title: "How does browser load videos" description: "" added: "July 27 2023" tags: [web]
HTTP range request is a widely used feature when it comes to file resource. File systems such as S3 have good support for this.
The HTTP 206 Partial Content may be sent from the server when the client has asked for a range (e.g. "give me the first 2MB of video data"). It is vital for downloading data in chunks which avoids fetching unused resources. Look at the outgoing request for a Range header (e.g., Range: bytes=200-1000).
The Range HTTP request header indicates the part of a document that the server should return. If the server sends back ranges, it uses the 206 Partial Content for the response. If the ranges are invalid, the server returns the 416 Range Not Satisfiable error. The server can also ignore the Range header and return the whole document with a 200 status code if it does not support range requests.
Besides HTTP 206 status code, the server should respond with a Content-Range header, specifying the start byte, end byte and total size of the resource. Content-Length should be set to the length of the returned range, not the total size of the resource. Note that the server is allowed to change the range that was requested and even ignore the fact that it was a range request.
By default when using video html tag, the Range header with value bytes=0- is sent. The client can know whether HTTP partial content is available by checking if the HTTP response header Accept-Ranges: bytes is included.
If you load an video, the metadata allows the browser to map a time code to a byte offset in the file. It assumes to look up the start and the footer data. (If the metadata is placed at the end of the file, it then sends another range request for the footer of the file.) Now that the browser knows the duration and other important data about the video, it can show the player controls and make a new request to buffer up the video data. It’s worth noting that MP4 can actually have the necessary metadata at the start, which will save you a round trip and will make your MP4 play earlier.
Look at this process in detail:
1. send a normal request. By reading the response header from stream,
Chrome find the server side supports range, then abort the connection
and start sending range requests.
Chrome Server
+------------------------+ ------------> +-------------------------------------+
| GET /a.mp4 HTTP/1.1 | close conn when | HTTP/1.1 200 OK |
| Host: example.com | <----x------- | Accept-Ranges: bytes |
+------------------------+ range support detected | Content-Length: 828908177 |
| ... |
| (body: some first bytes of a.mp4) |
+-------------------------------------+
2. send trivial range request, fetch head parts, verify server's support
Chrome Server
+------------------------+ ------------> +---------------------------------------------+
| GET /a.mp4 HTTP/1.1 | close conn when | HTTP/1.1 206 Partial Content |
| Host: example.com | <----x------- | Accept-Ranges: bytes |
| Range: [bytes=0-] | verify success | Content-Range: bytes 0-828908176/828908177 |
+------------------------+ | Content-Length: 828908177 |
| ... |
| (body: some first bytes of a.mp4) |
+---------------------------------------------+
Note that the data being sent from the server is only a small chunk, even though the Range header has value “bytes=0-”. Chrome and FireFox ask for ranges like bytes=300-, can server side return a smaller-range part, other than part from offset 300 to end of file? The answer is yes. When you play the video, browser will send range request for remaining bytes. Both Chrome and FireFox send range request using byte range (i.e bytes=1867776-) with last-byte-pos value absent. (The server sends TCP Keep-Alive to keep this TCP connection connected.)
If you skip ahead in the video, the browser will cancel the currently on-going response for the video content. It will then use the the video file’s metadata to map your desired new position to a byte offset and use it for a new range request (byte=offset-). When the buffer is full and the browser stops the server from sending more data, the request is technically still on-going, just no data is being sent.
MP4 and WebM formats are what we would call pseudo-streaming or "progressive download”. These formats do not support adaptive bitrate streaming (adjusts video quality based on network conditions). If you have ever taken an HTML video element and added a "src” attribute that points to an mp4, most players will progressively download the file. The good thing about progressive downloads is that you don’t have to wait for the player to download the entire file before you start watching. You can click play and start watching while the file is being downloaded in the background. Most players will also allow you to drag the playhead to specific places in the video timeline and the player will use byte-range requests to estimate which part of the file you are attempting to seek. What makes MP4 and WebM playback problematic is the lack of adaptive bitrate support. Every user who watches your content must have enough bandwidth available to download the file faster than it can playback the file.
HTTP Live Streaming sends audio and video as a series of small files, called media segment files. An index file, or playlist, provides an ordered list of the URLs of the media segment files. Index files for HTTP Live Streaming are saved as M3U8 playlists, an extension of the M3U format used for MP3 playlists.
