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 37 matching lines @@
     int32_t echo_delay_std_ms = -1;
     int32_t echo_return_loss = -100;
     int32_t echo_return_loss_enhancement = -100;
     float residual_echo_likelihood = -1.0f;
     bool typing_noise_detected = false;
   };
 
   struct Config {
     Config() = delete;
     explicit Config(Transport* send_transport);
+    ~Config();
     std::string ToString() const;
 
     // Send-stream specific RTP settings.
     struct Rtp {
       Rtp();
       ~Rtp();
       std::string ToString() const;
 
       // Sender SSRC.
       uint32_t ssrc = 0;
@@ skipping 17 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;
 
+    // Defines whether to turn on audio network adaptor, and defines its config
+    // string.
+    rtc::Optional<std::string> audio_network_adaptor_config;
+
     struct SendCodecSpec {
       SendCodecSpec();
       std::string ToString() const;
 
       bool operator==(const SendCodecSpec& rhs) const;
       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_