Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ )
Reason for revert:
Reverting all CLs related to moving the eventlog, as they break Chromium tests.
Original issue's description:
> Move RtcEventLog object from inside VoiceEngine to Call.
>
> In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
> The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.
>
> BUG=webrtc:4741,webrtc:5603,chromium:609749
> R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org
>
> Committed: https://crrev.com/1895526c6130e3d0e9b154f95079b8eda7567016
> Cr-Commit-Position: refs/heads/master@{#13321}
TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741,webrtc:5603,chromium:609749
Review-Url: https://codereview.webrtc.org/2111813002
Cr-Commit-Position: refs/heads/master@{#13340}
diff --git a/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc b/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
index f2a14ea..4f92a38 100644
--- a/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
+++ b/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
@@ -13,7 +13,6 @@
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/call/mock/mock_rtc_event_log.h"
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
#include "webrtc/modules/pacing/mock/mock_paced_sender.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -67,8 +66,8 @@
~BitrateControllerTest() {}
virtual void SetUp() {
- controller_ = BitrateController::CreateBitrateController(
- &clock_, &bitrate_observer_, &event_log_);
+ controller_ =
+ BitrateController::CreateBitrateController(&clock_, &bitrate_observer_);
controller_->SetStartBitrate(kStartBitrateBps);
EXPECT_EQ(kStartBitrateBps, bitrate_observer_.last_bitrate_);
controller_->SetMinMaxBitrate(kMinBitrateBps, kMaxBitrateBps);
@@ -92,7 +91,6 @@
TestBitrateObserver bitrate_observer_;
BitrateController* controller_;
RtcpBandwidthObserver* bandwidth_observer_;
- webrtc::MockRtcEventLog event_log_;
};
TEST_F(BitrateControllerTest, DefaultMinMaxBitrate) {
@@ -109,7 +107,6 @@
TEST_F(BitrateControllerTest, OneBitrateObserverOneRtcpObserver) {
// First REMB applies immediately.
- EXPECT_CALL(event_log_, LogBwePacketLossEvent(testing::Gt(0), 0, 0)).Times(8);
int64_t time_ms = 1001;
webrtc::ReportBlockList report_blocks;
report_blocks.push_back(CreateReportBlock(1, 2, 0, 1));
@@ -186,7 +183,6 @@
TEST_F(BitrateControllerTest, OneBitrateObserverTwoRtcpObservers) {
// REMBs during the first 2 seconds apply immediately.
- EXPECT_CALL(event_log_, LogBwePacketLossEvent(testing::Gt(0), 0, 0)).Times(9);
int64_t time_ms = 1;
webrtc::ReportBlockList report_blocks;
report_blocks.push_back(CreateReportBlock(1, 2, 0, 1));
@@ -282,13 +278,6 @@
}
TEST_F(BitrateControllerTest, OneBitrateObserverMultipleReportBlocks) {
- testing::Expectation first_calls =
- EXPECT_CALL(event_log_, LogBwePacketLossEvent(testing::Gt(0), 0, 0))
- .Times(7);
- EXPECT_CALL(event_log_,
- LogBwePacketLossEvent(testing::Gt(0), testing::Gt(0), 0))
- .Times(2)
- .After(first_calls);
uint32_t sequence_number[2] = {0, 0xFF00};
const int kStartBitrate = 200000;
const int kMinBitrate = 100000;