<img alt="m3u8 compatibility" src="https://raw.gitmirror.com/kexiZeroing/blog-images/main/51ec8e88-554e-4272-b3de-df878d9dede4.png" width="750">
<script src="https://cdn.jsdelivr.net/npm/hls.js@latest"></script>
<video id="video" controls></video>
<script>
if(Hls.isSupported()) {
var video = document.getElementById('video');
var hls = new Hls();
hls.loadSource('https://test-streams.mux.dev/x36xhzz/x36xhzz.m3u8');
hls.attachMedia(video);
hls.on(Hls.Events.MANIFEST_PARSED, function() {
video.play();
});
}
</script>
<img alt="blob-url-video" src="https://raw.gitmirror.com/kexiZeroing/blog-images/main/12643918-81ad-47c1-8ea2-45205d1f23a9.png" width="500">
All those websites actually still use the video tag. But instead of simply setting a video file in the src attribute, they make use of much more powerful web APIs, the Media Source Extensions (more often shortened to just “MSE”). Complex web-compatible video players are all based on MediaSource and SourceBuffer.
Those “extensions” add the MediaSource object to JavaScript. As its name suggests, this will be the source of the video, or put more simply, this is the object representing our video’s data. The URL.createObjectURL API allows creating an URL, which will actually refer not to a resource available online, but directly to a JavaScript object created on the client.
// Create a MediaSource and attach it to the video
const videoTag = document.getElementById("my-video");
const myMediaSource = new MediaSource();
const url = URL.createObjectURL(myMediaSource);
videoTag.src = url;
// 1. add source buffers
const audioSourceBuffer = myMediaSource
.addSourceBuffer('audio/mp4; codecs="mp4a.40.2"');
const videoSourceBuffer = myMediaSource
.addSourceBuffer('video/mp4; codecs="avc1.64001e"');
// 2. download and add our audio/video to the SourceBuffers
// fragmented mp4 (the advantage of fragmented MP4 is its ability to support DASH)
fetch("http://server.com/audio.mp4").then(function(response) {
return response.arrayBuffer();
}).then(function(audioData) {
audioSourceBuffer.appendBuffer(audioData);
});
// the same for the video SourceBuffer
fetch("http://server.com/video.mp4").then(function(response) {
return response.arrayBuffer();
}).then(function(videoData) {
videoSourceBuffer.appendBuffer(videoData);
});
What actually happens in the more advanced video players, is that video and audio data are split into multiple “segments”. These segments can come in various sizes, but they often represent between 2 to 10 seconds of content. Instead of pushing the whole content at once, we can just push progressively multiple segments. Now we do not have to wait for the whole audio or video content to be downloaded to begin playback.
// fetch a video or an audio segment, and returns it as an ArrayBuffer
function fetchSegment(url) {
return fetch(url).then(function(response) {
return response.arrayBuffer();
});
}
// fetching audio segments one after another
fetchSegment("http://server.com/audio/segment0.mp4")
.then(function(audioSegment0) {
audioSourceBuffer.appendBuffer(audioSegment0);
})
.then(function() {
return fetchSegment("http://server.com/audio/segment1.mp4");
})
.then(function(audioSegment1) {
audioSourceBuffer.appendBuffer(audioSegment1);
})
.then(function() {
return fetchSegment("http://server.com/audio/segment2.mp4");
})
.then(function(audioSegment2) {
audioSourceBuffer.appendBuffer(audioSegment2);
})
// same thing for video segments
fetchSegment("http://server.com/video/segment0.mp4")
.then(function(videoSegment0) {
videoSourceBuffer.appendBuffer(videoSegment0);
});
Many video players have an “auto quality” feature, where the quality is automatically chosen depending on the user’s network and processing capabilities. This behavior is also enabled thanks to the concept of media segments. On the server-side, the segments are actually encoded in multiple qualities, and a web player will then automatically choose the right segments to download as the network or CPU conditions change.
The most common transport protocols used in a web context:
For both HLS and DASH, players can adapt to the different renditions in real-time on a segment-by-segment basis.
./audio/
├── ./128kbps/
| ├── segment0.mp4
| ├── segment1.mp4
| └── segment2.mp4
└── ./320kbps/
├── segment0.mp4
├── segment1.mp4
└── segment2.mp4
./video/
├── ./240p/
| ├── segment0.mp4
| ├── segment1.mp4
| └── segment2.mp4
└── ./720p/
├── segment0.mp4
├── segment1.mp4
└── segment2.mp4
A regular MP4 video usually has a single moov chunk describing the video and a single mdat chunk containing the video. You wouldn’t be able to play a part of the video without having access to the whole video. Fragmented MP4 solves this issue, allowing us to split an MP4 video into segments. The first initialization segment contains the chunk describing the video. What follows are the media segments, each having a separate chunks containing a portion of the video which can be played on its own.
