Variable audio bitrate.

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 469098a32274973cb38189eb8094336920ae113c..2aa0552b9378a8b4df743f7db56e5dfeb9819d9e 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -1136,6 +1136,14 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
       stream_ = nullptr;
     }
     config_.rtp.extensions = extensions;
+    if (webrtc::field_trial::FindFullName("WebRTC-AdaptAudioBitrate") ==
+        "Enabled") {
+      // TODO(mflodman): Keep testing this and set proper values.
+      // Note: This is an early experiment currently only supported by Opus.
+      config_.min_bitrate_kbps = kOpusMinBitrate;
+      config_.max_bitrate_kbps = kOpusBitrateFb;
+    }
+
     RTC_DCHECK(!stream_);
     stream_ = call_->CreateAudioSendStream(config_);
     RTC_CHECK(stream_);