Hooking up audio network adaptor to VoE.

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

Index: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index a64330b2181bfa313873a747bf9d79051d7504da..9dc4fc5ea10fbd9ac6d814b1aee886b125e3e5e6 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -236,8 +236,10 @@ void AudioEncoderOpus::OnReceivedUplinkBandwidth(int uplink_bandwidth_bps) {
 
 void AudioEncoderOpus::OnReceivedUplinkPacketLossFraction(
     float uplink_packet_loss_fraction) {
-  if (!audio_network_adaptor_)
-    return;
+  if (!audio_network_adaptor_) {
+    return AudioEncoder::OnReceivedTargetAudioBitrate(
+        uplink_packet_loss_fraction);
+  }

[kwiberg-webrtc, 2016/10/06 09:46:13]

Better to just duplicate that one line from the default implementation, instead
of calling it explicitly from here. (Because it's easier to read, and that's the
behavior you want regardless of what the default behavior was. But see also the
question of whether this default behavior may be what everyone wants all the
time.)

[minyue-webrtc, 2016/10/11 09:00:22]

Done.

   audio_network_adaptor_->SetUplinkPacketLossFraction(
       uplink_packet_loss_fraction);
   ApplyAudioNetworkAdaptor();
@@ -246,7 +248,7 @@ void AudioEncoderOpus::OnReceivedUplinkPacketLossFraction(
 void AudioEncoderOpus::OnReceivedTargetAudioBitrate(
     int target_audio_bitrate_bps) {
   if (!audio_network_adaptor_)
-    return;
+    return AudioEncoder::OnReceivedTargetAudioBitrate(target_audio_bitrate_bps);

[kwiberg-webrtc, 2016/10/06 09:46:13]

Same.

[minyue-webrtc, 2016/10/11 09:00:22]

Done.

   audio_network_adaptor_->SetTargetAudioBitrate(target_audio_bitrate_bps);
   ApplyAudioNetworkAdaptor();
 }