commit 665174fdbb4e0540eccb27cf7412348f1b65534c
author Yves Gerey <yvesg@webrtc.org> Tue Jun 19 15:03:05 2018 +0200
committer Yves Gerey <yvesg@webrtc.org> Tue Jun 19 14:00:39 2018 +0000
tree 34a9eb770e6de54c282f62ad0b65d2cd69c917f6
parent b602123a5aae1a9f0b6b581e611acf44eb525c08

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}

video/rtp_streams_synchronizer.cc

diff --git a/video/rtp_streams_synchronizer.cc b/video/rtp_streams_synchronizer.cc
index cc7f893..e46a220 100644
--- a/video/rtp_streams_synchronizer.cc
+++ b/video/rtp_streams_synchronizer.cc
@@ -25,10 +25,9 @@
   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 @@
   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 @@
   }
 
   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;
   }