Rename 'mediaType' by 'mediaTypes'

versatica · Feb 3, 2013 · 42b3749 · 42b3749
1 parent 528d989
commit 42b3749
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Showing 2 changed files with 12 additions and 12 deletions.
diff --git a/src/MediaSession.js b/src/MediaSession.js
@@ -27,11 +27,11 @@ JsSIP.MediaSession.prototype = {
    * <br> -- If the user consents, create a peerConnection.
    * <br> -- If the user doesn't consent, fire onFailure callback.
    *
-   * @param {Object} mediaType {audio:true/false, video:true/false}
+   * @param {Object} mediaTypes {audio:true/false, video:true/false}
    * @param {Function} onSuccess
    * @param {Function} onFailure
    */
-  startCaller: function(mediaType, onSuccess, onFailure) {
+  startCaller: function(mediaTypes, onSuccess, onFailure) {
     var self = this;
 
     /** @private */
@@ -53,7 +53,7 @@ JsSIP.MediaSession.prototype = {
       onFailure(e);
     }
 
-    this.getUserMedia(mediaType, onGetUserMediaSuccess, onGetUserMediaFailure);
+    this.getUserMedia(mediaTypes, onGetUserMediaSuccess, onGetUserMediaFailure);
   },
 
   /**
@@ -177,11 +177,11 @@ JsSIP.MediaSession.prototype = {
   },
 
   /**
-  * @param {Object} mediaType
+  * @param {Object} mediaTypes
   * @param {Function} onSuccess
   * @param {Function} onFailure
   */
-  getUserMedia: function(mediaType, onSuccess, onFailure) {
+  getUserMedia: function(mediaTypes, onSuccess, onFailure) {
     var self = this;
 
     function getSuccess(stream) {
@@ -204,7 +204,7 @@ JsSIP.MediaSession.prototype = {
 
     // Get User Media
     console.log(JsSIP.C.LOG_MEDIA_SESSION +"Requesting access to local media.");
-    navigator.webkitGetUserMedia(mediaType, getSuccess, getFailure);
+    navigator.webkitGetUserMedia(mediaTypes, getSuccess, getFailure);
 
   },
 

diff --git a/src/Session.js b/src/Session.js
@@ -74,7 +74,7 @@ JsSIP.Session.prototype.init_incoming = function(request) {
 };
 
 JsSIP.Session.prototype.connect = function(target, views, options) {
-  var event, eventHandlers, request, selfView, remoteView, mediaType, extraHeaders, requestParams;
+  var event, eventHandlers, request, selfView, remoteView, mediaTypes, extraHeaders, requestParams;
 
   // Check UA Status
   JsSIP.Utils.checkUAStatus(this.ua);
@@ -100,7 +100,7 @@ JsSIP.Session.prototype.connect = function(target, views, options) {
   options = options || {};
   selfView = views.selfView || null;
   remoteView = views.remoteView || null;
-  mediaType = options.mediaType || {audio: true, video: true};
+  mediaTypes = options.mediaTypes || {audio: true, video: true};
   extraHeaders = options.extraHeaders || [];
   eventHandlers = options.eventHandlers || {};
 
@@ -155,7 +155,7 @@ JsSIP.Session.prototype.connect = function(target, views, options) {
 
   this.newSession('local', request, target);
   this.connecting('local', request, target);
-  this.sendInitialRequest(mediaType);
+  this.sendInitialRequest(mediaTypes);
 };
 
 /**
@@ -1003,7 +1003,7 @@ JsSIP.Session.prototype.sendDTMF = function(tones, options) {
 /**
  * @private
  */
-JsSIP.Session.prototype.sendInitialRequest = function(mediaType) {
+JsSIP.Session.prototype.sendInitialRequest = function(mediaTypes) {
   var
     self = this,
     request_sender = new JsSIP.RequestSender(self, this.ua);
@@ -1019,7 +1019,7 @@ JsSIP.Session.prototype.sendInitialRequest = function(mediaType) {
 
     // Hack to quit m=video section from sdp defined in http://code.google.com/p/webrtc/issues/detail?id=935
     // To be deleted when the fix arrives to chrome stable version
-    if (!mediaType.video) {
+    if (!mediaTypes.video) {
       if (self.request.body.indexOf('m=video') !== -1){
         self.request.body = self.request.body.substring(0, self.request.body.indexOf('m=video'));
       }
@@ -1038,7 +1038,7 @@ JsSIP.Session.prototype.sendInitialRequest = function(mediaType) {
     }
   }
 
-  self.mediaSession.startCaller(mediaType, onMediaSuccess, onMediaFailure);
+  self.mediaSession.startCaller(mediaTypes, onMediaSuccess, onMediaFailure);
 };