low-latency for now)!

Even without taking into account low-latency contents, I consider WebAssembly to be particularly adapted for an adaptive media player:

• Such software generally handle large objects (not only talking about media segments here). The relatively few controls JavaScript offers in terms of memory management can be problematic (I've seen GC-pressure related issues on another player I work on for example).

  With WebAssembly and a source language with enough control over memory management, GC-linked issues disappear (because no GC :p!) and the application's memory usage is much more explicit and controlled, which is usually what we want.

• Even if there's a lot of input/output and thus the need to interact with JavaScript, the performance profile is completely different than your usual JS front-end application: very few DOM access here, a lot of time is spent on parsing/transmuxing and other logic that can be completely implemented in WebAssembly.

Generating an HLS content

Because this is just the player part, the HLS content has to be prepared - through what we call the packaging step - separately by using what's called a "packager".

For example the shaka-packager is a relatively easy to use packager. With it and FFmpeg, you are armed with the right tools to produce HLS contents.

Why starting this project?

I'm currently working as the lead developper of another, featureful adaptive media player library, the open-source RxPlayer so this is not something totally out of the blue.

The reasons why I started this project are mainly:

• to see how IO-heavy logic (like we have here with many requests, media segments streaming, playback observation, network metrics etc.) using web APIs only exposed to JavaScript could be conjugated with WebAssembly and Rust.

• to experiment with a big-enough WebAssembly and Web Worker-based library: how should it interact with applications written in JavaScript?

• to learn more about the HLS streaming protocol, for which the RxPlayer does not provide first class support (yet?)

• to see how I would write a complex beast like a Media player if restarting from scratch (though the situation is very different here, due to the difference in the language used).

• to see if there's any real performance and/or memory related advantage (or disadvantage) in relying on WebAssembly for the core logic of a media player, in various situations (multiple players on the same page, large 4k segments, web workers, mse-in-worker).

• to play with a Web Worker-based media player, and find out its influence in terms of API definition, synchronization difficulties, performance issues etc.

• to work on and improve my Rust skills

What's done

It already has a lot of features but there's still some left work:

Type of contents:

• Play HLS VoD contents
• Play HLS live contents (for now require presence of EXT-X-PROGRAM-DATE-TIME tag in media playlist)
• Proper support of HLS low-latency contents. Priority: average

Worker-related features:

• Load content-related resources and run main logic loop in worker
• Use MSE-in-Worker when available
• Rely on main thread for MSE when MSE-in-Worker is not available

Adaptive BitRate:

• Choose variant based on throughtput-based estimates
• Allow application to list and select its own variant (quality) and know the current one
• Automatically filter out codecs not supported by the current environment.
• Urgent/non-urgent quality switches (Some quality switches lead to request for segments of the previous quality to be immediately interrupted, others await them before actually switching).
• Fast-switching (Push on top of already-loaded segments if they prove to be of higher quality - and are sufficiently far from playback to prevent rebuffering).
• Smart-switching (I just made-up the name here :D, but basically it's for the opposite situation than the one in which fast-switching is active: don't re-load segments who're already loaded or being pushed with a higher quality).
• Also choose variant based on buffer-based estimates. Priority: average
• Logic to detect sudden large fall in bandwidth before the end of a current request. Priority: average

Request Scheduling:

• Lazy Media Playlist downloading (only fetch and refresh them once they are needed)
• Media Playlist refreshing for live contents
• Buffer goal implementation (as in: stop loading segments once enough to fill the buffer up to a certain - configurable - point are loaded)
• Parallel audio and video segment loading
• Priorization between audio and video segment requests (to e.g. stop doing audio segment requests when video ones become urgent).
• Retry of failed requests with an exponential backoff.
• Perform range requests for segments if needed
• Parallel initialization segment and first media segment loading. Priority: average

Media demuxing:

• Media Segment Format: MPEG-2 Transport Streams
• Media Segment Format: Fragmented MPEG-4
• Media Segment Format: Packed Audio AAC with ADTS framing
• Transmux MPEG-2 Transport Streams to fmp4 on platforms not supporting the former like mostChrome or Firefox (through JS for now, Rust implementation pending).
• WebAssembly-based mpeg2-ts transmuxer. Priority: average
• Media Segment Format: Packed Audio MP3 Priority: low
• Media Segment Format: Packed Audio AC-3 Priority: low
• Media Segment Format: Packed Audio EAC-3 Priority: low
• Media Segment Format: WebVTT (subtitles not handled for now) Priority: low
• Media Segment Format: IMSC Subtitles (subtitles not handled for now) Priority: low

MSE API and buffer handling:

