Reformat the WebRTC code base
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}diff --git a/modules/bitrate_controller/bitrate_controller_unittest.cc b/modules/bitrate_controller/bitrate_controller_unittest.cc
index 8bd7800..ce930d1 100644
--- a/modules/bitrate_controller/bitrate_controller_unittest.cc
+++ b/modules/bitrate_controller/bitrate_controller_unittest.cc
@@ -26,28 +26,29 @@
using webrtc::PacedSender;
using webrtc::RtcpBandwidthObserver;
-uint8_t WeightedLoss(int num_packets1, uint8_t fraction_loss1,
- int num_packets2, uint8_t fraction_loss2) {
- int weighted_sum = num_packets1 * fraction_loss1 +
- num_packets2 * fraction_loss2;
+uint8_t WeightedLoss(int num_packets1,
+ uint8_t fraction_loss1,
+ int num_packets2,
+ uint8_t fraction_loss2) {
+ int weighted_sum =
+ num_packets1 * fraction_loss1 + num_packets2 * fraction_loss2;
int total_num_packets = num_packets1 + num_packets2;
return (weighted_sum + total_num_packets / 2) / total_num_packets;
}
webrtc::RTCPReportBlock CreateReportBlock(
- uint32_t remote_ssrc, uint32_t source_ssrc,
- uint8_t fraction_lost, uint32_t extended_high_sequence_number) {
+ uint32_t remote_ssrc,
+ uint32_t source_ssrc,
+ uint8_t fraction_lost,
+ uint32_t extended_high_sequence_number) {
return webrtc::RTCPReportBlock(remote_ssrc, source_ssrc, fraction_lost, 0,
extended_high_sequence_number, 0, 0, 0);
}
-class TestBitrateObserver: public BitrateObserver {
+class TestBitrateObserver : public BitrateObserver {
public:
TestBitrateObserver()
- : last_bitrate_(0),
- last_fraction_loss_(0),
- last_rtt_(0) {
- }
+ : last_bitrate_(0), last_fraction_loss_(0), last_rtt_(0) {}
virtual void OnNetworkChanged(uint32_t bitrate,
uint8_t fraction_loss,
@@ -76,8 +77,7 @@
bandwidth_observer_ = controller_.get();
}
- virtual void TearDown() {
- }
+ virtual void TearDown() {}
const int kMinBitrateBps = 100000;
const int kStartBitrateBps = 200000;
@@ -197,8 +197,8 @@
RtcpBandwidthObserver* second_bandwidth_observer = controller_.get();
report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 21)};
- second_bandwidth_observer->OnReceivedRtcpReceiverReport(
- report_blocks, 100, time_ms);
+ second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 100,
+ time_ms);
// Test start bitrate.
EXPECT_EQ(200000, bitrate_observer_.last_bitrate_);
@@ -212,8 +212,8 @@
time_ms += 500;
report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 21)};
- second_bandwidth_observer->OnReceivedRtcpReceiverReport(
- report_blocks, 100, time_ms);
+ second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 100,
+ time_ms);
EXPECT_EQ(217000, bitrate_observer_.last_bitrate_);
EXPECT_EQ(0, bitrate_observer_.last_fraction_loss_);
EXPECT_EQ(100, bitrate_observer_.last_rtt_);
@@ -221,8 +221,8 @@
// Extra report should not change estimate.
report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 31)};
- second_bandwidth_observer->OnReceivedRtcpReceiverReport(
- report_blocks, 100, time_ms);
+ second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 100,
+ time_ms);
EXPECT_EQ(217000, bitrate_observer_.last_bitrate_);
time_ms += 500;
@@ -232,34 +232,34 @@
// Second report should not change estimate.
report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 41)};
- second_bandwidth_observer->OnReceivedRtcpReceiverReport(
- report_blocks, 100, time_ms);
+ second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 100,
+ time_ms);
EXPECT_EQ(235360, bitrate_observer_.last_bitrate_);
time_ms += 1000;
// Reports from only one bandwidth observer is ok.
report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 61)};
- second_bandwidth_observer->OnReceivedRtcpReceiverReport(
- report_blocks, 50, time_ms);
+ second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 50,
+ time_ms);
EXPECT_EQ(255189, bitrate_observer_.last_bitrate_);
time_ms += 1000;
report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 81)};
- second_bandwidth_observer->OnReceivedRtcpReceiverReport(
- report_blocks, 50, time_ms);
+ second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 50,
+ time_ms);
EXPECT_EQ(276604, bitrate_observer_.last_bitrate_);
time_ms += 1000;
report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 121)};
- second_bandwidth_observer->OnReceivedRtcpReceiverReport(
- report_blocks, 50, time_ms);
+ second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 50,
+ time_ms);
EXPECT_EQ(299732, bitrate_observer_.last_bitrate_);
time_ms += 1000;
// Reach max cap.
report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 141)};
- second_bandwidth_observer->OnReceivedRtcpReceiverReport(
- report_blocks, 50, time_ms);
+ second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 50,
+ time_ms);
EXPECT_EQ(300000, bitrate_observer_.last_bitrate_);
// Test that a low REMB trigger immediately.