Index: webrtc/call/call.cc |
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc |
index 9b061623704b18a3bae3477ec53d07d7e5f83179..a30bccbfcc2e3de97cb4925538f9bd881e24d939 100644 |
--- a/webrtc/call/call.cc |
+++ b/webrtc/call/call.cc |
@@ -27,6 +27,12 @@ |
#include "webrtc/call/flexfec_receive_stream_impl.h" |
#include "webrtc/call/rtp_stream_receiver_controller.h" |
#include "webrtc/call/rtp_transport_controller_send.h" |
+#include "webrtc/logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h" |
+#include "webrtc/logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h" |
+#include "webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h" |
+#include "webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h" |
+#include "webrtc/logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h" |
+#include "webrtc/logging/rtc_event_log/events/rtc_event_video_send_stream_config.h" |
#include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
#include "webrtc/logging/rtc_event_log/rtc_stream_config.h" |
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" |
@@ -609,7 +615,8 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream( |
const webrtc::AudioSendStream::Config& config) { |
TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream"); |
RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
- event_log_->LogAudioSendStreamConfig(*CreateRtcLogStreamConfig(config)); |
+ event_log_->Log(rtc::MakeUnique<RtcEventAudioSendStreamConfig>( |
+ CreateRtcLogStreamConfig(config))); |
rtc::Optional<RtpState> suspended_rtp_state; |
{ |
@@ -675,7 +682,8 @@ webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream( |
const webrtc::AudioReceiveStream::Config& config) { |
TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream"); |
RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
- event_log_->LogAudioReceiveStreamConfig(*CreateRtcLogStreamConfig(config)); |
+ event_log_->Log(rtc::MakeUnique<RtcEventAudioReceiveStreamConfig>( |
+ CreateRtcLogStreamConfig(config))); |
AudioReceiveStream* receive_stream = new AudioReceiveStream( |
&audio_receiver_controller_, transport_send_->packet_router(), config, |
config_.audio_state, event_log_); |
@@ -735,8 +743,8 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream( |
video_send_delay_stats_->AddSsrcs(config); |
for (size_t ssrc_index = 0; ssrc_index < config.rtp.ssrcs.size(); |
++ssrc_index) { |
- event_log_->LogVideoSendStreamConfig( |
- *CreateRtcLogStreamConfig(config, ssrc_index)); |
+ event_log_->Log(rtc::MakeUnique<RtcEventVideoSendStreamConfig>( |
+ CreateRtcLogStreamConfig(config, ssrc_index))); |
} |
// TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if |
@@ -826,7 +834,8 @@ webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream( |
} |
receive_stream->SignalNetworkState(video_network_state_); |
UpdateAggregateNetworkState(); |
- event_log_->LogVideoReceiveStreamConfig(*CreateRtcLogStreamConfig(config)); |
+ event_log_->Log(rtc::MakeUnique<RtcEventVideoReceiveStreamConfig>( |
+ CreateRtcLogStreamConfig(config))); |
return receive_stream; |
} |
@@ -1302,8 +1311,10 @@ PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type, |
} |
} |
- if (rtcp_delivered) |
- event_log_->LogIncomingRtcpPacket(rtc::MakeArrayView(packet, length)); |
+ if (rtcp_delivered) { |
+ event_log_->Log(rtc::MakeUnique<RtcEventRtcpPacketIncoming>( |
+ rtc::MakeArrayView(packet, length))); |
+ } |
return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR; |
} |
@@ -1352,7 +1363,8 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type, |
if (audio_receiver_controller_.OnRtpPacket(*parsed_packet)) { |
received_bytes_per_second_counter_.Add(static_cast<int>(length)); |
received_audio_bytes_per_second_counter_.Add(static_cast<int>(length)); |
- event_log_->LogIncomingRtpHeader(*parsed_packet); |
+ event_log_->Log( |
+ rtc::MakeUnique<RtcEventRtpPacketIncoming>(*parsed_packet)); |
const int64_t arrival_time_ms = parsed_packet->arrival_time_ms(); |
if (!first_received_rtp_audio_ms_) { |
first_received_rtp_audio_ms_.emplace(arrival_time_ms); |
@@ -1364,7 +1376,8 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type, |
if (video_receiver_controller_.OnRtpPacket(*parsed_packet)) { |
received_bytes_per_second_counter_.Add(static_cast<int>(length)); |
received_video_bytes_per_second_counter_.Add(static_cast<int>(length)); |
- event_log_->LogIncomingRtpHeader(*parsed_packet); |
+ event_log_->Log( |
+ rtc::MakeUnique<RtcEventRtpPacketIncoming>(*parsed_packet)); |
const int64_t arrival_time_ms = parsed_packet->arrival_time_ms(); |
if (!first_received_rtp_video_ms_) { |
first_received_rtp_video_ms_.emplace(arrival_time_ms); |