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

@ -22,10 +22,9 @@ class AsyncStunTCPSocket : public rtc::AsyncTCPSocketBase {
// Binds and connects |socket| and creates AsyncTCPSocket for
// it. Takes ownership of |socket|. Returns NULL if bind() or
// connect() fail (|socket| is destroyed in that case).
static AsyncStunTCPSocket* Create(
rtc::AsyncSocket* socket,
const rtc::SocketAddress& bind_address,
const rtc::SocketAddress& remote_address);
static AsyncStunTCPSocket* Create(rtc::AsyncSocket* socket,
const rtc::SocketAddress& bind_address,
const rtc::SocketAddress& remote_address);
AsyncStunTCPSocket(rtc::AsyncSocket* socket, bool listen);
@ -39,8 +38,7 @@ class AsyncStunTCPSocket : public rtc::AsyncTCPSocketBase {
// This method returns the message hdr + length written in the header.
// This method also returns the number of padding bytes needed/added to the
// turn message. |pad_bytes| should be used only when |is_turn| is true.
size_t GetExpectedLength(const void* data, size_t len,
int* pad_bytes);
size_t GetExpectedLength(const void* data, size_t len, int* pad_bytes);
RTC_DISALLOW_COPY_AND_ASSIGN(AsyncStunTCPSocket);
};