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}
This commit is contained in:
Yves Gerey
2018-06-19 15:03:05 +02:00
parent b602123a5a
commit 665174fdbb
1569 changed files with 30495 additions and 30309 deletions

View File

@ -23,7 +23,7 @@
namespace rtc {
class Network;
struct PacketOptions;
}
} // namespace rtc
namespace cricket {
class Connection;
@ -68,13 +68,12 @@ class PortInterface {
virtual void PrepareAddress() = 0;
// Returns the connection to the given address or NULL if none exists.
virtual Connection* GetConnection(
const rtc::SocketAddress& remote_addr) = 0;
virtual Connection* GetConnection(const rtc::SocketAddress& remote_addr) = 0;
// Creates a new connection to the given address.
enum CandidateOrigin { ORIGIN_THIS_PORT, ORIGIN_OTHER_PORT, ORIGIN_MESSAGE };
virtual Connection* CreateConnection(
const Candidate& remote_candidate, CandidateOrigin origin) = 0;
virtual Connection* CreateConnection(const Candidate& remote_candidate,
CandidateOrigin origin) = 0;
// Functions on the underlying socket(s).
virtual int SetOption(rtc::Socket::Option opt, int value) = 0;
@ -87,25 +86,32 @@ class PortInterface {
// Sends the given packet to the given address, provided that the address is
// that of a connection or an address that has sent to us already.
virtual int SendTo(const void* data, size_t size,
virtual int SendTo(const void* data,
size_t size,
const rtc::SocketAddress& addr,
const rtc::PacketOptions& options, bool payload) = 0;
const rtc::PacketOptions& options,
bool payload) = 0;
// Indicates that we received a successful STUN binding request from an
// address that doesn't correspond to any current connection. To turn this
// into a real connection, call CreateConnection.
sigslot::signal6<PortInterface*, const rtc::SocketAddress&,
ProtocolType, IceMessage*, const std::string&,
bool> SignalUnknownAddress;
sigslot::signal6<PortInterface*,
const rtc::SocketAddress&,
ProtocolType,
IceMessage*,
const std::string&,
bool>
SignalUnknownAddress;
// Sends a response message (normal or error) to the given request. One of
// these methods should be called as a response to SignalUnknownAddress.
// NOTE: You MUST call CreateConnection BEFORE SendBindingResponse.
virtual void SendBindingResponse(StunMessage* request,
const rtc::SocketAddress& addr) = 0;
virtual void SendBindingErrorResponse(
StunMessage* request, const rtc::SocketAddress& addr,
int error_code, const std::string& reason) = 0;
virtual void SendBindingErrorResponse(StunMessage* request,
const rtc::SocketAddress& addr,
int error_code,
const std::string& reason) = 0;
// Signaled when this port decides to delete itself because it no longer has
// any usefulness.
@ -119,8 +125,9 @@ class PortInterface {
// through their respective connection and instead delivers every packet
// through this port.
virtual void EnablePortPackets() = 0;
sigslot::signal4<PortInterface*, const char*, size_t,
const rtc::SocketAddress&> SignalReadPacket;
sigslot::
signal4<PortInterface*, const char*, size_t, const rtc::SocketAddress&>
SignalReadPacket;
// Emitted each time a packet is sent on this port.
sigslot::signal1<const rtc::SentPacket&> SignalSentPacket;