• End of stream support (as in: actually end when playback reaches the end!)
• Multiple simultaneous type of buffers support (for now only audio and video, through MSE SourceBuffers)
• One and multiple initialization segments handling per rendition
• Lazy buffer memory management: Don't manually remove old buffers' media data if the browser thinks it's fine. Many players clean it up progressively as it also simplifies the logic (e.g. browser GC detection might become unneeded) but I like the idea of keeping it to e.g. allow seek-back without rebuffering if the current device allows it.
• Detect browser Garbage Collection of buffered media and re-load GCed segments if they are needed again.
• Discontinuity handling: Automatically skip "holes" in the buffer where it is known that no segment will be pushed to fill them.
• Freezing handling: Detect when the browser is not making progress in the content despite having media data to play and try to unstuck it. Priority: average
• Proper handling of QuotaExceededError after pushing segments (when low on memory). This is generally not needed as the browser should already handle some kind of garbage collection but some platforms still may have issues when memory is constrained. Priority: low

Tracks:

• Provide API to set an audio track
• Provide API to set a video track Priority: low
• Allow text track selection and support at least one text track format (TTML IMSC1 or webVTT) - through a JS library first? Priority: low

Miscellaneous:

• Error API
• Export embedded versions of the WebAssembly and Worker files to facilitate application's development code.
• Initial position API
• Delta playlist handling. Priority: low
• Content Steering handling. Priority: low
• Support content decryption. Priority: very low

Playlist tags specifically considered (unchecked ones are mainly just ignored, most of them are not needed for playback):

• EXT-X-ENDLIST: Parsed to know if a playlist needs to be refreshed or not, but also to detect if we're playing an unfinished live content to play close to the live edge by default.
• EXTINF: Only the indicated duration of a segment is considered, not the title for which there's no use for now. Both integer and float durations should be handled.
• EXT-X-PROGRAM-DATE-TIME: Used to determine the starting position of segments as stored by the player - which may be different than the actual media time once the corresponding segment is pushed. The units/scale indicated by this tag will be preferred over the real media time in the player APIs.
• EXT-X-BYTERANGE: Used for range requests
• EXT-X-PLAYLIST-TYPE: Used To know if a Playlist may be refreshed
• EXT-X-TARGETDURATION: Useful for heuristics for playlist refresh
• EXT-X-GAP: Those segments are just skipped, no variant switch or anything like that.
• EXT-X-START: Used to determine a default start time in the content.
• EXT-X-MAP:
  • URI: Used to fetch the initialization segment if one is present
  • BYTERANGE: To perform a range request for the initialization segment
• EXT-X-MEDIA:
  • TYPE: Both AUDIO and VIDEO are handled. SUBTITLES and CLOSED-CAPTIONS are just ignored for now.
  • URI
  • GROUP-ID
  • DEFAULT
  • AUTOSELECT
  • LANGUAGE: In audio track selection API
  • ASSOC-LANGUAGE: In audio track selection API
  • NAME: In audio track selection API
  • CHANNELS: Not so hard to implement, but I've been too lazy to parse that specific format from the Multivariant Playlist for now
  • CHARACTERISTICS: Soon...
  • FORCED: As the SUBTITLES TYPE is not handled yet, we don't have to use this one
  • INSTREAM-ID: As the CLOSED-CAPTIONS TYPE is not handled yet, we don't have to use this one.
  • STABLE-RENDITION-ID: Not really needed for now (only for content steering?)
• EXT-X-STREAM-INF:
  • BANDWIDTH: Used to select the right variant in function of the bandwidth
  • CODECS: Used for checking support (and filtering out if that's not the case, and for initializing buffers with the right info).
  • AUDIO
  • VIDEO: As no video track selection API exist yet, only the most prioritized video media playlist is considered
  • RESOLUTION: Used to describe variant in variant selection API
  • FRAME-RATE: Used to describe variant in variant selection API
  • SCORE: Considered both to select a variant and to determine if a quality is better when "fast-switching".
  • STABLE-VARIANT-ID: Not really needed for now (only for content steering?)
  • AVERAGE-BANDWIDTH: Not used yet. I don't know if it's useful yet for us.
  • SUPPLEMENTAL-CODECS: In our web use case, I'm not sure if this is only useful for track selection API or if filtering also needs to be done based on this.
  • SUBTITLES: No subtitles support for now
  • CLOSED-CAPTIONS: that one is just ignored for now
  • PATHWAY-ID: Content Steering not handled yet
  • HDCP-LEVEL: DRM are not handled for now
  • ALLOWED-CPC: DRM are not handled for now
• EXT-X-VERSION: Not specifically considered for now, most differences handled until now had compatible behaviors from version to version
• EXT-X-INDEPENDENT-SEGMENTS: Might needs to be considered once we're doing some manual cleaning?
• EXT-X-DEFINE: Seems rare enough, so may be supported if the time is taken...
• EXT-X-MEDIA-SEQUENCE: Not sure of what this allows. To check...
• EXT-X-I-FRAMES-ONLY: To handle one day, perhaps (very low priority)
• EXT-X-PART: low-latency related
• EXT-X-PART-INF: low-latency related
• EXT-X-SERVER-CONTROL: low-latency related?
• EXT-X-BITRATE
• EXT-X-DATERANGE: Might be used for an event emitting API?
• EXT-X-SKIP
• EXT-X-PRELOAD-HINT
• EXT-X-RENDITION-REPORT
• EXT-X-I-FRAME-STREAM-INF
• EXT-X-SESSION-DATA
• EXT-X-SESSION-KEY
• EXT-X-CONTENT-STEERING
• EXT-X-KEY: decryption and related tags are very low priority
• EXT-X-DISCONTINUITY: I'm under the impression in our scenario that it only is useful to increment discontinuity sequences, which we have no need for...
• EXT-X-DISCONTINUITY-SEQUENCE: I don't think we need this, at least I didn't encounter a case for it now that isn't handled by other tags.

