Set min and max rate on caller and on callee side.

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 71803cbee38c845e1cf4afbfa90c15ca94f062e3..169ff38f558428efcb6e357d1e5e80ec1f0b1fcc 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -1171,38 +1171,16 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
 
   void RecreateAudioSendStream(const SendCodecSpec& send_codec_spec) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-    if (stream_) {
-      call_->DestroyAudioSendStream(stream_);
-      stream_ = nullptr;
-    }
     config_.rtp.nack.rtp_history_ms =
         send_codec_spec.nack_enabled ? kNackRtpHistoryMs : 0;
-    RTC_DCHECK(!stream_);
-    stream_ = call_->CreateAudioSendStream(config_);
-    RTC_CHECK(stream_);
-    UpdateSendState();
+    RecreateAudioSendStream();
   }
 
   void RecreateAudioSendStream(
       const std::vector<webrtc::RtpExtension>& extensions) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-    if (stream_) {
-      call_->DestroyAudioSendStream(stream_);
-      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_);
-    UpdateSendState();
+    RecreateAudioSendStream();
   }
 
   bool SendTelephoneEvent(int payload_type, int event, int duration_ms) {
@@ -1316,6 +1294,25 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     }
   }
 
+  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") ==
+        "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;
+    }
+    stream_ = call_->CreateAudioSendStream(config_);
+    RTC_CHECK(stream_);
+    UpdateSendState();
+  }
+
   rtc::ThreadChecker worker_thread_checker_;
   rtc::RaceChecker audio_capture_race_checker_;
   webrtc::AudioTransport* const voe_audio_transport_ = nullptr;