| 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;
|
| }
|
|
|
|
|