Use vcm::VideoSender in ViEEncoder.

webrtc/video/video_send_stream.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 364 matching lines @@
       vie_encoder_(
           num_cpu_cores,
           config_.rtp.ssrcs,
           module_process_thread_,
           &stats_proxy_,
           config.pre_encode_callback,
           &overuse_detector_,
           congestion_controller_->pacer(),
           &payload_router_,
           config.post_encode_callback ? &encoded_frame_proxy_ : nullptr),
-      vcm_(vie_encoder_.vcm()),
+      video_sender_(vie_encoder_.video_sender()),
       bandwidth_observer_(congestion_controller_->GetBitrateController()
                               ->CreateRtcpBandwidthObserver()),
       rtp_rtcp_modules_(CreateRtpRtcpModules(
           config.send_transport,
           &encoder_feedback_,
           bandwidth_observer_.get(),
           congestion_controller_->GetTransportFeedbackObserver(),
           call_stats_->rtcp_rtt_stats(),
           congestion_controller_->pacer(),
           congestion_controller_->packet_router(),
@@ skipping 14 matching lines @@
 
   RTC_CHECK(vie_encoder_.Init());
   encoder_feedback_.Init(config_.rtp.ssrcs, &vie_encoder_);
 
   // RTP/RTCP initialization.
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
     module_process_thread_->RegisterModule(rtp_rtcp);
     congestion_controller_->packet_router()->AddRtpModule(rtp_rtcp);
   }
 
-  vcm_->RegisterProtectionCallback(this);
+  video_sender_->RegisterProtectionCallback(this);
 
   for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
     const std::string& extension = config_.rtp.extensions[i].name;
     int id = config_.rtp.extensions[i].id;
     // One-byte-extension local identifiers are in the range 1-14 inclusive.
     RTC_DCHECK_GE(id, 1);
     RTC_DCHECK_LE(id, 14);
     RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
     for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
       RTC_CHECK_EQ(0, rtp_rtcp->RegisterSendRtpHeaderExtension(
@@ skipping 109 matching lines @@
     }
     if (encoder_settings) {
       encoder_settings->video_codec.startBitrate =
           bitrate_allocator_->AddObserver(
               this, encoder_settings->video_codec.minBitrate * 1000,
               encoder_settings->video_codec.maxBitrate * 1000) /
           1000;
       vie_encoder_.SetEncoder(encoder_settings->video_codec,
                               encoder_settings->min_transmit_bitrate_bps);
       if (config_.suspend_below_min_bitrate) {
-        vcm_->SuspendBelowMinBitrate();
+        video_sender_->SuspendBelowMinBitrate();
         bitrate_allocator_->EnforceMinBitrate(false);
       }
       // We might've gotten new settings while configuring the encoder settings,
       // restart from the top to see if that's the case before trying to encode
       // a frame (which might correspond to the last frame size).
       encoder_wakeup_event_.Set();
       continue;
     }
 
     VideoFrame frame;
@@ skipping 196 matching lines @@
                           &module_nack_rate);
     *sent_video_rate_bps += module_video_rate;
     *sent_nack_rate_bps += module_nack_rate;
     *sent_fec_rate_bps += module_fec_rate;
   }
   return 0;
 }
 
 }  // namespace internal
 }  // namespace webrtc