Swapped force OPUS to iSAC

Chrome uses OPUS by default hence it makes sense to force ISAC instead.

BUG=279101
NOTRY=TRUE
TEST=Manual test verifying that iSAC is used instead of OPUS in chrome://webrtc-internals and that the sound is played back OK.

Review URL: https://chromiumcodereview.appspot.com/23460010

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@220550 0039d316-1c4b-4281-b951-d872f2087c98

chrome/test/data/webrtc/manual/peerconnection.html

@@ -109,8 +109,8 @@
       <input type="text" id="dtmf-tones-gap" size="10" value="50" />
       <button onclick="insertDtmfFromHere();">Send</button><br/>
       Options:
-      <input type="checkbox" id="force-opus" onclick="forceOpusChanged();"/>
-      Force Opus in Outgoing SDP<br/>
+      <input type="checkbox" id="force-isac" onclick="forceIsacChanged();"/>
+      Force iSAC in Outgoing SDP<br/>
       <button onclick="clearLog();">Clear Logs</button>
 
       <h2>Messages</h2>

chrome/test/data/webrtc/manual/peerconnection.js

@@ -117,10 +117,10 @@ function insertDtmfFromHere() {
   insertDtmfOnSender(tones, duration, gap);
 }
 
-function forceOpusChanged() {
-  var forceOpus = $('force-opus').checked;
-  if (forceOpus) {
-    forceOpus_();
+function forceIsacChanged() {
+  var forceIsac = $('force-isac').checked;
+  if (forceIsac) {
+    forceIsac_();
   } else {
     dontTouchSdp_();
   }
@@ -332,12 +332,13 @@ function preferOpus_() {
 }
 
 /** @private */
-function forceOpus_() {
+function forceIsac_() {
   setOutgoingSdpTransform(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 111\r\n');
-    sdp = sdp.replace(/a=rtpmap:(?!111)\d{1,3} (?!VP8|red|ulpfec).*\r\n/g, '');
+                      'm=audio $1 RTP/SAVPF 104\r\n');
+    sdp = sdp.replace('a=fmtp:111 minptime=10', 'a=fmtp:104 minptime=10');
+    sdp = sdp.replace(/a=rtpmap:(?!104)\d{1,3} (?!VP8|red|ulpfec).*\r\n/g, '');
     return sdp;
   });
 }