Adding build switch for Opus that supports 120ms ptime.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 480 matching lines @@
     if (IsCodec(*voe_codec, kG722CodecName)) {
       // If the DCHECK triggers, the codec definition in WebRTC VoiceEngine
       // has changed, and this special case is no longer needed.
       RTC_DCHECK(voe_codec->plfreq != new_plfreq);
       voe_codec->plfreq = new_plfreq;
     }
   }
 };
 
 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".
     {kG722CodecName, 8000, 1, 9, false, {10, 20, 30, 40, 50, 60}},
     {kIlbcCodecName, 8000, 1, 102, false, {20, 30, 40, 60}},
     {kPcmuCodecName, 8000, 1, 0, false, {10, 20, 30, 40, 50, 60}},
     {kPcmaCodecName, 8000, 1, 8, false, {10, 20, 30, 40, 50, 60}},
     {kCnCodecName, 32000, 1, 106, false, {}},
     {kCnCodecName, 16000, 1, 105, false, {}},
     {kCnCodecName, 8000, 1, 13, false, {}},
     {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,
                                       const webrtc::CodecInst& codec_inst) {
   const int bps = MinPositive(max_send_bitrate_bps, rtp_max_bitrate_bps);
   const int codec_rate = codec_inst.rate;
 
   if (bps <= 0) {
     return rtc::Optional<int>(codec_rate);
   }
@@ skipping 939 matching lines @@
         if (!packet_sizes_ms.empty()) {
           int max_packet_size_ms =
               *std::max_element(packet_sizes_ms.begin(), packet_sizes_ms.end());
           int min_packet_size_ms =
               *std::min_element(packet_sizes_ms.begin(), packet_sizes_ms.end());
 
           // Audio network adaptor will just use 20ms and 60ms frame lengths.
           // The adaptor will only be active for the Opus encoder.
           if (config_.audio_network_adaptor_config &&
               IsCodec(config_.send_codec_spec.codec_inst, kOpusCodecName)) {
             max_packet_size_ms = 60;

[hlundin-webrtc, 2017/02/01 14:12:34]

Is this relevant to change?

[michaelt, 2017/02/01 15:01:08]

Will be handled in https://codereview.webrtc.org/2669733002/

[hlundin-webrtc, 2017/02/01 15:10:59]

Acknowledged.

             min_packet_size_ms = 20;
           }
 
           // OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12)
           constexpr int kOverheadPerPacket = 20 + 8 + 10 + 12;
 
           int min_overhead_bps =
               kOverheadPerPacket * 8 * 1000 / max_packet_size_ms;
 
           int max_overhead_bps =
@@ skipping 1218 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = send_streams_.find(ssrc);
   if (it != send_streams_.end()) {
     return it->second->channel();
   }
   return -1;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE