Adding audio network adaptor to AudioEncoderOpus.

webrtc/modules/audio_coding/codecs/audio_encoder.cc

Index: webrtc/modules/audio_coding/codecs/audio_encoder.cc
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.cc b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
index c433dcdd0f1c9613cb7a9be44bf28d3f06237185..bd16f341d079eb53002f9c0faf4c1009de9aeae8 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.cc
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
@@ -67,4 +67,20 @@ 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