| 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),
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| vie_receiver_(vie_channel_.vie_receiver()),
|
| vie_encoder_(num_cpu_cores,
|
| + config_.rtp.ssrcs,
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| module_process_thread_,
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| &stats_proxy_,
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| config.pre_encode_callback,
|
| @@ -216,8 +217,6 @@ VideoSendStream::VideoSendStream(
|
|
|
| call_stats_->RegisterStatsObserver(vie_channel_.GetStatsObserver());
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|
|
| - vie_encoder_.SetSsrcs(std::vector<uint32_t>(1, config_.rtp.ssrcs[0]));
|
| -
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| for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
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| 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
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| - // 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));
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| - vie_encoder_.SetSsrcs(used_ssrcs);
|
| -
|
| // Restart the media flow
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| vie_encoder_.Restart();
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|
|
|
|