Added trailing slashes to links to avoid redirect to HTTP URL

Author: samdutton

README.md

@@ -1,4 +1,4 @@
-[![Build Status](https://travis-ci.org/webrtc/samples.svg?branch=master)](https://travis-ci.org/webrtc/samples)
+[![Build Status](https://travis-ci.org/webrtc/samples.svg?branch=master)](https://travis-ci.org/webrtc/samples/)
 
 # WebRTC code samples #
 
@@ -20,65 +20,65 @@ Patches and issues welcome! See [CONTRIBUTING](https://github.com/webrtc/samples
 
 ### getUserMedia ###
 
-[Basic getUserMedia demo](https://webrtc.github.io/samples/src/content/getusermedia/gum)
+[Basic getUserMedia demo](https://webrtc.github.io/samples/src/content/getusermedia/gum/)
 
-[getUserMedia + canvas](https://webrtc.github.io/samples/src/content/getusermedia/canvas)
+[getUserMedia + canvas](https://webrtc.github.io/samples/src/content/getusermedia/canvas/)
 
-[getUserMedia + canvas + CSS Filters](https://webrtc.github.io/samples/src/content/getusermedia/filter)
+[getUserMedia + canvas + CSS Filters](https://webrtc.github.io/samples/src/content/getusermedia/filter/)
 
-[getUserMedia with resolution constraints](https://webrtc.github.io/samples/src/content/getusermedia/resolution)
+[getUserMedia with resolution constraints](https://webrtc.github.io/samples/src/content/getusermedia/resolution/)
 
-[getUserMedia with camera, mic and speaker selection](https://webrtc.github.io/samples/src/content/getusermedia/source)
+[getUserMedia with camera, mic and speaker selection](https://webrtc.github.io/samples/src/content/getusermedia/source/)
 
-[Audio-only getUserMedia output to local audio element](https://webrtc.github.io/samples/src/content/getusermedia/audio)
+[Audio-only getUserMedia output to local audio element](https://webrtc.github.io/samples/src/content/getusermedia/audio/)
 
-[Audio-only getUserMedia displaying volume](https://webrtc.github.io/samples/src/content/getusermedia/volume)
+[Audio-only getUserMedia displaying volume](https://webrtc.github.io/samples/src/content/getusermedia/volume/)
 
-[Face tracking](https://webrtc.github.io/samples/src/content/getusermedia/face)
+[Face tracking](https://webrtc.github.io/samples/src/content/getusermedia/face/)
 
 ### Devices ###
 
-[Select camera, microphone and speaker](https://webrtc.github.io/samples/src/content/devices/input-output)
+[Select camera, microphone and speaker](https://webrtc.github.io/samples/src/content/devices/input-output/)
 
-[Select media source and audio output](https://webrtc.github.io/samples/src/content/devices/multi)
+[Select media source and audio output](https://webrtc.github.io/samples/src/content/devices/multi/)
 
 ### RTCPeerConnection ###
 
-[Basic peer connection](https://webrtc.github.io/samples/src/content/peerconnection/pc1)
+[Basic peer connection](https://webrtc.github.io/samples/src/content/peerconnection/pc1/)
 
-[Audio-only peer connection](https://webrtc.github.io/samples/src/content/peerconnection/audio)
+[Audio-only peer connection](https://webrtc.github.io/samples/src/content/peerconnection/audio/)
 
-[Multiple peer connections at once](https://webrtc.github.io/samples/src/content/peerconnection/multiple)
+[Multiple peer connections at once](https://webrtc.github.io/samples/src/content/peerconnection/multiple/)
 
-[Forward output of one peer connection into another](https://webrtc.github.io/samples/src/content/peerconnection/multiple-relay)
+[Forward output of one peer connection into another](https://webrtc.github.io/samples/src/content/peerconnection/multiple-relay/)
 
-[Munge SDP parameters](https://webrtc.github.io/samples/src/content/peerconnection/munge-sdp)
+[Munge SDP parameters](https://webrtc.github.io/samples/src/content/peerconnection/munge-sdp/)
 
