Injectable audio encoders: WebRtcVoiceEngine and company

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.
  */
 
@@ skipping 27 matching lines @@
   AudioSendStream(const webrtc::AudioSendStream::Config& config,
                   const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
                   rtc::TaskQueue* worker_queue,
                   PacketRouter* packet_router,
                   CongestionController* congestion_controller,
                   BitrateAllocator* bitrate_allocator,
                   RtcEventLog* event_log,
                   RtcpRttStats* rtcp_rtt_stats);
   ~AudioSendStream() override;
 
+  void Reconfigure(const webrtc::AudioSendStream::Config& config) override;

[the sun, 2017/04/04 23:02:54]

nit: looks like it should go under the below comment

+
   // webrtc::AudioSendStream implementation.
   void Start() override;
   void Stop() override;
   bool SendTelephoneEvent(int payload_type, int payload_frequency, 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,
                             int64_t probing_interval_ms) override;
 
   const webrtc::AudioSendStream::Config& config() const;
   void SetTransportOverhead(int transport_overhead_per_packet);
 
  private:
   VoiceEngine* voice_engine() const;
 
-  bool SetupSendCodec();
+  bool SetupSendCodec(const Config& new_config);
+  bool ReconfigureSendCodec(const Config& new_config);
+  void ReconfigureANA(const Config& new_config);
+  void ReconfigureCNG(const Config& new_config);
+  void ReconfigureBitrateObserver(const Config& new_config);
+
+  void ConfigureBitrateObserver(int min_bitrate_bps, int max_bitrate_bps);
+  void RemoveBitrateObserver();
 
   rtc::ThreadChecker thread_checker_;
   rtc::TaskQueue* worker_queue_;
-  const webrtc::AudioSendStream::Config config_;
+  webrtc::AudioSendStream::Config config_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   std::unique_ptr<voe::ChannelProxy> channel_proxy_;
+  RtcEventLog* const event_log_;
 
+  PacketRouter* const packet_router_;
   BitrateAllocator* const bitrate_allocator_;
   CongestionController* const congestion_controller_;
   std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_