Add 120ms frame ability to ANA

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 1460 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)) {
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+            max_packet_size_ms = 120;
+#else
             max_packet_size_ms = 60;
+#endif
             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