Index: webrtc/call/call.cc |
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc |
index 802778eb4d6eda4cf55c642fb6f35aaa85074de5..f9f1fb5eab5c0abebeb3c79809cafda92acc7f63 100644 |
--- a/webrtc/call/call.cc |
+++ b/webrtc/call/call.cc |
@@ -88,6 +88,25 @@ bool UseSendSideBwe(const FlexfecReceiveStream::Config& config) { |
return UseSendSideBwe(config.rtp_header_extensions, config.transport_cc); |
} |
+rtclog::StreamConfig CreateRtcLogStreamConfig( |
+ const VideoReceiveStream::Config& config) { |
+ rtclog::StreamConfig rtclog_config; |
+ rtclog_config.remote_ssrc = config.rtp.remote_ssrc; |
+ rtclog_config.local_ssrc = config.rtp.local_ssrc; |
+ rtclog_config.rtx_ssrc = config.rtp.rtx_ssrc; |
+ rtclog_config.rtcp_mode = config.rtp.rtcp_mode; |
+ rtclog_config.remb = config.rtp.remb; |
+ rtclog_config.rtp_extensions = config.rtp.extensions; |
+ |
+ for (const auto& d : config.decoders) { |
+ auto search = config.rtp.rtx_payload_types.find(d.payload_type); |
+ rtclog_config.codecs.emplace_back( |
+ d.payload_name, d.payload_type, |
+ search != config.rtp.rtx_payload_types.end() ? search->second : 0); |
+ } |
+ return rtclog_config; |
+} |
+ |
} // namespace |
namespace internal { |
@@ -710,7 +729,7 @@ webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream( |
} |
receive_stream->SignalNetworkState(video_network_state_); |
UpdateAggregateNetworkState(); |
- event_log_->LogVideoReceiveStreamConfig(config); |
+ event_log_->LogVideoReceiveStreamConfig(CreateRtcLogStreamConfig(config)); |
return receive_stream; |
} |