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Refactoring: move ownership of RtcEventLog from Call to PeerConnection

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This CL is a pure refactoring which should not result in any functinal
changes. It moves ownership of the RtcEventLog from webrtc::Call to the
webrtc::PeerConnection object.

This is done so that we can add RtcEventLog support for ICE events -
which will require the TransportController to have a pointer to the
RtcEventLog. PeerConnection is the closest common owner of both Call and
TransportController (through WebRtcSession).

BUG=webrtc:6393

Review-Url: https://codereview.webrtc.org/2353033005
Cr-Commit-Position: refs/heads/master@{#14578}
This commit is contained in:
skvlad
2016-10-07 11:53:05 -07:00
committed by Commit bot
parent 0d348d69e6
commit 11a9cbfa50
35 changed files with 162 additions and 118 deletions
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6
webrtc/call/call_perf_tests.cc
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@ -19,6 +19,7 @@
#include "webrtc/call.h"
#include "webrtc/call/transport_adapter.h"
#include "webrtc/config.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
@ -165,9 +166,9 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
AudioState::Config send_audio_state_config;
send_audio_state_config.voice_engine = voice_engine;
Call::Config sender_config;
Call::Config sender_config(&event_log_);
sender_config.audio_state = AudioState::Create(send_audio_state_config);
Call::Config receiver_config;
Call::Config receiver_config(&event_log_);
receiver_config.audio_state = sender_config.audio_state;
CreateCalls(sender_config, receiver_config);
@ -685,6 +686,7 @@ TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
Call::Config GetSenderCallConfig() override {
Call::Config config = EndToEndTest::GetSenderCallConfig();
config.event_log = &event_log_;
config.bitrate_config.start_bitrate_bps = kInitialBitrateKbps * 1000;
return config;
}