@twilio/webrtc - npm Package Compare versions

Comparing version 2.3.0 to 2.4.0

CHANGELOG.md

@@ -0,1 +1,14 @@
+2.4.0 (November 11, 2019)
+=========================
+
+New Features
+------------
+
+- ChromeRTCPeerConnection will now support Unified Plan SDPs in Chrome 72 and above. (JSDK-2313)
+
+Bug Fixes
+---------
+
+- Fixed a bug where `audioLevel` returned by getStats() was not in the range [0-32767]. (JSDK-2318)
+
 2.3.0 (March 18, 2019)
@@ -40,3 +53,3 @@ ======================
 - Removed workaround for this Chrome [bug](https://bugs.chromium.org/p/chromium/issues/detail?id=774303).
-  Now, we no longer suppress the RTCPeerConnection's native `RTCTrackEvent`. 
+  Now, we no longer suppress the RTCPeerConnection's native `RTCTrackEvent`.
 - Worked around the [deprecation](https://blog.mozilla.org/webrtc/getstats-isremote-65/) of the
@@ -155,3 +168,3 @@ `isRemote` property in `RTCInboundRTPStreamStats` and `RTCOutboundRTPStreamStats` in Firefox.
 
-- `RTCSessionDescription` properties are now read-only, and therefore standards-compliant. (JSDK-1503)  
+- `RTCSessionDescription` properties are now read-only, and therefore standards-compliant. (JSDK-1503)
 
@@ -158,0 +171,0 @@ 1.0.0 (August 17, 2017)

lib/getstats.js

@@ -17,2 +17,4 @@ 'use strict';
 
+var CHROME_LEGACY_MAX_AUDIO_LEVEL = 32767;
+
 /**
@@ -574,2 +576,3 @@ * Get the standardized {@link RTCPeerConnection} statistics.
   if (typeof audioLevel === 'number') {
+    audioLevel = Math.round(audioLevel * CHROME_LEGACY_MAX_AUDIO_LEVEL);
     if (isRemote) {
@@ -576,0 +579,0 @@ standardizedStats.audioOutputLevel = audioLevel;

lib/rtcpeerconnection/chrome.js

@@ -36,3 +36,3 @@ /* globals RTCDataChannel, RTCPeerConnection, RTCSessionDescription */
     ? { iceTransports: configuration.iceTransportPolicy }
-    : {}, { sdpSemantics: 'plan-b' }, configuration);
+    : {}, configuration);
 
@@ -151,8 +151,2 @@ util.interceptEvent(this, 'datachannel');
   if (RTCPeerConnection.prototype.addTrack) {
-    if (sdpFormat === 'unified') {
-      sender = getActiveSenders(this._peerConnection).get(track)
-        || this._peerConnection.addTrack.apply(this._peerConnection, args);
-      this._senders.set(track, sender);
-      return sender;
-    }
     return this._peerConnection.addTrack.apply(this._peerConnection, args);
@@ -170,18 +164,2 @@ }
 
