Variable audio bitrate.

webrtc/audio/audio_send_stream.h

 /*
  *  Copyright (c) 2015 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.
  */
 
 #ifndef WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
 #define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
 
 #include <memory>
 
 #include "webrtc/audio_send_stream.h"
 #include "webrtc/audio_state.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_checker.h"
+#include "webrtc/call/bitrate_allocator.h"
 
 namespace webrtc {
 class CongestionController;
 class VoiceEngine;
 
 namespace voe {
 class ChannelProxy;
 }  // namespace voe
 
 namespace internal {
-class AudioSendStream final : public webrtc::AudioSendStream {
+class AudioSendStream final : public webrtc::AudioSendStream,
+                              public webrtc::BitrateAllocatorObserver {
  public:
   AudioSendStream(const webrtc::AudioSendStream::Config& config,
                   const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
-                  CongestionController* congestion_controller);
+                  CongestionController* congestion_controller,
+                  BitrateAllocator* bitrate_allocator);
   ~AudioSendStream() override;
 
   // webrtc::AudioSendStream implementation.
   void Start() override;
   void Stop() override;
   bool SendTelephoneEvent(int payload_type, int event,
                           int duration_ms) override;
   void SetMuted(bool muted) override;
   webrtc::AudioSendStream::Stats GetStats() const override;
 
   void SignalNetworkState(NetworkState state);
   bool DeliverRtcp(const uint8_t* packet, size_t length);
+
+  // Implements BitrateAllocatorObserver.
+  uint32_t OnBitrateUpdated(uint32_t bitrate_bps,
+                            uint8_t fraction_loss,
+                            int64_t rtt) override;
+
   const webrtc::AudioSendStream::Config& config() const;
 
  private:
   VoiceEngine* voice_engine() const;
 
   rtc::ThreadChecker thread_checker_;
   const webrtc::AudioSendStream::Config config_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   std::unique_ptr<voe::ChannelProxy> channel_proxy_;
 
+  BitrateAllocator* const bitrate_allocator_;
+
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_