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/PacketLossTest.cc

diff --git a/modules/audio_coding/test/PacketLossTest.cc b/modules/audio_coding/test/PacketLossTest.cc
index d5fbfd1..c5cb396 100644
--- a/modules/audio_coding/test/PacketLossTest.cc
+++ b/modules/audio_coding/test/PacketLossTest.cc
@@ -23,11 +23,10 @@
       burst_length_(1),
       packet_counter_(0),
       lost_packet_counter_(0),
-      burst_lost_counter_(burst_length_) {
-}
+      burst_lost_counter_(burst_length_) {}
 
-void ReceiverWithPacketLoss::Setup(AudioCodingModule *acm,
-                                   RTPStream *rtpStream,
+void ReceiverWithPacketLoss::Setup(AudioCodingModule* acm,
+                                   RTPStream* rtpStream,
                                    std::string out_file_name,
                                    int channels,
                                    int loss_rate,
@@ -84,13 +83,14 @@
   return false;
 }
 
-SenderWithFEC::SenderWithFEC()
-    : expected_loss_rate_(0) {
-}
+SenderWithFEC::SenderWithFEC() : expected_loss_rate_(0) {}
 
-void SenderWithFEC::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
-                          std::string in_file_name, int sample_rate,
-                          int channels, int expected_loss_rate) {
+void SenderWithFEC::Setup(AudioCodingModule* acm,
+                          RTPStream* rtpStream,
+                          std::string in_file_name,
+                          int sample_rate,
+                          int channels,
+                          int expected_loss_rate) {
   Sender::Setup(acm, rtpStream, in_file_name, sample_rate, channels);
   EXPECT_TRUE(SetFEC(true));
   EXPECT_TRUE(SetPacketLossRate(expected_loss_rate));
@@ -111,18 +111,19 @@
   return false;
 }
 
-PacketLossTest::PacketLossTest(int channels, int expected_loss_rate,
-                               int actual_loss_rate, int burst_length)
+PacketLossTest::PacketLossTest(int channels,
+                               int expected_loss_rate,
+                               int actual_loss_rate,
+                               int burst_length)
     : channels_(channels),
-      in_file_name_(channels_ == 1 ? "audio_coding/testfile32kHz" :
-                    "audio_coding/teststereo32kHz"),
+      in_file_name_(channels_ == 1 ? "audio_coding/testfile32kHz"
+                                   : "audio_coding/teststereo32kHz"),
       sample_rate_hz_(32000),
       sender_(new SenderWithFEC),
       receiver_(new ReceiverWithPacketLoss),
       expected_loss_rate_(expected_loss_rate),
       actual_loss_rate_(actual_loss_rate),
-      burst_length_(burst_length) {
-}
+      burst_length_(burst_length) {}
 
 void PacketLossTest::Perform() {
 #ifndef WEBRTC_CODEC_OPUS