Add AudioEncoder::GetTargetBitrate

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 328698380a3a102938a1fcc7a923680687e6d41b..feed3c741150abeea51e994ae1f9735e4ef6dcee 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -129,10 +129,11 @@ int AudioEncoderOpus::Max10MsFramesInAPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
-void AudioEncoderOpus::SetTargetBitrate(int bits_per_second) {
+int AudioEncoderOpus::SetTargetBitrate(int bits_per_second) {
   bitrate_bps_ = std::max(std::min(bits_per_second, kMaxBitrateBps),
                           kMinBitrateBps);
   CHECK_EQ(WebRtcOpus_SetBitRate(inst_, bitrate_bps_), 0);
+  return bitrate_bps_;
 }
 
 void AudioEncoderOpus::SetProjectedPacketLossRate(double fraction) {