-// NOTE(mmalavalli): This shim works around a Chrome bug in "unified-plan"
-// SDPs where adding a MediaStreamTrack that was previously added and removed
-// generates an SDP where the MSID does not match the MediaStreamTrack ID.
-//
-// Chrome bug: https://bugs.chromium.org/p/chromium/issues/detail?id=894231
-//
-if (RTCPeerConnection.prototype.addTransceiver) {
-  ChromeRTCPeerConnection.prototype.addTransceiver = function addTransceiver() {
-    var transceiver = this._peerConnection.addTransceiver.apply(this._peerConnection, arguments);
-    var sender = transceiver.sender;
-    var track = sender.track;
-    this._senders.set(track, sender);
-    return transceiver;
-  };
-}
-
 // NOTE(mmalavalli): This shim supports our limited case of removing
@@ -191,8 +169,3 @@ // MediaStreamTracks from one MediaStream. It has been implemented this
 // supported natively in Chrome.
-// NOTE(mmalavalli): This shim also works around a Chrome bug in "unified-plan"
-// SDPs where adding a MediaStreamTrack that was previously added and removed
-// generates an SDP where the MSID does not match the MediaStreamTrack ID.
 //
-// Chrome bug: https://bugs.chromium.org/p/chromium/issues/detail?id=894231
-//
 ChromeRTCPeerConnection.prototype.removeTrack = function removeTrack(sender) {
@@ -205,9 +178,3 @@ if (this._peerConnection.signalingState === 'closed') {
     try {
-      if (sdpFormat === 'unified') {
-        if (sender.track) {
-          this._senders.set(sender.track, new RTCRtpSenderShim(null));
-        }
-      } else {
-        this._peerConnection.removeTrack(sender);
-      }
+      this._peerConnection.removeTrack(sender);
     }
@@ -237,3 +204,3 @@ catch (error) {
 ChromeRTCPeerConnection.prototype.getSenders = function getSenders() {
-  return this._peerConnection.getSenders && sdpFormat === 'planb'
+  return this._peerConnection.getSenders
     ? this._peerConnection.getSenders()
@@ -556,15 +523,2 @@ : Array.from(this._senders.values());
 /**
- * Gets the active RTCRtpSenders of the RTCPeerConnection.
- * @param peerConnection
- * @returns {Map<MediaStreamTrack, RTCRtpSender>}
- */
-function getActiveSenders(peerConnection) {
-  return new Map(peerConnection.getSenders().filter(function(sender) {
-    return sender.track;
-  }).map(function(sender) {
-    return [sender.track, sender];
-  }));
-}
-
-/**
  * Update the mappings from MediaStreamTrack IDs to SSRCs as indicated by both
@@ -571,0 +525,0 @@ * the Map from MediaStreamTrack IDs to SSRCs and the SDP itself. This method

lib/rtcpeerconnection/firefox.js

@@ -6,3 +6,2 @@ /* globals RTCPeerConnection */
 var FirefoxRTCSessionDescription = require('../rtcsessiondescription/firefox');
-var RTCRtpSenderShim = require('../rtcrtpsender');
 var inherits = require('util').inherits;
@@ -67,5 +66,2 @@ var updateTracksToSSRCs = require('../util/sdp').updateUnifiedPlanTrackIdsToSSRCs;
     },
-    _senders: {
-      value: new Map()
-    },
     _tracksToSSRCs: {
@@ -131,66 +127,11 @@ value: new Map()
 
-// NOTE(mmalavalli): Because we are not delegating to the native
-// RTCPeerConnection#removeTrack(), we have to manually maintain a list of added
-// tracks. So we disable the delegation to the native RTCPeerConnection#addTrack()
-// for now.
-FirefoxRTCPeerConnection.prototype.addTrack = function addTrack() {
-  var args = [].slice.call(arguments);
-  var track = args[0];
-  var sender = this._senders.get(track);
-  if (sender && sender.track) {
-    throw new Error('Cannot add MediaStreamTrack [' + track.id + ', '
-      + track.kind + ']: RTCPeerConnection already has it');
-  }
-  sender = getActiveSenders(this._peerConnection).get(track)
-    || this._peerConnection.addTrack.apply(this._peerConnection, args);
-
-  if (needsWorkaroundForBug1480277) {
+if (needsWorkaroundForBug1480277) {
+  FirefoxRTCPeerConnection.prototype.addTrack = function addTrack() {
+    var track = arguments[0];
+    var sender = this._peerConnection.addTrack.apply(this._peerConnection, arguments);
     sender.replaceTrack(track);
-  }
-
-  this._senders.set(track, sender);
-  return sender;
-};
-
-if (RTCPeerConnection.prototype.addTransceiver) {
-  FirefoxRTCPeerConnection.prototype.addTransceiver = function addTransceiver() {
-    var transceiver = this._peerConnection.addTransceiver.apply(this._peerConnection, arguments);
-    var sender = transceiver.sender;
-    var track = sender.track;
-    if (track && needsWorkaroundForBug1480277) {
-      sender.replaceTrack(track);
-    }
-    this._senders.set(track, sender);
-    return transceiver;
+    return sender;
   };
 }
 
