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Reformat the WebRTC code base

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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}
This commit is contained in:
Yves Gerey
2018-06-19 15:03:05 +02:00
parent b602123a5a
commit 665174fdbb
1569 changed files with 30495 additions and 30309 deletions
Download Patch File Download Diff File Expand all files Collapse all files

27
modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
View File

@ -49,8 +49,7 @@ class RtcpRttStatsTestImpl : public RtcpRttStats {
int64_t rtt_ms_;
};
class SendTransport : public Transport,
public RtpData {
class SendTransport : public Transport, public RtpData {
public:
SendTransport()
: receiver_(nullptr),
@ -61,9 +60,7 @@ class SendTransport : public Transport,
keepalive_payload_type_(0),
num_keepalive_sent_(0) {}
void SetRtpRtcpModule(ModuleRtpRtcpImpl* receiver) {
receiver_ = receiver;
}
void SetRtpRtcpModule(ModuleRtpRtcpImpl* receiver) { receiver_ = receiver; }
void SimulateNetworkDelay(int64_t delay_ms, SimulatedClock* clock) {
clock_ = clock;
delay_ms_ = delay_ms;
@ -156,9 +153,7 @@ class RtpRtcpModule : public RtcpPacketTypeCounterObserver {
// Received RTCP stats for (own) local SSRC.
return counter_map_[impl_->SSRC()];
}
int RtpSent() {
return transport_.rtp_packets_sent_;
}
int RtpSent() { return transport_.rtp_packets_sent_; }
uint16_t LastRtpSequenceNumber() {
return transport_.last_rtp_header_.sequenceNumber;
}
@ -250,8 +245,8 @@ class RtpRtcpImplTest : public ::testing::Test {
const uint8_t payload[100] = {0};
EXPECT_EQ(true, module->impl_->SendOutgoingData(
kVideoFrameKey, codec_.plType, 0, 0, payload,
sizeof(payload), nullptr, &rtp_video_header, nullptr));
kVideoFrameKey, codec_.plType, 0, 0, payload,
sizeof(payload), nullptr, &rtp_video_header, nullptr));
}
void IncomingRtcpNack(const RtpRtcpModule* module, uint16_t sequence_number) {
@ -348,16 +343,16 @@ TEST_F(RtpRtcpImplTest, Rtt) {
int64_t avg_rtt;
int64_t min_rtt;
int64_t max_rtt;
EXPECT_EQ(0,
sender_.impl_->RTT(kReceiverSsrc, &rtt, &avg_rtt, &min_rtt, &max_rtt));
EXPECT_EQ(
0, sender_.impl_->RTT(kReceiverSsrc, &rtt, &avg_rtt, &min_rtt, &max_rtt));
EXPECT_NEAR(2 * kOneWayNetworkDelayMs, rtt, 1);
EXPECT_NEAR(2 * kOneWayNetworkDelayMs, avg_rtt, 1);
EXPECT_NEAR(2 * kOneWayNetworkDelayMs, min_rtt, 1);
EXPECT_NEAR(2 * kOneWayNetworkDelayMs, max_rtt, 1);
// No RTT from other ssrc.
EXPECT_EQ(-1,
sender_.impl_->RTT(kReceiverSsrc+1, &rtt, &avg_rtt, &min_rtt, &max_rtt));
EXPECT_EQ(-1, sender_.impl_->RTT(kReceiverSsrc + 1, &rtt, &avg_rtt, &min_rtt,
&max_rtt));
// Verify RTT from rtt_stats config.
EXPECT_EQ(0, sender_.rtt_stats_.LastProcessedRtt());
@ -464,8 +459,8 @@ TEST_F(RtpRtcpImplTest, AddStreamDataCounters) {
rtp.transmitted.header_bytes = 2;
rtp.transmitted.padding_bytes = 3;
EXPECT_EQ(rtp.transmitted.TotalBytes(), rtp.transmitted.payload_bytes +
rtp.transmitted.header_bytes +
rtp.transmitted.padding_bytes);
rtp.transmitted.header_bytes +
rtp.transmitted.padding_bytes);
StreamDataCounters rtp2;
rtp2.first_packet_time_ms = -1;