Rename the adapt audio bitrate experiment.

webrtc/media/engine/webrtcvoiceengine.cc

Index: webrtc/media/engine/webrtcvoiceengine.cc
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 35629da72227b7d75d361bee5bad0dfa6d01063e..49a0f1cee71fd42b1c040c7907b0fd3d7da31d59 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -1388,7 +1388,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
       stream_ = nullptr;
     }
     RTC_DCHECK(!stream_);
-    if (webrtc::field_trial::FindFullName("WebRTC-AdaptAudioBitrate") ==
+    if (webrtc::field_trial::FindFullName("WebRTC-Audio-SendSideBwe") ==
         "Enabled") {
       // TODO(mflodman): Keep testing this and set proper values.
       // Note: This is an early experiment currently only supported by Opus.