Fix issue with conflicting behavior if setting a max BW with b=AS on both audio and video.

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 1620 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendParameters: "
                << params.ToString();
   // TODO(pthatcher): Refactor this to be more clean now that we have
   // all the information at once.
 
   if (!SetSendCodecs(params.codecs)) {
     return false;
   }
 
-  if (params.max_bandwidth_bps >= 0) {
-    // Note that max_bandwidth_bps intentionally takes priority over the
-    // bitrate config for the codec.
-    bitrate_config_.max_bitrate_bps =
-        params.max_bandwidth_bps == 0 ? -1 : params.max_bandwidth_bps;
-  }
   call_->SetBitrateConfig(bitrate_config_);

[minyue-webrtc, 2017/03/27 09:22:06]

I think there is no need to call this if we remove resetting of bitrate_config_

we may move this to  SetSendCodecs(), and the need of bitrate_config_ as member
var goes away.

[stefan-webrtc, 2017/03/27 10:10:27]

Done.

 
   if (!ValidateRtpExtensions(params.extensions)) {
     return false;
   }
   std::vector<webrtc::RtpExtension> filtered_extensions =
       FilterRtpExtensions(params.extensions,
                           webrtc::RtpExtension::IsSupportedForAudio, true);
   if (send_rtp_extensions_ != filtered_extensions) {
     send_rtp_extensions_.swap(filtered_extensions);
     for (auto& it : send_streams_) {
@@ skipping 996 matching lines @@
                       ssrc);
   if (it != unsignaled_recv_ssrcs_.end()) {
     unsignaled_recv_ssrcs_.erase(it);
     return true;
   }
   return false;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE