Rename the adapt audio bitrate experiment.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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 1370 matching lines @@
     }
   }
 
   void RecreateAudioSendStream() {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     if (stream_) {
       call_->DestroyAudioSendStream(stream_);
       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.
       config_.min_bitrate_bps = kOpusMinBitrateBps;
       config_.max_bitrate_bps = kOpusBitrateFbBps;
     }
     stream_ = call_->CreateAudioSendStream(config_);
     RTC_CHECK(stream_);
     UpdateSendState();
   }
@@ skipping 1168 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = send_streams_.find(ssrc);
   if (it != send_streams_.end()) {
     return it->second->channel();
   }
   return -1;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE