Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(20)

Unified Diff: webrtc/call/call.cc

Issue 3016473002: Remove encoding code from RtcEventLogImpl and use RtcEventLogEncoder instead (Closed)
Patch Set: Created 3 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « no previous file | webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
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);
« no previous file with comments | « no previous file | webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698