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In order to tap into these types of controls, developers needed to use an external JavaScript library that provided UI components, or else use an alternative development framework such as Adobe Flex, Microsoft Silverlight, or JavaFX. HTML5 aims to fill some of the gaps left by its predecessor in this space by providing a whole range of new form input types.
In addition to these new input types, HTML5 also supports two major new features for form fields.
The second enhancement is autofocus, which tells a browser to automatically give focus to a particular form field when the page has rendered, without requiring JavaScript code to do so.
The first improvement HTML5 brings to web forms is the ability to set placeholder text in an input field. Because the autofocusing is done with JavaScript, it can be tricky to handle all of these edge cases, and there is little recourse for people who don't want a web page to steal the focus. As it happens to the placeholder, browsers that don't support the autofocus attribute will simply ignore it. Detect if the browser supports the autofocus attribute, and only run our own autofocus script if the browser doesn't support autofocus natively.
However, HTML5 defines several new field types, and for reasons that will become clear in a moment, there is no reason not to start using them.
The HTML5 specification doesn't mandate any particular user interface for the new input types.
Browsers that don't support HTML5 will treat type="url" exactly like type="text", so there's no downside to using it for all your web-address-inputting needs. As with all the other input types, browsers that don't support type="number" will treat it as type="text". Modernizr is an open source, MIT-licensed JavaScript library that detects support for many HTML5 & CSS3 features. All the available attributes are the same as type="number", min, max, step, value, and they mean the same thing. Tips for Troubleshooting HTML5 Web Pages- Tutorial: A picture of a sample HTML5 web page that contains many common coding mistakes that you should look for when troubleshooting HTML5 web pages. The header part is the most important as here you will be communicating with client browser, tell it that you can accept range request and check if browser made range requests or not and decide the amount of data to stream. Hope this small tutorial on PHP HTML5 Video Streaming will help you to get started with your next video streaming app. Did you compare the load time with direct media file loading and via PHP streaming(if so, give some link to test)? I afraid, this will take both S3 and local server bandwidth, unless you cache the s3 on local machine. Great article, I am also trying to stream to a player but the video is taking over my whole page how do i make the video is just added to my html5 player. You can test the videos using the following links, the first streams the local video, the second streams the video from S3. I want to stream a live event from a camera, is it possible to stream it with the php script? Hi thanks a lot for your wonderful tutorial , I have question , what is difference between show the video directly through html5 video tag and through PHP code ? Hi, I am using your script to play a video, when i play two files one by one in refresh, it is playing the old one only, some times even I eventually gave a video file it does not exist, it is playing the old one only. I use the script on my site to hide the mp3 url so it cannot be downloaded without permission from the artist or band that uploaded it. Membuat Slideshow Dengan Css dan Jquery itu mudah dan tidak sesulit yang dibayangkan, pada tutorial kali ini akan kita bahas secara singkat cara membuat nya. Jquery bukan hal asing lagi ditelinga kita, jquery datang dengan banyak efek untuk website kita dan hampir menggantikan flash yang dulu juga di gunakan agar website kita terasa hidup dengan efek yang bisa kita ciptakan. Menjadikan web browser sebagai editor seperti notepad rasanya tidak mungkin bisa, tapi nyatanya kita bisa menggunakan Browser yang kita miliki agar bisa menjadi sebuah teks editor untuk membuat catatan, atau membuat file html. Memutar video di halaman website adalah bahasan kita kali ini, dimana kita akan mencoba memutar file dengan format flv, mp4, wmv, dan lainnya di halaman website dengan bantuan html5 dan plugin javascript. Membuat menu bersuara atau ada efek suara nya ketika mouse hover dengan menggunakan HTML5 Audio adalah bahasan kita kali ini. Belajar HTML5 Dengan melihat video tutorial dan contoh adalah sesuatu yang baik, ada banyak cara untuk belajar, diataranya dengan membaca, bertanya, mendengar dan melihat. HTML5 dan CSS3 bukanlah barang baru, sudah beberapa tahun belakangan ini kita mendengar kedua kata itu. Imagine a world where your phone, TV and computer could all communicate on a common platform. Open the URL displayed at the bottom of the page in a new tab or, better still, on a different computer. Alternatively, jump straight into our WebRTC codelab: a step-by-step guide that explains how to build a complete video chat app, including a simple signaling server. One of the last major challenges for the web is to enable human communication via voice and video: Real Time Communication, RTC for short. Historically, RTC has been corporate and complex, requiring expensive audio and video technologies to be licensed or developed in house.
