Add AudioEncoder::GetTargetBitrate

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 111 matching lines @@
 }
 
 int AudioEncoderOpus::Num10MsFramesInNextPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
 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) {
   DCHECK_GE(fraction, 0.0);
   DCHECK_LE(fraction, 1.0);
   // Optimize the loss rate to configure Opus. Basically, optimized loss rate is
   // the input loss rate rounded down to various levels, because a robustly good
   // audio quality is achieved by lowering the packet loss down.
   // Additionally, to prevent toggling, margins are used, i.e., when jumping to
   // a loss rate from below, a higher threshold is used than jumping to the same
@@ skipping 110 matching lines @@
   return Reconstruct(conf);
 }
 
 bool AudioEncoderMutableOpus::SetMaxPlaybackRate(int frequency_hz) {
   auto conf = config();
   conf.max_playback_rate_hz = frequency_hz;
   return Reconstruct(conf);
 }
 
 }  // namespace webrtc