| Index: content/test/data/media/peerconnection-call-audio.html
|
| diff --git a/content/test/data/media/peerconnection-call-audio.html b/content/test/data/media/peerconnection-call-audio.html
|
| index 3c996887f97a42943df0824e3809df531db2f63b..275c19ef9700093f94b63362a7233954244dad64 100644
|
| --- a/content/test/data/media/peerconnection-call-audio.html
|
| +++ b/content/test/data/media/peerconnection-call-audio.html
|
| @@ -64,7 +64,7 @@
|
| sdp = sdp.replace(/m=audio (\d+) RTP\/SAVPF.*\r\n/g,
|
| 'm=audio $1 RTP/SAVPF 103 126\r\n');
|
| sdp = sdp.replace('a=fmtp:111 minptime=10', 'a=fmtp:103 minptime=10');
|
| - if (sdp.search('a=rtpmap:103 ISAC/16000') == -1)
|
| + if (sdp.search(/a=rtpmap:103 ISAC\/16000/i) == -1)
|
| failTest('Missing iSAC 16K codec on Android; cannot force codec.');
|
|
|
| return sdp;
|
|
|