Add 120ms frame ability to ANA

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc

 /*
  *  Copyright (c) 2014 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 13 matching lines @@
 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 
 namespace {
 
 constexpr int kSampleRateHz = 48000;
 constexpr int kMinBitrateBps = 500;
 constexpr int kMaxBitrateBps = 512000;
+
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+constexpr int kSupportedFrameLengths[] = {20, 60, 120};
+#else
 constexpr int kSupportedFrameLengths[] = {20, 60};
+#endif
 
 // PacketLossFractionSmoother uses an exponential filter with a time constant
 // of -1.0 / ln(0.9999) = 10000 ms.
 constexpr float kAlphaForPacketLossFractionSmoother = 0.9999f;
 
 AudioEncoderOpus::Config CreateConfig(const CodecInst& codec_inst) {
   AudioEncoderOpus::Config config;
   config.frame_size_ms = rtc::CheckedDivExact(codec_inst.pacsize, 48);
   config.num_channels = codec_inst.channels;
   config.bitrate_bps = rtc::Optional<int>(codec_inst.rate);
@@ skipping 506 matching lines @@
             config_.uplink_bandwidth_update_interval_ms) {
       rtc::Optional<float> smoothed_bitrate = bitrate_smoother_->GetAverage();
       if (smoothed_bitrate)
         audio_network_adaptor_->SetUplinkBandwidth(*smoothed_bitrate);
       bitrate_smoother_last_update_time_ = rtc::Optional<int64_t>(now_ms);
     }
   }
 }
 
 }  // namespace webrtc