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}
diff --git a/modules/audio_coding/test/TestVADDTX.cc b/modules/audio_coding/test/TestVADDTX.cc
index 5865638..d211a6b 100644
--- a/modules/audio_coding/test/TestVADDTX.cc
+++ b/modules/audio_coding/test/TestVADDTX.cc
@@ -84,8 +84,11 @@
 
 // Encoding a file and see if the numbers that various packets occur follow
 // the expectation.
-void TestVadDtx::Run(std::string in_filename, int frequency, int channels,
-                     std::string out_filename, bool append,
+void TestVadDtx::Run(std::string in_filename,
+                     int frequency,
+                     int channels,
+                     std::string out_filename,
+                     bool append,
                      const int* expects) {
   monitor_->ResetStatistics();
 
@@ -146,13 +149,10 @@
 
 // Following is the implementation of TestWebRtcVadDtx.
 TestWebRtcVadDtx::TestWebRtcVadDtx()
-    : vad_enabled_(false),
-      dtx_enabled_(false),
-      output_file_num_(0) {
-}
+    : vad_enabled_(false), dtx_enabled_(false), output_file_num_(0) {}
 
 void TestWebRtcVadDtx::Perform() {
-  // Go through various test cases.
+// Go through various test cases.
 #ifdef WEBRTC_CODEC_ISAC
   // Register iSAC WB as send codec
   RegisterCodec(kIsacWb);
@@ -206,15 +206,14 @@
     output_file_num_++;
   }
   std::stringstream out_filename;
-  out_filename << webrtc::test::OutputPath()
-               << "testWebRtcVadDtx_outFile_"
-               << output_file_num_
-               << ".pcm";
-  Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
-      32000, 1, out_filename.str(), !new_outfile, expects);
+  out_filename << webrtc::test::OutputPath() << "testWebRtcVadDtx_outFile_"
+               << output_file_num_ << ".pcm";
+  Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"), 32000, 1,
+      out_filename.str(), !new_outfile, expects);
 }
 
-void TestWebRtcVadDtx::SetVAD(bool enable_dtx, bool enable_vad,
+void TestWebRtcVadDtx::SetVAD(bool enable_dtx,
+                              bool enable_vad,
                               ACMVADMode vad_mode) {
   ACMVADMode mode;
   EXPECT_EQ(0, acm_send_->SetVAD(enable_dtx, enable_vad, vad_mode));
@@ -227,10 +226,10 @@
     enable_dtx = enable_vad = false;
   }
 
-  EXPECT_EQ(dtx_enabled_ , enable_dtx); // DTX should be set as expected.
+  EXPECT_EQ(dtx_enabled_, enable_dtx);  // DTX should be set as expected.
 
   if (dtx_enabled_) {
-    EXPECT_TRUE(vad_enabled_); // WebRTC DTX cannot run without WebRTC VAD.
+    EXPECT_TRUE(vad_enabled_);  // WebRTC DTX cannot run without WebRTC VAD.
   } else {
     // Using no DTX should not affect setting of VAD.
     EXPECT_EQ(enable_vad, vad_enabled_);
@@ -250,19 +249,19 @@
   int expects[] = {0, 1, 0, 0, 0};
 
   // Register Opus as send codec
-  std::string out_filename = webrtc::test::OutputPath() +
-      "testOpusDtx_outFile_mono.pcm";
+  std::string out_filename =
+      webrtc::test::OutputPath() + "testOpusDtx_outFile_mono.pcm";
   RegisterCodec(kOpus);
   EXPECT_EQ(0, acm_send_->DisableOpusDtx());
 
-  Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
-      32000, 1, out_filename, false, expects);
+  Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"), 32000, 1,
+      out_filename, false, expects);
 
   EXPECT_EQ(0, acm_send_->EnableOpusDtx());
   expects[kEmptyFrame] = 1;
   expects[kAudioFrameCN] = 1;
-  Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
-      32000, 1, out_filename, true, expects);
+  Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"), 32000, 1,
+      out_filename, true, expects);
 
   // Register stereo Opus as send codec
   out_filename = webrtc::test::OutputPath() + "testOpusDtx_outFile_stereo.pcm";
@@ -270,15 +269,15 @@
   EXPECT_EQ(0, acm_send_->DisableOpusDtx());
   expects[kEmptyFrame] = 0;
   expects[kAudioFrameCN] = 0;
-  Run(webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm"),
-      32000, 2, out_filename, false, expects);
+  Run(webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm"), 32000,
+      2, out_filename, false, expects);
 
   EXPECT_EQ(0, acm_send_->EnableOpusDtx());
 
   expects[kEmptyFrame] = 1;
   expects[kAudioFrameCN] = 1;
-  Run(webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm"),
-      32000, 2, out_filename, true, expects);
+  Run(webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm"), 32000,
+      2, out_filename, true, expects);
 #endif
 }