Move BitrateAllocator reference from ViEEncoder to VideoSendStream.

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 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 376 matching lines @@
 webrtc::VideoSendStream* Call::CreateVideoSendStream(
     const webrtc::VideoSendStream::Config& config,
     const VideoEncoderConfig& encoder_config) {
   TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
 
   // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if
   // the call has already started.
   VideoSendStream* send_stream = new VideoSendStream(
       num_cpu_cores_, module_process_thread_.get(), call_stats_.get(),
-      congestion_controller_.get(), &remb_, bitrate_allocator_.get(), config,
+      congestion_controller_.get(), bitrate_allocator_.get(), &remb_, config,
       encoder_config, suspended_video_send_ssrcs_);
 
   if (!network_enabled_)
     send_stream->SignalNetworkState(kNetworkDown);
 
   WriteLockScoped write_lock(*send_crit_);
   for (uint32_t ssrc : config.rtp.ssrcs) {
     RTC_DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end());
     video_send_ssrcs_[ssrc] = send_stream;
   }
@@ skipping 336 matching lines @@
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread());
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
 }  // namespace internal
 }  // namespace webrtc