All WebRTC content browsertests will now force iSAC 16K 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..20129ad26cfc85edd6fb120c941ea32e7ba88b71 100644
--- a/content/test/data/media/peerconnection-call.html
+++ b/content/test/data/media/peerconnection-call.html
@@ -21,6 +21,22 @@
   var transformCandidate = function(candidate) { return candidate; };
   var onLocalDescriptionError = function(error) { };
 
+  // Temporary measure to be able to force iSAC 16K where needed, particularly
+  // on Android. This applies to every test which is why it's implemented like
+  // this.
+  var maybeForceIsac16K = function(sdp) { return sdp; };
+  function forceIsac16KInSdp() {
+    maybeForceIsac16K = function(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;
+    };
+  }
+
   // When using external SDES, the crypto key is chosen by javascript.
   var EXTERNAL_SDES_LINES = {
     'audio': 'a=crypto:1 AES_CM_128_HMAC_SHA1_80 ' +
@@ -36,6 +52,7 @@
   var EXTERNAL_GICE_PWD = '123456789012345678901234';
 
   setAllEventsOccuredHandler(function() {
+    // The C++ tests look for this 'OK' in the title.
     document.title = 'OK';
   });
 
@@ -248,20 +265,7 @@
     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);
@@ -368,7 +372,7 @@
       local_stream.addTrack(gLocalStream.getAudioTracks()[0]);
       local_stream.removeTrack(local_stream.getVideoTracks()[0]);
       negotiate();
-    });  // End of setAllEventsOccuredHandler.
+    });
   }
 
   // This function is used for setting up a test that:
@@ -604,7 +608,7 @@
   }
 
   function onOfferCreated(offer, caller, callee) {
-    offer.sdp = transformSdp(offer.sdp);
+    offer.sdp = maybeForceIsac16K(transformSdp(offer.sdp));
     caller.setLocalDescription(offer, function() {
       expectEquals('have-local-offer', caller.signalingState);
       receiveOffer(offer.sdp, caller, callee);
@@ -670,7 +674,7 @@
   }
 
   function onAnswerCreated(answer, caller, callee) {
-    answer.sdp = transformSdp(answer.sdp);
+    answer.sdp = maybeForceIsac16K(transformSdp(answer.sdp));
     callee.setLocalDescription(answer);
     expectEquals('stable', callee.signalingState);
     receiveAnswer(answer.sdp, caller);