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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

19
video/rtp_streams_synchronizer.cc
View File

@ -25,10 +25,9 @@ bool UpdateMeasurements(StreamSynchronization::Measurements* stream,
stream->latest_timestamp = info.latest_received_capture_timestamp;
stream->latest_receive_time_ms = info.latest_receive_time_ms;
bool new_rtcp_sr = false;
if (!stream->rtp_to_ntp.UpdateMeasurements(info.capture_time_ntp_secs,
info.capture_time_ntp_frac,
info.capture_time_source_clock,
&new_rtcp_sr)) {
if (!stream->rtp_to_ntp.UpdateMeasurements(
info.capture_time_ntp_secs, info.capture_time_ntp_frac,
info.capture_time_source_clock, &new_rtcp_sr)) {
return false;
}
return true;
@ -63,7 +62,7 @@ int64_t RtpStreamsSynchronizer::TimeUntilNextProcess() {
RTC_DCHECK_RUN_ON(&process_thread_checker_);
const int64_t kSyncIntervalMs = 1000;
return kSyncIntervalMs -
(rtc::TimeNanos() - last_sync_time_) / rtc::kNumNanosecsPerMillisec;
(rtc::TimeNanos() - last_sync_time_) / rtc::kNumNanosecsPerMillisec;
}
void RtpStreamsSynchronizer::Process() {
@ -100,18 +99,16 @@ void RtpStreamsSynchronizer::Process() {
}
TRACE_COUNTER1("webrtc", "SyncCurrentVideoDelay",
video_info->current_delay_ms);
video_info->current_delay_ms);
TRACE_COUNTER1("webrtc", "SyncCurrentAudioDelay",
audio_info->current_delay_ms);
audio_info->current_delay_ms);
TRACE_COUNTER1("webrtc", "SyncRelativeDelay", relative_delay_ms);
int target_audio_delay_ms = 0;
int target_video_delay_ms = video_info->current_delay_ms;
// Calculate the necessary extra audio delay and desired total video
// delay to get the streams in sync.
if (!sync_->ComputeDelays(relative_delay_ms,
audio_info->current_delay_ms,
&target_audio_delay_ms,
&target_video_delay_ms)) {
if (!sync_->ComputeDelays(relative_delay_ms, audio_info->current_delay_ms,
&target_audio_delay_ms, &target_video_delay_ms)) {
return;
}