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BSD-3-Clause

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WebRTC code samples

This is a repository for the WebRTC Javascript code samples.

Some of the samples use new browser features. They may only work in Chrome Canary and/or Firefox Beta, and may require flags to be set.

All of the samples use adapter.js, a shim to insulate apps from spec changes and prefix differences. In fact, the standards and protocols used for WebRTC implementations are highly stable, and there are only a few prefixed names. For full interop information, see webrtc.org/web-apis/interop.

In Chrome and Opera, all samples that use getUserMedia() must be run from a server. Calling getUserMedia() from a file:// URL will work in Firefox, but fail silently in Chrome and Opera.

webrtc.org/testing lists command line flags useful for development and testing with Chrome.

For more information about WebRTC, we maintain a list of WebRTC Resources. If you've never worked with WebRTC, we recommend you start with the 2013 Google I/O WebRTC presentation.

Patches and issues welcome! See CONTRIBUTING for instructions. All contributors must sign a contributor license agreement before code can be accepted. Please complete the agreement for an individual or a corporation as appropriate. The Developer's Guide for this repo has more information about code style, structure and validation. Head over to test/README.md and get started developing.

The demos

getUserMedia

Basic getUserMedia demo

getUserMedia + canvas

getUserMedia + canvas + CSS Filters

getUserMedia with resolution constraints

getUserMedia with camera, mic and speaker selection

Audio-only getUserMedia output to local audio element

Audio-only getUserMedia displaying volume

Face tracking

Record stream

Stream capture

Stream from a canvas element to a video element

Stream from a canvas element to a peer connection

Devices

Select camera, microphone and speaker

Select media source and audio output

RTCPeerConnection

Basic peer connection

Audio-only peer connection

Multiple peer connections at once

Forward output of one peer connection into another

Munge SDP parameters

Use pranswer when setting up a peer connection

Adjust constraints, view stats

Display createOffer output

Use RTCDTMFSender

Display peer connection states

ICE candidate gathering from STUN/TURN servers

Do an ICE restart

Web Audio output as input to peer connection

Peer connection as input to Web Audio

RTCDataChannel

Transmit text

Transfer a file

Transfer data

Video chat

AppRTC video chat client powered by Google App Engine

AppRTC URL parameters

Copyright (c) 2014, The WebRTC project authors. All rights reserved. Redistribution and use in source and binary forms, with or without modification, are permitted provided that the following conditions are met: * Redistributions of source code must retain the above copyright notice, this list of conditions and the following disclaimer. * Redistributions in binary form must reproduce the above copyright notice, this list of conditions and the following disclaimer in the documentation and/or other materials provided with the distribution. * Neither the name of Google nor the names of its contributors may be used to endorse or promote products derived from this software without specific prior written permission. THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.

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