Chromium Blog
News and developments from the open source browser project
Introducing “Find Your Way to Oz”, a new Chrome Experiment
Tuesday, February 5, 2013
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.
Posted by Max Heinritz, Product Magician (Manager)
Hello Firefox, this is Chrome calling!
Monday, February 4, 2013
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:
Posted by Maire Reavy, Firefox Media Product Lead and Serge Lachapelle, Chrome Product Manager
Introducing getUserMedia and the Javascript Gamepad API
Monday, July 9, 2012
Today’s
Chrome Beta release
includes two new APIs: the getUserMedia API and the Gamepad Javascript API.
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.
In addition, getUserMedia can be combined with other platform features like CSS filters and WebGL to render effects as the <video> is captured. For example, you can
rotate the video and add hipstery filters
,
play a xylophone with motion detection
,
try on glasses with face detection
, and
step into a photobooth
with crazy effects like “Snow” and “Fire.”
Follow WebRTC on Google+
for the occasional awesome demo update, and check out the video below for an in depth discussion of WebRTC at Google I/O.
The Gamepad Javascript API helps developers access input from any standard gamepad connected to the user’s machine, creating a richer gameplay experience with these controllers. Gamepad access was made available for NaCl in May, and since its introduction has enabled awesome games like
AirMech
. We’re excited to see what developers will create in JavaScript.
Tommy Widenflycht, Software Engineer and Real-Time Communicator
Chrome’s WebRTC roadmap
Tuesday, April 10, 2012
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.
What's in:
JSEP
JSEP
(Javascript Session Establishment Protocol) is an API for signaling that allows for much more powerful apps and flexibility in choice of signaling protocols. To abstract the complexity, we provide and maintain a
Javascript lib
that makes browser to browser calls a few lines of Javascript.
Topologies
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.
DTLS-SRTP
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.
What’s next?
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.
Comments, questions, kudos or tomatoes? Let us know on our
discussion list
.
Posted by Serge Lachapelle, Product Manager
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