Gmail video chat became popular in 2008, and in 2011 Google introduced Hangouts, which use the Google Talk service (as does Gmail). WebRTC has now implemented open standards for real-time, plugin-free video, audio and data communication. Plugins can be difficult to deploy, debug, troubleshoot, test and maintain—and may require licensing and integration with complex, expensive technology. The guiding principles of the WebRTC project are that its APIs should be open source, free, standardized, built into web browsers and more efficient than existing technologies. RTCPeerConnection is in Chrome (on desktop and for Android), Opera (on desktop and in the latest Android Beta) and in Firefox.
Get network information such as IP addresses and ports, and exchange this with other WebRTC clients (known as peers) to enable connection, even through NATs and firewalls. RTCPeerConnection: audio or video calling, with facilities for encryption and bandwidth management.
Likewise on pages using HTTPS: permission only has to be granted once for for getUserMedia() (in Chrome at least). Also, Chrome will deprecate HTTP access for getUserMedia() at the end of 2015 due to it being classified as a Powerful feature. The intention is eventually to enable a MediaStream for any streaming data source, not just a camera or microphone.
Webcam Toy is a photobooth app that uses WebGL to add weird and wonderful effects to photos which can be shared or saved locally.
One gotcha: getUserMedia constraints set in one browser tab affect constraints for all tabs opened subsequently. WebRTC uses RTCPeerConnection to communicate streaming data between browsers (aka peers), but also needs a mechanism to coordinate communication and to send control messages, a process known as signaling. Instead, WebRTC app developers can choose whatever messaging protocol they prefer, such as SIP or XMPP, and any appropriate duplex (two-way) communication channel.
Media capabilities: what codecs and resolutions can be handled by my browser and the browser it wants to communicate with? The exchange of information via signaling must have completed successfully before peer-to-peer streaming can begin. Alice sends serialized candidate data to Bob, via whatever signaling channel they are using: WebSocket or some other mechanism.
When Bob gets a candidate message from Alice, he calls addIceCandidate, to add the candidate to the remote peer description.
WebRTC clients (known as peers, aka Alice and Bob) also need to ascertain and exchange local and remote audio and video media information, such as resolution and codec capabilities. In the callback, Alice sets the local description using setLocalDescription() and then sends this session description to Bob via their signaling channel. Bob sets the description Alice sent him as the remote description using setRemoteDescription(). Bob runs the RTCPeerConnection createAnswer() method, passing it the remote description he got from Alice, so a local session can be generated that is compatible with hers.

When Alice gets Bob's session description, she sets that as the remote description with setRemoteDescription. RTCSessionDescription objects are blobs that conform to the Session Description Protocol, SDP. The acquisition and exchange of network and media information can be done simultaneously, but both processes must have completed before audio and video streaming between peers can begin. Once the signaling process has completed successfully, data can be streamed directly peer to peer, between the caller and callee—or if that fails, via an intermediary relay server (more about that below). RTCPeerConnection is the WebRTC component that handles stable and efficient communication of streaming data between peers. From a JavaScript perspective, the main thing to understand from this diagram is that RTCPeerConnection shields web developers from the myriad complexities that lurk beneath. In this example, pc1 represents the local peer (caller) and pc2 represents the remote peer (callee).
Create an offer and set it as the local description for pc1 and as the remote description for pc2. NAT traversal, peer-to-peer networking, and the requirements for building a server app for user discovery and signaling, are beyond the scope of this article. The code is deliberately verbose in its logging: check the console to understand the order of events. This code initializes variables for the HTML video elements that will display video streams from the local camera (localVideo) and from the camera on the remote client (remoteVideo).
For signaling, this demo uses the Google App Engine Channel API, which enables messaging between JavaScript clients without polling. This causes video from the local camera to be displayed in the localVideo element, by creating an object (Blob) URL for the camera's data stream and then setting that URL as the src for the element. The underlying purpose is to set up a connection, using a STUN server, with onIceCandidate() as the callback (see above for an explanation of ICE, STUN and 'candidate'). The offer creation process here is similar to the no-signaling example above but, in addition, a message is sent to the remote peer, giving a serialized SessionDescription for the offer. XHR works fine for sending signaling messages from the client to the server, but some mechanism is needed for server-to-client messaging: this application uses the Google App Engine Channel API. If the message is an answer from a peer (a response to an offer), RTCPeerConnection sets the remote SessionDescription and communication can begin.
Many existing WebRTC apps only demonstrate communication between web browsers, but gateway servers can enable a WebRTC app running on a browser to interact with devices such as telephones (aka PSTN) and with VOIP systems. As well as audio and video, WebRTC supports real-time communication for other types of data. The RTCDataChannel API enables peer-to-peer exchange of arbitrary data, with low latency and high throughput. Communication occurs directly between browsers, so RTCDataChannel can be much faster than WebSocket even if a relay (TURN) server is required when 'hole punching' to cope with firewalls and NATs fails. There are several ways a real-time communication application or plugin might compromise security.