-// NOTE(mmalavalli): RTCPeerConnection#removeTrack() has a bug in the
-// Firefox <--> Chrome interop case, which is mentioned below. So we disable
-// its delegation for now. Also, we maintain only one RTCRtpSender per
-// MediaStreamTrack for our use case, and not worry about multiple RTCRtpSenders
-// due to replaceTrack().
-// Bugzilla: https://bugzilla.mozilla.org/show_bug.cgi?id=1133874
-FirefoxRTCPeerConnection.prototype.removeTrack = function removeTrack(sender) {
-  if (this._isClosed) {
-    throw new Error('Cannot remove MediaStreamTrack: RTCPeerConnection is closed');
-  }
-  var track = sender.track;
-  if (!track) {
-    return;
-  }
-  sender = this._senders.get(track);
-  if (sender && sender.track) {
-    this._senders.set(track, new RTCRtpSenderShim(null));
-  }
-};
-
-// NOTE(mmalavalli): Because we are not delegating to the native
-// RTCPeerConnection#removeTrack(), we have to manually maintain a list of added
-// tracks. So we disable the delegation to the native RTCPeerConnection#getSenders()
-// for now.
-FirefoxRTCPeerConnection.prototype.getSenders = function getSenders() {
-  return Array.from(this._senders.values());
-};
-
 FirefoxRTCPeerConnection.prototype.createAnswer = function createAnswer() {
@@ -393,15 +334,2 @@ var args = [].slice.call(arguments);
 
-/**
- * Gets the active RTCRtpSenders of the RTCPeerConnection.
- * @param peerConnection
- * @returns {Map<MediaStreamTrack, RTCRtpSender>}
- */
-function getActiveSenders(peerConnection) {
-  return new Map(peerConnection.getSenders().filter(function(sender) {
-    return sender.track;
-  }).map(function(sender) {
-    return [sender.track, sender];
-  }));
-}
-
 module.exports = FirefoxRTCPeerConnection;

lib/rtcpeerconnection/safari.js

@@ -7,3 +7,2 @@ /* globals RTCPeerConnection, RTCSessionDescription */
 var Latch = require('../util/latch');
-var RTCRtpSenderShim = require('../rtcrtpsender');
 var sdpUtils = require('../util/sdp');
@@ -52,5 +51,2 @@ var util = require('../util');
     },
-    _senders: {
-      value: new Map()
-    },
     _signalingStateLatch: {
@@ -246,77 +242,11 @@ value: new Latch()
 
