Adding audio network adaptor to AudioEncoderOpus.

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

Index: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
index 48fb494dbe180bb7b151c5fe36767c83c84411be..6bc1eecda6c5275a29d1927e7c8878d926dd45e9 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -20,6 +20,7 @@
 
 namespace webrtc {
 
+class AudioNetworkAdaptor;
 struct CodecInst;
 
 class AudioEncoderOpus final : public AudioEncoder {
@@ -47,6 +48,7 @@ class AudioEncoderOpus final : public AudioEncoder {
     int max_playback_rate_hz = 48000;
     int complexity = kDefaultComplexity;
     bool dtx_enabled = false;
+    bool audio_network_adaptor_enabled = false;
 
    private:
 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
@@ -82,6 +84,11 @@ class AudioEncoderOpus final : public AudioEncoder {
   void SetProjectedPacketLossRate(double fraction) override;
   void SetTargetBitrate(int target_bps) override;
 
+  void OnReceivedUplinkBandwidth(int uplink_bandwidth_bps) override;
+  void OnReceivedUplinkPacketLossFraction(float uplink_packet_loss_fraction) override;
+  void OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) override;
+  void OnReceivedRtt(int rtt_ms) override;
+
   // Getters for testing.
   double packet_loss_rate() const { return packet_loss_rate_; }
   ApplicationMode application() const { return config_.application; }
@@ -96,12 +103,19 @@ class AudioEncoderOpus final : public AudioEncoder {
   size_t SamplesPer10msFrame() const;
   size_t SufficientOutputBufferSize() const;
   bool RecreateEncoderInstance(const Config& config);
+  void SetFrameLength(int frame_length_ms);
+  void SetNumChannelsToEncode(size_t num_channels_to_encode);
+  void ApplyAudioNetworkAdaptor();
 
   Config config_;
   double packet_loss_rate_;
   std::vector<int16_t> input_buffer_;
   OpusEncInst* inst_;
   uint32_t first_timestamp_in_buffer_;
+  size_t num_channels_to_encode_;
+  int next_frame_size_ms_;
+  std::unique_ptr<AudioNetworkAdaptor> audio_network_adaptor_;
+
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus);
 };