-[Use pranswer when setting up a peer connection](https://webrtc.github.io/samples/src/content/peerconnection/pr-answer)
+[Use pranswer when setting up a peer connection](https://webrtc.github.io/samples/src/content/peerconnection/pr-answer/)
 
-[Adjust constraints, view stats](https://webrtc.github.io/samples/src/content/peerconnection/constraints)
+[Adjust constraints, view stats](https://webrtc.github.io/samples/src/content/peerconnection/constraints/)
 
-[Display createOffer output](https://webrtc.github.io/samples/src/content/peerconnection/create-offer)
+[Display createOffer output](https://webrtc.github.io/samples/src/content/peerconnection/create-offer/)
 
-[Use RTCDTMFSender](https://webrtc.github.io/samples/src/content/peerconnection/dtmf)
+[Use RTCDTMFSender](https://webrtc.github.io/samples/src/content/peerconnection/dtmf/)
 
-[Display peer connection states](https://webrtc.github.io/samples/src/content/peerconnection/states)
+[Display peer connection states](https://webrtc.github.io/samples/src/content/peerconnection/states/)
 
-[ICE candidate gathering from STUN/TURN servers](https://webrtc.github.io/samples/src/content/peerconnection/trickle-ice)
+[ICE candidate gathering from STUN/TURN servers](https://webrtc.github.io/samples/src/content/peerconnection/trickle-ice/)
 
-[Web Audio output as input to peer connection](https://webrtc.github.io/samples/src/content/peerconnection/webaudio-input)
+[Web Audio output as input to peer connection](https://webrtc.github.io/samples/src/content/peerconnection/webaudio-input/)
 
 ### RTCDataChannel ###
 
-[Transmit text](https://webrtc.github.io/samples/src/content/datachannel/basic)
+[Transmit text](https://webrtc.github.io/samples/src/content/datachannel/basic/)
 
-[Transfer a file](https://webrtc.github.io/samples/src/content/datachannel/filetransfer)
+[Transfer a file](https://webrtc.github.io/samples/src/content/datachannel/filetransfer/)
 
-[Transfer data](https://webrtc.github.io/samples/src/content/datachannel/datatransfer)
+[Transfer data](https://webrtc.github.io/samples/src/content/datachannel/datatransfer/)
 
 ### Video chat ###
 
-[AppRTC video chat client](https://apprtc.appspot.com) powered by Google App Engine
+[AppRTC video chat client](https://apprtc.appspot.com/) powered by Google App Engine
 
 [AppRTC URL parameters](https://apprtc.appspot.com/params.html)
 

index.html

@@ -75,67 +75,67 @@ <h2 id="the-demos">The demos</h2>
 
       <h3 id="getusermedia">getUserMedia</h3>
 
-      <p><a href="src/content/getusermedia/gum">Basic getUserMedia demo</a></p>
+      <p><a href="src/content/getusermedia/gum/">Basic getUserMedia demo</a></p>
 
-      <p><a href="src/content/getusermedia/canvas">Use getUserMedia with canvas</a></p>
+      <p><a href="src/content/getusermedia/canvas/">Use getUserMedia with canvas</a></p>
 
-      <p><a href="src/content/getusermedia/filter">Use getUserMedia with canvas and CSS filters</a></p>
+      <p><a href="src/content/getusermedia/filter/">Use getUserMedia with canvas and CSS filters</a></p>
 
-      <p><a href="src/content/getusermedia/resolution">Choose camera resolution</a></p>
+      <p><a href="src/content/getusermedia/resolution/">Choose camera resolution</a></p>
 
 
 
-      <p><a href="src/content/getusermedia/audio">Audio-only getUserMedia() output to local audio element</a></p>
+      <p><a href="src/content/getusermedia/audio/">Audio-only getUserMedia() output to local audio element</a></p>
 
-      <p><a href="src/content/getusermedia/volume">Audio-only getUserMedia() displaying volume</a></p>
+      <p><a href="src/content/getusermedia/volume/">Audio-only getUserMedia() displaying volume</a></p>
 
-      <p><a href="src/content/getusermedia/face">Face tracking, using getUserMedia and canvas</a></p>
+      <p><a href="src/content/getusermedia/face/">Face tracking, using getUserMedia and canvas</a></p>
 
