Using AudioOption to enable audio network adaptor.

webrtc/api/call/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 71 matching lines @@
     // TODO(solenberg): Remove when VoiceEngine channels are created outside
     // of Call.
     int voe_channel_id = -1;
 
     // Bitrate limits used for variable audio bitrate streams. Set both to -1 to
     // disable audio bitrate adaptation.
     // Note: This is still an experimental feature and not ready for real usage.
     int min_bitrate_kbps = -1;
     int max_bitrate_kbps = -1;
 
+    bool enable_audio_network_adaptor = false;
+    std::string audio_network_adaptor_config;
+
     struct SendCodecSpec {
       SendCodecSpec() {
         webrtc::CodecInst empty_inst = {0};
         codec_inst = empty_inst;
         codec_inst.pltype = -1;
       }
       bool operator==(const SendCodecSpec& rhs) const {
         {
           if (nack_enabled != rhs.nack_enabled) {

[michaelt, 2016/10/20 09:10:04]

wouldn't this be the cleaner impl. ? 

if (nack_enabled == rhs.nack_enabled &&
    transport_cc_enabled == rhs.transport_cc_enabled &&
    ...)
 return true;
return false

[minyue-webrtc, 2016/10/20 09:22:11]

Yes, that can be an improvement.

             return false;
           }
           if (transport_cc_enabled != rhs.transport_cc_enabled) {
             return false;
           }
           if (enable_codec_fec != rhs.enable_codec_fec) {
             return false;
           }
           if (enable_opus_dtx != rhs.enable_opus_dtx) {
             return false;
           }
           if (opus_max_playback_rate != rhs.opus_max_playback_rate) {
             return false;
           }
           if (cng_payload_type != rhs.cng_payload_type) {
             return false;
           }
           if (cng_plfreq != rhs.cng_plfreq) {
             return false;
           }
+          if (max_ptime_ms != rhs.max_ptime_ms) {
+            return false;
+          }
+          if (min_ptime_ms != rhs.min_ptime_ms) {
+            return false;
+          }
           if (codec_inst != rhs.codec_inst) {
             return false;
           }
           return true;
         }
       }
       bool operator!=(const SendCodecSpec& rhs) const {
         return !(*this == rhs);
       }
 
       bool nack_enabled = false;
       bool transport_cc_enabled = false;
       bool enable_codec_fec = false;
       bool enable_opus_dtx = false;
       int opus_max_playback_rate = 0;
       int cng_payload_type = -1;
       int cng_plfreq = -1;
+      int max_ptime_ms = -1;
+      int min_ptime_ms = -1;
       webrtc::CodecInst codec_inst;
     } send_codec_spec;
   };
 
   // Starts stream activity.
   // When a stream is active, it can receive, process and deliver packets.
   virtual void Start() = 0;
   // Stops stream activity.
   // When a stream is stopped, it can't receive, process or deliver packets.
   virtual void Stop() = 0;
 
   // TODO(solenberg): Make payload_type a config property instead.
   virtual bool SendTelephoneEvent(int payload_type, int event,
                                   int duration_ms) = 0;
 
   virtual void SetMuted(bool muted) = 0;
 
   virtual Stats GetStats() const = 0;
 
  protected:
   virtual ~AudioSendStream() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_