Simplify SSRC usage inside 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 170 matching lines @@
                        ->CreateRtcpBandwidthObserver(),
                    congestion_controller_->GetTransportFeedbackObserver(),
                    nullptr,
                    call_stats_->rtcp_rtt_stats(),
                    congestion_controller_->pacer(),
                    congestion_controller_->packet_router(),
                    config_.rtp.ssrcs.size(),
                    true),
       vie_receiver_(vie_channel_.vie_receiver()),
       vie_encoder_(num_cpu_cores,
+                   config_.rtp.ssrcs,
                    module_process_thread_,
                    &stats_proxy_,
                    config.pre_encode_callback,
                    &overuse_detector_,
                    congestion_controller_->pacer(),
                    &payload_router_,
                    bitrate_allocator),
       vcm_(vie_encoder_.vcm()),
       input_(&vie_encoder_,
              config_.local_renderer,
              &stats_proxy_,
              &overuse_detector_) {
   LOG(LS_INFO) << "VideoSendStream: " << config_.ToString();
 
   RTC_DCHECK(!config_.rtp.ssrcs.empty());
   RTC_DCHECK(module_process_thread_);
   RTC_DCHECK(call_stats_);
   RTC_DCHECK(congestion_controller_);
   RTC_DCHECK(remb_);
 
   RTC_CHECK(vie_encoder_.Init());
   encoder_feedback_.Init(config_.rtp.ssrcs, &vie_encoder_);
   RTC_CHECK(vie_channel_.Init() == 0);
 
   vcm_->RegisterProtectionCallback(vie_channel_.vcm_protection_callback());
 
   call_stats_->RegisterStatsObserver(vie_channel_.GetStatsObserver());
 
-  vie_encoder_.SetSsrcs(std::vector<uint32_t>(1, config_.rtp.ssrcs[0]));
-
   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);
     if (extension == RtpExtension::kTOffset) {
       RTC_CHECK_EQ(0, vie_channel_.SetSendTimestampOffsetStatus(true, id));
     } else if (extension == RtpExtension::kAbsSendTime) {
       RTC_CHECK_EQ(0, vie_channel_.SetSendAbsoluteSendTimeStatus(true, id));
@@ skipping 364 matching lines @@
   if (vie_encoder_.SetEncoder(video_codec) != 0) {
     LOG(LS_ERROR) << "Failed to set encoder.";
     return false;
   }
 
   if (vie_channel_.SetSendCodec(video_codec, false) != 0) {
     LOG(LS_ERROR) << "Failed to set send codec.";
     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));
-  vie_encoder_.SetSsrcs(used_ssrcs);
-
   // Restart the media flow
   vie_encoder_.Restart();
 
   return true;
 }
 }  // namespace internal
 }  // namespace webrtc