Using AudioOption to enable audio network adaptor.

webrtc/api/call/audio_send_stream.h

Index: webrtc/api/call/audio_send_stream.h
diff --git a/webrtc/api/call/audio_send_stream.h b/webrtc/api/call/audio_send_stream.h
index ae7531cbcb372333e803b7ceda91a78cf6514ea6..de64fc4d048464d693c7f971c0d61a38b03fc114 100644
--- a/webrtc/api/call/audio_send_stream.h
+++ b/webrtc/api/call/audio_send_stream.h
@@ -89,6 +89,10 @@ class AudioSendStream {
     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() {
         webrtc::CodecInst empty_inst = {0};
@@ -118,6 +122,12 @@ class AudioSendStream {
           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;
           }
@@ -135,6 +145,8 @@ class AudioSendStream {
       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;
   };