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}

modules/audio_coding/test/target_delay_unittest.cc

diff --git a/modules/audio_coding/test/target_delay_unittest.cc b/modules/audio_coding/test/target_delay_unittest.cc
index 89bf34f..7579d62 100644
--- a/modules/audio_coding/test/target_delay_unittest.cc
+++ b/modules/audio_coding/test/target_delay_unittest.cc
@@ -73,8 +73,8 @@
 
   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 @@
     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 @@
   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 @@
     return stats.preferredBufferSize;
   }
 
-  int RequiredDelay() {
-    return acm_->LeastRequiredDelayMs();
-  }
+  int RequiredDelay() { return acm_->LeastRequiredDelayMs(); }
 
   std::unique_ptr<AudioCodingModule> acm_;
   WebRtcRTPHeader rtp_info_;