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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

16
modules/audio_coding/test/target_delay_unittest.cc
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@ -73,8 +73,8 @@ class TargetDelayTest : public ::testing::Test {
void WithTargetDelayBufferNotChanging() {
// A target delay that is one packet larger than jitter.
const int kTargetDelayMs = (kInterarrivalJitterPacket + 1) *
kNum10msPerFrame * 10;
const int kTargetDelayMs =
(kInterarrivalJitterPacket + 1) * kNum10msPerFrame * 10;
ASSERT_EQ(0, SetMinimumDelay(kTargetDelayMs));
for (int n = 0; n < 30; ++n) // Run enough iterations to fill the buffer.
Run(true);
@ -91,8 +91,8 @@ class TargetDelayTest : public ::testing::Test {
int clean_optimal_delay = GetCurrentOptimalDelayMs();
// A relatively large delay.
const int kTargetDelayMs = (kInterarrivalJitterPacket + 10) *
kNum10msPerFrame * 10;
const int kTargetDelayMs =
(kInterarrivalJitterPacket + 10) * kNum10msPerFrame * 10;
ASSERT_EQ(0, SetMinimumDelay(kTargetDelayMs));
for (int n = 0; n < 300; ++n) // Run enough iterations to fill the buffer.
Run(true);
@ -146,8 +146,8 @@ class TargetDelayTest : public ::testing::Test {
void Push() {
rtp_info_.header.timestamp += kFrameSizeSamples;
rtp_info_.header.sequenceNumber++;
ASSERT_EQ(0, acm_->IncomingPacket(payload_, kFrameSizeSamples * 2,
rtp_info_));
ASSERT_EQ(0,
acm_->IncomingPacket(payload_, kFrameSizeSamples * 2, rtp_info_));
}
// Pull audio equivalent to the amount of audio in one RTP packet.
@ -195,9 +195,7 @@ class TargetDelayTest : public ::testing::Test {
return stats.preferredBufferSize;
}
int RequiredDelay() {
return acm_->LeastRequiredDelayMs();
}
int RequiredDelay() { return acm_->LeastRequiredDelayMs(); }
std::unique_ptr<AudioCodingModule> acm_;
WebRtcRTPHeader rtp_info_;