Setup

If you want to contribute or build the Wasp-hls locally, you will need to have nodejs and rust installed.

Then you need to install node dependencies by calling in your shell:

# Install all node dependencies (needs npm, generally installed with nodejs)
npm install

You also need to add the rust wasm32 target and some rust dependencies:

# Add wasm32 target (needs rustup that you most likely took with rust)
rustup target add wasm32-unknown-unknown

# Add wasm-bindgen CLI (needs cargo, generally installed with rust)
cargo install wasm-bindgen-cli

# Optionally, you may also need clippy, for checking Rust code mistakes
rustup component add clippy

Build

The building of the Wasp-hls player may be performed by module, if you just updated one area of the code (the Rust code for example), or as a whole.

The Rust (WebAssembly) code

To build only the Rust code in src/rs-core/ to its destination WebAssembly file (build/wasp_hls_bg.wasm), you can run any of the following commands:

# Build in debug mode, which leads to a bigger file and slower code, but is
# more useful and quicker to build when developping
npm run build:wasm

# Build in release mode, which is the actual delivered result
npm run build:wasm:release

The TypeScript Worker code

To build only the Worker code in src/ts-worker to its destination JavaScript file (build/worker.js), you can run any of the following commands:

# Build in debug mode, which leads to a bigger file though much easier to debug
npm run build:worker

# Build in release mode, which is the actual delivered minified result
npm run build:worker:release

The Main (API) code

To build only the code running in the main thread present in src/ts-main to its destination JavaScript file (build/main.js), you can run any of the following commands:

# Build in debug mode, which leads to a bigger file though much easier to debug
npm run build:main

# Build in release mode, which is the actual delivered minified result
npm run build:main:release

The Demo

To build only the demo application showcasing the Wasp-hls player, whose code is present in the (demo/) directory, to its destination JavaScript file (build/demo.js), you can run any of the following commands:

# Build in debug mode, which leads to a bigger file though much easier to debug
npm run build:demo

# Build in debug mode, with a "watcher" rebuilding each time one of its files
# changes
npm run build:demo:watch

# Build in release mode, which is the actual delivered minified result
npm run build:demo:release

Then to perform your tests, you generally want to serve the demo. You can do so with:

npm run serve

Combinations

If you just want to build the whole Wasp-hls player code, without the demo, you may call:

npm run build:all

If you want to build all that code AND the demo:

npm run build:all && npm run build:demo

Though what you most likely want to do here is build the full code used by the demo to perform your tests, here just write:

npm run build:all:demo

That last script bypass the generation of the build/main.js file, as the demo file (build/demo.js) already includes the content of that file anyway.

To build everything in release mode, for an actual release or for tests in production conditions, write:

# WebAssembly + Worker + Main in release mode
npm run build:release

# If you also want the demo in release mode
npm run build:demo:release

The Documentation

The documentation, written in doc/ may also be built to its final directory (build/doc), through the following command:

npm run doc

It may then be served, so it can be read on a web browser, through:

npm run serve

And then requesting the /doc path.

Update the code

You're welcome to read the code which should be hopefully documented enough and readable enough to dive into. The source code of the player is in the src directory, if you would prefer to work on the demo, it's in the demo directory, as for the documentation, it's in the demo directory.

Check the code

To check the TypeScript types of TypeScript files and their code style with the eslint package, you can run:

# Check all TypeScript files in the project
npm run check

# OR, check only the Worker code
npm run check:worker

# OR, check only the Main code
npm run check:main

# OR, check only the Demo code
npm run check:demo

To check the Rust code, with clippy, you can run:

# Check all Rust files in the project
npm run clippy

You also might want to format automatically the code before commiting:

# Format all TypeScript, JavaScript, Markdown and HTML files in the project
npm run fmtt

# Format all Rust code
npm run fmtr

# Do both
npm run fmt