Remove UpdateSsrcs from EncoderStateFeedback.

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.
  */
 
 #include "webrtc/video/video_send_stream.h"
 
 #include <algorithm>
 #include <sstream>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/system_wrappers/interface/trace_event.h"
 #include "webrtc/video/video_capture_input.h"
-#include "webrtc/video_engine/encoder_state_feedback.h"
 #include "webrtc/video_engine/vie_channel.h"
 #include "webrtc/video_engine/vie_channel_group.h"
 #include "webrtc/video_engine/vie_defines.h"
 #include "webrtc/video_engine/vie_encoder.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 std::string
 VideoSendStream::Config::EncoderSettings::ToString() const {
   std::stringstream ss;
@@ skipping 80 matching lines @@
       encoded_frame_proxy_(config.post_encode_callback),
       config_(config),
       suspended_ssrcs_(suspended_ssrcs),
       module_process_thread_(module_process_thread),
       channel_group_(channel_group),
       channel_id_(channel_id),
       use_config_bitrate_(true),
       stats_proxy_(Clock::GetRealTimeClock(), config) {
   DCHECK(!config_.rtp.ssrcs.empty());
   CHECK(channel_group->CreateSendChannel(channel_id_, 0, &transport_adapter_,
-                                         num_cpu_cores,
-                                         config_.rtp.ssrcs.size(), true));
+                                         num_cpu_cores, config_.rtp.ssrcs,
+                                         true));
   vie_channel_ = channel_group_->GetChannel(channel_id_);
   vie_encoder_ = channel_group_->GetEncoder(channel_id_);
 
   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.
     DCHECK_GE(id, 1);
     DCHECK_LE(id, 14);
     if (extension == RtpExtension::kTOffset) {
@@ skipping 352 matching lines @@
     return false;
   }
 
   // Not all configured SSRCs have to be utilized (simulcast senders don't have
   // to send on all SSRCs at once etc.)
   std::vector<uint32_t> used_ssrcs = config_.rtp.ssrcs;
   used_ssrcs.resize(static_cast<size_t>(video_codec.numberOfSimulcastStreams));
 
   // Update used SSRCs.
   vie_encoder_->SetSsrcs(used_ssrcs);
-  EncoderStateFeedback* encoder_state_feedback =
-      channel_group_->GetEncoderStateFeedback();
-  encoder_state_feedback->UpdateSsrcs(used_ssrcs, vie_encoder_);
 
   // Update the protection mode, we might be switching NACK/FEC.
   vie_encoder_->UpdateProtectionMethod(vie_encoder_->nack_enabled(),
                                        vie_channel_->IsSendingFecEnabled());
 
   // Restart the media flow
   vie_encoder_->Restart();
 
   return true;
 }
 
 }  // namespace internal
 }  // namespace webrtc