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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

26
modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -79,10 +79,14 @@ const char* FrameTypeToString(FrameType frame_type) {
switch (frame_type) {
case kEmptyFrame:
return "empty";
case kAudioFrameSpeech: return "audio_speech";
case kAudioFrameCN: return "audio_cn";
case kVideoFrameKey: return "video_key";
case kVideoFrameDelta: return "video_delta";
case kAudioFrameSpeech:
return "audio_speech";
case kAudioFrameCN:
return "audio_cn";
case kVideoFrameKey:
return "video_key";
case kVideoFrameDelta:
return "video_delta";
}
return "";
}
@ -260,8 +264,8 @@ int32_t RTPSender::RegisterPayload(
RTC_DCHECK(payload);
// Check if it's the same as we already have.
if (RtpUtility::StringCompare(
payload->name, payload_name, RTP_PAYLOAD_NAME_SIZE - 1)) {
if (RtpUtility::StringCompare(payload->name, payload_name,
RTP_PAYLOAD_NAME_SIZE - 1)) {
if (audio_configured_ && payload->typeSpecific.is_audio()) {
auto& p = payload->typeSpecific.audio_payload();
if (rtc::SafeEq(p.format.clockrate_hz, frequency) &&
@ -439,8 +443,8 @@ bool RTPSender::SendOutgoingData(FrameType frame_type,
result = audio_->SendAudio(frame_type, payload_type, rtp_timestamp,
payload_data, payload_size);
} else {
TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", capture_time_ms,
"Send", "type", FrameTypeToString(frame_type));
TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", capture_time_ms, "Send", "type",
FrameTypeToString(frame_type));
if (frame_type == kEmptyFrame)
return true;
@ -983,9 +987,9 @@ void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
rtc::CritScope cs(&statistics_crit_);
// TODO(holmer): Compute this iteratively instead.
send_delays_[now_ms] = now_ms - capture_time_ms;
send_delays_.erase(send_delays_.begin(),
send_delays_.lower_bound(now_ms -
kSendSideDelayWindowMs));
send_delays_.erase(
send_delays_.begin(),
send_delays_.lower_bound(now_ms - kSendSideDelayWindowMs));
int num_delays = 0;
for (auto it = send_delays_.upper_bound(now_ms - kSendSideDelayWindowMs);
it != send_delays_.end(); ++it) {