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/rtp_rtcp/include/receive_statistics.h

@@ -16,8 +16,8 @@
 
 #include "modules/include/module.h"
 #include "modules/include/module_common_types.h"
-#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
 #include "typedefs.h"  // NOLINT(build/include)
 
 namespace webrtc {

modules/rtp_rtcp/include/remote_ntp_time_estimator.h

@@ -33,7 +33,9 @@ class RemoteNtpTimeEstimator {
 
   // Updates the estimator with round trip time |rtt|, NTP seconds |ntp_secs|,
   // NTP fraction |ntp_frac| and RTP timestamp |rtcp_timestamp|.
-  bool UpdateRtcpTimestamp(int64_t rtt, uint32_t ntp_secs, uint32_t ntp_frac,
+  bool UpdateRtcpTimestamp(int64_t rtt,
+                           uint32_t ntp_secs,
+                           uint32_t ntp_frac,
                            uint32_t rtp_timestamp);
 
   // Estimates the NTP timestamp in local timebase from |rtp_timestamp|.

modules/rtp_rtcp/include/rtp_payload_registry.h

@@ -68,8 +68,8 @@ class RTPPayloadRegistry {
   std::map<int, RtpUtility::Payload> payload_type_map_;
   int8_t last_received_payload_type_;
 
-  // As a first step in splitting this class up in separate cases for audio and
-  // video, DCHECK that no instance is used for both audio and video.
+// As a first step in splitting this class up in separate cases for audio and
+// video, DCHECK that no instance is used for both audio and video.
 #if RTC_DCHECK_IS_ON
   bool used_for_audio_ RTC_GUARDED_BY(crit_sect_) = false;
   bool used_for_video_ RTC_GUARDED_BY(crit_sect_) = false;

modules/rtp_rtcp/include/rtp_rtcp_defines.h

@@ -22,8 +22,8 @@
 #include "system_wrappers/include/clock.h"
 #include "typedefs.h"  // NOLINT(build/include)
 
-#define RTCP_CNAME_SIZE 256    // RFC 3550 page 44, including null termination
-#define IP_PACKET_SIZE 1500    // we assume ethernet
+#define RTCP_CNAME_SIZE 256  // RFC 3550 page 44, including null termination
+#define IP_PACKET_SIZE 1500  // we assume ethernet
 #define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10
 
 namespace webrtc {
@@ -97,15 +97,9 @@ class PayloadUnion {
 
 enum RTPAliveType { kRtpDead = 0, kRtpNoRtp = 1, kRtpAlive = 2 };
 
-enum ProtectionType {
-  kUnprotectedPacket,
-  kProtectedPacket
-};
+enum ProtectionType { kUnprotectedPacket, kProtectedPacket };
 
-enum StorageType {
-  kDontRetransmit,
-  kAllowRetransmission
-};
+enum StorageType { kDontRetransmit, kAllowRetransmission };
 
 enum RTPExtensionType {
   kRtpExtensionNone,
@@ -166,10 +160,10 @@ enum RetransmissionMode : uint8_t {
 };
 
 enum RtxMode {
-  kRtxOff                 = 0x0,
-  kRtxRetransmitted       = 0x1,  // Only send retransmissions over RTX.
-  kRtxRedundantPayloads   = 0x2   // Preventively send redundant payloads
-                                  // instead of padding.
+  kRtxOff = 0x0,
+  kRtxRetransmitted = 0x1,     // Only send retransmissions over RTX.
+  kRtxRedundantPayloads = 0x2  // Preventively send redundant payloads
+                               // instead of padding.
 };
 
 const size_t kRtxHeaderSize = 2;