FFMPEG stands for Fast Forward Moving Picture Experts Group. It is a free and open source software project that offers many tools for video and audio processing. It's designed to run on a command line interface, and has many different libraries and programs to manipulate and handle video files. Most video programs include FFMPEG as a part of the video processing pipeline. (FFmpeg powers all online video - Youtube, Facebook, Instagram, Disney+, Netflix etc, all run FFmpeg underneath.)
WebAssembly enables developers to bring new performant functionality to the web from other languages. FFmpeg.wasm (WebAssembly / JavaScript port of FFmpeg) is one of a showcasing of the new functionality being made available thanks to WebAssembly. It enables video & audio record, convert and stream right inside browsers. There are two components inside ffmpeg.wasm: @ffmpeg/ffmpeg and @ffmpeg/core. @ffmpeg/ffmpeg contains kind of a wrapper to handle the complexity of loading core and calling low-level APIs. @ffmpeg/core contains WebAssembly code which is transpiled from original FFmpeg C code with minor modifications.
// AVI to MP4 Demo
import { writeFile } from 'fs/promises';
import { createFFmpeg, fetchFile } from '@ffmpeg/ffmpeg';
const ffmpeg = createFFmpeg({ log: true });
(async () => {
await ffmpeg.load();
ffmpeg.FS('writeFile', 'test.avi', await fetchFile('./test.avi'));
await ffmpeg.run('-i', 'test.avi', 'test.mp4');
await fs.promises.writeFile('./test.mp4', ffmpeg.FS('readFile', 'test.mp4'));
process.exit(0);
})();
- ffmpeg-online is an online version of ffmpeg based on
ffmpeg.wasm, which can be used to process audio and video online. The most straightforward exampleffmpeg -i input.mp4 output.aviis used to convert an input media file to a different format.- ffmpeg-video-processing shares examples using FFmpeg to optimize uploaded videos.
A codec is a hardware or software tool that is used to compress (and decompress) video files. Codec is a blend of coder/decoder. Common video codecs include h.264, h.265, VP8, VP9 and AV1. An efficient codec can deliver high-quality video at lower bitrates.
The bitrate of a file is measured by the number of bits being transmitted over a period of time. For video it is typically measured in kilobytes per second (kbps) or megabytes per second (mbps). Video bitrate is often confused with video resolution terms like 720p, 1080p, 4K, etc. Video resolution is the number of pixels that make up an image on your screen; video bitrate is the amount of information per second in video. A higher bitrate results in better quality but also larger file sizes.
VOD(Video on Demand) is videos that can be accessed on viewer request. Unlike live streaming, VODs are prerecorded programs. With VOD, viewers can watch content they enjoy as frequently as they like. They can also pause, rewind, and view additional content that was not previously available. Streaming is one of two ways to access Video On Demand. The other way is to permanently download video files to a device’s memory. VOD systems typically distribute media using internet connections, so good bandwidth is important for best results for viewers. Popular platforms include Netflix, Hulu, Disney, Amazon Prime Video and many others.
Some of the most popular streaming protocols include RTMP, HLS, and WebRTC:
RTMP stands for Real-Time Messaging Protocol and it's been used for streaming video and audio on the internet for many years owned by Adobe. The RTMP streaming protocol is TCP-based and designed to maintain constant, low-latency connections between a video player and server. The design allows RTMP to provide smooth and reliable streaming for viewers. To send your broadcast to a destination using RTMP, you need the RTMP Server URL as a unique web address that carries your live video stream every time you broadcast, and a Stream Key which is the private code that will allow your RTMP feed to connect to the exact location that you are streaming to.
HTTP Live Streaming (HLS) is one of the most widely used video streaming protocols created by Apple. It breaks down video files into smaller downloadable HTTP files and delivers them using the HTTP protocol. One advantage of HLS is that all Internet-connected devices support HTTP, making it simpler to implement than streaming protocols that require the use of specialized servers. Another advantage is that an HLS stream can increase or decrease video quality depending on network conditions without interrupting playback. This is why video quality may get better or worse in the middle of a video as a user is watching it. This feature is known as "adaptive bitrate streaming".
WebRTC is an abbrevation of "Web Real Time Communication." The set of standards that comprise WebRTC makes it possible to share data and perform teleconferencing peer-to-peer, without requiring that the user install plug-ins or any other third-party software. The connections are peer-to-peer rather than sent and aggregated by a central location, which means that the video and audio is shared to each device communicating in the conversation. This keeps the communictaion latency very low.