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webrtc

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webrtc - npm Package Compare versions

Comparing version 0.4.0 to 0.4.1

9

package.json
{
"name": "webrtc",
"version": "0.4.0",
"version": "0.4.1",
"keywords": ["webrtc", "browser"],

@@ -12,7 +12,8 @@ "repository": {

"dependencies": {
"webrtcsupport": "0.3.2",
"webrtcsupport": "0.4.0",
"wildemitter": "0.0.5",
"rtcpeerconnection": "0.2.0",
"getusermedia": "0.2.0",
"hark": "0.1.1"
"getusermedia": "0.2.1",
"hark": "0.1.1",
"mediastream-gain": "0.0.1"
},

@@ -19,0 +20,0 @@ "devDependencies": {

@@ -8,2 +8,3 @@ (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;

var hark = require('hark');
var GainController = require('mediastream-gain');
var log;

@@ -90,4 +91,6 @@

}
self.localStream = self.setupMicVolumeControl(stream);
self.localStream = stream;
self.gainController = new GainController(stream);
// start out somewhat muted if we can track audio

@@ -140,27 +143,2 @@ self.setMicIfEnabled(0.5);

WebRTC.prototype.setupMicVolumeControl = function (stream) {
if (!webrtc.webAudio || !this.config.autoAdjustMic) return stream;
var context = new webkitAudioContext();
var microphone = context.createMediaStreamSource(stream);
var gainFilter = this.gainFilter = context.createGainNode();
var destination = context.createMediaStreamDestination();
var outputStream = destination.stream;
microphone.connect(gainFilter);
gainFilter.connect(destination);
stream.removeTrack(stream.getAudioTracks()[0]);
stream.addTrack(outputStream.getAudioTracks()[0]);
return stream;
};
// sets the gain input on the microphone if web audio
// is available.
WebRTC.prototype.setMicVolume = function (volume) {
if (!webrtc.webAudio) return;
this.gainFilter.gain.value = volume;
};
// We do this as a seperate method in order to

@@ -171,3 +149,3 @@ // still leave the "setMicVolume" as a working

if (!this.config.autoAdjustMic) return;
this.setMicVolume(volume);
this.gainController.setGain(volume);
};

@@ -343,3 +321,3 @@

},{"getusermedia":4,"hark":6,"rtcpeerconnection":3,"webrtcsupport":2,"wildemitter":5}],2:[function(require,module,exports){
},{"getusermedia":3,"hark":6,"mediastream-gain":7,"rtcpeerconnection":4,"webrtcsupport":2,"wildemitter":5}],2:[function(require,module,exports){
// created by @HenrikJoreteg

@@ -357,3 +335,3 @@ var PC = window.mozRTCPeerConnection || window.webkitRTCPeerConnection || window.RTCPeerConnection;

var screenSharing = navigator.userAgent.match('Chrome') && parseInt(navigator.userAgent.match(/Chrome\/(.*) /)[1], 10) >= 26;
var webAudio = !!window.webkitAudioContext;
var AudioContext = window.webkitAudioContext || window.AudioContext;

@@ -365,4 +343,5 @@ // export support flags and constructors.prototype && PC

prefix: prefix,
webAudio: webAudio,
webAudio: !!AudioContext.prototype.createMediaStreamSource,
screenSharing: screenSharing,
AudioContext: AudioContext,
PeerConnection: PC,

@@ -373,3 +352,3 @@ SessionDescription: SessionDescription,

},{}],4:[function(require,module,exports){
},{}],3:[function(require,module,exports){
// getUserMedia helper by @HenrikJoreteg

@@ -575,3 +554,30 @@ var func = (navigator.getUserMedia ||

},{}],3:[function(require,module,exports){
},{}],8:[function(require,module,exports){
// created by @HenrikJoreteg
var PC = window.mozRTCPeerConnection || window.webkitRTCPeerConnection || window.RTCPeerConnection;
var IceCandidate = window.mozRTCIceCandidate || window.RTCIceCandidate;
var SessionDescription = window.mozRTCSessionDescription || window.RTCSessionDescription;
var prefix = function () {
if (window.mozRTCPeerConnection) {
return 'moz';
} else if (window.webkitRTCPeerConnection) {
return 'webkit';
}
}();
var screenSharing = navigator.userAgent.match('Chrome') && parseInt(navigator.userAgent.match(/Chrome\/(.*) /)[1], 10) >= 26;
var webAudio = !!window.webkitAudioContext;
// export support flags and constructors.prototype && PC
module.exports = {
support: !!PC,
dataChannel: !!(PC && PC.prototype && PC.prototype.createDataChannel),
prefix: prefix,
webAudio: webAudio,
screenSharing: screenSharing,
PeerConnection: PC,
SessionDescription: SessionDescription,
IceCandidate: IceCandidate
};
},{}],4:[function(require,module,exports){
var WildEmitter = require('wildemitter');

@@ -714,3 +720,3 @@ var webrtc = require('webrtcsupport');

},{"webrtcsupport":2,"wildemitter":5}],6:[function(require,module,exports){
},{"webrtcsupport":8,"wildemitter":5}],6:[function(require,module,exports){
var WildEmitter = require('wildemitter');

@@ -808,4 +814,51 @@

},{"wildemitter":5}]},{},[1])(1)
},{"wildemitter":5}],7:[function(require,module,exports){
var support = require('webrtcsupport');
function GainController(stream) {
this.support = support.webAudio;
// set our starting value
this.gain = 1;
if (this.support) {
var context = this.context = new support.AudioContext();
this.microphone = context.createMediaStreamSource(stream);
this.gainFilter = context.createGain();
this.destination = context.createMediaStreamDestination();
this.outputStream = this.destination.stream;
this.microphone.connect(this.gainFilter);
this.gainFilter.connect(this.destination);
stream.removeTrack(stream.getAudioTracks()[0]);
stream.addTrack(this.outputStream.getAudioTracks()[0]);
}
this.stream = stream;
}
// setting
GainController.prototype.setGain = function (val) {
// check for support
if (!this.support) return;
this.gainFilter.gain.value = val;
this.gain = val;
};
GainController.prototype.getGain = function () {
return this.gain;
};
GainController.prototype.off = function () {
return this.setGain(0);
};
GainController.prototype.on = function () {
this.setGain(1);
};
module.exports = GainController;
},{"webrtcsupport":2}]},{},[1])(1)
});
;

@@ -6,2 +6,3 @@ var webrtc = require('webrtcsupport');

var hark = require('hark');
var GainController = require('mediastream-gain');
var log;

@@ -88,4 +89,6 @@

}
self.localStream = self.setupMicVolumeControl(stream);
self.localStream = stream;
self.gainController = new GainController(stream);
// start out somewhat muted if we can track audio

@@ -138,27 +141,2 @@ self.setMicIfEnabled(0.5);

WebRTC.prototype.setupMicVolumeControl = function (stream) {
if (!webrtc.webAudio || !this.config.autoAdjustMic) return stream;
var context = new webkitAudioContext();
var microphone = context.createMediaStreamSource(stream);
var gainFilter = this.gainFilter = context.createGainNode();
var destination = context.createMediaStreamDestination();
var outputStream = destination.stream;
microphone.connect(gainFilter);
gainFilter.connect(destination);
stream.removeTrack(stream.getAudioTracks()[0]);
stream.addTrack(outputStream.getAudioTracks()[0]);
return stream;
};
// sets the gain input on the microphone if web audio
// is available.
WebRTC.prototype.setMicVolume = function (volume) {
if (!webrtc.webAudio) return;
this.gainFilter.gain.value = volume;
};
// We do this as a seperate method in order to

@@ -169,3 +147,3 @@ // still leave the "setMicVolume" as a working

if (!this.config.autoAdjustMic) return;
this.setMicVolume(volume);
this.gainController.setGain(volume);
};

@@ -172,0 +150,0 @@

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