| Index: webrtc/video/video_send_stream.cc | 
| diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc | 
| index e6674c8531d5ec99945a38b753fc37a3c7fe96fc..db00adf24edc1a92f379df6a6c42d2918f150de6 100644 | 
| --- a/webrtc/video/video_send_stream.cc | 
| +++ b/webrtc/video/video_send_stream.cc | 
| @@ -188,6 +188,7 @@ VideoSendStream::VideoSendStream( | 
| true), | 
| vie_receiver_(vie_channel_.vie_receiver()), | 
| vie_encoder_(num_cpu_cores, | 
| +                   config_.rtp.ssrcs, | 
| module_process_thread_, | 
| &stats_proxy_, | 
| config.pre_encode_callback, | 
| @@ -216,8 +217,6 @@ VideoSendStream::VideoSendStream( | 
|  | 
| 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; | 
| @@ -602,12 +601,6 @@ bool VideoSendStream::SetSendCodec(VideoCodec video_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(); | 
|  | 
|  |