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(); |