Adding audio network adaptor to AudioEncoderOpus.

webrtc/modules/audio_coding/codecs/audio_encoder.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 49 matching lines @@
 
 void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {}
 
 void AudioEncoder::SetProjectedPacketLossRate(double fraction) {}
 
 void AudioEncoder::SetTargetBitrate(int target_bps) {}
 
 rtc::ArrayView<std::unique_ptr<AudioEncoder>>
 AudioEncoder::ReclaimContainedEncoders() { return nullptr; }
 
+bool AudioEncoder::SetAudioNetworkAdaptor(bool enable) {
+  return false;
+}

[kwiberg-webrtc, 2016/09/27 09:35:54]

Wouldn't

  return !enable;

be a better choice for the default implementation?

[minyue-webrtc, 2016/09/27 16:02:32]

True. Will change.

[minyue-webrtc, 2016/09/29 15:34:24]

since I split it into two functions now. The problem is self-solved.

+
+void AudioEncoder::OnReceivedUplinkBandwidth(int uplink_bandwidth_bps) {}
+
+void AudioEncoder::OnReceivedUplinkPacketLossFraction(
+    float uplink_packet_loss_fraction) {}
+
+void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) {}
+
+void AudioEncoder::OnReceivedRtt(int rtt_ms) {}
+
+void AudioEncoder::SetReceiverFrameLengthRange(int min_frame_length_ms,
+                                               int max_frame_length_ms) {}
+
 }  // namespace webrtc