Unencrypted media or data might be intercepted en route between browsers, or between a browser and a server. Malware or viruses might be installed alongside an apparently innocuous plugin or application.
WebRTC is not a plugin: its components run in the browser sandbox and not in a separate process, components do not require separate installation, and are updated whenever the browser is updated. Camera and microphone access must be granted explicitly and, when the camera or microphone are running, this is clearly shown by the user interface.
The APIs and standards of WebRTC can democratize and decentralize tools for content creation and communication—for telephony, gaming, video production, music making, news gathering and many other applications. We look forward to what JavaScript developers make of WebRTC as it becomes widely implemented. With a couple of important exceptions (which we will get into later) all tags have to be opened and closed. Everything between the opening tag and closing tag are inside that tag and therefore have the attributes that tag gives them. Tags or elements may represent headings, paragraphs, hypertext links, lists, embedded media, and a variety of other structures. Search boxes, text inputs, and other you-type-here fields get better controls for focusing, validating data, interacting with other page elements, sending through email, and more. The first of these is the placeholder attribute, which allows the developer to define the text that will appear in a textbox-based control when its contents are empty. Placeholder text is displayed inside the input field as long as the field is empty and not focused.
While this is convenient for most people, it can be annoying for power users or people with special needs. The autofocus attribute does exactly what it says on the tin: as soon as the page loads, it moves the input focus to a particular input field. They may not do anything special with it, but browsers that don't recognize type="email" will treat it as type="text" and render it as a plain text field. The default value will show up in the field (since it's stored in the value attribute), but the other attributes like min and max will be ignored. For example, you can use HTML5 forms to find out details about your visitors through surveys and feedback, or engage in e-commerce by selling your goods and services to people. After a user inputs all of the form information, they submit the HTML5 form by using the “Submit” button you create within the form. CGI (Common Gateway Interface) is a script written in a language such as Java, or Perl, which runs on a Web server. If a user clicks on a link and it doesn’t take them where you intend, it is considered a “broken link”.
Though it can stream with a given public video file url, sometimes you will need to control the streaming from your server script to provide additional facility like authentication, resume support, sending in partial chunks etc. It can either give it a local path, or even an amazon s3 file location as well(explained later). However, I realized that the script from which I’m building the VideoStream object and executing start() is re-executed 2 or 3 times. But I don’t really get why I should use this script and not simply embedd it with the HTML5 tag? 100% working code, I only had to pass the correct path to my video ))) What a luck, thank you!
Sebuah slide show adalah presentasi dari serangkaian gambar diam pada perangkat layar proyeksi atau layar elektronik, biasanya dalam urutan yang telah diatur sebelumnya. Memutar video di website kadang memang menyulitkan, namun jika sudah mengetahui plugin yang benar dan powerful maka semua akan terasa lebih mudah. Dahulu kita perlu menggunakan Flash untuk membuat efek suara seperti ini, terlihat sangat canggih.
Kali ini kita akan mencoba belajar dengan melihat video tentang HTML5 yang mungkin belum kita ketahui. Intinya disini kita akan belajar bagaimana menghapus dan menambah class pada tag html ketika menu di scroll dengan bantuan jquery.
Responsive web design saat ini sedang banyak dibicarakan oleh web designer dan web developer diseluruh dunia. Imagine it was easy to add video chat and peer-to-peer data sharing to your web application. Integrating RTC technology with existing content, data and services has been difficult and time consuming, particularly on the web. Google bought GIPS, a company which had developed many components required for RTC, such as codecs and echo cancellation techniques.
These includes Skype, Facebook (which uses Skype) and Google Hangouts (which use the Google Talk plugin). Take a look at the cross-browser demo at demo and Chris Wilson's amazing examples using getUserMedia as input for Web Audio.
Often this actually just means that getUserMedia is supported, but not any of the other RTC components.

For example, a stream taken from camera and microphone input has synchronized video and audio tracks.
This would enable streaming from disc, or from arbitrary data sources such as sensors or other inputs. Note that screen capture requires HTTPS and should only be used for development due to it being enabled via a command line flag as explaind in this discuss-webrtc post.
Signaling methods and protocols are not specified by WebRTC: signaling is not part of the RTCPeerConnection API. Here's a code sample from the WebRTC W3C Working Draft, which shows the signaling process in action. The callback argument of this is passed an RTCSessionDescription: Alice's local session description.
The createAnswer() callback is passed an RTCSessionDescription: Bob sets that as the local description and sends it to Alice. Suffice to say that the STUN protocol and its extension TURN are used by the ICE framework to enable RTCPeerConnection to cope with NAT traversal and other network vagaries. This step-by-step guide explains how to build a complete video chat application, including a simple signaling server built with running on a Node server. The Channel API is used in this demo, but other methods (such as WebSocket) could be used instead. So—if a connection hasn't already been made, and a local stream is available, and a channel is ready for signaling, a connection is created and passed the local video stream.
Handlers are then set for each of the RTCPeerConnection events: when a session is connecting or open, and when a remote stream is added or removed. The caller and callee have discovered each other and exchanged information about their capabilities, a call session is initiated, and real-time data communication can begin. The magnificent Cube Slam game uses the API to communicate game state: play a friend or play the bear! An example of this would be a search box where the developer would prefer not to use a label outside the box itself.
If you press the space bar expecting to scroll the page, the page will not scroll because the focus is already in a form input field. But because it's just markup instead of script, the behavior will be consistent across all web sites. Other HTML5 browsers like Safari and Chrome simply render it as a text box - exactly like type="text" - so your users will never know the difference (unless they view-source). You may only want certain kinds of numbers within that range - maybe whole numbers but not fractions or decimals, or something more esoteric like numbers divisible by 10.
We're free to implement them ourselves, or we could reuse one of the many JavaScript frameworks that have already implemented spinbox controls.
When it runs, it creates a global object called Modernizr, which contains a set of Boolean properties for each feature it can detect. Instead of a field for typing, browsers are expected to render type="range" as a slider control.
Most Web servers accommodate the processing of CGI scripts, but you should check first with your web-hosting provider to make sure. In a recent project, I have something similar, where a video was needed to be streamed through PHP to client browser’s HTML5 player. First attempt to load video will be streamed directly from s3 and keep a backup on local machine. When I log off and login again, then the changed file(or second file) is played, but again if change to next file it is playing the previos on only. HTML5 akan bisa  semakin canggih jika digabungkan dengan CSS3 dan Javascript seperti Jquery. Banyak orang bilang HTML5 adalah masa depan website, fitur-fitur baru dari HTML5 cukup banyak. Beberapa website skala besar juga telah mulai migrasi website nya menjadi responsive, agar bisa dibuka di hampir semua device, terutama handphone…. Without it, we're limited in our ability to innovate and develop new ways for people to interact. Google open sourced the technologies developed by GIPS and engaged with relevant standards bodies at the IETF and W3C to ensure industry consensus. The intention is to implement support for other constraints such as aspect ratio, facing mode (front or back camera), frame rate, height and width, along with an applyConstraints() method. For screencast, code and more information, see the HTML5 Rocks update: Screensharing with WebRTC. The code assumes the existence of some signaling mechanism, created in the createSignalingChannel() method. In this process, STUN servers have a single task: to enable a peer behind a NAT to find out its public address and port. In other words, ICE will first use STUN with UDP to directly connect peers and, if that fails, will fall back to a TURN relay server. Once that happens, started is set to true, so a connection won't be started more than once. The placeholder attribute allows the developer to specify text that will show when the value of the control is empty and the control does not have focus. The default step value is 1, and we can omit the step attribute unless we need a different step value.
For example, if your browser supports the canvas API, the Modernizr.canvas property will be true. You can use a CGI script to manage the data, send the data to a database, or even receive the data via e-mail. In addition, you will want to find the location of the server’s CGI-bin within your web server.
If it does not, verify that the path and file name are correct and that the file the link points to is actually uploaded to your Web server. Examining other developers’ HTML pages can provide you with fresh ideas and inspiration in your own design endeavors. Here I will try to share my experience and code that will help you to implement PHP HTML5 video streaming easily and efficiently.
However, at this moment, mp4 is the most widely supported format across major browsers, so using this format is very much recommended.
Also, Feel free to share your experience, if you become successful to solve it, on this post by commenting so that other readers get help if they suffer from similar issue. In my app user selects list of files to play but, only first time selected file only played always. Pada postingan kali ini kami akan menampilkan beberapa website canggih yang dibuat dengan menggunakan HTML5 tersebut.
HTML5 dan CSS3 adalah Masa depan website, kedepan nya HTML5 dan CSS3 akan semakin banyak digunakan….
In this case, there is only one video track and no audio, but it is easy to imagine use cases where there are more: for example, a chat application that gets streams from the front camera, rear camera, microphone, and a 'screenshared' application. The expression 'finding candidates' refers to the process of finding network interfaces and ports. If there's no default value, then the value attribute can be blank or even omitted altogether.
If your browser does not support the canvas API, the Modernizr.canvas property will be false.
The CGI-bin is a directory where CGI scripting is stored, and is where you will need to store the CGI script you create or use within the HTML5 form. This app uses adapter.js, a JavaScript shim, maintained Google with help from the WebRTC community, that abstracts away browser differences and spec changes. You will need to make the necessary changes to the CGI script you use (such as changing script variables, path names, etc.) and then upload the CGI script to your web-hosting server.

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