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Reformat the WebRTC code base

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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
Download Patch File Download Diff File Expand all files Collapse all files

21
rtc_base/asyncpacketsocket.h
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@ -28,9 +28,9 @@ struct PacketTimeUpdateParams {
~PacketTimeUpdateParams();
int rtp_sendtime_extension_id = -1; // extension header id present in packet.
std::vector<char> srtp_auth_key; // Authentication key.
int srtp_auth_tag_len = -1; // Authentication tag length.
int64_t srtp_packet_index = -1; // Required for Rtp Packet authentication.
std::vector<char> srtp_auth_key; // Authentication key.
int srtp_auth_tag_len = -1; // Authentication tag length.
int64_t srtp_packet_index = -1; // Required for Rtp Packet authentication.
};
// This structure holds meta information for the packet which is about to send
@ -57,7 +57,7 @@ struct PacketTime {
PacketTime(int64_t timestamp, int64_t not_before)
: timestamp(timestamp), not_before(not_before) {}
int64_t timestamp; // Receive time after socket delivers the data.
int64_t timestamp; // Receive time after socket delivers the data.
// Earliest possible time the data could have arrived, indicating the
// potential error in the |timestamp| value, in case the system, is busy. For
@ -93,8 +93,10 @@ class AsyncPacketSocket : public sigslot::has_slots<> {
virtual SocketAddress GetRemoteAddress() const = 0;
// Send a packet.
virtual int Send(const void *pv, size_t cb, const PacketOptions& options) = 0;
virtual int SendTo(const void *pv, size_t cb, const SocketAddress& addr,
virtual int Send(const void* pv, size_t cb, const PacketOptions& options) = 0;
virtual int SendTo(const void* pv,
size_t cb,
const SocketAddress& addr,
const PacketOptions& options) = 0;
// Close the socket.
@ -114,9 +116,12 @@ class AsyncPacketSocket : public sigslot::has_slots<> {
// Emitted each time a packet is read. Used only for UDP and
// connected TCP sockets.
sigslot::signal5<AsyncPacketSocket*, const char*, size_t,
sigslot::signal5<AsyncPacketSocket*,
const char*,
size_t,
const SocketAddress&,
const PacketTime&> SignalReadPacket;
const PacketTime&>
SignalReadPacket;
// Emitted each time a packet is sent.
sigslot::signal2<AsyncPacketSocket*, const SentPacket&> SignalSentPacket;