Adding build switch for Opus that supports 120ms ptime.

webrtc/modules/audio_coding/codecs/opus/opus_interface.c

Index: webrtc/modules/audio_coding/codecs/opus/opus_interface.c
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
index e2b4fdbeee79b83eab36120077e5fabf60ede292..d544d850a9307bf5ef75ec4f121030e8656aac8d 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
@@ -17,8 +17,13 @@
 #include <string.h>
 
 enum {
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+  /* Maximum supported frame size in WebRTC is 120 ms. */
+  kWebRtcOpusMaxEncodeFrameSizeMs = 120,
+#else
   /* Maximum supported frame size in WebRTC is 60 ms. */
   kWebRtcOpusMaxEncodeFrameSizeMs = 60,
+#endif
 
   /* The format allows up to 120 ms frames. Since we don't control the other
    * side, we must allow for packets of that size. NetEq is currently limited