Earlier today we unveiled Find Your Way To Oz, a new Chrome Experiment inspired by the upcoming feature film Oz The Great and Powerful. Developed by UNIT9, this experiment brings together Disney’s unique storytelling tradition and the power of the web platform, allowing users to interact with the web in a completely new way.
The desktop version of “Find Your Way To Oz” uses many of the open web’s more advanced features:
- Immersive Graphics: The experiment uses WebGL for the main 3D environment, CSS3 features such as CSS Transitions for various visual embellishments, and GLSL shaders for the tornado’s ominous look and feel.
- Rich Audio: As the user explores the experiment, the 3D sound dynamically adapts thanks to the Web Audio API. The same API powers the experiment’s music composing section.
- Camera-based interactions: Through WebRTC’s getUserMedia API, users can become circus characters or record their own mini-movies.
The experiment’s mobile web version also uses cutting-edge web technologies. These include graphics features such as accelerated 3D transforms and sprite sheets as well as mobile hardware features like camera, multi-touch, gyroscope and accelerometer. Together they create an experience that can normally only be found in native apps.
To learn more about how this experiment was built, read our technical case study and join us for a Google Developers Live event on February 11th at 11 a.m. GMT where we’ll be talking to the team behind the project. Alternatively, use Chrome’s developer tools to see how the experiment works on your own, perhaps finding in the process your own path to the yellow brick road.
For the first time, Chrome and Firefox can “talk” to each other via WebRTC. WebRTC is a new set of technologies that brings clear crisp voice, sharp high-definition (HD) video and low-delay communication to the web browser.
From the very beginning, this joint WebRTC effort was embraced by the open web community, including engineers from the Chrome and Firefox teams. The common goal was to help developers offer rich, secure communications, integrated directly into their web applications.
In order to succeed, a web-based communications platform needs to work across browsers. Thanks to the work and participation of the W3C and IETF communities in developing the platform, Chrome and Firefox can now communicate by using standard technologies such as the Opus and VP8 codecs for audio and video, DTLS-SRTP for encryption, and ICE for networking.
To try this yourself, you’ll need desktop Chrome 25 Beta and Firefox Nightly for Desktop. In Firefox, you'll need to go to about:config and set the media.peerconnection.enabled pref to "true”. Then head over to the WebRTC demo site and start calling.
For developers looking to include this functionality in their own apps, there are a few places you can go to get more information. You can look at the source code of the AppRTC demo, a library that makes writing cross-browser WebRTC apps a snap, and a document detailing some of the minor differences between browsers.
You can read more from Mozilla’s hacks blog here and view our first “Official” call at the video below:
The getUserMedia API lets users grant web apps access to their camera and microphone without a plug-in. This is the first step in enabling high quality video and audio communication as part of WebRTC, a powerful new real-time communications standard for the open web platform.
Last January, Chrome was the first major browser to preview WebRTC, HTML5's new real time audio and video stack. Since then, we've been hard at work keeping up with the evolving specification, fixing bugs and listening to the web community’s feedback.
The main parts of the WebRTC specification are now stable and are coming soon to all 200M+ Chrome users. With this blog post, we want to help developers plan for what will be introduced in this first stable release later this year.
Our implementation will support multiple independent PeerConnections, each capable of sending and receiving multiple independent media sources.
ICE / STUN / TURN
ICE and STUN are standardized methods for establishing a peer-to-peer connection on the Internet, even if the two end points are behind private network addresses (NAT). Chrome’s current stack deviates from the official current standards. We are working to fix this.
We will also support TURN servers to allow connections through tougher firewalls, where relaying and encapsulation are needed. Exactly what type of TURN will be supported is TBD.
Encryption will be mandatory for all usage of WebRTC in Chrome. For our first stable release, we will implement DTLS-SRTP.
VP8, iSAC, iLBC, G.711
The video codec support by Chrome will be VP8. We've made several major improvements inside and around VP8 to ensure it can deliver a great real time experience. On the audio side, we will initially support iSAC, iLBC, G.711, and DTMF, with iSAC being the default. It is a royalty free wideband codec optimized for speech, open sourced at webrtc.org.
More functionality and features will appear in future versions of Chrome. We’ll work on prioritizing them once we get the basics right:
- Data API. Implementation will start once the network stack is ready.
- Screen sharing.
- PeerConnection proxying. The ability to relay a stream to a third party will not make our first version.
- Recording. MediaRecorder specification work has not been completed yet.