 
       <h3 id="devices">Devices</h3>
 
-      <p><a href="src/content/devices/input-output">Choose camera, microphone and speaker</a></p>
+      <p><a href="src/content/devices/input-output/">Choose camera, microphone and speaker</a></p>
 
-      <p><a href="src/content/devices/multi">Choose media source and audio output</a></p>
+      <p><a href="src/content/devices/multi/">Choose media source and audio output</a></p>
 
 
       <h3 id="peerconnection">RTCPeerConnection</h3>
 
-      <p><a href="src/content/peerconnection/pc1">Basic peer connection demo</a></p>
+      <p><a href="src/content/peerconnection/pc1/">Basic peer connection demo</a></p>
 
-      <p><a href="src/content/peerconnection/audio">Audio-only peer connection demo</a></p>
+      <p><a href="src/content/peerconnection/audio/">Audio-only peer connection demo</a></p>
 
-      <p><a href="src/content/peerconnection/multiple">Multiple peer connections at once</a></p>
+      <p><a href="src/content/peerconnection/multiple/">Multiple peer connections at once</a></p>
 
-      <p><a href="src/content/peerconnection/multiple-relay">Forward the output of one PC into another</a></p>
+      <p><a href="src/content/peerconnection/multiple-relay/">Forward the output of one PC into another</a></p>
 
-      <p><a href="src/content/peerconnection/munge-sdp">Munge SDP parameters</a></p>
+      <p><a href="src/content/peerconnection/munge-sdp/">Munge SDP parameters</a></p>
 
-      <p><a href="src/content/peerconnection/pr-answer">Use pranswer when setting up a peer connection</a></p>
+      <p><a href="src/content/peerconnection/pr-answer/">Use pranswer when setting up a peer connection</a></p>
 
-      <p><a href="src/content/peerconnection/constraints">Constraints and stats</a></p>
+      <p><a href="src/content/peerconnection/constraints/">Constraints and stats</a></p>
 
-      <p><a href="src/content/peerconnection/create-offer">Display createOffer output for various scenarios</a></p>
+      <p><a href="src/content/peerconnection/create-offer/">Display createOffer output for various scenarios</a></p>
 
-      <p><a href="src/content/peerconnection/dtmf">Use RTCDTMFSender</a></p>
+      <p><a href="src/content/peerconnection/dtmf/">Use RTCDTMFSender</a></p>
 
-      <p><a href="src/content/peerconnection/states">Display peer connection states</a></p>
+      <p><a href="src/content/peerconnection/states/">Display peer connection states</a></p>
 
-      <p><a href="src/content/peerconnection/trickle-ice">ICE candidate gathering from STUN/TURN servers</a></p>
+      <p><a href="src/content/peerconnection/trickle-ice/">ICE candidate gathering from STUN/TURN servers</a></p>
 
-      <p><a href="src/content/peerconnection/webaudio-input">Web Audio output as input to peer connection</a></p>
+      <p><a href="src/content/peerconnection/webaudio-input/">Web Audio output as input to peer connection</a></p>
 
       <h3 id="datachannel">RTCDataChannel</h3>
 
-      <p><a href="src/content/datachannel/basic">Transmit text</a></p>
+      <p><a href="src/content/datachannel/basic/">Transmit text</a></p>
 
-      <p><a href="src/content/datachannel/filetransfer">Transfer a file</a></p>
+      <p><a href="src/content/datachannel/filetransfer/">Transfer a file</a></p>
 
-      <p><a href="https://webrtc.github.io/samples/src/content/datachannel/datatransfer">Transfer data</a></p>
+      <p><a href="src/content/datachannel/datatransfer/">Transfer data</a></p>
 
       <h3 id="videoChat">Video chat</h3>
 
-      <p><a href="//apprtc.appspot.com">AppRTC video chat client</a> powered by Google App Engine</p>
+      <p><a href="//apprtc.appspot.com/">AppRTC video chat client</a> powered by Google App Engine</p>
 
       <p><a href="//apprtc.appspot.com/params.html">AppRTC URL parameters</a></p>