webrtc - npm Package Compare versions

Comparing version 1.0.0 to 1.0.1

package.json

 {
   "name": "webrtc",
-  "version": "1.0.0",
+  "version": "1.0.1",
   "keywords": ["webrtc", "browser"],
@@ -14,3 +14,3 @@ "repository": {
     "wildemitter": "0.0.5",
-    "rtcpeerconnection": "1.0.0",
+    "rtcpeerconnection": "1.0.1",
     "getusermedia": "0.2.1",
@@ -17,0 +17,0 @@ "hark": "0.1.1",

webrtc.bundle.js

@@ -649,4 +649,97 @@ (function(e){if("function"==typeof bootstrap)bootstrap("webrtc",e);else if("object"==typeof exports)module.exports=e();else if("function"==typeof define&&define.amd)define(e);else if("undefined"!=typeof ses){if(!ses.ok())return;ses.makeWebRTC=e}else"undefined"!=typeof window?window.WebRTC=e():global.WebRTC=e()})(function(){var define,ses,bootstrap,module,exports;
 
-},{}],4:[function(require,module,exports){
+},{}],6:[function(require,module,exports){
 var WildEmitter = require('wildemitter');
+
+function getMaxVolume (analyser, fftBins) {
+  var maxVolume = -Infinity;
+  analyser.getFloatFrequencyData(fftBins);
+
+  for(var i=0, ii=fftBins.length; i < ii; i++) {
+    if (fftBins[i] > maxVolume && fftBins[i] < 0) {
+      maxVolume = fftBins[i];
+    }
+  };
+
+  return maxVolume;
+}
+
+
+module.exports = function(stream, options) {
+  var harker = new WildEmitter();
+
+  // make it not break in non-supported browsers
+  if (!window.webkitAudioContext) return harker;
+
+  //Config
+  var options = options || {},
+      smoothing = (options.smoothing || 0.5),
+      interval = (options.interval || 100),
+      threshold = options.threshold,
+      play = options.play;
+
+  //Setup Audio Context
+  var audioContext = new webkitAudioContext();
+  var sourceNode, fftBins, analyser;
+
+  analyser = audioContext.createAnalyser();
+  analyser.fftSize = 512;
+  analyser.smoothingTimeConstant = smoothing;
+  fftBins = new Float32Array(analyser.fftSize);
+
+  if (stream.jquery) stream = stream[0];
+  if (stream instanceof HTMLAudioElement) {
+    //Audio Tag
+    sourceNode = audioContext.createMediaElementSource(stream);
+    if (typeof play === 'undefined') play = true;
+    threshold = threshold || -65;
+  } else {
+    //WebRTC Stream
+    sourceNode = audioContext.createMediaStreamSource(stream);
+    threshold = threshold || -45;
+  }
+
+  sourceNode.connect(analyser);
+  if (play) analyser.connect(audioContext.destination);
+
+  harker.speaking = false;
+
+  harker.setThreshold = function(t) {
+    threshold = t;
+  };
+
+  harker.setInterval = function(i) {
+    interval = i;
+  };
+
+  // Poll the analyser node to determine if speaking
+  // and emit events if changed
+  var looper = function() {
+    setTimeout(function() {
+      var currentVolume = getMaxVolume(analyser, fftBins);
+
+      harker.emit('volume_change', currentVolume, threshold);
+
+      if (currentVolume > threshold) {
+        if (!harker.speaking) {
+          harker.speaking = true;
+          harker.emit('speaking');
+        }
+      } else {
+        if (harker.speaking) {
+          harker.speaking = false;
+          harker.emit('stopped_speaking');
+        }
+      }
+
+      looper();
+    }, interval);
+  };
+  looper();
+
+
+  return harker;
+}
+
+},{"wildemitter":5}],4:[function(require,module,exports){
+var WildEmitter = require('wildemitter');
 var webrtc = require('webrtcsupport');
@@ -656,2 +749,3 @@
 function PeerConnection(config, constraints) {
+    var item;
     this.pc = new webrtc.PeerConnection(config, constraints);
@@ -671,3 +765,14 @@ WildEmitter.call(this);
 