-// NOTE(mmalavalli): Because we are not delegating to the native
-// RTCPeerConnection#removeTrack(), we have to manually maintain a list of added
-// tracks. So we disable the delegation to the native RTCPeerConnection#addTrack()
-// for now. Also, we maintain only one RTCRtpSender per MediaStreamTrack for our
-// use case, and not worry about multiple RTCRtpSenders due to replaceTrack().
-SafariRTCPeerConnection.prototype.addTrack = function addTrack() {
-  var args = [].slice.call(arguments);
-  var track = args[0];
-  var sender = this._senders.get(track);
-  if (sender && sender.track) {
-    throw new Error('Cannot add MediaStreamTrack [' + track.id + ', '
-      + track.kind + ']: RTCPeerConnection already has it');
-  }
-  sender = getActiveSenders(this._peerConnection).get(track)
-    || this._peerConnection.addTrack.apply(this._peerConnection, args);
-
-  // NOTE(mmalavalli): webrtc-adapter has a bug where the "addTrack" shim
-  // does not return an RTCRtpSender and returns undefined instead. An issue
-  // [https://github.com/webrtc/adapter/issues/714] has been filed. For now,
-  // we manually get the RTCRtpSender associated with the added track and
-  // return it.
-  sender = sender || getActiveSenders(this._peerConnection).get(track);
-  this._senders.set(track, sender);
-  return sender;
-};
-
-// NOTE(mmalavalli): This shim works around a Safari bug in "unified-plan"
-// SDPs where adding a MediaStreamTrack that was previously added and removed
-// generates an SDP where the MSID does not match the MediaStreamTrack ID.
+// NOTE(mmalavalli): This shim works around the following bug in
+// RTCPeerConnection#removeTrack():
+// https://bugs.webkit.org/show_bug.cgi?id=174327
 //
-// Safari bug: https://bugs.webkit.org/show_bug.cgi?id=192101
-//
-if (RTCPeerConnection.prototype.addTransceiver) {
-  SafariRTCPeerConnection.prototype.addTransceiver = function addTransceiver() {
-    var transceiver = this._peerConnection.addTransceiver.apply(this._peerConnection, arguments);
-    var sender = transceiver.sender;
-    var track = sender.track;
-    this._senders.set(track, sender);
-    return transceiver;
-  };
-}
-
-// NOTE(mroberts): We can't really remove tracks right now, at least if we
-// ever want to add them back...
-//
-//     https://bugs.webkit.org/show_bug.cgi?id=174327
-//
-// NOTE(mmalavalli): This shim also works around a Safari bug in "unified-plan"
-// SDPs where adding a MediaStreamTrack that was previously added and removed
-// generates an SDP where the MSID does not match the MediaStreamTrack ID.
-//
-// Safari bug: https://bugs.webkit.org/show_bug.cgi?id=192101
-//
 SafariRTCPeerConnection.prototype.removeTrack = function removeTrack(sender) {
-  if (this._isClosed) {
-    throw new Error('Cannot remove MediaStreamTrack: RTCPeerConnection is closed');
-  }
-  var track = sender.track;
-  if (!track) {
-    return;
-  }
-  sender = this._senders.get(track);
-  if (sender && sender.track) {
-    this._senders.set(track, new RTCRtpSenderShim(null));
-  }
+  sender.replaceTrack(null);
+  this._peerConnection.removeTrack(sender);
 };
 
-// NOTE(mmalavalli): Because we are not delegating to the native
-// RTCPeerConnection#removeTrack(), we have to manually maintain a list of added
-// tracks. So we disable the delegation to the native RTCPeerConnection#getSenders()
-// for now.
-SafariRTCPeerConnection.prototype.getSenders = function getSenders() {
-  return Array.from(this._senders.values());
-};
-
 util.delegateMethods(
@@ -428,15 +358,2 @@ RTCPeerConnection.prototype,
 
-/**
- * Gets the active RTCRtpSenders of the RTCPeerConnection.
- * @param peerConnection
- * @returns {Map<MediaStreamTrack, RTCRtpSender>}
- */
-function getActiveSenders(peerConnection) {
-  return new Map(peerConnection.getSenders().filter(function(sender) {
-    return sender.track;
-  }).map(function(sender) {
-    return [sender.track, sender];
-  }));
-}
-
 module.exports = SafariRTCPeerConnection;

lib/util/sdp.js

@@ -7,27 +7,2 @@ /* globals RTCPeerConnection, RTCRtpTransceiver */
 
-// NOTE(mmalavalli): We cache Chrome's sdpSemantics support in order to prevent
-// instantiation of more than one RTCPeerConnection.
-var isSdpSemanticsSupported;
-
-/**
- * Check if Chrome supports specifying sdpSemantics for an RTCPeerConnection.
- * @return {boolean}
- */
-function checkIfSdpSemanticsIsSupported() {
-  if (typeof isSdpSemanticsSupported === 'boolean') {
-    return isSdpSemanticsSupported;
-  }
-  if (typeof RTCPeerConnection === 'undefined') {
-    isSdpSemanticsSupported = false;
-    return isSdpSemanticsSupported;
-  }
-  try {
-    new RTCPeerConnection({ sdpSemantics: 'foo' });
-    isSdpSemanticsSupported = false;
-  } catch (e) {
-    isSdpSemanticsSupported = true;
-  }
-  return isSdpSemanticsSupported;
-}
-
 // NOTE(mmalavalli): We cache Chrome's SDP format in order to prevent
@@ -42,13 +17,15 @@ // instantiation of more than one RTCPeerConnection.
 function getChromeSdpFormat() {
-  if (typeof chromeSdpFormat === 'string') {
-    return chromeSdpFormat;
+  if (!chromeSdpFormat) {
+    if (typeof RTCPeerConnection !== 'undefined'
+      && 'addTransceiver' in RTCPeerConnection.prototype) {
+      try {
+        new RTCPeerConnection().addTransceiver('audio');
+        chromeSdpFormat = 'unified';
+      } catch (e) {
+        chromeSdpFormat = 'planb';
+      }
+    } else {
+      chromeSdpFormat = 'planb';
+    }
   }
-  if (checkIfSdpSemanticsIsSupported()) {
-    chromeSdpFormat = 'planb';
-    return chromeSdpFormat;
-  }
-  chromeSdpFormat = typeof RTCPeerConnection !== 'undefined'
-  && 'addStream' in RTCPeerConnection.prototype
-    ? 'planb'
-    : 'unified';
   return chromeSdpFormat;
@@ -55,0 +32,0 @@ }

package.json

 {
   "name": "@twilio/webrtc",
-  "version": "2.3.0",
+  "version": "2.4.0",
   "description": "WebRTC-related APIs and shims used by twilio-video.js",
@@ -5,0 +5,0 @@ "scripts": {

README.md

@@ -80,5 +80,2 @@ twilio-webrtc.js
   more information.
-* Works around a [this bug](https://bugs.chromium.org/p/chromium/issues/detail?id=894231)
-  in "unified-plan" SDPs where adding a `MediaStreamTrack` that was previously added and
-  removed generates an SDP where the MSID does not match the `MediaStreamTrack` ID.
 * Does not depend on the native "track" event for "plan-b" RTCPeerConnection because
@@ -99,7 +96,2 @@ of [this bug](https://bugs.chromium.org/p/chromium/issues/detail?id=774303), which partly
   where the browser throws when `RTCPeerConnection.prototype.peerIdentity` is accessed.
-* Provides a shim for the `removeTrack` method in order to work around [this bug](https://bugzilla.mozilla.org/show_bug.cgi?id=1133874).
-* Provides a shim for the `addTrack` method. Since `removeTrack` is shimmed, there is a necessity to
-  maintain an explicit list of added `RTCRtpSender`s.
-* Provides a shim for the `getSenders` method. Since `removeTrack` is shimmed, there is a necessity to
-  maintain an explicit list of added `RTCRtpSender`s.
 * Works around Firefox [Bug 1480277](https://bugzilla.mozilla.org/show_bug.cgi?id=1480277).
@@ -112,9 +104,3 @@
   `MediaStreamTrack`.
-* Provides a shim for the `removeTrack` method in order to work around [this bug](https://bugs.webkit.org/show_bug.cgi?id=174327).
-* Provides a shim for the `addTrack` method. Since `removeTrack` is shimmed, there is a necessity to
-  maintain an explicit list of added `RTCRtpSender`s.
-* Provides a workaround for [this bug](https://github.com/webrtc/adapter/issues/714), where webrtc-adapter's shimmed
-  `addTrack` method does not return the `RTCRtpSender` associated with the added track.
-* Provides a shim for the `getSenders` method. Since `removeTrack` is shimmed, there is a necessity to
-  maintain an explicit list of added `RTCRtpSender`s.
+* Provides a workaround for this `removeTrack` [bug](https://bugs.webkit.org/show_bug.cgi?id=174327).
 
@@ -121,0 +107,0 @@ ### RTCSessionDescription

No alert changes

Worsened metrics

Total package byte prevSize

113356

decreased by-7.57%

Lines of code

2630

decreased by-7.3%

Number of lines in readme file

115

decreased by-10.85%

No dependency changes