Adding build switch for Opus that supports 120ms ptime.

webrtc/media/engine/webrtcvoiceengine.cc

Index: webrtc/media/engine/webrtcvoiceengine.cc
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index b9a725b0a0519bce6ccd006dd1d6ed4ec3de6378..b81173032d7e6852990caed6e7aafdbcbf0b07bf 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -498,7 +498,12 @@ class WebRtcVoiceCodecs final {
 };
 
 const WebRtcVoiceCodecs::CodecPref WebRtcVoiceCodecs::kCodecPrefs[14] = {
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+    {kOpusCodecName, 48000, 2, 111, true, {10, 20, 40, 60, 120},
+     kOpusMaxBitrateBps},
+#else
     {kOpusCodecName, 48000, 2, 111, true, {10, 20, 40, 60}, kOpusMaxBitrateBps},
+#endif
     {kIsacCodecName, 16000, 1, 103, true, {30, 60}, kIsacMaxBitrateBps},
     {kIsacCodecName, 32000, 1, 104, true, {30}, kIsacMaxBitrateBps},
     // G722 should be advertised as 8000 Hz because of the RFC "bug".
@@ -512,8 +517,7 @@ const WebRtcVoiceCodecs::CodecPref WebRtcVoiceCodecs::kCodecPrefs[14] = {
     {kDtmfCodecName, 48000, 1, 110, false, {}},
     {kDtmfCodecName, 32000, 1, 112, false, {}},
     {kDtmfCodecName, 16000, 1, 113, false, {}},
-    {kDtmfCodecName, 8000, 1, 126, false, {}}
-};
+    {kDtmfCodecName, 8000, 1, 126, false, {}}};
 
 rtc::Optional<int> ComputeSendBitrate(int max_send_bitrate_bps,
                                       int rtp_max_bitrate_bps,