-    if (config.debug) {
+    // whether to use SDP hack for faster data transfer
+    this.config = {
+        debug: false,
+        sdpHack: true
+    };
+
+    // apply our config
+    for (item in config) {
+        this.config[item] = config[item];
+    }
+
+    if (this.config.debug) {
         this.on('*', function (eventName, event) {
@@ -712,6 +817,7 @@ var logger = config.logger || console;
     this.pc.createOffer(
-        function (sessionDescription) {
-            self.pc.setLocalDescription(sessionDescription);
-            self.emit('offer', sessionDescription);
-            if (callback) callback(null, sessionDescription);
+        function (offer) {
+            offer.sdp = self._applySdpHack(offer.sdp);
+            self.pc.setLocalDescription(offer);
+            self.emit('offer', offer);
+            if (callback) callback(null, offer);
         },
@@ -779,6 +885,7 @@ function (err) {
     this.pc.createAnswer(
-        function (sessionDescription) {
-            self.pc.setLocalDescription(sessionDescription);
-            self.emit('answer', sessionDescription);
-            if (cb) cb(null, sessionDescription);
+        function (answer) {
+            answer.sdp = self._applySdpHack(answer.sdp);
+            self.pc.setLocalDescription(answer);
+            self.emit('answer', answer);
+            if (cb) cb(null, answer);
         }, function (err) {
@@ -813,2 +920,14 @@ self.emit('error', err);
 
+// SDP hack for increasing AS (application specific) data transfer speed allowed in chrome
+PeerConnection.prototype._applySdpHack = function (sdp) {
+    if (!this.config.sdpHack) return sdp;
+    var parts = sdp.split('b=AS:30');
+    if (parts.length === 2) {
+        // increase max data transfer bandwidth to 100 Mbps
+        return parts[0] + 'b=AS:102400' + parts[1];
+    } else {
+        return sdp;
+    }
+};
+
 module.exports = PeerConnection;
@@ -863,97 +982,4 @@
 
-},{"webrtcsupport":2}],6:[function(require,module,exports){
-var WildEmitter = require('wildemitter');
-
-function getMaxVolume (analyser, fftBins) {
-  var maxVolume = -Infinity;
-  analyser.getFloatFrequencyData(fftBins);
-
-  for(var i=0, ii=fftBins.length; i < ii; i++) {
-    if (fftBins[i] > maxVolume && fftBins[i] < 0) {
-      maxVolume = fftBins[i];
-    }
-  };
-
-  return maxVolume;
-}
-
-
-module.exports = function(stream, options) {
-  var harker = new WildEmitter();
-
-  // make it not break in non-supported browsers
-  if (!window.webkitAudioContext) return harker;
-
-  //Config
-  var options = options || {},
-      smoothing = (options.smoothing || 0.5),
-      interval = (options.interval || 100),
-      threshold = options.threshold,
-      play = options.play;
-
-  //Setup Audio Context
-  var audioContext = new webkitAudioContext();
-  var sourceNode, fftBins, analyser;
-
-  analyser = audioContext.createAnalyser();
-  analyser.fftSize = 512;
-  analyser.smoothingTimeConstant = smoothing;
-  fftBins = new Float32Array(analyser.fftSize);
-
-  if (stream.jquery) stream = stream[0];
-  if (stream instanceof HTMLAudioElement) {
-    //Audio Tag
-    sourceNode = audioContext.createMediaElementSource(stream);
-    if (typeof play === 'undefined') play = true;
-    threshold = threshold || -65;
-  } else {
-    //WebRTC Stream
-    sourceNode = audioContext.createMediaStreamSource(stream);
-    threshold = threshold || -45;
-  }
-
-  sourceNode.connect(analyser);
-  if (play) analyser.connect(audioContext.destination);
-
-  harker.speaking = false;
-
-  harker.setThreshold = function(t) {
-    threshold = t;
-  };
-
-  harker.setInterval = function(i) {
-    interval = i;
-  };
-
-  // Poll the analyser node to determine if speaking
-  // and emit events if changed
-  var looper = function() {
-    setTimeout(function() {
-      var currentVolume = getMaxVolume(analyser, fftBins);
-
-      harker.emit('volume_change', currentVolume, threshold);
-
-      if (currentVolume > threshold) {
-        if (!harker.speaking) {
-          harker.speaking = true;
-          harker.emit('speaking');
-        }
-      } else {
-        if (harker.speaking) {
-          harker.speaking = false;
-          harker.emit('stopped_speaking');
-        }
-      }
-
-      looper();
-    }, interval);
-  };
-  looper();
-
-
-  return harker;
-}
-
-},{"wildemitter":5}]},{},[1])(1)
+},{"webrtcsupport":2}]},{},[1])(1)
 });
 ;

No alert changes

Improved metrics

Total package byte prevSize

44002

increased by1.55%

Lines of code

1184

increased by1.98%

Dependency changes

• + Addedrtcpeerconnection@1.0.1(transitive)

• - Removedrtcpeerconnection@1.0.0(transitive)

• Updatedrtcpeerconnection@1.0.1