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/RTPFile.cc b/modules/audio_coding/test/RTPFile.cc
index 8cc5bd9..a1329e7 100644
--- a/modules/audio_coding/test/RTPFile.cc
+++ b/modules/audio_coding/test/RTPFile.cc
@@ -14,9 +14,9 @@
 #include <limits>
 
 #ifdef WIN32
-#   include <Winsock2.h>
+#include <Winsock2.h>
 #else
-#   include <arpa/inet.h>
+#include <arpa/inet.h>
 #endif
 
 #include "modules/include/module_common_types.h"
@@ -29,18 +29,22 @@
 void RTPStream::ParseRTPHeader(WebRtcRTPHeader* rtpInfo,
                                const uint8_t* rtpHeader) {
   rtpInfo->header.payloadType = rtpHeader[1];
-  rtpInfo->header.sequenceNumber = (static_cast<uint16_t>(rtpHeader[2]) << 8) |
-      rtpHeader[3];
+  rtpInfo->header.sequenceNumber =
+      (static_cast<uint16_t>(rtpHeader[2]) << 8) | rtpHeader[3];
   rtpInfo->header.timestamp = (static_cast<uint32_t>(rtpHeader[4]) << 24) |
-      (static_cast<uint32_t>(rtpHeader[5]) << 16) |
-      (static_cast<uint32_t>(rtpHeader[6]) << 8) | rtpHeader[7];
+                              (static_cast<uint32_t>(rtpHeader[5]) << 16) |
+                              (static_cast<uint32_t>(rtpHeader[6]) << 8) |
+                              rtpHeader[7];
   rtpInfo->header.ssrc = (static_cast<uint32_t>(rtpHeader[8]) << 24) |
-      (static_cast<uint32_t>(rtpHeader[9]) << 16) |
-      (static_cast<uint32_t>(rtpHeader[10]) << 8) | rtpHeader[11];
+                         (static_cast<uint32_t>(rtpHeader[9]) << 16) |
+                         (static_cast<uint32_t>(rtpHeader[10]) << 8) |
+                         rtpHeader[11];
 }
 
-void RTPStream::MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
-                              int16_t seqNo, uint32_t timeStamp,
+void RTPStream::MakeRTPheader(uint8_t* rtpHeader,
+                              uint8_t payloadType,
+                              int16_t seqNo,
+                              uint32_t timeStamp,
                               uint32_t ssrc) {
   rtpHeader[0] = 0x80;
   rtpHeader[1] = payloadType;
@@ -56,8 +60,11 @@
   rtpHeader[11] = ssrc & 0xFF;
 }
 
-RTPPacket::RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo,
-                     const uint8_t* payloadData, size_t payloadSize,
+RTPPacket::RTPPacket(uint8_t payloadType,
+                     uint32_t timeStamp,
+                     int16_t seqNo,
+                     const uint8_t* payloadData,
+                     size_t payloadSize,
                      uint32_t frequency)
     : payloadType(payloadType),
       timeStamp(timeStamp),
@@ -82,20 +89,25 @@
   delete _queueRWLock;
 }
 
-void RTPBuffer::Write(const uint8_t payloadType, const uint32_t timeStamp,
-                      const int16_t seqNo, const uint8_t* payloadData,
-                      const size_t payloadSize, uint32_t frequency) {
-  RTPPacket *packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData,
+void RTPBuffer::Write(const uint8_t payloadType,
+                      const uint32_t timeStamp,
+                      const int16_t seqNo,
+                      const uint8_t* payloadData,
+                      const size_t payloadSize,
+                      uint32_t frequency) {
+  RTPPacket* packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData,
                                     payloadSize, frequency);
   _queueRWLock->AcquireLockExclusive();
   _rtpQueue.push(packet);
   _queueRWLock->ReleaseLockExclusive();
 }
 
-size_t RTPBuffer::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
-                       size_t payloadSize, uint32_t* offset) {
+size_t RTPBuffer::Read(WebRtcRTPHeader* rtpInfo,
+                       uint8_t* payloadData,
+                       size_t payloadSize,
+                       uint32_t* offset) {
   _queueRWLock->AcquireLockShared();
-  RTPPacket *packet = _rtpQueue.front();
+  RTPPacket* packet = _rtpQueue.front();
   _rtpQueue.pop();
   _queueRWLock->ReleaseLockShared();
   rtpInfo->header.markerBit = 1;
@@ -120,7 +132,7 @@
   return eof;
 }
 
-void RTPFile::Open(const char *filename, const char *mode) {
+void RTPFile::Open(const char* filename, const char* mode) {
   if ((_rtpFile = fopen(filename, mode)) == NULL) {
     printf("Cannot write file %s.\n", filename);
     ADD_FAILURE() << "Unable to write file";
@@ -165,9 +177,12 @@
   padding = ntohs(padding);
 }
 
-void RTPFile::Write(const uint8_t payloadType, const uint32_t timeStamp,
-                    const int16_t seqNo, const uint8_t* payloadData,
-                    const size_t payloadSize, uint32_t frequency) {
+void RTPFile::Write(const uint8_t payloadType,
+                    const uint32_t timeStamp,
+                    const int16_t seqNo,
+                    const uint8_t* payloadData,
+                    const size_t payloadSize,
+                    uint32_t frequency) {
   /* write RTP packet to file */
   uint8_t rtpHeader[12];
   MakeRTPheader(rtpHeader, payloadType, seqNo, timeStamp, 0);
@@ -185,8 +200,10 @@
   EXPECT_EQ(payloadSize, fwrite(payloadData, 1, payloadSize, _rtpFile));
 }
 
-size_t RTPFile::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
-                     size_t payloadSize, uint32_t* offset) {
+size_t RTPFile::Read(WebRtcRTPHeader* rtpInfo,
+                     uint8_t* payloadData,
+                     size_t payloadSize,
+                     uint32_t* offset) {
   uint16_t lengthBytes;
   uint16_t plen;
   uint8_t rtpHeader[12];