🎓 Getting Started
-
⏯ Demo
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📖 API documentation

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?

$3

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.

!de18e941-81de-482f-843d-834a4dd3aa71
_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.

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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 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.

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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.

!twitch-wasp
_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).

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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.

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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:

- [x] Play HLS VoD contents
- [x] 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:

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

Adaptive BitRate:

- [x] Choose variant based on throughtput-based estimates
- [x] Allow application to list and select its own variant (quality) and know
the current one
- [x] Automatically filter out codecs not supported by the current environment.
- [x] 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).
- [x] 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).
- [x] 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:

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

Media demuxing:

- [x] Media Segment Format: MPEG-2 Transport Streams
- [x] Media Segment Format: Fragmented MPEG-4
- [x] Media Segment Format: Packed Audio AAC with ADTS framing
- [x] 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:

- [x] End of stream support (as in: actually end when playback reaches the end!)
- [x] Multiple simultaneous type of buffers support (for now only audio and
video, through MSE SourceBuffers)
- [x] One and multiple initialization segments handling per rendition
- [x] 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.
- [x] Detect browser Garbage Collection of buffered media and re-load GCed
segments if they are needed again.
- [x] 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:

- [x] 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:

- [x] Error API
- [x] Export embedded versions of the WebAssembly and Worker files to facilitate
application's development code.
- [x] 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):

- [x] 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.
- [x] 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.
- [x] 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.
- [x] EXT-X-BYTERANGE: Used for range requests
- [x] EXT-X-PLAYLIST-TYPE: Used To know if a Playlist may be refreshed
- [x] EXT-X-TARGETDURATION: Useful for heuristics for playlist refresh
- [x] EXT-X-GAP: Those segments are just skipped, no variant switch or
anything like that.
- [x] EXT-X-START: Used to determine a default start time in the content.
- EXT-X-MAP:
- [x] URI: Used to fetch the initialization segment if one is present
- [x] BYTERANGE: To perform a range request for the initialization segment
- EXT-X-MEDIA:
- [x] TYPE: Both AUDIO and VIDEO are handled. SUBTITLES and CLOSED-CAPTIONS
are just ignored for now.
- [x] URI
- [x] GROUP-ID
- [x] DEFAULT
- [x] AUTOSELECT
- [x] LANGUAGE: In audio track selection API
- [x] ASSOC-LANGUAGE: In audio track selection API
- [x] 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:
- [x] BANDWIDTH: Used to select the right variant in function of the
bandwidth
- [x] CODECS: Used for checking support (and filtering out if that's not the
case, and for initializing buffers with the right info).
- [x] AUDIO
- [x] VIDEO: As no video track selection API exist yet, only the most
prioritized video media playlist is considered
- [x] RESOLUTION: Used to describe variant in variant selection API
- [x] FRAME-RATE: Used to describe variant in variant selection API
- [x] 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:

``sh


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:

`sh


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.

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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:

`sh


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
`

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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:

`sh


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
`

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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:

`sh


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
`

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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:

`sh


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:

`sh
npm run serve
`

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

`sh
npm run build:all
`

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

`sh
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:

`sh
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:

`sh


WebAssembly + Worker + Main in release mode

npm run build:release

If you also want the demo in release mode

npm run build:demo:release
`

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

`sh
npm run doc
`

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

`sh
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:

`sh


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:

`sh


Check all Rust files in the project


npm run clippy
`

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

`sh


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
``