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

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 1151 matching lines @@
     config_.rtp.extensions = extensions;
     RecreateAudioSendStream(send_codec_spec);
   }
 
   ~WebRtcAudioSendStream() override {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     ClearSource();
     call_->DestroyAudioSendStream(stream_);
   }
 
   void RecreateAudioSendStream(const SendCodecSpec& send_codec_spec) {

[the sun, 2016/10/11 15:22:15]

Thread check should happen before mutating internal state.

[michaelt, 2016/10/11 15:27:48]

Acknowledged.

+    config_.rtp.nack.rtp_history_ms =
+        send_codec_spec.nack_enabled ? kNackRtpHistoryMs : 0;
+    RecreateAudioSendStream();
+  }
+
+  void RecreateAudioSendStream(
+      const std::vector<webrtc::RtpExtension>& extensions) {
+    config_.rtp.extensions = extensions;
+    RecreateAudioSendStream();
+  }
+
+  void RecreateAudioSendStream() {

[the sun, 2016/10/11 15:22:15]

This is an internal utility - make it private. There should be no reason for a
client to recreate the stream without having set new state.

[michaelt, 2016/10/11 15:27:48]

Acknowledged.

     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();
-  }
-
-  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();
   }
 
   bool SendTelephoneEvent(int payload_type, int event, int duration_ms) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     RTC_DCHECK(stream_);
     return stream_->SendTelephoneEvent(payload_type, event, duration_ms);
   }
@@ skipping 1451 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