Enable some webrtc content_browsertests on Android.

content/test/data/media/peerconnection-call.html

Index: content/test/data/media/peerconnection-call.html
diff --git a/content/test/data/media/peerconnection-call.html b/content/test/data/media/peerconnection-call.html
index afd8b0318795ca2a962104034c842b913431fccf..758e57c40bf68614d0ac47c5dda15b9c2b15a1ce 100644
--- a/content/test/data/media/peerconnection-call.html
+++ b/content/test/data/media/peerconnection-call.html
@@ -12,11 +12,12 @@
   var gTestWithoutMsid = false;
   var gLocalStream = null;
   var gSentTones = '';
+  var gForceIsac16K = false;
 
   var gRemoteStreams = {};
 
   // Default transform functions, overridden by some test cases.
-  var transformSdp = function(sdp) { return sdp; };
+  var transformSdp = function(sdp) { return gForceIsac16K ? forceIsac16k_(sdp) : sdp; };
   var transformRemoteSdp = function(sdp) { return sdp; };
   var transformCandidate = function(candidate) { return candidate; };
   var onLocalDescriptionError = function(error) { };
@@ -35,10 +36,25 @@
   var EXTERNAL_GICE_UFRAG = '1234567890123456';
   var EXTERNAL_GICE_PWD = '123456789012345678901234';
 
+  /** @private */
+  function forceIsac16k_(sdp) {
+    // Remove all other codecs (not the video codecs though).
+    sdp = sdp.replace(/m=audio (\d+) RTP\/SAVPF.*\r\n/g,
+                      'm=audio $1 RTP/SAVPF 103\r\n');
+    sdp = sdp.replace('a=fmtp:111 minptime=10', 'a=fmtp:103 minptime=10');
+    sdp = sdp.replace(/a=rtpmap:(?!103)\d{1,3} (?!VP8|red|ulpfec).*\r\n/g, '');
+    return sdp;
+  }
+
   setAllEventsOccuredHandler(function() {
     document.title = 'OK';
   });
 
+  // Set gForceIsac16K.
+  function setForceIsac16K(force_isac_16k) {
+    gForceIsac16K = force_isac_16k;
+  }
+
   // Test that we can setup call with an audio and video track.
   function call(constraints) {
     createConnections(null);
@@ -248,20 +264,7 @@
     detectVideoPlaying('remote-view-2', onCallEstablished);
   }
 
-  //TODO(phoglund): do this for all tests on android if this works on bots.
-  /** @private */
-  function forceIsac16k_(sdp) {
-    // Remove all other codecs (not the video codecs though).
-    sdp = sdp.replace(/m=audio (\d+) RTP\/SAVPF.*\r\n/g,
-                      'm=audio $1 RTP/SAVPF 103\r\n');
-    sdp = sdp.replace('a=fmtp:111 minptime=10', 'a=fmtp:103 minptime=10');
-    sdp = sdp.replace(/a=rtpmap:(?!103)\d{1,3} (?!VP8|red|ulpfec).*\r\n/g, '');
-    return sdp;
-  }
-
-  function callAndEnsureAudioIsPlaying(force_isac_16k) {
-    if (force_isac_16k)
-      transformSdp = forceIsac16k_;
+  function callAndEnsureAudioIsPlaying() {
     createConnections(null);
     navigator.webkitGetUserMedia({audio: true, video: true},
       addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
@@ -279,8 +282,8 @@
     detectVideoPlaying('remote-view-2', onCallEstablished);
   }
 
-  function callAndEnsureAudioMutingWorks(force_isac_16k) {
-    callAndEnsureAudioIsPlaying(force_isac_16k);
+  function callAndEnsureAudioMutingWorks() {
+    callAndEnsureAudioIsPlaying();
     setAllEventsOccuredHandler(function() {
       var audioTrack =
         gSecondConnection.getRemoteStreams()[0].getAudioTracks()[0];
@@ -634,17 +637,18 @@
   function removeVideoCodec(offerSdp) {
     offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n',
                                 'a=rtpmap:100 XVP8/90000\r\n');
-    return offerSdp;
+    return gForceIsac16K ? forceIsac16k_(offerSdp) : offerSdp;
   }
 
   function removeCrypto(offerSdp) {
     offerSdp = offerSdp.replace(/a=crypto.*\r\n/g, 'a=Xcrypto\r\n');
     offerSdp = offerSdp.replace(/a=fingerprint.*\r\n/g, '');
-    return offerSdp;
+    return gForceIsac16K ? forceIsac16k_(offerSdp) : offerSdp;
   }
 
   function removeBundle(sdp) {
-    return sdp.replace(/a=group:BUNDLE .*\r\n/g, '');
+    sdp = sdp.replace(/a=group:BUNDLE .*\r\n/g, '');
+    return gForceIsac16K ? forceIsac16k_(sdp) : sdp;
   }
 
   function useGice(sdp) {